From 988aec3de5f0fa848f26fbf64f9e83364d6b3c25 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Wed, 1 Aug 2012 16:05:39 +0200 Subject: ALSA: isa: Move snd_legacy_find_free_ioport to initval.h Move snd_legacy_find_free_ioport() function back to initval.h as it is used by two drivers. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 15 +-------------- sound/isa/opti9xx/opti92x-ad1848.c | 14 +------------- 2 files changed, 2 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c24594c866f4..3d1afb612b35 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -37,6 +37,7 @@ #include #include #include +#define SNDRV_LEGACY_FIND_FREE_IOPORT #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include @@ -770,20 +771,6 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return 0; } -static long snd_legacy_find_free_ioport(long *port_table, long size) -{ - while (*port_table != -1) { - struct resource *res; - if ((res = request_region(*port_table, size, - "ALSA test")) != NULL) { - release_and_free_resource(res); - return *port_table; - } - port_table++; - } - return -1; -} - static int __devinit snd_miro_init(struct snd_miro *chip, unsigned short hardware) { diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index f8fbe22515c9..2899c9fd1ceb 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -39,6 +39,7 @@ #ifndef OPTi93X #include #endif +#define SNDRV_LEGACY_FIND_FREE_IOPORT #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include @@ -185,19 +186,6 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; - -static long __devinit snd_legacy_find_free_ioport(long *port_table, long size) -{ - while (*port_table != -1) { - if (request_region(*port_table, size, "ALSA test")) { - release_region(*port_table, size); - return *port_table; - } - port_table++; - } - return -1; -} - static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, unsigned short hardware) { -- cgit v1.2.3 From d77ae3329292baebfc6eced97d2e12b66349f83c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 23 Jul 2012 12:39:51 +0300 Subject: ASoC: omap-mcpdm: Convert to use devm_* Switch to use devm_* te make the probe/remove code more cleaner. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 52 ++++++++++++--------------------------------- 1 file changed, 13 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 2c66e2498a45..f7babb374a37 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -445,9 +445,8 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) { struct omap_mcpdm *mcpdm; struct resource *res; - int ret = 0; - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL); if (!mcpdm) return -ENOMEM; @@ -456,55 +455,30 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_res; - } + if (res == NULL) + return -ENOMEM; - if (!request_mem_region(res->start, resource_size(res), "McPDM")) { - ret = -EBUSY; - goto err_res; - } + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), "McPDM")) + return -EBUSY; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_iomap; - } + mcpdm->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcpdm->io_base) + return -ENOMEM; mcpdm->irq = platform_get_irq(pdev, 0); - if (mcpdm->irq < 0) { - ret = mcpdm->irq; - goto err_irq; - } + if (mcpdm->irq < 0) + return mcpdm->irq; mcpdm->dev = &pdev->dev; - ret = snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); - if (!ret) - return 0; - -err_irq: - iounmap(mcpdm->io_base); -err_iomap: - release_mem_region(res->start, resource_size(res)); -err_res: - kfree(mcpdm); - return ret; + return snd_soc_register_dai(&pdev->dev, &omap_mcpdm_dai); } static int __devexit asoc_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm = platform_get_drvdata(pdev); - struct resource *res; - snd_soc_unregister_dai(&pdev->dev); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - iounmap(mcpdm->io_base); - release_mem_region(res->start, resource_size(res)); - - kfree(mcpdm); return 0; } -- cgit v1.2.3 From 0651322bfcf3ca51802c6d8d161d6d1c9f3013eb Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 23 Jul 2012 17:25:15 +0530 Subject: ASoC: isabelle: Remove version.h header file inclusion version.h header file inclusion is no longer needed for this file. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5d8f39e32978..1bf55602c9eb 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -13,7 +13,6 @@ */ #include #include -#include #include #include #include -- cgit v1.2.3 From fbfe69836c088bcc0c5a0f32e788d3aef5f66aca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jul 2012 20:14:43 +0100 Subject: ASoC: wm8994: Implement support for self-oscillation mode in the FLL The FLLs in the WM8994 series devices can be started without any reference being supplied, mainly for use in analogue bypass cases. Implement support for this mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 21 +++++++++++++++------ sound/soc/codecs/wm8994.h | 9 +++++---- 2 files changed, 20 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef03175c51..2c9b8b7fdf3d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2102,6 +2102,10 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, case WM8994_FLL_SRC_LRCLK: case WM8994_FLL_SRC_BCLK: break; + case WM8994_FLL_SRC_INTERNAL: + freq_in = 12000000; + freq_out = 12000000; + break; default: return -EINVAL; } @@ -2164,9 +2168,11 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, - WM8958_FLL1_BYP | + WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | WM8994_FLL1_REFCLK_DIV_MASK | WM8994_FLL1_REFCLK_SRC_MASK, + ((src == WM8994_FLL_SRC_INTERNAL) + << WM8994_FLL1_FRC_NCO_SHIFT) | (fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) | (src - 1)); @@ -2192,13 +2198,16 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, } } + reg = WM8994_FLL1_ENA; + if (fll.k) - reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC; - else - reg = WM8994_FLL1_ENA; + reg |= WM8994_FLL1_FRAC; + if (src == WM8994_FLL_SRC_INTERNAL) + reg |= WM8994_FLL1_OSC_ENA; + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, - WM8994_FLL1_ENA | WM8994_FLL1_FRAC, - reg); + WM8994_FLL1_ENA | WM8994_FLL1_OSC_ENA | + WM8994_FLL1_FRAC, reg); if (wm8994->fll_locked_irq) { timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index d77e06f0a675..19068d8fa301 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -28,10 +28,11 @@ #define WM8994_FLL1 1 #define WM8994_FLL2 2 -#define WM8994_FLL_SRC_MCLK1 1 -#define WM8994_FLL_SRC_MCLK2 2 -#define WM8994_FLL_SRC_LRCLK 3 -#define WM8994_FLL_SRC_BCLK 4 +#define WM8994_FLL_SRC_MCLK1 1 +#define WM8994_FLL_SRC_MCLK2 2 +#define WM8994_FLL_SRC_LRCLK 3 +#define WM8994_FLL_SRC_BCLK 4 +#define WM8994_FLL_SRC_INTERNAL 5 enum wm8994_vmid_mode { WM8994_VMID_NORMAL, -- cgit v1.2.3 From 0dcd47426abde223b2386165470ec45d9777478e Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Thu, 26 Jul 2012 11:28:37 +0100 Subject: ASoC: ux500: Strengthen error checking after memory allocation Currently there is no out-of-memory error checking after attempting to allocate memory for the ux500_msp or ux500_msp_i2s_drvdata data structures. Instead we go about populating them regardless. This patch applies the necessary error checking to prevent a panic. Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 3 +++ sound/soc/ux500/ux500_msp_i2s.c | 2 ++ 2 files changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 057e28ef770e..772cb19d2fb3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -760,6 +760,9 @@ static int __devinit ux500_msp_drv_probe(struct platform_device *pdev) drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp_i2s_drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + drvdata->fmt = 0; drvdata->slots = 1; drvdata->tx_mask = 0x01; diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 5c472f335a64..36be11e47ad6 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -673,6 +673,8 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; + if (!msp) + return -ENOMEM; msp->id = platform_data->id; msp->dev = &pdev->dev; -- cgit v1.2.3 From 5ef75e710b4950439f953c4897e4a871c2f9dc8f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 25 Jul 2012 16:09:11 +0300 Subject: ASoC: omap-abe-twl6040: Add device tree support When the board boots with device tree the driver will receive the name of the card, DAPM routing map, phandle for the audio components described in the dts file, mclk speed, and the possibility of detecting the jack detection. The card will be set up based on this information. Since the routing is provided via DT we can mark the card fully routed so core can take care of disconnecting the unused pins. Signed-off-by: Peter Ujfalusi Reviwed-by: Mark Brown Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-abe-twl6040.txt | 91 +++++++++++++ sound/soc/omap/omap-abe-twl6040.c | 145 ++++++++++++++++----- 2 files changed, 206 insertions(+), 30 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt new file mode 100644 index 000000000000..65dec876cb2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt @@ -0,0 +1,91 @@ +* Texas Instruments OMAP4+ and twl6040 based audio setups + +Required properties: +- compatible: "ti,abe-twl6040" +- ti,model: Name of the sound card ( for example "SDP4430") +- ti,mclk-freq: MCLK frequency for HPPLL operation +- ti,mcpdm: phandle for the McPDM node +- ti,twl6040: phandle for the twl6040 core node +- ti,audio-routing: List of connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. + +Optional properties: +- ti,dmic: phandle for the OMAP dmic node if the machine have it connected +- ti,jack_detection: Need to be set to <1> if the board capable to detect jack + insertion, removal. + +Available audio endpoints for the audio-routing table: + +Board connectors: + * Headset Stereophone + * Earphone Spk + * Ext Spk + * Line Out + * Vibrator + * Headset Mic + * Main Handset Mic + * Sub Handset Mic + * Line In + * Digital Mic + +twl6040 pins: + * HSOL + * HSOR + * EP + * HFL + * HFR + * AUXL + * AUXR + * VIBRAL + * VIBRAR + * HSMIC + * MAINMIC + * SUBMIC + * AFML + * AFMR + + * Headset Mic Bias + * Main Mic Bias + * Digital Mic1 Bias + * Digital Mic2 Bias + +Digital mic pins: + * DMic + +Example: + +sound { + compatible = "ti,abe-twl6040"; + ti,model = "SDP4430"; + + ti,jack-detection = <1>; + ti,mclk-freq = <38400000>; + + ti,mcpdm = <&mcpdm>; + ti,dmic = <&dmic>; + + ti,twl6040 = <&twl6040>; + + /* Audio routing */ + ti,audio-routing = + "Headset Stereophone", "HSOL", + "Headset Stereophone", "HSOR", + "Earphone Spk", "EP", + "Ext Spk", "HFL", + "Ext Spk", "HFR", + "Line Out", "AUXL", + "Line Out", "AUXR", + "Vibrator", "VIBRAL", + "Vibrator", "VIBRAR", + "HSMIC", "Headset Mic", + "Headset Mic", "Headset Mic Bias", + "MAINMIC", "Main Handset Mic", + "Main Handset Mic", "Main Mic Bias", + "SUBMIC", "Sub Handset Mic", + "Sub Handset Mic", "Main Mic Bias", + "AFML", "Line In", + "AFMR", "Line In", + "DMic", "Digital Mic", + "Digital Mic", "Digital Mic1 Bias"; +}; diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 9d93793d3077..be525dfe9faa 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include @@ -43,6 +44,8 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ + + struct platform_device *dmic_codec_dev; }; static int omap_abe_hw_params(struct snd_pcm_substream *substream, @@ -185,17 +188,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) int hs_trim; int ret = 0; - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - /* * Configure McPDM offset cancellation based on the HSOTRIM value from * twl6040. @@ -216,6 +208,24 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } + /* + * NULL pdata means we booted with DT. In this case the routing is + * provided and the card is fully routed, no need to mark pins. + */ + if (!pdata) + return ret; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); + return ret; } @@ -270,52 +280,116 @@ static struct snd_soc_card omap_abe_card = { static __devinit int omap_abe_probe(struct platform_device *pdev) { struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; struct abe_twl6040 *priv; int num_links = 0; - int ret; + int ret = 0; card->dev = &pdev->dev; - if (!pdata) { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; - } - priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (pdata->card_name) { - card->name = pdata->card_name; + priv->dmic_codec_dev = ERR_PTR(-EINVAL); + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + ret = snd_soc_of_parse_audio_routing(card, + "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, + "Error while parsing DAPM routing\n"); + return ret; + } + + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; + + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; + + priv->dmic_codec_dev = platform_device_register_simple( + "dmic-codec", -1, NULL, 0); + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, + "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); + } + } else { + num_links = 1; + } + + of_property_read_u32(node, "ti,jack-detection", + &priv->jack_detection); + of_property_read_u32(node, "ti,mclk-freq", + &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; + } + + omap_abe_card.fully_routed = 1; + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + if (pdata->has_dmic) + num_links = 2; + else + num_links = 1; + + priv->jack_detection = pdata->jack_detection; + priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Card name is not provided\n"); + dev_err(&pdev->dev, "Missing pdata\n"); return -ENODEV; } - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; - if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - return -ENODEV; + ret = -ENODEV; + goto err_unregister; } - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - card->dai_link = abe_twl6040_dai_links; card->num_links = num_links; snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_unregister; + } + + return 0; + +err_unregister: + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -323,17 +397,28 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) static int __devexit omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); + struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); + if (!IS_ERR(priv->dmic_codec_dev)) + platform_device_unregister(priv->dmic_codec_dev); + return 0; } +static const struct of_device_id omap_abe_of_match[] = { + {.compatible = "ti,abe-twl6040", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_abe_of_match); + static struct platform_driver omap_abe_driver = { .driver = { .name = "omap-abe-twl6040", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = omap_abe_of_match, }, .probe = omap_abe_probe, .remove = __devexit_p(omap_abe_remove), -- cgit v1.2.3 From acaf24f015b2bdd34032188d26c3092d6ca749b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jul 2012 22:57:35 +0100 Subject: ASoC: jack: Always update jack state even for noop changes Now that DAPM is very cheap for most updates we've no need to avoid trying to run it so always notify even if we don't think there are any changes. This avoids potential issues with bootstrapping state like the pin state or other notifiers when there's nothing in the jack. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-jack.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7f8b3b7428bb..2ca3c734a288 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -83,11 +83,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) jack->status &= ~mask; jack->status |= status & mask; - /* The DAPM sync is expensive enough to be worth skipping. - * However, empty mask means pin synchronization is desired. */ - if (mask && (jack->status == oldstatus)) - goto out; - trace_snd_soc_jack_notify(jack, status); list_for_each_entry(pin, &jack->pins, list) { @@ -109,7 +104,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_jack_report(jack->jack, jack->status); -out: mutex_unlock(&jack->mutex); } EXPORT_SYMBOL_GPL(snd_soc_jack_report); -- cgit v1.2.3 From 8cb8e83bfa7cb63ad4b3c3b79410766da397124b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jul 2012 18:10:03 +0100 Subject: ASoC: wm_hubs: Move CODEC stored in private data into wm_hubs Further wm_hubs code will use this. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 28 ++++++++++++++++------------ sound/soc/codecs/wm8994.c | 28 +++++++++++++--------------- sound/soc/codecs/wm8994.h | 1 - sound/soc/codecs/wm_hubs.c | 2 ++ sound/soc/codecs/wm_hubs.h | 2 ++ 5 files changed, 33 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 1332692ef81b..00121ba36597 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -946,7 +946,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_texts = kmalloc(sizeof(char *) * pdata->num_mbc_cfgs, GFP_KERNEL); if (!wm8994->mbc_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d MBC config texts\n", pdata->num_mbc_cfgs); return; @@ -958,9 +958,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_enum.max = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add MBC mode controls: %d\n", ret); } @@ -974,7 +975,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_texts = kmalloc(sizeof(char *) * pdata->num_vss_cfgs, GFP_KERNEL); if (!wm8994->vss_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d VSS config texts\n", pdata->num_vss_cfgs); return; @@ -986,9 +987,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_enum.max = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add VSS mode controls: %d\n", ret); } @@ -1003,7 +1005,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_texts = kmalloc(sizeof(char *) * pdata->num_vss_hpf_cfgs, GFP_KERNEL); if (!wm8994->vss_hpf_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d VSS HPF config texts\n", pdata->num_vss_hpf_cfgs); return; @@ -1015,9 +1017,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add VSS HPFmode controls: %d\n", ret); } @@ -1033,7 +1036,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_texts = kmalloc(sizeof(char *) * pdata->num_enh_eq_cfgs, GFP_KERNEL); if (!wm8994->enh_eq_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d enhanced EQ config texts\n", pdata->num_enh_eq_cfgs); return; @@ -1045,9 +1048,10 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + control, 1); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add enhanced EQ controls: %d\n", ret); } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2c9b8b7fdf3d..1237c11c8c35 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3036,7 +3036,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) { - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; struct wm8994_pdata *pdata = wm8994->pdata; struct snd_kcontrol_new controls[] = { SOC_ENUM_EXT("AIF1.1 EQ Mode", @@ -3094,16 +3094,16 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add ReTune Mobile controls: %d\n", ret); } static void wm8994_handle_pdata(struct wm8994_priv *wm8994) { - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; struct wm8994_pdata *pdata = wm8994->pdata; int ret, i; @@ -3132,10 +3132,10 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) }; /* We need an array of texts for the enum API */ - wm8994->drc_texts = devm_kzalloc(wm8994->codec->dev, + wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev, sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL); if (!wm8994->drc_texts) { - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to allocate %d DRC config texts\n", pdata->num_drc_cfgs); return; @@ -3147,10 +3147,10 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) wm8994->drc_enum.max = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; - ret = snd_soc_add_codec_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); if (ret != 0) - dev_err(wm8994->codec->dev, + dev_err(wm8994->hubs.codec->dev, "Failed to add DRC mode controls: %d\n", ret); for (i = 0; i < WM8994_NUM_DRC; i++) @@ -3163,7 +3163,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) if (pdata->num_retune_mobile_cfgs) wm8994_handle_retune_mobile_pdata(wm8994); else - snd_soc_add_codec_controls(wm8994->codec, wm8994_eq_controls, + snd_soc_add_codec_controls(wm8994->hubs.codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { @@ -3318,7 +3318,7 @@ static void wm8994_mic_work(struct work_struct *work) static irqreturn_t wm8994_mic_irq(int irq, void *data) { struct wm8994_priv *priv = data; - struct snd_soc_codec *codec = priv->codec; + struct snd_soc_codec *codec = priv->hubs.codec; #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); @@ -3431,7 +3431,7 @@ static void wm8958_default_micdet(u16 status, void *data) static irqreturn_t wm1811_jackdet_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; int reg; bool present; @@ -3609,7 +3609,7 @@ EXPORT_SYMBOL_GPL(wm8958_mic_detect); static irqreturn_t wm8958_mic_irq(int irq, void *data) { struct wm8994_priv *wm8994 = data; - struct snd_soc_codec *codec = wm8994->codec; + struct snd_soc_codec *codec = wm8994->hubs.codec; int reg, count; /* @@ -3699,13 +3699,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) unsigned int reg; int ret, i; - wm8994->codec = codec; + wm8994->hubs.codec = codec; codec->control_data = control->regmap; snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - wm8994->codec = codec; - mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 19068d8fa301..e6d8209b8f29 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -73,7 +73,6 @@ struct wm8994; struct wm8994_priv { struct wm_hubs_data hubs; struct wm8994 *wm8994; - struct snd_soc_codec *codec; int sysclk[2]; int sysclk_rate[2]; int mclk[2]; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 61baa48823cb..728a18010a49 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1112,6 +1112,8 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + hubs->codec = codec; + INIT_LIST_HEAD(&hubs->dcs_cache); init_completion(&hubs->dcs_done); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index da2dc899ce6d..a5a09e6f87d5 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -46,6 +46,8 @@ struct wm_hubs_data { bool dcs_done_irq; struct completion dcs_done; + + struct snd_soc_codec *codec; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); -- cgit v1.2.3 From 99af79dff5a609fe886d271bbc91e1a95eca3066 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jul 2012 23:03:36 +0100 Subject: ASoC: wm8994: Ensure we get a notification on startup for jackdet Since jackdet only reports deltas it won't generate an interrupt on startup when a jack is not present. This doesn't make a difference to userspace but does mean we don't generate a notification via the internal notifier chains. Fix that by scheduling a work to poll the chip after the clock is enabled. Use an extremely large timeout since there's no urgency and we don't want to report a false negative. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8994.h | 2 ++ 2 files changed, 38 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1237c11c8c35..7bb0c2c824cc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -789,11 +789,27 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: return configure_clock(codec); + case SND_SOC_DAPM_POST_PMU: + /* + * JACKDET won't run until we start the clock and it + * only reports deltas, make sure we notify the state + * up the stack on startup. Use a *very* generous + * timeout for paranoia, there's no urgency and we + * don't want false reports. + */ + if (wm8994->jackdet && !wm8994->clk_has_run) { + schedule_delayed_work(&wm8994->jackdet_bootstrap, + msecs_to_jiffies(1000)); + wm8994->clk_has_run = true; + } + break; + case SND_SOC_DAPM_POST_PMD: configure_clock(codec); break; @@ -1632,7 +1648,8 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), @@ -3508,10 +3525,22 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + /* Since we only report deltas force an update, ensures we + * avoid bootstrapping issues with the core. */ + snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0); + pm_runtime_put(codec->dev); return IRQ_HANDLED; } +static void wm1811_jackdet_bootstrap(struct work_struct *work) +{ + struct wm8994_priv *wm8994 = container_of(work, + struct wm8994_priv, + jackdet_bootstrap.work); + wm1811_jackdet_irq(0, wm8994); +} + /** * wm8958_mic_detect - Enable microphone detection via the WM8958 IRQ * @@ -3582,6 +3611,10 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, * otherwise jump straight to microphone detection. */ if (wm8994->jackdet) { + /* Disable debounce for the initial detect */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, 0); + snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); @@ -3706,6 +3739,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); + INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, + wm1811_jackdet_bootstrap); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index e6d8209b8f29..f142ec198db3 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -81,6 +81,7 @@ struct wm8994_priv { struct completion fll_locked[2]; bool fll_locked_irq; bool fll_byp; + bool clk_has_run; int vmid_refcount; int active_refcount; @@ -134,6 +135,7 @@ struct wm8994_priv { int btn_mask; bool jackdet; int jackdet_mode; + struct delayed_work jackdet_bootstrap; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- cgit v1.2.3 From fae4efa23ac012a57d45682bc22d540271c54532 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jul 2012 19:49:06 +0100 Subject: ASoC: wm_hubs: Factor out DC servo readback code It's currently only used in one place but another user will be added shortly and there's an argument it's clearer anyway. Also add support for readback in mode 1, though it's not currently used. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 65 ++++++++++++++++++++++++++++++---------------- 1 file changed, 43 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 728a18010a49..6ab69f32f249 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -199,6 +199,47 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) list_add_tail(&cache->list, &hubs->dcs_cache); } +static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, + u16 *reg_l, u16 *reg_r) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + u16 dcs_reg, reg; + + switch (hubs->dcs_readback_mode) { + case 2: + dcs_reg = WM8994_DC_SERVO_4E; + break; + case 1: + dcs_reg = WM8994_DC_SERVO_READBACK; + break; + default: + dcs_reg = WM8993_DC_SERVO_3; + break; + } + + /* Different chips in the family support different readback + * methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + *reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK; + *reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 2: + case 1: + reg = snd_soc_read(codec, dcs_reg); + *reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + *reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method\n"); + return; + } +} + /* * Startup calibration of the DC servo */ @@ -207,7 +248,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct wm_hubs_dcs_cache *cache; s8 offset; - u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg; + u16 reg_l, reg_r, dcs_cfg, dcs_reg; switch (hubs->dcs_readback_mode) { case 2: @@ -245,27 +286,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_STARTUP_1); } - /* Different chips in the family support different readback - * methods. - */ - switch (hubs->dcs_readback_mode) { - case 0: - reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) - & WM8993_DCS_INTEG_CHAN_0_MASK; - reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) - & WM8993_DCS_INTEG_CHAN_1_MASK; - break; - case 2: - case 1: - reg = snd_soc_read(codec, dcs_reg); - reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) - >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; - break; - default: - WARN(1, "Unknown DCS readback method\n"); - return; - } + wm_hubs_read_dc_servo(codec, ®_l, ®_r); dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); -- cgit v1.2.3 From a7892c35cfad4d6c6ccf1242b55c1004b0d5d1d1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jul 2012 19:50:45 +0100 Subject: ASoC: wm_hubs: Rename calibrate_dc_servo() Really we're enabling it here and the name will become very confusing shortly. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 6ab69f32f249..05a02e1b7e92 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -243,7 +243,7 @@ static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, /* * Startup calibration of the DC servo */ -static void calibrate_dc_servo(struct snd_soc_codec *codec) +static void enable_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct wm_hubs_dcs_cache *cache; @@ -556,7 +556,7 @@ static int hp_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_TIMER_PERIOD_01_MASK, 0); - calibrate_dc_servo(codec); + enable_dc_servo(codec); reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT | WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT; -- cgit v1.2.3 From 7435d4eec76ee9debffb070f3e0d67615a828673 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jul 2012 14:49:11 +0100 Subject: ASoC: wm8994: Fix some indentation issues Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7bb0c2c824cc..5fc31797994a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2182,7 +2182,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, WM8994_FLL1_N_MASK, - fll.n << WM8994_FLL1_N_SHIFT); + fll.n << WM8994_FLL1_N_SHIFT); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset, WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP | @@ -3371,7 +3371,7 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, 0, wm8994->btn_mask | - SND_JACK_HEADSET); + SND_JACK_HEADSET); } return; } -- cgit v1.2.3 From d95e933730b3eb7b06bd778dc1d8f0ab3702b607 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jul 2012 15:25:48 +0100 Subject: ASoC: ab8500: Remove pointless cast There's never any need to cast away from void. Signed-off-by: Mark Brown Acked-by: Lee Jones --- sound/soc/codecs/ab8500-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 23b40186f9b8..b7836503dc69 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2404,12 +2404,12 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); - /* Setup AB8500 according to board-settings */ - pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); - /* Inform SoC Core that we have our own I/O arrangements. */ codec->control_data = (void *)true; + /* Setup AB8500 according to board-settings */ + pdata = dev_get_platdata(dev->parent); + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); -- cgit v1.2.3 From d9f34df782b2aa7d233cb08850c8b12fdb37d18a Mon Sep 17 00:00:00 2001 From: Chris Rattray Date: Tue, 31 Jul 2012 14:51:34 +0100 Subject: ASoC: wm8994: enable mic and short detect debounce. Signed-off-by: Chris Rattray Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5fc31797994a..02080da8b451 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3262,6 +3262,12 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg); + /* enable MICDET and MICSHRT deboune */ + snd_soc_update_bits(codec, WM8994_IRQ_DEBOUNCE, + WM8994_MIC1_DET_DB_MASK | WM8994_MIC1_SHRT_DB_MASK | + WM8994_MIC2_DET_DB_MASK | WM8994_MIC2_SHRT_DB_MASK, + WM8994_MIC1_DET_DB | WM8994_MIC1_SHRT_DB); + snd_soc_dapm_sync(&codec->dapm); return 0; -- cgit v1.2.3 From 85d07e4d625d6511934799f7df93e9111ac2c88b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 25 Jul 2012 15:28:34 +0200 Subject: ASoC: add DT bindings for cs4270 Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4270.txt | 16 ++++++++++++++++ sound/soc/codecs/cs4270.c | 11 +++++++++++ 2 files changed, 27 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt new file mode 100644 index 000000000000..7f0bfd84d3fc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -0,0 +1,16 @@ +CS4270 audio CODEC + +The driver for this device currently only supports I2C. + +Required properties: + + - compatible : "cirrus,cs4270" + + - reg : the I2C address of the device for I2C + +Example: + +codec: cs4270@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; +}; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 047917f0b8ae..4b71b01ecbcd 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,6 +29,7 @@ #include #include #include +#include /* * The codec isn't really big-endian or little-endian, since the I2S @@ -639,6 +640,15 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .reg_cache_default = cs4270_default_reg_cache, }; +/* + * cs4270_of_match - the device tree bindings + */ +static const struct of_device_id cs4270_of_match[] = { + { .compatible = "cirrus,cs4270", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4270_of_match); + /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object @@ -718,6 +728,7 @@ static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", .owner = THIS_MODULE, + .of_match_table = cs4270_of_match, }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, -- cgit v1.2.3 From 02286190f3ec86f03025a60c4d3f747ff1047248 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 25 Jul 2012 15:28:35 +0200 Subject: ASoC: Add reset-gpio DT property to cs4270 driver In the process of moving over from static board files to the device tree, reset pins of peripheral reset pins should be handled by their corresponding drivers. Add a reset-gpio DT property to the cs4270 driver, and de-assert it before probing the chip. The logic could be augmented some day to re-assert it when codec is put to suspend. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/cs4270.txt | 5 +++++ sound/soc/codecs/cs4270.c | 17 +++++++++++++++++ 2 files changed, 22 insertions(+) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt index 7f0bfd84d3fc..6b222f9b8ef5 100644 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ b/Documentation/devicetree/bindings/sound/cs4270.txt @@ -8,6 +8,11 @@ Required properties: - reg : the I2C address of the device for I2C +Optional properties: + + - reset-gpio : a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + Example: codec: cs4270@48 { diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4b71b01ecbcd..fd11bb646d40 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -30,6 +30,7 @@ #include #include #include +#include /* * The codec isn't really big-endian or little-endian, since the I2S @@ -660,9 +661,25 @@ MODULE_DEVICE_TABLE(of, cs4270_of_match); static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { + struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; int ret; + /* See if we have a way to bring the codec out of reset */ + if (np) { + enum of_gpio_flags flags; + int gpio = of_get_named_gpio_flags(np, "reset-gpio", 0, &flags); + + if (gpio_is_valid(gpio)) { + ret = devm_gpio_request_one(&i2c_client->dev, gpio, + flags & OF_GPIO_ACTIVE_LOW ? + GPIOF_OUT_INIT_LOW : GPIOF_OUT_INIT_HIGH, + "cs4270 reset"); + if (ret < 0) + return ret; + } + } + /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); -- cgit v1.2.3 From ad3ab1bba9bf3fcd13a4e3f868a438013174dcc1 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 27 Jul 2012 14:32:10 -0300 Subject: ASoC: imx-ssi: Use devm functions Using devm_ functions can make the code simpler and smaller. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c7cb1c..e174c1767c2d 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -523,7 +523,7 @@ static int imx_ssi_probe(struct platform_device *pdev) int ret = 0; struct snd_soc_dai_driver *dai; - ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + ssi = devm_kzalloc(&pdev->dev, sizeof(*ssi), GFP_KERNEL); if (!ssi) return -ENOMEM; dev_set_drvdata(&pdev->dev, ssi); @@ -536,7 +536,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->irq = platform_get_irq(pdev, 0); - ssi->clk = clk_get(&pdev->dev, NULL); + ssi->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi->clk)) { ret = PTR_ERR(ssi->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -551,23 +551,17 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_get_resource; } - if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - ssi->base = ioremap(res->start, resource_size(res)); + ssi->base = devm_request_and_ioremap(&pdev->dev, res); if (!ssi->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_register; } if (ssi->flags & IMX_SSI_USE_AC97) { if (ac97_ssi) { ret = -EBUSY; - goto failed_ac97; + goto failed_register; } ac97_ssi = ssi; setup_channel_to_ac97(ssi); @@ -636,15 +630,10 @@ failed_pdev_fiq_add: failed_pdev_fiq_alloc: snd_soc_unregister_dai(&pdev->dev); failed_register: -failed_ac97: - iounmap(ssi->base); -failed_ioremap: release_mem_region(res->start, resource_size(res)); failed_get_resource: clk_disable_unprepare(ssi->clk); - clk_put(ssi->clk); failed_clk: - kfree(ssi); return ret; } @@ -662,11 +651,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - iounmap(ssi->base); release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); - clk_put(ssi->clk); - kfree(ssi); return 0; } -- cgit v1.2.3 From 9a37eae230e7350ba803801404a022e098016e56 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:37:30 +0100 Subject: ASoC: wm9712: Fix funky indentation Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f16fb361a4eb..3fefa6e8e451 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -684,8 +684,8 @@ static int __devexit wm9712_remove(struct platform_device *pdev) static struct platform_driver wm9712_codec_driver = { .driver = { - .name = "wm9712-codec", - .owner = THIS_MODULE, + .name = "wm9712-codec", + .owner = THIS_MODULE, }, .probe = wm9712_probe, -- cgit v1.2.3 From 689185b78ba6fbe0042f662a468b5565909dff7a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:37:29 +0100 Subject: ASoC: wm9712: Fix name of Capture Switch Help UIs associate it with the matching gain control. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 3fefa6e8e451..c9c696ca76f7 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -146,7 +146,7 @@ SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), -SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), +SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), -- cgit v1.2.3 From 1427cc37b6c073e83309565bfebad25fb6cd9182 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Aug 2012 19:25:50 +0100 Subject: ASoC: sta529: Staticise non-exported codec driver struct Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 0c225cd569d2..9e3144862386 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -358,7 +358,7 @@ static int sta529_resume(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sta529_codec_driver = { +static const struct snd_soc_codec_driver sta529_codec_driver = { .probe = sta529_probe, .remove = sta529_remove, .set_bias_level = sta529_set_bias_level, -- cgit v1.2.3 From a0f1e98b34f22bb4055aebfc528bc9080b259f8f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jul 2012 18:34:07 +0100 Subject: ASoC: imx-ssi: Say if we fail to register a second AC'97 bus Saves anyone wondering what happened if they run into this error. Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index e174c1767c2d..3c520c46fa4a 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -560,6 +560,7 @@ static int imx_ssi_probe(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) { if (ac97_ssi) { + dev_err(&pdev->dev, "AC'97 SSI already registered\n"); ret = -EBUSY; goto failed_register; } -- cgit v1.2.3 From 793ea49c476ebacfefabf78af9ea88a19da6ecdb Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Thu, 2 Aug 2012 16:52:41 +0300 Subject: ALSA: print small buffers via %*ph[C] Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai --- sound/isa/gus/interwave.c | 5 ++--- sound/usb/6fire/firmware.c | 5 ++--- 2 files changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index a76bc8d27c1d..3fc8b66fd167 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,9 +443,8 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %02x:%02x:%02x:%02x:%02x:%02x:%02x:%02x\n", bank_pos, - iwave[0], iwave[1], iwave[2], iwave[3], - iwave[4], iwave[5], iwave[6], iwave[7]); + printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, + 8, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 56ad923bf6b5..a1d9b0792a1e 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,11 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 4)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: " - "%02x %02x %02x %02x. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - version[0], version[1], version[2], version[3]); + 4, version); return -EINVAL; } -- cgit v1.2.3 From 3b09efd1decb2b36ba2c7eb88ae4893f0581e470 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:34 +0530 Subject: ASoC: tlv320aic32x4: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index b0a73d37ed52..f230292ba96b 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -746,24 +746,7 @@ static struct i2c_driver aic32x4_i2c_driver = { .id_table = aic32x4_i2c_id, }; -static int __init aic32x4_modinit(void) -{ - int ret = 0; - - ret = i2c_add_driver(&aic32x4_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(aic32x4_modinit); - -static void __exit aic32x4_exit(void) -{ - i2c_del_driver(&aic32x4_i2c_driver); -} -module_exit(aic32x4_exit); +module_i2c_driver(aic32x4_i2c_driver); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin "); -- cgit v1.2.3 From a3627e9c0a22283cb1f73fa1170f70fe604315d9 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:35 +0530 Subject: ASoC: max9877: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 3a2ba3d8fd6d..d15e5943c85e 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -291,17 +291,7 @@ static struct i2c_driver max9877_i2c_driver = { .id_table = max9877_i2c_id, }; -static int __init max9877_init(void) -{ - return i2c_add_driver(&max9877_i2c_driver); -} -module_init(max9877_init); - -static void __exit max9877_exit(void) -{ - i2c_del_driver(&max9877_i2c_driver); -} -module_exit(max9877_exit); +module_i2c_driver(max9877_i2c_driver); MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); MODULE_AUTHOR("Joonyoung Shim "); -- cgit v1.2.3 From 96124c2910a5317b5ea9fbfd9e07b135fdc1dd28 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:39 +0530 Subject: ASoC: wm9090: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2c2346fdd637..c7ddc56175d1 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -695,17 +695,7 @@ static struct i2c_driver wm9090_i2c_driver = { .id_table = wm9090_id, }; -static int __init wm9090_init(void) -{ - return i2c_add_driver(&wm9090_i2c_driver); -} -module_init(wm9090_init); - -static void __exit wm9090_exit(void) -{ - i2c_del_driver(&wm9090_i2c_driver); -} -module_exit(wm9090_exit); +module_i2c_driver(wm9090_i2c_driver); MODULE_AUTHOR("Mark Brown "); MODULE_DESCRIPTION("WM9090 ASoC driver"); -- cgit v1.2.3 From 38ece8db99e1c7954608a6452bbe8b331d269ad6 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:40 +0530 Subject: ASoC: wm8991: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 9ac31ba9b82e..b9dbfebbda13 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1400,23 +1400,7 @@ static struct i2c_driver wm8991_i2c_driver = { .id_table = wm8991_i2c_id, }; -static int __init wm8991_modinit(void) -{ - int ret; - ret = i2c_add_driver(&wm8991_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", - ret); - } - return 0; -} -module_init(wm8991_modinit); - -static void __exit wm8991_exit(void) -{ - i2c_del_driver(&wm8991_i2c_driver); -} -module_exit(wm8991_exit); +module_i2c_driver(wm8991_i2c_driver); MODULE_DESCRIPTION("ASoC WM8991 driver"); MODULE_AUTHOR("Graeme Gregory"); -- cgit v1.2.3 From 5e383f53e80f4c644bc49a17b5590b110bad5832 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:41 +0530 Subject: ASoC: cs4270: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index fd11bb646d40..44a176f74170 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -752,17 +752,7 @@ static struct i2c_driver cs4270_i2c_driver = { .remove = cs4270_i2c_remove, }; -static int __init cs4270_init(void) -{ - return i2c_add_driver(&cs4270_i2c_driver); -} -module_init(cs4270_init); - -static void __exit cs4270_exit(void) -{ - i2c_del_driver(&cs4270_i2c_driver); -} -module_exit(cs4270_exit); +module_i2c_driver(cs4270_i2c_driver); MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); -- cgit v1.2.3 From fd39d14b9676cfd3dbd5b7bfdefe3ec6149b9e1a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:42 +0530 Subject: ASoC: tlv320aic3x: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index dc78f5a4bcbf..01485bd51404 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1499,23 +1499,7 @@ static struct i2c_driver aic3x_i2c_driver = { .id_table = aic3x_i2c_id, }; -static int __init aic3x_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC3x I2C driver: %d\n", - ret); - } - return ret; -} -module_init(aic3x_modinit); - -static void __exit aic3x_exit(void) -{ - i2c_del_driver(&aic3x_i2c_driver); -} -module_exit(aic3x_exit); +module_i2c_driver(aic3x_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); MODULE_AUTHOR("Vladimir Barinov"); -- cgit v1.2.3 From 0ead1136bda75d04cc134960bd265eebe210f74b Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:43 +0530 Subject: ASoC: sta32x: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/sta32x.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 8d717f4b5a87..51b7313a4c14 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1006,17 +1006,7 @@ static struct i2c_driver sta32x_i2c_driver = { .id_table = sta32x_i2c_id, }; -static int __init sta32x_init(void) -{ - return i2c_add_driver(&sta32x_i2c_driver); -} -module_init(sta32x_init); - -static void __exit sta32x_exit(void) -{ - i2c_del_driver(&sta32x_i2c_driver); -} -module_exit(sta32x_exit); +module_i2c_driver(sta32x_i2c_driver); MODULE_DESCRIPTION("ASoC STA32X driver"); MODULE_AUTHOR("Johannes Stezenbach "); -- cgit v1.2.3 From 63a47a7544c65f0d4ca28f3ffa54468bc5f6cc6c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:44 +0530 Subject: ASoC: tlv320dac33: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 0dd41077ab79..d2e16c5d7d1f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1621,24 +1621,7 @@ static struct i2c_driver tlv320dac33_i2c_driver = { .id_table = tlv320dac33_i2c_id, }; -static int __init dac33_module_init(void) -{ - int r; - r = i2c_add_driver(&tlv320dac33_i2c_driver); - if (r < 0) { - printk(KERN_ERR "DAC33: driver registration failed\n"); - return r; - } - return 0; -} -module_init(dac33_module_init); - -static void __exit dac33_module_exit(void) -{ - i2c_del_driver(&tlv320dac33_i2c_driver); -} -module_exit(dac33_module_exit); - +module_i2c_driver(tlv320dac33_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); MODULE_AUTHOR("Peter Ujfalusi "); -- cgit v1.2.3 From beb22de07e87a2f6802cc0e916b2f5c6aeb3f59f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:45 +0530 Subject: ASoC: adau1701: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 3d50fc8646b6..51f2f3cd8136 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -527,17 +527,7 @@ static struct i2c_driver adau1701_i2c_driver = { .id_table = adau1701_i2c_id, }; -static int __init adau1701_init(void) -{ - return i2c_add_driver(&adau1701_i2c_driver); -} -module_init(adau1701_init); - -static void __exit adau1701_exit(void) -{ - i2c_del_driver(&adau1701_i2c_driver); -} -module_exit(adau1701_exit); +module_i2c_driver(adau1701_i2c_driver); MODULE_DESCRIPTION("ASoC ADAU1701 SigmaDSP driver"); MODULE_AUTHOR("Cliff Cai "); -- cgit v1.2.3 From 4abdc8c8fd25fa2f85d86babcbdb4fbbf759c86a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:46 +0530 Subject: ASoC: max9850: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max9850.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index a1913091f56c..efe535c37b39 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -369,17 +369,7 @@ static struct i2c_driver max9850_i2c_driver = { .id_table = max9850_i2c_id, }; -static int __init max9850_init(void) -{ - return i2c_add_driver(&max9850_i2c_driver); -} -module_init(max9850_init); - -static void __exit max9850_exit(void) -{ - i2c_del_driver(&max9850_i2c_driver); -} -module_exit(max9850_exit); +module_i2c_driver(max9850_i2c_driver); MODULE_AUTHOR("Christian Glindkamp "); MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); -- cgit v1.2.3 From 28285b96c9e75a79b7698bc4286aa1cb94e5c9cb Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:47 +0530 Subject: ASoC: wm8971: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index eef783f6b6d6..5ce647758443 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -721,23 +721,7 @@ static struct i2c_driver wm8971_i2c_driver = { .id_table = wm8971_i2c_id, }; -static int __init wm8971_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8971_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8971 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8971_modinit); - -static void __exit wm8971_exit(void) -{ - i2c_del_driver(&wm8971_i2c_driver); -} -module_exit(wm8971_exit); +module_i2c_driver(wm8971_i2c_driver); MODULE_DESCRIPTION("ASoC WM8971 driver"); MODULE_AUTHOR("Lab126"); -- cgit v1.2.3 From 0ecbbb4fe5f0aae4cd70ec02383fd6b96aedb052 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:48 +0530 Subject: ASoC: ak4671: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 5fb7c2a80e6d..2b457976a7bf 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -696,17 +696,7 @@ static struct i2c_driver ak4671_i2c_driver = { .id_table = ak4671_i2c_id, }; -static int __init ak4671_modinit(void) -{ - return i2c_add_driver(&ak4671_i2c_driver); -} -module_init(ak4671_modinit); - -static void __exit ak4671_exit(void) -{ - i2c_del_driver(&ak4671_i2c_driver); -} -module_exit(ak4671_exit); +module_i2c_driver(ak4671_i2c_driver); MODULE_DESCRIPTION("ASoC AK4671 codec driver"); MODULE_AUTHOR("Joonyoung Shim "); -- cgit v1.2.3 From f6ec139f2dc5380c542fa3100dbe1c73324be432 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:49 +0530 Subject: ASoC: lm4857: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index ba4fafb93e56..81a328c78838 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -250,17 +250,7 @@ static struct i2c_driver lm4857_i2c_driver = { .id_table = lm4857_i2c_id, }; -static int __init lm4857_init(void) -{ - return i2c_add_driver(&lm4857_i2c_driver); -} -module_init(lm4857_init); - -static void __exit lm4857_exit(void) -{ - i2c_del_driver(&lm4857_i2c_driver); -} -module_exit(lm4857_exit); +module_i2c_driver(lm4857_i2c_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("LM4857 amplifier driver"); -- cgit v1.2.3 From 0b34ac810ac079735f9e7e2c58d467f849a67ede Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:50 +0530 Subject: ASoC: wm8978: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a5be3adecf75..5421fd9fbcb5 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1105,23 +1105,7 @@ static struct i2c_driver wm8978_i2c_driver = { .id_table = wm8978_i2c_id, }; -static int __init wm8978_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8978_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8978 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8978_modinit); - -static void __exit wm8978_exit(void) -{ - i2c_del_driver(&wm8978_i2c_driver); -} -module_exit(wm8978_exit); +module_i2c_driver(wm8978_i2c_driver); MODULE_DESCRIPTION("ASoC WM8978 codec driver"); MODULE_AUTHOR("Guennadi Liakhovetski "); -- cgit v1.2.3 From 2342a07f2ca81c8e076ed6d5c6d19ac36794c848 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:51 +0530 Subject: ASoC: max98088: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index af7324b79dd0..3264a5169306 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2107,23 +2107,7 @@ static struct i2c_driver max98088_i2c_driver = { .id_table = max98088_i2c_id, }; -static int __init max98088_init(void) -{ - int ret; - - ret = i2c_add_driver(&max98088_i2c_driver); - if (ret) - pr_err("Failed to register max98088 I2C driver: %d\n", ret); - - return ret; -} -module_init(max98088_init); - -static void __exit max98088_exit(void) -{ - i2c_del_driver(&max98088_i2c_driver); -} -module_exit(max98088_exit); +module_i2c_driver(max98088_i2c_driver); MODULE_DESCRIPTION("ALSA SoC MAX98088 driver"); MODULE_AUTHOR("Peter Hsiang, Jesse Marroquin"); -- cgit v1.2.3 From 07c9c32be01db1705db2655922ee66173594b230 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:52 +0530 Subject: ASoC: wm8955: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 61fe97433e73..2f1c075755b1 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1071,23 +1071,7 @@ static struct i2c_driver wm8955_i2c_driver = { .id_table = wm8955_i2c_id, }; -static int __init wm8955_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8955_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8955_modinit); - -static void __exit wm8955_exit(void) -{ - i2c_del_driver(&wm8955_i2c_driver); -} -module_exit(wm8955_exit); +module_i2c_driver(wm8955_i2c_driver); MODULE_DESCRIPTION("ASoC WM8955 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3 From a9418ddca69db1c4b5d2d7b5f091e50893387be3 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:53 +0530 Subject: ASoC: wm2200: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 32682c1b7cde..71debd0a3822 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2270,17 +2270,7 @@ static struct i2c_driver wm2200_i2c_driver = { .id_table = wm2200_i2c_id, }; -static int __init wm2200_modinit(void) -{ - return i2c_add_driver(&wm2200_i2c_driver); -} -module_init(wm2200_modinit); - -static void __exit wm2200_exit(void) -{ - i2c_del_driver(&wm2200_i2c_driver); -} -module_exit(wm2200_exit); +module_i2c_driver(wm2200_i2c_driver); MODULE_DESCRIPTION("ASoC WM2200 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3 From 3a4bfd88af87b065e8a650211a7730b3f1e58e3e Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:54 +0530 Subject: ASoC: wm2000: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 3fd5b29dc933..89cd6fcad015 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -858,17 +858,7 @@ static struct i2c_driver wm2000_i2c_driver = { .id_table = wm2000_i2c_id, }; -static int __init wm2000_init(void) -{ - return i2c_add_driver(&wm2000_i2c_driver); -} -module_init(wm2000_init); - -static void __exit wm2000_exit(void) -{ - i2c_del_driver(&wm2000_i2c_driver); -} -module_exit(wm2000_exit); +module_i2c_driver(wm2000_i2c_driver); MODULE_DESCRIPTION("ASoC WM2000 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3 From 794836b959e95b4e4705e11e7bdf2a688c72e2f7 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:55 +0530 Subject: ASoC: wm8940: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 481a3d9cfe48..b20aa4e7c3f9 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -785,23 +785,7 @@ static struct i2c_driver wm8940_i2c_driver = { .id_table = wm8940_i2c_id, }; -static int __init wm8940_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8940_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8940 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8940_modinit); - -static void __exit wm8940_exit(void) -{ - i2c_del_driver(&wm8940_i2c_driver); -} -module_exit(wm8940_exit); +module_i2c_driver(wm8940_i2c_driver); MODULE_DESCRIPTION("ASoC WM8940 driver"); MODULE_AUTHOR("Jonathan Cameron"); -- cgit v1.2.3 From 8b08eb28c761aea8434e0228a0f080e31e16d791 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:56 +0530 Subject: ASoC: wm8961: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 01edbcc754d2..719fb69a17c8 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1114,23 +1114,7 @@ static struct i2c_driver wm8961_i2c_driver = { .id_table = wm8961_i2c_id, }; -static int __init wm8961_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8961_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8961 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8961_modinit); - -static void __exit wm8961_exit(void) -{ - i2c_del_driver(&wm8961_i2c_driver); -} -module_exit(wm8961_exit); +module_i2c_driver(wm8961_i2c_driver); MODULE_DESCRIPTION("ASoC WM8961 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3 From 5c86ea44bba82641b6173419c88c0de7cf09b60a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:57 +0530 Subject: ASoC: wm8903: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 73f1c8d7bafb..839414f9e2ed 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2241,23 +2241,7 @@ static struct i2c_driver wm8903_i2c_driver = { .id_table = wm8903_i2c_id, }; -static int __init wm8903_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8903_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8903 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8903_modinit); - -static void __exit wm8903_exit(void) -{ - i2c_del_driver(&wm8903_i2c_driver); -} -module_exit(wm8903_exit); +module_i2c_driver(wm8903_i2c_driver); MODULE_DESCRIPTION("ASoC WM8903 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3 From d0b2d4fabb262dd7200382ee834d4292f6d76b1e Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:58 +0530 Subject: ASoC: adau1373: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 44f59064d8de..704544bfc90d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1392,17 +1392,7 @@ static struct i2c_driver adau1373_i2c_driver = { .id_table = adau1373_i2c_id, }; -static int __init adau1373_init(void) -{ - return i2c_add_driver(&adau1373_i2c_driver); -} -module_init(adau1373_init); - -static void __exit adau1373_exit(void) -{ - i2c_del_driver(&adau1373_i2c_driver); -} -module_exit(adau1373_exit); +module_i2c_driver(adau1373_i2c_driver); MODULE_DESCRIPTION("ASoC ADAU1373 driver"); MODULE_AUTHOR("Lars-Peter Clausen "); -- cgit v1.2.3 From 3c010e60ee54ee19dc2f39b4efa43dea03d65aaa Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:36 +0530 Subject: ASoC: wm8960: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 96518ac8e24c..804f4116912f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1010,23 +1010,7 @@ static struct i2c_driver wm8960_i2c_driver = { .id_table = wm8960_i2c_id, }; -static int __init wm8960_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8960_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8960_modinit); - -static void __exit wm8960_exit(void) -{ - i2c_del_driver(&wm8960_i2c_driver); -} -module_exit(wm8960_exit); +module_i2c_driver(wm8960_i2c_driver); MODULE_DESCRIPTION("ASoC WM8960 driver"); MODULE_AUTHOR("Liam Girdwood"); -- cgit v1.2.3 From 2be59418f76dac590b98027586ac1714be17fcae Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:59 +0530 Subject: ASoC: wm8974: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d93c03f820c9..9a39511af52a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -659,23 +659,7 @@ static struct i2c_driver wm8974_i2c_driver = { .id_table = wm8974_i2c_id, }; -static int __init wm8974_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8974_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8974 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8974_modinit); - -static void __exit wm8974_exit(void) -{ - i2c_del_driver(&wm8974_i2c_driver); -} -module_exit(wm8974_exit); +module_i2c_driver(wm8974_i2c_driver); MODULE_DESCRIPTION("ASoC WM8974 driver"); MODULE_AUTHOR("Liam Girdwood"); -- cgit v1.2.3 From a8af02cf62e32644c02566adc462bba9ae148154 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:26:00 +0530 Subject: ASoC: max98095: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 7cd508e16a5c..38d43c59d3f4 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2533,23 +2533,7 @@ static struct i2c_driver max98095_i2c_driver = { .id_table = max98095_i2c_id, }; -static int __init max98095_init(void) -{ - int ret; - - ret = i2c_add_driver(&max98095_i2c_driver); - if (ret) - pr_err("Failed to register max98095 I2C driver: %d\n", ret); - - return ret; -} -module_init(max98095_init); - -static void __exit max98095_exit(void) -{ - i2c_del_driver(&max98095_i2c_driver); -} -module_exit(max98095_exit); +module_i2c_driver(max98095_i2c_driver); MODULE_DESCRIPTION("ALSA SoC MAX98095 driver"); MODULE_AUTHOR("Peter Hsiang"); -- cgit v1.2.3 From cee4fcfa9dba7f13b0b45810832208df48ff4ca4 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:37 +0530 Subject: ASoC: cs42l51: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 091d0193f507..1e0fa3b5f79a 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -614,24 +614,7 @@ static struct i2c_driver cs42l51_i2c_driver = { .remove = cs42l51_i2c_remove, }; -static int __init cs42l51_init(void) -{ - int ret; - - ret = i2c_add_driver(&cs42l51_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "%s: can't add i2c driver\n", __func__); - return ret; - } - return 0; -} -module_init(cs42l51_init); - -static void __exit cs42l51_exit(void) -{ - i2c_del_driver(&cs42l51_i2c_driver); -} -module_exit(cs42l51_exit); +module_i2c_driver(cs42l51_i2c_driver); MODULE_AUTHOR("Arnaud Patard "); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); -- cgit v1.2.3 From f062e2b64153e9769adf5370103f787971c9cd95 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 6 Aug 2012 17:25:38 +0530 Subject: ASoC: tpa6130a2: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tpa6130a2.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 6fe4aa3ac544..565ff39ad3a3 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -487,19 +487,8 @@ static struct i2c_driver tpa6130a2_i2c_driver = { .id_table = tpa6130a2_id, }; -static int __init tpa6130a2_init(void) -{ - return i2c_add_driver(&tpa6130a2_i2c_driver); -} - -static void __exit tpa6130a2_exit(void) -{ - i2c_del_driver(&tpa6130a2_i2c_driver); -} +module_i2c_driver(tpa6130a2_i2c_driver); MODULE_AUTHOR("Peter Ujfalusi "); MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); MODULE_LICENSE("GPL"); - -module_init(tpa6130a2_init); -module_exit(tpa6130a2_exit); -- cgit v1.2.3 From 209ffe19ff98f5c0133bd98a689fc4fb42202de3 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 7 Aug 2012 12:07:27 +0530 Subject: ASoC: cs42l52: Remove duplicate inclusion of slab.h header file slab.h header file was included twice. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 628daf6a1d97..61599298fb26 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.3 From 5f800080ca6840326b3048202fb63fc4a71a4c49 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 7 Aug 2012 10:24:13 +0300 Subject: ASoC: core: Set dapm->idle_bias_off for DAIs not mapped with a codec The idle_bias_off flag is not configured for DAIs not mapped with a codec. This causes the pm counter to be increased at probe time for the CPU dai which unbalances the pm counter handling. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c5976b961..f10f00b5d29f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3715,6 +3715,9 @@ int snd_soc_register_dai(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); @@ -3803,6 +3806,9 @@ int snd_soc_register_dais(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); -- cgit v1.2.3 From 730963f8190b7650b0445a76e701fdef20c31cfb Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 7 Aug 2012 01:29:43 -0300 Subject: ASoC: mxs-saif: Use devm_clk_get() Using devm_clk_get can make the code simpler and smaller. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b3030718c228..aa037b292f3d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -704,7 +704,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->clk = clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -717,8 +717,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_get_resource; + return -ENODEV; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -731,7 +730,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) &saif->dma_param.chan_num); if (ret) { dev_err(&pdev->dev, "failed to get dma channel\n"); - goto failed_get_resource; + return ret; } } else { saif->dma_param.chan_num = dmares->start; @@ -742,7 +741,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } saif->dev = &pdev->dev; @@ -750,7 +749,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_resource; + return ret; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -758,7 +757,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } platform_set_drvdata(pdev, saif); @@ -766,7 +765,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_get_resource; + return ret; } ret = mxs_pcm_platform_register(&pdev->dev); @@ -779,19 +778,14 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_get_resource: - clk_put(saif->clk); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct mxs_saif *saif = platform_get_drvdata(pdev); - mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_put(saif->clk); return 0; } -- cgit v1.2.3 From 8f245499791a4701bfe1ce9b0df90cea9d2f13e5 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Wed, 8 Aug 2012 20:40:30 +0530 Subject: ASoC: Davinci: McASP: remove unused header include Defines or parameters from isn't used anywhere. Hence remove the header include. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 10a2d8c788b7..c80c20a89b11 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -24,7 +24,6 @@ #include #include -#include #include "davinci-pcm.h" #include "davinci-i2s.h" -- cgit v1.2.3 From 10884347f18842aa9b8ae18ebb16272d9b7fafa2 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Wed, 8 Aug 2012 20:40:32 +0530 Subject: ASoC: McASP: Convert driver to use Runtime PM API * Add Runtime PM support to McASP host controller. * Use Runtime PM API to enable/disable McASP clock. This was tested on AM18x Board using suspend/resume Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 40 +++++++++++++++++++-------------------- sound/soc/davinci/davinci-mcasp.h | 3 +-- 2 files changed, 20 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 95441bfc8190..d919fb8de7a3 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -21,7 +21,7 @@ #include #include #include -#include +#include #include #include @@ -776,20 +776,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!dev->clk_active) { - clk_enable(dev->clk); - dev->clk_active = 1; - } + ret = pm_runtime_get_sync(dev->dev); + if (IS_ERR_VALUE(ret)) + dev_err(dev->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(dev, substream->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: davinci_mcasp_stop(dev, substream->stream); - if (dev->clk_active) { - clk_disable(dev->clk); - dev->clk_active = 0; - } - + ret = pm_runtime_put_sync(dev->dev); + if (IS_ERR_VALUE(ret)) + dev_err(dev->dev, "pm_runtime_put_sync() failed\n"); break; case SNDRV_PCM_TRIGGER_STOP: @@ -886,12 +883,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } pdata = pdev->dev.platform_data; - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return -ENODEV; + pm_runtime_enable(&pdev->dev); - clk_enable(dev->clk); - dev->clk_active = 1; + ret = pm_runtime_get_sync(&pdev->dev); + if (IS_ERR_VALUE(ret)) { + dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); + return ret; + } dev->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!dev->base) { @@ -908,6 +906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->version = pdata->version; dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; + dev->dev = &pdev->dev; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; @@ -949,19 +948,18 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err_release_clk: - clk_disable(dev->clk); - clk_put(dev->clk); + pm_runtime_put_sync(&pdev->dev); + pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - struct davinci_audio_dev *dev = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_disable(dev->clk); - clk_put(dev->clk); - dev->clk = NULL; + + pm_runtime_put_sync(&pdev->dev); + pm_runtime_disable(&pdev->dev); return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 4681acc63606..51479f9ee909 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -40,9 +40,8 @@ struct davinci_audio_dev { struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; - struct clk *clk; + struct device *dev; unsigned int codec_fmt; - u8 clk_active; /* McASP specific data */ int tdm_slots; -- cgit v1.2.3 From 4918cdab497d693f4de288a576fb22e8ff9df21e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 12:33:28 +0200 Subject: ALSA: hda - Load firmware in hda_intel.c This is a preliminary work for the deferred probing for request_firmware() errors at init. This patch moves the call of request_firmware() to hda_intel.c, and call it in the earlier stage of probing rather than azx_probe_continue(). Tested-by: Thierry Reding Reviewed-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_hwdep.c | 39 +++++++++++++-------------------------- sound/pci/hda/hda_intel.c | 29 ++++++++++++++++++++++++----- 3 files changed, 38 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d330ca54..39e43753ddbf 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1072,7 +1072,7 @@ static inline void snd_hda_power_down(struct hda_codec *codec) {} /* * patch firmware */ -int snd_hda_load_patch(struct hda_bus *bus, const char *patch); +int snd_hda_load_patch(struct hda_bus *bus, size_t size, const void *buf); #endif /* diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 6b2efb8cb1f9..b9a644ca03b3 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include #include "hda_codec.h" @@ -747,18 +746,21 @@ static int parse_line_mode(char *buf, struct hda_bus *bus) * * the spaces at the beginning and the end of the line are stripped */ -static int get_line_from_fw(char *buf, int size, struct firmware *fw) +static int get_line_from_fw(char *buf, int size, size_t *fw_size_p, + const void **fw_data_p) { int len; - const char *p = fw->data; - while (isspace(*p) && fw->size) { + size_t fw_size = *fw_size_p; + const char *p = *fw_data_p; + + while (isspace(*p) && fw_size) { p++; - fw->size--; + fw_size--; } - if (!fw->size) + if (!fw_size) return 0; - for (len = 0; len < fw->size; len++) { + for (len = 0; len < fw_size; len++) { if (!*p) break; if (*p == '\n') { @@ -770,8 +772,8 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw) *buf++ = *p++; } *buf = 0; - fw->size -= len; - fw->data = p; + *fw_size_p = fw_size - len; + *fw_data_p = p; remove_trail_spaces(buf); return 1; } @@ -779,29 +781,15 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw) /* * load a "patch" firmware file and parse it */ -int snd_hda_load_patch(struct hda_bus *bus, const char *patch) +int snd_hda_load_patch(struct hda_bus *bus, size_t fw_size, const void *fw_buf) { - int err; - const struct firmware *fw; - struct firmware tmp; char buf[128]; struct hda_codec *codec; int line_mode; - struct device *dev = bus->card->dev; - - if (snd_BUG_ON(!dev)) - return -ENODEV; - err = request_firmware(&fw, patch, dev); - if (err < 0) { - printk(KERN_ERR "hda-codec: Cannot load the patch '%s'\n", - patch); - return err; - } - tmp = *fw; line_mode = LINE_MODE_NONE; codec = NULL; - while (get_line_from_fw(buf, sizeof(buf) - 1, &tmp)) { + while (get_line_from_fw(buf, sizeof(buf) - 1, &fw_size, &fw_buf)) { if (!*buf || *buf == '#' || *buf == '\n') continue; if (*buf == '[') @@ -810,7 +798,6 @@ int snd_hda_load_patch(struct hda_bus *bus, const char *patch) (codec || !patch_items[line_mode].need_codec)) patch_items[line_mode].parser(buf, bus, &codec); } - release_firmware(fw); return 0; } EXPORT_SYMBOL_HDA(snd_hda_load_patch); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced182fd1..b25d5392c3b2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,6 +55,7 @@ #include #include #include +#include #include "hda_codec.h" @@ -470,6 +471,10 @@ struct azx { struct snd_dma_buffer rb; struct snd_dma_buffer posbuf; +#ifdef CONFIG_SND_HDA_PATCH_LOADER + const struct firmware *fw; +#endif + /* flags */ int position_fix[2]; /* for both playback/capture streams */ int poll_count; @@ -2639,6 +2644,10 @@ static int azx_free(struct azx *chip) pci_release_regions(chip->pci); pci_disable_device(chip->pci); kfree(chip->azx_dev); +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (chip->fw) + release_firmware(chip->fw); +#endif kfree(chip); return 0; @@ -3167,7 +3176,6 @@ static int __devinit azx_probe(struct pci_dev *pci, return err; } - /* set this here since it's referred in snd_hda_load_patch() */ snd_card_set_dev(card, &pci->dev); err = azx_create(card, pci, dev, pci_id->driver_data, &chip); @@ -3175,6 +3183,16 @@ static int __devinit azx_probe(struct pci_dev *pci, goto out_free; card->private_data = chip; +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (patch[dev] && *patch[dev]) { + snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", + patch[dev]); + err = request_firmware(&chip->fw, patch[dev], &pci->dev); + if (err < 0) + goto out_free; + } +#endif /* CONFIG_SND_HDA_PATCH_LOADER */ + if (!chip->disabled) { err = azx_probe_continue(chip); if (err < 0) @@ -3205,12 +3223,13 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) if (err < 0) goto out_free; #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (patch[dev] && *patch[dev]) { - snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", - patch[dev]); - err = snd_hda_load_patch(chip->bus, patch[dev]); + if (chip->fw) { + err = snd_hda_load_patch(chip->bus, chip->fw->size, + chip->fw->data); if (err < 0) goto out_free; + release_firmware(chip->fw); /* no longer needed */ + chip->fw = NULL; } #endif if ((probe_only[dev] & 1) == 0) { -- cgit v1.2.3 From 5cb543dba9867588786f87af2e64fca371b69283 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 13:49:23 +0200 Subject: ALSA: hda - Deferred probing with request_firmware_nowait() For processing the firmware handling properly for built-in kernels, implement an asynchronous firmware loading with request_firmware_nowait(). This means that the codec probing is deferred when the patch option is specified. Tested-by: Thierry Reding Reviewed-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 42 ++++++++++++++++++++++++++++++++++++++---- 2 files changed, 39 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20eb7a2..408babc10437 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4184,7 +4184,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) * * This function returns 0 if successful, or a negative error code. */ -int __devinit snd_hda_build_pcms(struct hda_bus *bus) +int snd_hda_build_pcms(struct hda_bus *bus) { struct hda_codec *codec; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b25d5392c3b2..3de3a5cbf1dc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -564,13 +564,17 @@ enum { * VGA-switcher support */ #ifdef SUPPORT_VGA_SWITCHEROO +#define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) +#else +#define use_vga_switcheroo(chip) 0 +#endif + +#if defined(SUPPORT_VGA_SWITCHEROO) || defined(CONFIG_SND_HDA_PATCH_LOADER) #define DELAYED_INIT_MARK #define DELAYED_INITDATA_MARK -#define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) #else #define DELAYED_INIT_MARK __devinit #define DELAYED_INITDATA_MARK __devinitdata -#define use_vga_switcheroo(chip) 0 #endif static char *driver_short_names[] DELAYED_INITDATA_MARK = { @@ -3155,12 +3159,38 @@ static void power_down_all_codecs(struct azx *chip) #endif } +/* callback from request_firmware_nowait() */ +static void azx_firmware_cb(const struct firmware *fw, void *context) +{ + struct snd_card *card = context; + struct azx *chip = card->private_data; + struct pci_dev *pci = chip->pci; + + if (!fw) { + snd_printk(KERN_ERR SFX "Cannot load firmware, aborting\n"); + goto error; + } + + chip->fw = fw; + if (!chip->disabled) { + /* continue probing */ + if (azx_probe_continue(chip)) + goto error; + } + return; /* OK */ + + error: + snd_card_free(card); + pci_set_drvdata(pci, NULL); +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { static int dev; struct snd_card *card; struct azx *chip; + bool probe_now; int err; if (dev >= SNDRV_CARDS) @@ -3182,18 +3212,22 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; card->private_data = chip; + probe_now = !chip->disabled; #ifdef CONFIG_SND_HDA_PATCH_LOADER if (patch[dev] && *patch[dev]) { snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", patch[dev]); - err = request_firmware(&chip->fw, patch[dev], &pci->dev); + err = request_firmware_nowait(THIS_MODULE, true, patch[dev], + &pci->dev, GFP_KERNEL, card, + azx_firmware_cb); if (err < 0) goto out_free; + probe_now = false; /* continued in azx_firmware_cb() */ } #endif /* CONFIG_SND_HDA_PATCH_LOADER */ - if (!chip->disabled) { + if (probe_now) { err = azx_probe_continue(chip); if (err < 0) goto out_free; -- cgit v1.2.3 From 97c6a3d17b978092d0df94d5d68dbfc7c74f0902 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Aug 2012 17:40:46 +0200 Subject: ALSA: hda - Fix forgotten ifdef CONFIG_SND_HDA_PATCH_LOADER The firmware callback must be protected by that ifdef. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3de3a5cbf1dc..23ad7e216a6b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3159,6 +3159,7 @@ static void power_down_all_codecs(struct azx *chip) #endif } +#ifdef CONFIG_SND_HDA_PATCH_LOADER /* callback from request_firmware_nowait() */ static void azx_firmware_cb(const struct firmware *fw, void *context) { @@ -3183,6 +3184,7 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) snd_card_free(card); pci_set_drvdata(pci, NULL); } +#endif static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) -- cgit v1.2.3 From 28d528c8dbab5c6b3cf2fc4f0e91470a9a63dbc0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 9 Aug 2012 18:45:23 +0100 Subject: ASoC: core: Remove pointless error on card registration failure If we fail to register the card we should say why somewhere else so there's no point in repeating the same thing with less information. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f10f00b5d29f..3a43fa6ba2ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1830,13 +1830,7 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - ret = snd_soc_register_card(card); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to register card\n"); - return ret; - } - - return 0; + return snd_soc_register_card(card); } static int soc_cleanup_card_resources(struct snd_soc_card *card) -- cgit v1.2.3 From cbd840dadeb03826b6cc074e38f380bbd4faaea5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Aug 2012 17:52:44 +0100 Subject: ASoC: arizona: Implement OPCLK support Arizona devices support two output system clocks. Provide support for configuring these via set_sysclk(). Once the clock API is more useful we should migrate over to that. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 66 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 +++-- 2 files changed, 70 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9cacaf2d52..5e96a0a1669c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -229,6 +229,69 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); +static unsigned int arizona_sysclk_48k_rates[] = { + 6144000, + 12288000, + 22579200, + 49152000, +}; + +static unsigned int arizona_sysclk_44k1_rates[] = { + 5644800, + 11289600, + 24576000, + 45158400, +}; + +static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, + unsigned int freq) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + unsigned int *rates; + int ref, div, refclk; + + switch (clk) { + case ARIZONA_CLK_OPCLK: + reg = ARIZONA_OUTPUT_SYSTEM_CLOCK; + refclk = priv->sysclk; + break; + case ARIZONA_CLK_ASYNC_OPCLK: + reg = ARIZONA_OUTPUT_ASYNC_CLOCK; + refclk = priv->asyncclk; + break; + default: + return -EINVAL; + } + + if (refclk % 8000) + rates = arizona_sysclk_44k1_rates; + else + rates = arizona_sysclk_48k_rates; + + for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + rates[ref] <= refclk; ref++) { + div = 1; + while (rates[ref] / div >= freq && div < 32) { + if (rates[ref] / div == freq) { + dev_dbg(codec->dev, "Configured %dHz OPCLK\n", + freq); + snd_soc_update_bits(codec, reg, + ARIZONA_OPCLK_DIV_MASK | + ARIZONA_OPCLK_SEL_MASK, + (div << + ARIZONA_OPCLK_DIV_SHIFT) | + ref); + return 0; + } + div++; + } + } + + dev_err(codec->dev, "Unable to generate %dHz OPCLK\n", freq); + return -EINVAL; +} + int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -252,6 +315,9 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, reg = ARIZONA_ASYNC_CLOCK_1; clk = &priv->asyncclk; break; + case ARIZONA_CLK_OPCLK: + case ARIZONA_CLK_ASYNC_OPCLK: + return arizona_set_opclk(codec, clk_id, freq); default: return -EINVAL; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 59caca8865e8..eb66b52777c9 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -17,8 +17,10 @@ #include -#define ARIZONA_CLK_SYSCLK 1 -#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_OPCLK 3 +#define ARIZONA_CLK_ASYNC_OPCLK 4 #define ARIZONA_CLK_SRC_MCLK1 0x0 #define ARIZONA_CLK_SRC_MCLK2 0x1 -- cgit v1.2.3 From c665d1a8c4c391a7918f53c8a02d909626266773 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Aug 2012 17:53:01 +0100 Subject: ASoC: wm5102: Enable output clocks Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6537f16d383e..4c2fc361051c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -285,6 +285,10 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -- cgit v1.2.3 From 94237f8e8ed5c2bfc5d8a28cdda241170eda6994 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 9 Aug 2012 19:14:43 +0100 Subject: ASoC: wm5110: Enable output clocks Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 8033f7065189..3db6e6e7a591 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -305,6 +305,10 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -- cgit v1.2.3 From b545dd924b4ffaf1e4fdd73fe7e9b6eb01e45aea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Aug 2012 20:03:02 +0100 Subject: ASoC: bells: Add machine driver for Wolfson Bells boards The Wolfson Bells board takes submodules for various audio functions but since the system integrations are virtually identical for most of them we can support the overwhemling majority from the same machine driver. Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/bells.c | 346 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 356 insertions(+) create mode 100644 sound/soc/samsung/bells.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fe3995ce9b38..fb5600083612 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -199,6 +199,14 @@ config SND_SOC_TOBERMORY select SND_SAMSUNG_I2S select SND_SOC_WM8962 +config SND_SOC_BELLS + tristate "Audio support for Wolfson Bells" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM5102 + select SND_SOC_WM5110 + select SND_SOC_WM9081 + config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 9d03beb40c86..709f6059ad67 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -42,6 +42,7 @@ snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o +snd-soc-bells-objs := bells.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -65,3 +66,4 @@ obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o +obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c new file mode 100644 index 000000000000..5dc10dfc0d42 --- /dev/null +++ b/sound/soc/samsung/bells.c @@ -0,0 +1,346 @@ +/* + * Bells audio support + * + * Copyright 2012 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include + +#include "../codecs/wm5102.h" +#include "../codecs/wm9081.h" + +/* + * 44.1kHz based clocks for the SYSCLK domain, use a very high clock + * to allow all the DSP functionality to be enabled if desired. + */ +#define SYSCLK_RATE (44100 * 1024) + +/* 48kHz based clocks for the ASYNC domain */ +#define ASYNCCLK_RATE (48000 * 512) + +/* BCLK2 is fixed at this currently */ +#define BCLK2_RATE (64 * 8000) + +/* + * Expect a 24.576MHz crystal if one is fitted (the driver will function + * if this is not fitted). + */ +#define MCLK_RATE 24576000 + +#define WM9081_AUDIO_RATE 44100 +#define WM9081_MCLK_RATE (WM9081_AUDIO_RATE * 256) + +static int bells_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + SYSCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, + ARIZONA_FLL_SRC_AIF2BCLK, + BCLK2_RATE, + ASYNCCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + } + break; + + default: + break; + } + + return 0; +} + +static int bells_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int bells_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_dai *aif3_dai = card->rtd[2].cpu_dai; + struct snd_soc_dai *wm9081_dai = card->rtd[2].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); + if (ret != 0) { + dev_err(aif2_dai->dev, "Failed to set AIF2 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, + ARIZONA_CLK_SRC_FLL1, SYSCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_OPCLK, 0, + WM9081_MCLK_RATE, SND_SOC_CLOCK_OUT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set OPCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK, + ARIZONA_CLK_SRC_FLL2, ASYNCCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(wm9081_dai->codec, WM9081_SYSCLK_MCLK, + 0, WM9081_MCLK_RATE, 0); + if (ret != 0) { + dev_err(wm9081_dai->dev, "Failed to set MCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = WM9081_AUDIO_RATE, + .rate_max = WM9081_AUDIO_RATE, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link bells_dai_wm5102[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5102-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5102-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5102-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_dai_link bells_dai_wm5110[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5110-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5110-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5110-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_codec_conf bells_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_dapm_route bells_routes[] = { + { "Sub CLK_SYS", NULL, "OPCLK" }, +}; + +static struct snd_soc_card bells_cards[] = { + { + .name = "Bells WM5102", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5102, + .num_links = ARRAY_SIZE(bells_dai_wm5102), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, + { + .name = "Bells WM5110", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5110, + .num_links = ARRAY_SIZE(bells_dai_wm5110), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, +}; + + +static __devinit int bells_probe(struct platform_device *pdev) +{ + int ret; + + bells_cards[pdev->id].dev = &pdev->dev; + + ret = snd_soc_register_card(&bells_cards[pdev->id]); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + bells_cards[pdev->id].name, ret); + return ret; + } + + return 0; +} + +static int __devexit bells_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bells_cards[pdev->id]); + + return 0; +} + +static struct platform_driver bells_driver = { + .driver = { + .name = "bells", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bells_probe, + .remove = __devexit_p(bells_remove), +}; + +module_platform_driver(bells_driver); + +MODULE_DESCRIPTION("Bells audio support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bells"); -- cgit v1.2.3 From 3876566a368b30908ec4195d58f04a411f22f736 Mon Sep 17 00:00:00 2001 From: Jerry Snitselaar Date: Thu, 9 Aug 2012 23:16:26 -0700 Subject: ASoC: core: remove unused variable in soc_probe() in linux-next With commit 28d528c8 "ASoC: core: Remove pointless error on card registration failure", the variable ret is no longer used in soc_probe() and generates an unused variable warning during a build. Signed-off-by: Jerry Snitselaar Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a43fa6ba2ee..f585d0239551 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1814,7 +1814,6 @@ base_error: static int soc_probe(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - int ret = 0; /* * no card, so machine driver should be registering card -- cgit v1.2.3 From 6152597971db4055b7b03ccc60afcd62214d41f4 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 13 Aug 2012 15:43:49 +0800 Subject: ALSA: hda - show ICT/KAE control bits Enable two debug options for S/PDIF Converter Control. KAE: Keep Alive Enable; ICT: IEC Coding Type. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 39e43753ddbf..d772c2521cd9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -386,6 +386,10 @@ enum { /* DIGITAL2 bits */ #define AC_DIG2_CC (0x7f<<0) +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<0) +#define AC_DIG3_KAE (1<<7) + /* Pin widget control - 8bit */ #define AC_PINCTL_EPT (0x3<<0) #define AC_PINCTL_EPT_NATIVE 0 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258fc700..32bb3d9bfc62 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -402,6 +402,9 @@ static void print_digital_conv(struct snd_info_buffer *buffer, { unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + unsigned char digi2 = digi1 >> 8; + unsigned char digi3 = digi1 >> 16; + snd_iprintf(buffer, " Digital:"); if (digi1 & AC_DIG1_ENABLE) snd_iprintf(buffer, " Enabled"); @@ -419,9 +422,13 @@ static void print_digital_conv(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Pro"); if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); + if (digi3 & AC_DIG3_KAE) + snd_iprintf(buffer, " KAE"); snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Digital category: 0x%x\n", - (digi1 >> 8) & AC_DIG2_CC); + digi2 & AC_DIG2_CC); + snd_iprintf(buffer, " IEC Coding Type: 0x%x\n", + digi3 & AC_DIG3_ICT); } static const char *get_pwr_state(u32 state) -- cgit v1.2.3 From 7ccbde57ce312ff1388c2990699f8863280808ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 18:10:09 +0200 Subject: ALSA: hda - Fix possible compile warnings regarding CONFIG_PM Replace with a proper ifdef check of CONFIG_PM_SLEEP in hda_intel.c. But other places in HD-audio driver are still marked with CONFIG_PM, since these can be called for power-saving even without CONFIG_PM_SLEEP. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 23ad7e216a6b..464ea16bdd9e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2409,11 +2409,10 @@ static void azx_power_notify(struct hda_bus *bus) } #endif /* CONFIG_SND_HDA_POWER_SAVE */ -#ifdef CONFIG_PM +#if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management */ - static int azx_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -2471,10 +2470,8 @@ static int azx_resume(struct device *dev) static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); #define AZX_PM_OPS &azx_pm #else -#define azx_suspend(dev) -#define azx_resume(dev) #define AZX_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ /* -- cgit v1.2.3 From c7561cd80469f2fe4a6be0984db57832ee7f2a3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 18:12:04 +0200 Subject: ALSA: PCI: Replace CONFIG_PM with CONFIG_PM_SLEEP Otherwise we may get compile warnings due to unused functions. Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 4 ++-- sound/pci/ali5451/ali5451.c | 10 +++++----- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 4 ++-- sound/pci/asihpi/asihpi.c | 2 +- sound/pci/atiixp.c | 6 +++--- sound/pci/atiixp_modem.c | 6 +++--- sound/pci/azt3328.c | 6 +++--- sound/pci/ca0106/ca0106.h | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- sound/pci/cmipci.c | 6 +++--- sound/pci/cs4281.c | 6 +++--- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs46xx/cs46xx.h | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 8 ++++---- sound/pci/cs46xx/cs46xx_lib.h | 2 +- sound/pci/cs46xx/dsp_spos.c | 8 ++++---- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/cs5535audio/Makefile | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/ctatc.c | 4 ++-- sound/pci/ctxfi/ctatc.h | 2 +- sound/pci/ctxfi/cthardware.h | 2 +- sound/pci/ctxfi/cthw20k1.c | 4 ++-- sound/pci/ctxfi/cthw20k2.c | 4 ++-- sound/pci/ctxfi/ctmixer.c | 4 ++-- sound/pci/ctxfi/ctmixer.h | 2 +- sound/pci/ctxfi/ctpcm.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/echoaudio.c | 12 ++++++------ sound/pci/echoaudio/echoaudio.h | 2 +- sound/pci/emu10k1/emu10k1.c | 4 ++-- sound/pci/emu10k1/emu10k1_main.c | 8 ++++---- sound/pci/emu10k1/emufx.c | 2 +- sound/pci/emu10k1/p16v.c | 2 +- sound/pci/ens1370.c | 4 ++-- sound/pci/es1938.c | 6 +++--- sound/pci/es1968.c | 12 ++++++------ sound/pci/fm801.c | 6 +++--- sound/pci/ice1712/aureon.c | 4 ++-- sound/pci/ice1712/ice1712.h | 2 +- sound/pci/ice1712/ice1724.c | 4 ++-- sound/pci/ice1712/juli.c | 4 ++-- sound/pci/ice1712/prodigy_hifi.c | 4 ++-- sound/pci/intel8x0.c | 4 ++-- sound/pci/intel8x0m.c | 4 ++-- sound/pci/maestro3.c | 10 +++++----- sound/pci/nm256/nm256.c | 4 ++-- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/oxygen.h | 2 +- sound/pci/oxygen/oxygen_lib.c | 4 ++-- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/sis7019.c | 6 +++--- sound/pci/trident/trident.c | 2 +- sound/pci/trident/trident_main.c | 4 ++-- sound/pci/via82xx.c | 10 +++++----- sound/pci/via82xx_modem.c | 4 ++-- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- sound/pci/ymfpci/ymfpci.h | 2 +- sound/pci/ymfpci/ymfpci_main.c | 8 ++++---- 63 files changed, 135 insertions(+), 135 deletions(-) (limited to 'sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 4f865df42f0f..1a33f48ebe78 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1788,7 +1788,7 @@ struct snd_emu10k1 { unsigned int efx_voices_mask[2]; unsigned int next_free_voice; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP unsigned int *saved_ptr; unsigned int *saved_gpr; unsigned int *tram_val_saved; @@ -1856,7 +1856,7 @@ unsigned short snd_emu10k1_ac97_read(struct snd_ac97 *ac97, unsigned short reg); void snd_emu10k1_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short data); unsigned int snd_emu10k1_rate_to_pitch(unsigned int rate); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu); void snd_emu10k1_resume_init(struct snd_emu10k1 *emu); void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index ee895f3c8605..c7e3c533316e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -270,7 +270,7 @@ struct snd_ali { spinlock_t reg_lock; spinlock_t voice_alloc; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP struct snd_ali_image *image; #endif }; @@ -1883,7 +1883,7 @@ static int __devinit snd_ali_mixer(struct snd_ali * codec) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int ali_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -1989,7 +1989,7 @@ static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume); #define ALI_PM_OPS &ali_pm #else #define ALI_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int snd_ali_free(struct snd_ali * codec) { @@ -2000,7 +2000,7 @@ static int snd_ali_free(struct snd_ali * codec) if (codec->port) pci_release_regions(codec->pci); pci_disable_device(codec->pci); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP kfree(codec->image); #endif pci_dev_put(codec->pci_m1533); @@ -2232,7 +2232,7 @@ static int __devinit snd_ali_create(struct snd_card *card, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP codec->image = kmalloc(sizeof(*codec->image), GFP_KERNEL); if (!codec->image) snd_printk(KERN_WARNING "can't allocate apm buffer\n"); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 68c4469c6d19..00f157a2cf64 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -765,7 +765,7 @@ static int __devinit snd_als300_create(struct snd_card *card, return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_als300_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 0eeca49c5754..feb2a1436830 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -987,7 +987,7 @@ static void __devexit snd_card_als4000_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_als4000_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -1040,7 +1040,7 @@ static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume #define SND_ALS4000_PM_OPS &snd_als4000_pm #else #define SND_ALS4000_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver als4000_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e8de831f98bc..a51e3ce3c800 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -2968,7 +2968,7 @@ static struct pci_driver driver = { .id_table = asihpi_pci_tbl, .probe = snd_asihpi_probe, .remove = __devexit_p(snd_asihpi_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* .suspend = snd_asihpi_suspend, .resume = snd_asihpi_resume, */ #endif diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 31020d2a868b..c744df5bb1c6 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -535,7 +535,7 @@ static int snd_atiixp_aclink_reset(struct atiixp *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_atiixp_aclink_down(struct atiixp *chip) { // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ @@ -1458,7 +1458,7 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp *chip, int clock, } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1533,7 +1533,7 @@ static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); #define SND_ATIIXP_PM_OPS &snd_atiixp_pm #else #define SND_ATIIXP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef CONFIG_PROC_FS diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 79e204ec623f..6fc03d9f2cff 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -511,7 +511,7 @@ static int snd_atiixp_aclink_reset(struct atiixp_modem *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_atiixp_aclink_down(struct atiixp_modem *chip) { // if (atiixp_read(chip, MODEM_MIRROR) & 0x1) /* modem running, too? */ @@ -1113,7 +1113,7 @@ static int __devinit snd_atiixp_mixer_new(struct atiixp_modem *chip, int clock) } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1169,7 +1169,7 @@ static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume); #define SND_ATIIXP_PM_OPS &snd_atiixp_pm #else #define SND_ATIIXP_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef CONFIG_PROC_FS /* diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 4dddd871548b..c03b66b784a3 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -365,7 +365,7 @@ struct snd_azf3328 { * CONFIG_PM register storage below, but that's slightly difficult. */ u16 shadow_reg_ctrl_6AH; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* register value containers for power management * Note: not always full I/O range preserved (similar to Win driver!) */ u32 saved_regs_ctrl[AZF_ALIGN(AZF_IO_SIZE_CTRL_PM) / 4]; @@ -2729,7 +2729,7 @@ snd_azf3328_remove(struct pci_dev *pci) snd_azf3328_dbgcallleave(); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static inline void snd_azf3328_suspend_regs(unsigned long io_addr, unsigned count, u32 *saved_regs) { @@ -2866,7 +2866,7 @@ static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume #define SND_AZF3328_PM_OPS &snd_azf3328_pm #else #define SND_AZF3328_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver azf3328_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index e8e8ccc96403..04402c14cb23 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -710,7 +710,7 @@ struct snd_ca0106 { u16 spi_dac_reg[16]; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP #define NUM_SAVED_VOLUMES 9 unsigned int saved_vol[NUM_SAVED_VOLUMES]; #endif @@ -733,7 +733,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); int snd_ca0106_spi_write(struct snd_ca0106 * emu, unsigned int data); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); #else diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 83277b747b36..fc6787699ba9 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1871,7 +1871,7 @@ static void __devexit snd_ca0106_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_ca0106_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 84f3f92436b5..68eacf7002d6 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -907,7 +907,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP struct ca0106_vol_tbl { unsigned int channel_id; unsigned int reg; @@ -953,4 +953,4 @@ void snd_ca0106_mixer_resume(struct snd_ca0106 *chip) if (chip->details->i2c_adc) ca0106_set_capture_mic_line_in(chip); } -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index b7d6f2b886ef..7c25906b5f88 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -504,7 +504,7 @@ struct cmipci { spinlock_t reg_lock; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP unsigned int saved_regs[0x20]; unsigned char saved_mixers[0x20]; #endif @@ -3315,7 +3315,7 @@ static void __devexit snd_cmipci_remove(struct pci_dev *pci) } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -3403,7 +3403,7 @@ static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume); #define SND_CMIPCI_PM_OPS &snd_cmipci_pm #else #define SND_CMIPCI_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver cmipci_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 45a8317085f4..8e86ec0031fc 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -486,7 +486,7 @@ struct cs4281 { struct gameport *gameport; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 suspend_regs[SUSPEND_REGISTERS]; #endif @@ -1977,7 +1977,7 @@ static void __devexit snd_cs4281_remove(struct pci_dev *pci) /* * Power Management */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int saved_regs[SUSPEND_REGISTERS] = { BA0_JSCTL, @@ -2089,7 +2089,7 @@ static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume); #define CS4281_PM_OPS &cs4281_pm #else #define CS4281_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver cs4281_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 1e007c736a8b..575bed0836ff 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -166,7 +166,7 @@ static struct pci_driver cs46xx_driver = { .id_table = snd_cs46xx_ids, .probe = snd_card_cs46xx_probe, .remove = __devexit_p(snd_card_cs46xx_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &snd_cs46xx_pm, }, diff --git a/sound/pci/cs46xx/cs46xx.h b/sound/pci/cs46xx/cs46xx.h index 29d8a8da1ba7..fc339ef0a0ae 100644 --- a/sound/pci/cs46xx/cs46xx.h +++ b/sound/pci/cs46xx/cs46xx.h @@ -1721,7 +1721,7 @@ struct snd_cs46xx { unsigned int play_ctl; #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 *saved_regs; #endif }; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ffdfdfb..6b111d0929b1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2797,7 +2797,7 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP kfree(chip->saved_regs); #endif @@ -3590,7 +3590,7 @@ static struct cs_card_type __devinitdata cards[] = { /* * APM support */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static unsigned int saved_regs[] = { BA0_ACOSV, /*BA0_ASER_FADDR,*/ @@ -3711,7 +3711,7 @@ static int snd_cs46xx_resume(struct device *dev) } SIMPLE_DEV_PM_OPS(snd_cs46xx_pm, snd_cs46xx_suspend, snd_cs46xx_resume); -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ /* @@ -3868,7 +3868,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card, snd_cs46xx_proc_init(card, chip); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * ARRAY_SIZE(saved_regs), GFP_KERNEL); if (!chip->saved_regs) { diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index b5189495d58a..86f14620f817 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -90,7 +90,7 @@ static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip); void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip); int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int cs46xx_dsp_resume(struct snd_cs46xx * chip); #endif struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 56fec0bc0efb..1686b4f4c44f 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -287,7 +287,7 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP kfree(ins->scbs[i].data); #endif } @@ -1019,7 +1019,7 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 { struct dsp_scb_descriptor * desc; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* copy the data for resume */ scb_data = kmemdup(scb_data, SCB_BYTES, GFP_KERNEL); if (!scb_data) @@ -1032,7 +1032,7 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP kfree(scb_data); #endif } @@ -1937,7 +1937,7 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int cs46xx_dsp_resume(struct snd_cs46xx * chip) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index c2c695b07f8c..409e8764fbeb 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -203,7 +203,7 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * remove_symbol (chip,scb->scb_symbol); ins->scbs[scb->index].deleted = 1; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP kfree(ins->scbs[scb->index].data); ins->scbs[scb->index].data = NULL; #endif diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile index ccc642269b9e..a8f75f8dfda9 100644 --- a/sound/pci/cs5535audio/Makefile +++ b/sound/pci/cs5535audio/Makefile @@ -3,7 +3,7 @@ # snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o -snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o +snd-cs5535audio-$(CONFIG_PM_SLEEP) += cs5535audio_pm.o snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.o # Toplevel Module Dependency diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 51f64ba5facf..4915efa551fc 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -399,7 +399,7 @@ static struct pci_driver cs5535audio_driver = { .id_table = snd_cs5535audio_ids, .probe = snd_cs5535audio_probe, .remove = __devexit_p(snd_cs5535audio_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &snd_cs5535audio_pm, }, diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 8e40262d4117..58b235c46e86 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1536,7 +1536,7 @@ static void atc_connect_resources(struct ct_atc *atc) } } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int atc_suspend(struct ct_atc *atc) { int i; @@ -1647,7 +1647,7 @@ static struct ct_atc atc_preset __devinitdata = { .output_switch_put = atc_output_switch_put, .mic_source_switch_get = atc_mic_source_switch_get, .mic_source_switch_put = atc_mic_source_switch_put, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .suspend = atc_suspend, .resume = atc_resume, #endif diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 653e813ad142..69b51f9d345e 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -143,7 +143,7 @@ struct ct_atc { struct ct_timer *timer; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int (*suspend)(struct ct_atc *atc); int (*resume)(struct ct_atc *atc); #define NUM_PCMS (NUM_CTALSADEVS - 1) diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index c56fe533b3f3..5977e9a24b5c 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -72,7 +72,7 @@ struct hw { int (*card_init)(struct hw *hw, struct card_conf *info); int (*card_stop)(struct hw *hw); int (*pll_init)(struct hw *hw, unsigned int rsr); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int (*suspend)(struct hw *hw); int (*resume)(struct hw *hw, struct card_conf *info); #endif diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index dc1969bc67d4..4507f7088b24 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -2085,7 +2085,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2180,7 +2180,7 @@ static struct hw ct20k1_preset __devinitdata = { .is_adc_source_selected = hw_is_adc_input_selected, .select_adc_source = hw_adc_input_select, .capabilities = hw_capabilities, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .suspend = hw_suspend, .resume = hw_resume, #endif diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 9d1231dc4ae2..b9c9349058bc 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -2201,7 +2201,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int hw_suspend(struct hw *hw) { struct pci_dev *pci = hw->pci; @@ -2250,7 +2250,7 @@ static struct hw ct20k2_preset __devinitdata = { .output_switch_put = hw_output_switch_put, .mic_source_switch_get = hw_mic_source_switch_get, .mic_source_switch_put = hw_mic_source_switch_put, -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .suspend = hw_suspend, .resume = hw_resume, #endif diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 0cc13eeef8da..48fe0e39c2be 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -1118,7 +1118,7 @@ mixer_set_input_right(struct ct_mixer *mixer, return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int mixer_resume(struct ct_mixer *mixer) { int i, state; @@ -1188,7 +1188,7 @@ int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer) mixer->get_output_ports = mixer_get_output_ports; mixer->set_input_left = mixer_set_input_left; mixer->set_input_right = mixer_set_input_right; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP mixer->resume = mixer_resume; #endif diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h index b009e989e77d..be881c639fee 100644 --- a/sound/pci/ctxfi/ctmixer.h +++ b/sound/pci/ctxfi/ctmixer.h @@ -56,7 +56,7 @@ struct ct_mixer { enum MIXER_PORT_T type, struct rsc *rsc); int (*set_input_right)(struct ct_mixer *mixer, enum MIXER_PORT_T type, struct rsc *rsc); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int (*resume)(struct ct_mixer *mixer); #endif }; diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 2c8622617c8c..d021876901bb 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -427,7 +427,7 @@ int ct_alsa_pcm_create(struct ct_atc *atc, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(atc->pci), 128*1024, 128*1024); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP atc->pcms[device] = pcm; #endif diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index e002183ef8b2..07c07d752fd8 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -125,7 +125,7 @@ static void __devexit ct_card_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int ct_card_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0ff754f180d0..abb0b86c41c9 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -46,7 +46,7 @@ static int get_firmware(const struct firmware **fw_entry, int err; char name[30]; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP if (chip->fw_cache[fw_index]) { DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); *fw_entry = chip->fw_cache[fw_index]; @@ -59,7 +59,7 @@ static int get_firmware(const struct firmware **fw_entry, err = request_firmware(fw_entry, name, pci_device(chip)); if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP else chip->fw_cache[fw_index] = *fw_entry; #endif @@ -70,7 +70,7 @@ static int get_firmware(const struct firmware **fw_entry, static void free_firmware(const struct firmware *fw_entry) { -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); #else release_firmware(fw_entry); @@ -82,7 +82,7 @@ static void free_firmware(const struct firmware *fw_entry) static void free_firmware_cache(struct echoaudio *chip) { -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int i; for (i = 0; i < 8 ; i++) @@ -2203,7 +2203,7 @@ ctl_error: -#if defined(CONFIG_PM) +#if defined(CONFIG_PM_SLEEP) static int snd_echo_suspend(struct device *dev) { @@ -2313,7 +2313,7 @@ static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume); #define SND_ECHO_PM_OPS &snd_echo_pm #else #define SND_ECHO_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static void __devexit snd_echo_remove(struct pci_dev *pci) diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 1df974dcb5f4..e158369f5faa 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -449,7 +449,7 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP const struct firmware *fw_cache[8]; /* Cached firmwares */ #endif diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index ddac4e6d660d..b7c1875ba90e 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -206,7 +206,7 @@ static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci) } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_emu10k1_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -268,7 +268,7 @@ static SIMPLE_DEV_PM_OPS(snd_emu10k1_pm, snd_emu10k1_suspend, snd_emu10k1_resume #define SND_EMU10K1_PM_OPS &snd_emu10k1_pm #else #define SND_EMU10K1_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver emu10k1_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 754924081d0a..bed4485f34f6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1241,7 +1241,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) * Create the EMU10K1 instance */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int alloc_pm_buffer(struct snd_emu10k1 *emu); static void free_pm_buffer(struct snd_emu10k1 *emu); #endif @@ -1275,7 +1275,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) snd_dma_free_pages(&emu->ptb_pages); vfree(emu->page_ptr_table); vfree(emu->page_addr_table); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP free_pm_buffer(emu); #endif if (emu->port) @@ -1971,7 +1971,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, err = snd_emu10k1_init(emu, enable_ir, 0); if (err < 0) goto error; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP err = alloc_pm_buffer(emu); if (err < 0) goto error; @@ -2000,7 +2000,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static unsigned char saved_regs[] = { CPF, PTRX, CVCF, VTFT, Z1, Z2, PSST, DSL, CCCA, CCR, CLP, FXRT, MAPA, MAPB, ENVVOL, ATKHLDV, DCYSUSV, LFOVAL1, ENVVAL, diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index dae4050ede5c..52419959178c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2646,7 +2646,7 @@ int __devinit snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device, struct return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int __devinit snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu) { int len; diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index a81dc44228ea..88cec6b7dd41 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -893,7 +893,7 @@ int __devinit snd_p16v_mixer(struct snd_emu10k1 *emu) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP #define NUM_CHS 1 /* up to 4, but only first channel is used */ diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index f7e6f73186e1..2ba58d365959 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2032,7 +2032,7 @@ static void snd_ensoniq_chip_init(struct ensoniq *ensoniq) synchronize_irq(ensoniq->irq); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_ensoniq_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -2094,7 +2094,7 @@ static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume #define SND_ENSONIQ_PM_OPS &snd_ensoniq_pm #else #define SND_ENSONIQ_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int __devinit snd_ensoniq_create(struct snd_card *card, struct pci_dev *pci, diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index dbb81807bc1a..394c5d413530 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -236,7 +236,7 @@ struct es1938 { #ifdef SUPPORT_JOYSTICK struct gameport *gameport; #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP unsigned char saved_regs[SAVED_REG_SIZE]; #endif }; @@ -1456,7 +1456,7 @@ static void snd_es1938_chip_init(struct es1938 *chip) outb(0, SLDM_REG(chip, DMACLEAR)); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * PM support */ @@ -1536,7 +1536,7 @@ static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume); #define ES1938_PM_OPS &es1938_pm #else #define ES1938_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef SUPPORT_JOYSTICK static int __devinit snd_es1938_create_gameport(struct es1938 *chip) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index fb4c90b99c00..5d0e568fdea1 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -491,7 +491,7 @@ struct esschan { /* linked list */ struct list_head list; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u16 wc_map[4]; #endif }; @@ -544,7 +544,7 @@ struct es1968 { struct list_head substream_list; spinlock_t substream_lock; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u16 apu_map[NR_APUS][NR_APU_REGS]; #endif @@ -706,7 +706,7 @@ static void __apu_set_register(struct es1968 *chip, u16 channel, u8 reg, u16 dat { if (snd_BUG_ON(channel >= NR_APUS)) return; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->apu_map[channel][reg] = data; #endif reg |= (channel << 4); @@ -993,7 +993,7 @@ static void snd_es1968_program_wavecache(struct es1968 *chip, struct esschan *es /* set the wavecache control reg */ wave_set_register(chip, es->apu[channel] << 3, tmpval); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP es->wc_map[channel] = tmpval; #endif } @@ -2377,7 +2377,7 @@ static void snd_es1968_start_irq(struct es1968 *chip) outw(w, chip->io_port + ESM_PORT_HOST_IRQ); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * PM support */ @@ -2461,7 +2461,7 @@ static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume); #define ES1968_PM_OPS &es1968_pm #else #define ES1968_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef SUPPORT_JOYSTICK #define JOYSTICK_ADDR 0x200 diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 522c8706f244..ce3e548de41d 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -205,7 +205,7 @@ struct fm801 { struct snd_tea575x tea; #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u16 saved_regs[0x20]; #endif }; @@ -1361,7 +1361,7 @@ static void __devexit snd_card_fm801_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static unsigned char saved_regs[] = { FM801_PCM_VOL, FM801_I2S_VOL, FM801_FM_VOL, FM801_REC_SRC, FM801_PLY_CTRL, FM801_PLY_COUNT, FM801_PLY_BUF1, FM801_PLY_BUF2, @@ -1421,7 +1421,7 @@ static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume); #define SND_FM801_PM_OPS &snd_fm801_pm #else #define SND_FM801_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver fm801_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 3e4f8c12ffce..20bcddea2eab 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -2103,7 +2103,7 @@ static int aureon_reset(struct snd_ice1712 *ice) /* * suspend/resume */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int aureon_resume(struct snd_ice1712 *ice) { struct aureon_spec *spec = ice->spec; @@ -2160,7 +2160,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP ice->pm_resume = aureon_resume; ice->pm_suspend_enabled = 1; #endif diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 0da778a69ef8..d0e7d87f09f0 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -384,7 +384,7 @@ struct snd_ice1712 { char **ext_clock_names; int ext_clock_count; void (*pro_open)(struct snd_ice1712 *, struct snd_pcm_substream *); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); unsigned int pm_suspend_enabled:1; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index bed9f34f4efe..3050a5279253 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2792,7 +2792,7 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_vt1724_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -2878,7 +2878,7 @@ static SIMPLE_DEV_PM_OPS(snd_vt1724_pm, snd_vt1724_suspend, snd_vt1724_resume); #define SND_VT1724_PM_OPS &snd_vt1724_pm #else #define SND_VT1724_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static struct pci_driver vt1724_driver = { .name = KBUILD_MODNAME, diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 98bc3b7681b5..14fd536b6452 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -486,7 +486,7 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) * suspend/resume * */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int juli_resume(struct snd_ice1712 *ice) { struct snd_akm4xxx *ak = ice->akm; @@ -652,7 +652,7 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice->spdif.ops.open = juli_spdif_in_open; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP ice->pm_resume = juli_resume; ice->pm_suspend = juli_suspend; ice->pm_suspend_enabled = 1; diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 764cc93dbca4..7f2b63f97e61 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1099,7 +1099,7 @@ static void ak4396_init(struct snd_ice1712 *ice) ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ @@ -1140,7 +1140,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) return -ENOMEM; ice->spec = spec; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP ice->pm_resume = &prodigy_hd2_resume; ice->pm_suspend_enabled = 1; #endif diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index cd553f592e2d..5c4115289a9a 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2620,7 +2620,7 @@ static int snd_intel8x0_free(struct intel8x0 *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -2741,7 +2741,7 @@ static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume); #define INTEL8X0_PM_OPS &intel8x0_pm #else #define INTEL8X0_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #define INTEL8X0_TESTBUF_SIZE 32768 /* enough large for one shot */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index da44bb3f8e7a..4d551736531e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1008,7 +1008,7 @@ static int snd_intel8x0m_free(struct intel8x0m *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1067,7 +1067,7 @@ static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume); #define INTEL8X0M_PM_OPS &intel8x0m_pm #else #define INTEL8X0M_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef CONFIG_PROC_FS static void snd_intel8x0m_proc_read(struct snd_info_entry * entry, diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index c85d1ffcc955..eb3cd3a4315e 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -789,7 +789,7 @@ struct snd_m3 { unsigned int in_suspend; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u16 *suspend_mem; #endif @@ -2368,7 +2368,7 @@ static int snd_m3_free(struct snd_m3 *chip) outw(0, chip->iobase + HOST_INT_CTRL); /* disable ints */ } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP vfree(chip->suspend_mem); #endif @@ -2390,7 +2390,7 @@ static int snd_m3_free(struct snd_m3 *chip) /* * APM support */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int m3_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -2485,7 +2485,7 @@ static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume); #define M3_PM_OPS &m3_pm #else #define M3_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ #ifdef CONFIG_SND_MAESTRO3_INPUT static int __devinit snd_m3_input_register(struct snd_m3 *chip) @@ -2656,7 +2656,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, } chip->irq = pci->irq; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->suspend_mem = vmalloc(sizeof(u16) * (REV_B_CODE_MEMORY_LENGTH + REV_B_DATA_MEMORY_LENGTH)); if (chip->suspend_mem == NULL) snd_printk(KERN_WARNING "can't allocate apm buffer\n"); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 465cff25b146..e80e9a1e84aa 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1377,7 +1377,7 @@ snd_nm256_peek_for_sig(struct nm256 *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * APM event handler, so the card is properly reinitialized after a power * event. @@ -1441,7 +1441,7 @@ static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume); #define NM256_PM_OPS &nm256_pm #else #define NM256_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int snd_nm256_free(struct nm256 *chip) { diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 37520a2b4dcf..2becae155a48 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -872,7 +872,7 @@ static struct pci_driver oxygen_driver = { .id_table = oxygen_ids, .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &oxygen_pci_pm, }, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 7112a89fb8bd..09a24b24958b 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -161,7 +161,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, ) ); void oxygen_pci_remove(struct pci_dev *pci); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP extern const struct dev_pm_ops oxygen_pci_pm; #endif void oxygen_pci_shutdown(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index ab8738e21ad1..25697584b94c 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -726,7 +726,7 @@ void oxygen_pci_remove(struct pci_dev *pci) } EXPORT_SYMBOL(oxygen_pci_remove); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int oxygen_pci_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -824,7 +824,7 @@ static int oxygen_pci_resume(struct device *dev) SIMPLE_DEV_PM_OPS(oxygen_pci_pm, oxygen_pci_suspend, oxygen_pci_resume); EXPORT_SYMBOL(oxygen_pci_pm); -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ void oxygen_pci_shutdown(struct pci_dev *pci) { diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index d3b606b69f3b..3d71423b23bc 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -93,7 +93,7 @@ static struct pci_driver xonar_driver = { .id_table = xonar_ids, .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &oxygen_pci_pm, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 760ee467cd9a..7d291542c5ba 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -464,7 +464,7 @@ struct snd_riptide { unsigned long received_irqs; unsigned long handled_irqs; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP int in_suspend; #endif }; @@ -1150,7 +1150,7 @@ static void riptide_handleirq(unsigned long dev_id) } } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int riptide_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -1193,7 +1193,7 @@ static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume); #define RIPTIDE_PM_OPS &riptide_pm #else #define RIPTIDE_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) { diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 512434efcc31..535efe295075 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -103,7 +103,7 @@ struct voice { * we're not doing power management, we still need to allocate a page * for the silence buffer. */ -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP #define SIS_SUSPEND_PAGES 4 #else #define SIS_SUSPEND_PAGES 1 @@ -1208,7 +1208,7 @@ static int sis_chip_init(struct sis7019 *sis) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int sis_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -1305,7 +1305,7 @@ static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume); #define SIS_PM_OPS &sis_pm #else #define SIS_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int sis_alloc_suspend(struct sis7019 *sis) { diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d36e6ca147e1..8a6f1f76e870 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -177,7 +177,7 @@ static struct pci_driver trident_driver = { .id_table = snd_trident_ids, .probe = snd_trident_probe, .remove = __devexit_p(snd_trident_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &snd_trident_pm, }, diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 94011dcae731..06b10d1a76e5 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3919,7 +3919,7 @@ static void snd_trident_clear_voices(struct snd_trident * trident, unsigned shor } } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_trident_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); @@ -3983,4 +3983,4 @@ static int snd_trident_resume(struct device *dev) } SIMPLE_DEV_PM_OPS(snd_trident_pm, snd_trident_suspend, snd_trident_resume); -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 0eb7245dd362..e3d32e2d574e 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -362,7 +362,7 @@ struct via82xx { unsigned char old_legacy; unsigned char old_legacy_cfg; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP unsigned char legacy_saved; unsigned char legacy_cfg_saved; unsigned char spdif_ctrl_saved; @@ -2038,7 +2038,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) if (mpu_port >= 0x200) { /* force MIDI */ mpu_port &= 0xfffc; pci_write_config_dword(chip->pci, 0x18, mpu_port | 0x01); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->mpu_port_saved = mpu_port; #endif } else { @@ -2090,7 +2090,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) snd_via686_create_gameport(chip, &legacy); -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->legacy_saved = legacy; chip->legacy_cfg_saved = legacy_cfg; #endif @@ -2238,7 +2238,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -2313,7 +2313,7 @@ static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume #define SND_VIA82XX_PM_OPS &snd_via82xx_pm #else #define SND_VIA82XX_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int snd_via82xx_free(struct via82xx *chip) { diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index e886bc16999d..8e0efc416f22 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1019,7 +1019,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) return 0; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -1076,7 +1076,7 @@ static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume #define SND_VIA82XX_PM_OPS &snd_via82xx_pm #else #define SND_VIA82XX_PM_OPS NULL -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ static int snd_via82xx_free(struct via82xx_modem *chip) { diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index b89e7a86e9d8..fdfbaf857233 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -257,7 +257,7 @@ static void __devexit snd_vx222_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int snd_vx222_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 4810356b97ba..e01fe34db9ec 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -355,7 +355,7 @@ static struct pci_driver ymfpci_driver = { .id_table = snd_ymfpci_ids, .probe = snd_card_ymfpci_probe, .remove = __devexit_p(snd_card_ymfpci_remove), -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP .driver = { .pm = &snd_ymfpci_pm, }, diff --git a/sound/pci/ymfpci/ymfpci.h b/sound/pci/ymfpci/ymfpci.h index bddc4052286b..4631a2348915 100644 --- a/sound/pci/ymfpci/ymfpci.h +++ b/sound/pci/ymfpci/ymfpci.h @@ -363,7 +363,7 @@ struct snd_ymfpci { const struct firmware *dsp_microcode; const struct firmware *controller_microcode; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP u32 *saved_regs; u32 saved_ydsxgr_mode; u16 saved_dsxg_legacy; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 62b23635b754..ee8b6366e48d 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2242,7 +2242,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) pci_set_power_state(chip->pci, 3); #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP vfree(chip->saved_regs); #endif if (chip->irq >= 0) @@ -2272,7 +2272,7 @@ static int snd_ymfpci_dev_free(struct snd_device *device) return snd_ymfpci_free(chip); } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int saved_regs_index[] = { /* spdif */ YDSXGR_SPDIFOUTCTRL, @@ -2374,7 +2374,7 @@ static int snd_ymfpci_resume(struct device *dev) } SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume); -#endif /* CONFIG_PM */ +#endif /* CONFIG_PM_SLEEP */ int __devinit snd_ymfpci_create(struct snd_card *card, struct pci_dev * pci, @@ -2452,7 +2452,7 @@ int __devinit snd_ymfpci_create(struct snd_card *card, return err; } -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP chip->saved_regs = vmalloc(YDSXGR_NUM_SAVED_REGS * sizeof(u32)); if (chip->saved_regs == NULL) { snd_ymfpci_free(chip); -- cgit v1.2.3 From 1c86845268dc91fa6a53de9a4479b407cd4ee903 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Aug 2012 11:09:35 +0200 Subject: ALSA: hda - Add 3stack-automute model to AD1882 codec Added a simple support of automute for the front HP jack to AD1882 stack model. Such an addition is basically an exception -- we really want to avoid the static quirk codes, but AD1882 parser isn't still ready for moving to the BIOS auto-parser yet. So, as a quick fix, I merged it for now. In near future, we really need the big clean up of patch_analog.c to move on to the auto-parser... Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 ++- sound/pci/hda/patch_analog.c | 40 ++++++++++++++++++++++++++++ 2 files changed, 42 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a92bba816843..16dfe57f1731 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -74,7 +74,8 @@ CMI9880 AD1882 / AD1882A ================ - 3stack 3-stack mode (default) + 3stack 3-stack mode + 3stack-automute 3-stack with automute front HP (default) 6stack 6-stack mode AD1884A / AD1883 / AD1984A / AD1984B diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0208fa121e5a..21218853366d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4814,6 +4814,32 @@ static const struct snd_kcontrol_new ad1882_3stack_mixers[] = { { } /* end */ }; +/* simple auto-mute control for AD1882 3-stack board */ +#define AD1882_HP_EVENT 0x01 + +static void ad1882_3stack_automute(struct hda_codec *codec) +{ + bool mute = snd_hda_jack_detect(codec, 0x11); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + mute ? 0 : PIN_OUT); +} + +static int ad1882_3stack_automute_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1882_3stack_automute(codec); + return 0; +} + +static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch (res >> 26) { + case AD1882_HP_EVENT: + ad1882_3stack_automute(codec); + break; + } +} + static const struct snd_kcontrol_new ad1882_6stack_mixers[] = { HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), @@ -4928,6 +4954,11 @@ static const struct hda_verb ad1882_init_verbs[] = { { } /* end */ }; +static const struct hda_verb ad1882_3stack_automute_verbs[] = { + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT}, + { } /* end */ +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE static const struct hda_amp_list ad1882_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ @@ -4942,12 +4973,14 @@ static const struct hda_amp_list ad1882_loopbacks[] = { enum { AD1882_3STACK, AD1882_6STACK, + AD1882_3STACK_AUTOMUTE, AD1882_MODELS }; static const char * const ad1882_models[AD1986A_MODELS] = { [AD1882_3STACK] = "3stack", [AD1882_6STACK] = "6stack", + [AD1882_3STACK_AUTOMUTE] = "3stack-automute", }; @@ -5002,6 +5035,7 @@ static int patch_ad1882(struct hda_codec *codec) switch (board_config) { default: case AD1882_3STACK: + case AD1882_3STACK_AUTOMUTE: spec->num_mixers = 3; spec->mixers[2] = ad1882_3stack_mixers; spec->channel_mode = ad1882_modes; @@ -5009,6 +5043,12 @@ static int patch_ad1882(struct hda_codec *codec) spec->need_dac_fix = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; + if (board_config != AD1882_3STACK) { + spec->init_verbs[spec->num_init_verbs++] = + ad1882_3stack_automute_verbs; + codec->patch_ops.unsol_event = ad1882_3stack_unsol_event; + codec->patch_ops.init = ad1882_3stack_automute_init; + } break; case AD1882_6STACK: spec->num_mixers = 3; -- cgit v1.2.3 From fff8491c8b8cce5fc9190e025d1a665f2ee71a4f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Aug 2012 12:07:56 +0300 Subject: ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec Machine driver to handle simple devices using twl4030 as audio codec. The driver supports the following boards: - Beagleboard or Devkit8000 - Gumstix Overo or CompuLab CM-T35/CM-T3730 - IGEP v2 - OMAP3EVM All of these boards can be switched to use this driver since their setup is identical. Devicetree support for the omap-twl4030 machine driver also implemented. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-twl4030.txt | 17 ++ include/linux/platform_data/omap-twl4030.h | 32 ++++ sound/soc/omap/Kconfig | 13 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-twl4030.c | 188 +++++++++++++++++++++ 5 files changed, 252 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 include/linux/platform_data/omap-twl4030.h create mode 100644 sound/soc/omap/omap-twl4030.c (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/omap-twl4030.txt b/Documentation/devicetree/bindings/sound/omap-twl4030.txt new file mode 100644 index 000000000000..6fae51c7f766 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-twl4030.txt @@ -0,0 +1,17 @@ +* Texas Instruments SoC with twl4030 based audio setups + +Required properties: +- compatible: "ti,omap-twl4030" +- ti,model: Name of the sound card (for example "omap3beagle") +- ti,mcbsp: phandle for the McBSP node +- ti,codec: phandle for the twl4030 audio node + +Example: + +sound { + compatible = "ti,omap-twl4030"; + ti,model = "omap3beagle"; + + ti,mcbsp = <&mcbsp2>; + ti,codec = <&twl_audio>; +}; diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h new file mode 100644 index 000000000000..c7bef788daab --- /dev/null +++ b/include/linux/platform_data/omap-twl4030.h @@ -0,0 +1,32 @@ +/** + * omap-twl4030.h - ASoC machine driver for TI SoC based boards with twl4030 + * codec, header. + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * All rights reserved. + * + * Author: Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#ifndef _OMAP_TWL4030_H_ +#define _OMAP_TWL4030_H_ + +struct omap_tw4030_pdata { + const char *card_name; +}; + +#endif /* _OMAP_TWL4030_H_ */ diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 57a2fa751085..fc83d748625f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -95,6 +95,19 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. +config SND_OMAP_SOC_OMAP_TWL4030 + tristate "SoC Audio support for TI SoC based boards with twl4030 codec" + depends on TWL4030_CORE && SND_OMAP_SOC + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on TI SoC based boards + using twl4030 as c codec. This driver currently supports: + - Beagleboard or Devkit8000 + - Gumstix Overo or CompuLab CM-T35/CM-T3730 + - IGEP v2 + - OMAP3EVM + config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0e14dd322565..861e640e2be9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -21,6 +21,7 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o +snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c new file mode 100644 index 000000000000..3b97b87971f5 --- /dev/null +++ b/sound/soc/omap/omap-twl4030.c @@ -0,0 +1,188 @@ +/* + * omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * All rights reserved. + * + * Author: Peter Ujfalusi + * + * This driver replaces the following machine drivers: + * omap3beagle (Author: Steve Sakoman ) + * omap3evm (Author: Anuj Aggarwal ) + * overo (Author: Steve Sakoman ) + * igep0020 (Author: Enric Balletbo i Serra ) + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include + +#include +#include +#include + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +static int omap_twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + unsigned int fmt; + int ret; + + switch (params_channels(params)) { + case 2: /* Stereo I2S mode */ + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + case 4: /* Four channel TDM mode */ + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + break; + default: + return -EINVAL; + } + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(card->dev, "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(card->dev, "can't set cpu DAI configuration\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap_twl4030_ops = { + .hw_params = omap_twl4030_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap_twl4030_dai_links[] = { + { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai_name = "omap-mcbsp.2", + .codec_dai_name = "twl4030-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "twl4030-codec", + .ops = &omap_twl4030_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card omap_twl4030_card = { + .owner = THIS_MODULE, + .dai_link = omap_twl4030_dai_links, + .num_links = ARRAY_SIZE(omap_twl4030_dai_links), +}; + +static __devinit int omap_twl4030_probe(struct platform_device *pdev) +{ + struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev); + struct device_node *node = pdev->dev.of_node; + struct snd_soc_card *card = &omap_twl4030_card; + int ret = 0; + + card->dev = &pdev->dev; + + if (node) { + struct device_node *dai_node; + + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + + dai_node = of_parse_phandle(node, "ti,mcbsp", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McBSP node is not provided\n"); + return -EINVAL; + } + omap_twl4030_dai_links[0].cpu_dai_name = NULL; + omap_twl4030_dai_links[0].cpu_of_node = dai_node; + + } else if (pdata) { + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } + } else { + dev_err(&pdev->dev, "Missing pdata\n"); + return -ENODEV; + } + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit omap_twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id omap_twl4030_of_match[] = { + {.compatible = "ti,omap-twl4030", }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_twl4030_of_match); + +static struct platform_driver omap_twl4030_driver = { + .driver = { + .name = "omap-twl4030", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = omap_twl4030_of_match, + }, + .probe = omap_twl4030_probe, + .remove = __devexit_p(omap_twl4030_remove), +}; + +module_platform_driver(omap_twl4030_driver); + +MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_DESCRIPTION("ALSA SoC for TI SoC based boards with twl4030 codec"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-twl4030"); -- cgit v1.2.3 From 152c6e56f6a8577bd291f6f4ca897e5758332a1b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Aug 2012 12:07:59 +0300 Subject: ASoC: Remove obsolete OMAP3 machine drivers The new omap-twl4030 handles the boards used the following drivers: igep0020, omap3beagle, omap3evm and overo. Remove these drivers since they are mostly identical and we already have drop in replacement for all of them. Note: Earlier patch added the needed code for the board files to retain the audio support for boards I can identify that used one of the removed drivers. If I missed something please take a look at for example: arch/arm/mach-omap2/board-omap3beagle.c on how add support for omap-twl4030 audio. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 35 ---------- sound/soc/omap/Makefile | 8 --- sound/soc/omap/igep0020.c | 120 ---------------------------------- sound/soc/omap/omap3beagle.c | 150 ------------------------------------------- sound/soc/omap/omap3evm.c | 118 ---------------------------------- sound/soc/omap/overo.c | 122 ----------------------------------- 6 files changed, 553 deletions(-) delete mode 100644 sound/soc/omap/igep0020.c delete mode 100644 sound/soc/omap/omap3beagle.c delete mode 100644 sound/soc/omap/omap3evm.c delete mode 100644 sound/soc/omap/overo.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fc83d748625f..2c484a52ef92 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -60,23 +60,6 @@ config SND_OMAP_SOC_OSK5912 help Say Y if you want to add support for SoC audio on osk5912. -config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" - depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on the - Gumstix Overo or CompuLab CM-T35 - -config SND_OMAP_SOC_OMAP3EVM - tristate "SoC Audio support for OMAP3EVM board" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on the omap3evm board. - config SND_OMAP_SOC_AM3517EVM tristate "SoC Audio support for OMAP3517 / AM3517 EVM" depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C @@ -140,16 +123,6 @@ config SND_OMAP_SOC_OMAP3_PANDORA help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. -config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" - depends on TWL4030_CORE && SND_OMAP_SOC - depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for SoC audio on the Beagleboard or - the clone Devkit8000. - config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 @@ -157,11 +130,3 @@ config SND_OMAP_SOC_ZOOM2 select SND_SOC_TWL4030 help Say Y if you want to add support for Soc audio on Zoom2 board. - -config SND_OMAP_SOC_IGEP0020 - tristate "SoC Audio support for IGEP v2" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 - select SND_OMAP_SOC_MCBSP - select SND_SOC_TWL4030 - help - Say Y if you want to add support for Soc audio on IGEP v2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 861e640e2be9..19637e55ea48 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -16,31 +16,23 @@ snd-soc-n810-objs := n810.o snd-soc-rx51-objs := rx51.o snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o -snd-soc-overo-objs := overo.o -snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap-twl4030-objs := omap-twl4030.o snd-soc-omap3pandora-objs := omap3pandora.o -snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o -snd-soc-igep0020-objs := igep0020.o snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o -obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o -obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c deleted file mode 100644 index e8357819175b..000000000000 --- a/sound/soc/omap/igep0020.c +++ /dev/null @@ -1,120 +0,0 @@ -/* - * igep0020.c -- SoC audio for IGEP v2 - * - * Based on sound/soc/omap/overo.c by Steve Sakoman - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int igep2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops igep2_ops = { - .hw_params = igep2_hw_params, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link igep2_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &igep2_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_card_igep2 = { - .name = "igep2", - .owner = THIS_MODULE, - .dai_link = &igep2_dai, - .num_links = 1, -}; - -static struct platform_device *igep2_snd_device; - -static int __init igep2_soc_init(void) -{ - int ret; - - if (!machine_is_igep0020()) - return -ENODEV; - printk(KERN_INFO "IGEP v2 SoC init\n"); - - igep2_snd_device = platform_device_alloc("soc-audio", -1); - if (!igep2_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(igep2_snd_device, &snd_soc_card_igep2); - - ret = platform_device_add(igep2_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(igep2_snd_device); - - return ret; -} -module_init(igep2_soc_init); - -static void __exit igep2_soc_exit(void) -{ - platform_device_unregister(igep2_snd_device); -} -module_exit(igep2_soc_exit); - -MODULE_AUTHOR("Enric Balletbo i Serra "); -MODULE_DESCRIPTION("ALSA SoC IGEP v2"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c deleted file mode 100644 index 2830dfd05661..000000000000 --- a/sound/soc/omap/omap3beagle.c +++ /dev/null @@ -1,150 +0,0 @@ -/* - * omap3beagle.c -- SoC audio for OMAP3 Beagle - * - * Author: Steve Sakoman - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int omap3beagle_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int fmt; - int ret; - - switch (params_channels(params)) { - case 2: /* Stereo I2S mode */ - fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - break; - case 4: /* Four channel TDM mode */ - fmt = SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM; - break; - default: - return -EINVAL; - } - - /* Set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return ret; - } - - /* Set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return ret; - } - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops omap3beagle_ops = { - .hw_params = omap3beagle_hw_params, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link omap3beagle_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .platform_name = "omap-pcm-audio", - .codec_dai_name = "twl4030-hifi", - .codec_name = "twl4030-codec", - .ops = &omap3beagle_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_omap3beagle = { - .name = "omap3beagle", - .owner = THIS_MODULE, - .dai_link = &omap3beagle_dai, - .num_links = 1, -}; - -static struct platform_device *omap3beagle_snd_device; - -static int __init omap3beagle_soc_init(void) -{ - int ret; - - if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) - return -ENODEV; - pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); - - omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap3beagle_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(omap3beagle_snd_device, &snd_soc_omap3beagle); - - ret = platform_device_add(omap3beagle_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap3beagle_snd_device); - - return ret; -} - -static void __exit omap3beagle_soc_exit(void) -{ - platform_device_unregister(omap3beagle_snd_device); -} - -module_init(omap3beagle_soc_init); -module_exit(omap3beagle_soc_exit); - -MODULE_AUTHOR("Steve Sakoman "); -MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c deleted file mode 100644 index 3d468c9179d7..000000000000 --- a/sound/soc/omap/omap3evm.c +++ /dev/null @@ -1,118 +0,0 @@ -/* - * omap3evm.c -- ALSA SoC support for OMAP3 EVM - * - * Author: Anuj Aggarwal - * - * Based on sound/soc/omap/beagle.c by Steve Sakoman - * - * Copyright (C) 2008 Texas Instruments, Incorporated - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation version 2. - * - * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, - * whether express or implied; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int omap3evm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "Can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops omap3evm_ops = { - .hw_params = omap3evm_hw_params, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link omap3evm_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &omap3evm_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_omap3evm = { - .name = "omap3evm", - .owner = THIS_MODULE, - .dai_link = &omap3evm_dai, - .num_links = 1, -}; - -static struct platform_device *omap3evm_snd_device; - -static int __init omap3evm_soc_init(void) -{ - int ret; - - if (!machine_is_omap3evm()) - return -ENODEV; - pr_info("OMAP3 EVM SoC init\n"); - - omap3evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap3evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(omap3evm_snd_device, &snd_soc_omap3evm); - ret = platform_device_add(omap3evm_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap3evm_snd_device); - - return ret; -} - -static void __exit omap3evm_soc_exit(void) -{ - platform_device_unregister(omap3evm_snd_device); -} - -module_init(omap3evm_soc_init); -module_exit(omap3evm_soc_exit); - -MODULE_AUTHOR("Anuj Aggarwal "); -MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c deleted file mode 100644 index 6ac3e0c3c282..000000000000 --- a/sound/soc/omap/overo.c +++ /dev/null @@ -1,122 +0,0 @@ -/* - * overo.c -- SoC audio for Gumstix Overo - * - * Author: Steve Sakoman - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include "omap-mcbsp.h" -#include "omap-pcm.h" - -static int overo_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* Set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - return 0; -} - -static struct snd_soc_ops overo_ops = { - .hw_params = overo_hw_params, -}; - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link overo_dai = { - .name = "TWL4030", - .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp.2", - .codec_dai_name = "twl4030-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "twl4030-codec", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &overo_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_card_overo = { - .name = "overo", - .owner = THIS_MODULE, - .dai_link = &overo_dai, - .num_links = 1, -}; - -static struct platform_device *overo_snd_device; - -static int __init overo_soc_init(void) -{ - int ret; - - if (!(machine_is_overo() || machine_is_cm_t35())) { - pr_debug("Incomatible machine!\n"); - return -ENODEV; - } - printk(KERN_INFO "overo SoC init\n"); - - overo_snd_device = platform_device_alloc("soc-audio", -1); - if (!overo_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(overo_snd_device, &snd_soc_card_overo); - - ret = platform_device_add(overo_snd_device); - if (ret) - goto err1; - - return 0; - -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(overo_snd_device); - - return ret; -} -module_init(overo_soc_init); - -static void __exit overo_soc_exit(void) -{ - platform_device_unregister(overo_snd_device); -} -module_exit(overo_soc_exit); - -MODULE_AUTHOR("Steve Sakoman "); -MODULE_DESCRIPTION("ALSA SoC overo"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From a3150f09174ace7878bf4bbbf23d3ba25cc01261 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Aug 2012 20:28:18 +0100 Subject: ASoC: wm5102: Add AEC routing control Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4c2fc361051c..2e6f1ffc9fd4 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -280,6 +280,27 @@ ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const char *wm5102_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", + "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm5102_aec_loopback_values[] = { + 0, 1, 2, 3, 4, 6, 7, 8, 9, +}; + +static const struct soc_enum wm5102_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, + ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARRAY_SIZE(wm5102_aec_loopback_texts), + wm5102_aec_loopback_texts, + wm5102_aec_loopback_values); + +static const struct snd_kcontrol_new wm5102_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm5102_aec_loopback); + static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), @@ -435,6 +456,9 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), @@ -532,6 +556,7 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "Noise Generator", "Noise Generator" }, \ { name, "Tone Generator 1", "Tone Generator 1" }, \ { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "AEC", "AEC Loopback" }, \ { name, "IN1L", "IN1L PGA" }, \ { name, "IN1R", "IN1R PGA" }, \ { name, "IN2L", "IN2L PGA" }, \ @@ -692,21 +717,30 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "EPOUT", "OUT3L" }, { "EPOUTN", NULL, "OUT3L" }, { "EPOUTP", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, }; -- cgit v1.2.3 From 45a690f6bcd5506d9988d0d069811ac9380750ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 15 Aug 2012 19:20:54 +0100 Subject: ASoC: wm8994: Add bytes controls for DRC If DRC coefficients are not configured via platform data then add bytes controls for them instead so they can be configured by applications. This is the normal means of controlling things like this for newer systems, we maintain compatibility with platform data to avoid disruption to existing systems. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 02080da8b451..353612eec8bf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -671,6 +671,18 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const struct snd_kcontrol_new wm8994_drc_controls[] = { +SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5, + WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5, + WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA | + WM8994_AIF1ADC2R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5, + WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA | + WM8994_AIF2ADCR_DRC_ENA), +}; + static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; @@ -3166,14 +3178,19 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); - if (ret != 0) - dev_err(wm8994->hubs.codec->dev, - "Failed to add DRC mode controls: %d\n", ret); - for (i = 0; i < WM8994_NUM_DRC; i++) wm8994_set_drc(codec, i); + } else { + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + wm8994_drc_controls, + ARRAY_SIZE(wm8994_drc_controls)); } + if (ret != 0) + dev_err(wm8994->hubs.codec->dev, + "Failed to add DRC mode controls: %d\n", ret); + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", pdata->num_retune_mobile_cfgs); -- cgit v1.2.3 From d14a13d3d94bae80c3f44ff35a3c76192fb3408c Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Fri, 10 Aug 2012 18:06:08 -0400 Subject: ASoC: bf5xx-ad1836: convert to use snd_soc_register_card Cpu dai and codec name are passed in through platform data. Signed-off-by: Scott Jiang Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1836.c | 73 ++++++++++++++++++++------------------- 1 file changed, 37 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d542d4063771..16b9c9efd19a 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -59,62 +59,63 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.0", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.1", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; -static struct platform_device *bfxx_ad1836_snd_device; - -static int __init bf5xx_ad1836_init(void) +static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &bf5xx_ad1836; + const char **link_name; int ret; - bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1836_snd_device) - return -ENOMEM; + link_name = pdev->dev.platform_data; + if (!link_name) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; + bf5xx_ad1836_dai.codec_name = link_name[1]; - platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836); - ret = platform_device_add(bfxx_ad1836_snd_device); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(bfxx_ad1836_snd_device); - + dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static void __exit bf5xx_ad1836_exit(void) +static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev) { - platform_device_unregister(bfxx_ad1836_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(bf5xx_ad1836_init); -module_exit(bf5xx_ad1836_exit); +static struct platform_driver bf5xx_ad1836_driver = { + .driver = { + .name = "bfin-snd-ad1836", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bf5xx_ad1836_driver_probe, + .remove = __devexit_p(bf5xx_ad1836_driver_remove), +}; +module_platform_driver(bf5xx_ad1836_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); -- cgit v1.2.3 From c86b6b452a6b2a80a2c9ffa3c8f7d80eea0fa196 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 19 Aug 2012 23:27:19 +0200 Subject: ALSA: snd-ad1816a: remove useless struct snd_card_ad1816a struct snd_card_ad1816a is only set but the values are never used then. Removing it allows struct snd_card's private_data to be used for struct snd_ad1816a, simplifying the code. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- include/sound/ad1816a.h | 2 +- sound/isa/ad1816a/ad1816a.c | 38 ++++++++++++-------------------------- sound/isa/ad1816a/ad1816a_lib.c | 10 +--------- 3 files changed, 14 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index d010858c33c2..62da41e508e1 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -165,7 +165,7 @@ struct snd_ad1816a { extern int snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **chip); + struct snd_ad1816a *chip); extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm); extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 94b83b6e46a3..1a374e61ab2b 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -63,11 +63,6 @@ MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard."); module_param_array(clockfreq, int, NULL, 0444); MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0)."); -struct snd_card_ad1816a { - struct pnp_dev *dev; - struct pnp_dev *devmpu; -}; - static struct pnp_card_device_id snd_ad1816a_pnpids[] = { /* Analog Devices AD1815 */ { .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } }, @@ -99,25 +94,16 @@ MODULE_DEVICE_TABLE(pnp_card, snd_ad1816a_pnpids); #define DRIVER_NAME "snd-card-ad1816a" -static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acard, - struct pnp_card_link *card, +static int __devinit snd_card_ad1816a_pnp(int dev, struct pnp_card_link *card, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; int err; - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + pdev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devmpu == NULL) { - mpu_port[dev] = -1; - snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); - } - - pdev = acard->dev; - err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); @@ -130,16 +116,17 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - if (acard->devmpu == NULL) + pdev = pnp_request_card_device(card, id->devs[1].id, NULL); + if (pdev == NULL) { + mpu_port[dev] = -1; + snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); return 0; - - pdev = acard->devmpu; + } err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); mpu_port[dev] = -1; - acard->devmpu = NULL; } else { mpu_port[dev] = pnp_port_start(pdev, 0); mpu_irq[dev] = pnp_irq(pdev, 0); @@ -153,18 +140,17 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard { int error; struct snd_card *card; - struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; struct snd_timer *timer; error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_ad1816a), &card); + sizeof(struct snd_ad1816a), &card); if (error < 0) return error; - acard = card->private_data; + chip = card->private_data; - if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { + if ((error = snd_card_ad1816a_pnp(dev, pcard, pid))) { snd_card_free(card); return error; } @@ -174,7 +160,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) { + chip)) < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 177eed3271bc..de5cc1c26279 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -548,7 +548,6 @@ static int snd_ad1816a_free(struct snd_ad1816a *chip) snd_dma_disable(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } @@ -573,19 +572,13 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) int __devinit snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **rchip) + struct snd_ad1816a *chip) { static struct snd_device_ops ops = { .dev_free = snd_ad1816a_dev_free, }; int error; - struct snd_ad1816a *chip; - *rchip = NULL; - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -631,7 +624,6 @@ int __devinit snd_ad1816a_create(struct snd_card *card, return error; } - *rchip = chip; return 0; } -- cgit v1.2.3 From 6f0fa66051e92f361bd293432466f5e62832adbf Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 19 Aug 2012 23:27:26 +0200 Subject: ALSA: snd-ad1816a: Implement suspend/resume Implement suspend/resume support for AD1816 chips. Tested with Terratec SoundSystem Base-1. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- include/sound/ad1816a.h | 7 +++++++ sound/isa/ad1816a/ad1816a.c | 26 +++++++++++++++++++++++++- sound/isa/ad1816a/ad1816a_lib.c | 28 +++++++++++++++++++++++++++- 3 files changed, 59 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index 62da41e508e1..2a89f0d71440 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -147,6 +147,9 @@ struct snd_ad1816a { unsigned int c_dma_size; struct snd_timer *timer; +#ifdef CONFIG_PM + unsigned short image[48]; +#endif }; @@ -171,5 +174,9 @@ extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer); +#ifdef CONFIG_PM +extern void snd_ad1816a_suspend(struct snd_ad1816a *chip); +extern void snd_ad1816a_resume(struct snd_ad1816a *chip); +#endif #endif /* __SOUND_AD1816A_H */ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 1a374e61ab2b..2c2f829c3fd7 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -244,13 +244,37 @@ static void __devexit snd_ad1816a_pnp_remove(struct pnp_card_link * pcard) pnp_set_card_drvdata(pcard, NULL); } +#ifdef CONFIG_PM +static int snd_ad1816a_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_ad1816a_suspend(card->private_data); + return 0; +} + +static int snd_ad1816a_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_ad1816a_resume(card->private_data); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct pnp_card_driver ad1816a_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "ad1816a", .id_table = snd_ad1816a_pnpids, .probe = snd_ad1816a_pnp_detect, .remove = __devexit_p(snd_ad1816a_pnp_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_ad1816a_pnp_suspend, + .resume = snd_ad1816a_pnp_resume, +#endif }; static int __init alsa_card_ad1816a_init(void) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index de5cc1c26279..db64df6023e0 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -491,7 +491,7 @@ static int snd_ad1816a_capture_close(struct snd_pcm_substream *substream) } -static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) +static void snd_ad1816a_init(struct snd_ad1816a *chip) { unsigned long flags; @@ -511,6 +511,32 @@ static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) spin_unlock_irqrestore(&chip->lock, flags); } +#ifdef CONFIG_PM +void snd_ad1816a_suspend(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_pcm_suspend_all(chip->pcm); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + chip->image[reg] = snd_ad1816a_read(chip, reg); + spin_unlock_irqrestore(&chip->lock, flags); +} + +void snd_ad1816a_resume(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_ad1816a_init(chip); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + snd_ad1816a_write(chip, reg, chip->image[reg]); + spin_unlock_irqrestore(&chip->lock, flags); +} +#endif + static int __devinit snd_ad1816a_probe(struct snd_ad1816a *chip) { unsigned long flags; -- cgit v1.2.3 From b244d335609f9bbf4c0204a365bf03b7ac53c3c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 10:22:25 +0200 Subject: ALSA: hda - Add tracepoints at snd_hda_power_up/down entrances. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_trace.h | 24 ++++++++++++++++++++++++ 2 files changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 408babc10437..ad0101e2d4ff 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4417,6 +4417,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_lock(&codec->power_lock); codec->power_count++; + trace_hda_power_count(codec); /* Return if power_on or transitioning to power_on, unless currently * powering down. */ if ((codec->power_on || codec->power_transition > 0) && @@ -4496,6 +4497,7 @@ void snd_hda_power_down(struct hda_codec *codec) { spin_lock(&codec->power_lock); --codec->power_count; + trace_hda_power_count(codec); if (!codec->power_on || codec->power_count || codec->power_transition) { spin_unlock(&codec->power_lock); return; diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index 9884871ddb00..9a34b867d8c2 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -87,6 +87,30 @@ DEFINE_EVENT(hda_power, hda_power_up, TP_ARGS(codec) ); +TRACE_EVENT(hda_power_count, + TP_PROTO(struct hda_codec *codec), + TP_ARGS(codec), + TP_STRUCT__entry( + __field( unsigned int, card ) + __field( unsigned int, addr ) + __field( int, power_count ) + __field( int, power_on ) + __field( int, power_transition ) + ), + + TP_fast_assign( + __entry->card = (codec)->bus->card->number; + __entry->addr = (codec)->addr; + __entry->power_count = (codec)->power_count; + __entry->power_on = (codec)->power_on; + __entry->power_transition = (codec)->power_transition; + ), + + TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", + __entry->card, __entry->addr, __entry->power_count, + __entry->power_on, __entry->power_transition) +); + TRACE_EVENT(hda_unsol_event, TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex), -- cgit v1.2.3 From c376e2c72ba2d510486d7dea41163dce5f5606ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 17:12:47 +0200 Subject: ALSA: hda - Implement snd_hda_power_sync() helper function Added a new helper function snd_hda_power_sync() to trigger the power-saving manually. It's an inline function call to snd_hda_power_save() helper function. Together with this addition, snd_hda_power_up*() and snd_hda_power_down() functions are inlined to a call of the same snd_hda_power_save() helper function. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 82 ++++++++++++++++------------------------------- sound/pci/hda/hda_codec.h | 60 ++++++++++++++++++++++++++++++---- 2 files changed, 82 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ad0101e2d4ff..5e0bdea6c5c8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4411,20 +4411,16 @@ void snd_hda_update_power_acct(struct hda_codec *codec) /* Transition to powered up, if wait_power_down then wait for a pending * transition to D3 to complete. A pending D3 transition is indicated * with power_transition == -1. */ +/* call this with codec->power_lock held! */ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; - spin_lock(&codec->power_lock); - codec->power_count++; - trace_hda_power_count(codec); /* Return if power_on or transitioning to power_on, unless currently * powering down. */ if ((codec->power_on || codec->power_transition > 0) && - !(wait_power_down && codec->power_transition < 0)) { - spin_unlock(&codec->power_lock); + !(wait_power_down && codec->power_transition < 0)) return; - } spin_unlock(&codec->power_lock); cancel_delayed_work_sync(&codec->power_work); @@ -4433,10 +4429,9 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) /* If the power down delayed work was cancelled above before starting, * then there is no need to go through power up here. */ - if (codec->power_on) { - spin_unlock(&codec->power_lock); + if (codec->power_on) return; - } + trace_hda_power_up(codec); snd_hda_update_power_acct(codec); codec->power_on = 1; @@ -4450,66 +4445,45 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_lock(&codec->power_lock); codec->power_transition = 0; - spin_unlock(&codec->power_lock); } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) -{ - __snd_hda_power_up(codec, false); -} -EXPORT_SYMBOL_HDA(snd_hda_power_up); +#define power_save(codec) \ + ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) -/** - * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending - * D3 transition to complete. This differs from snd_hda_power_up() when - * power_transition == -1. snd_hda_power_up sees this case as a nop, - * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers - * back up. - * @codec: HD-audio codec - * - * Cancel any power down operation hapenning on the work queue, then power up. - */ -void snd_hda_power_up_d3wait(struct hda_codec *codec) +/* Transition to powered down */ +static void __snd_hda_power_down(struct hda_codec *codec) { - /* This will cancel and wait for pending power_work to complete. */ - __snd_hda_power_up(codec, true); -} -EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + if (!codec->power_on || codec->power_count || codec->power_transition) + return; -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) + if (power_save(codec)) { + codec->power_transition = -1; /* avoid reentrance */ + queue_delayed_work(codec->bus->workq, &codec->power_work, + msecs_to_jiffies(power_save(codec) * 1000)); + } +} /** - * snd_hda_power_down - Power-down the codec + * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec + * @delta: the counter delta to change * - * Decrement the power-up counter and schedules the power-off work if - * the counter rearches to zero. + * Change the power-up counter via @delta, and power up or down the hardware + * appropriately. For the power-down, queue to the delayed action. + * Passing zero to @delta means to synchronize the power state. */ -void snd_hda_power_down(struct hda_codec *codec) +void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) { spin_lock(&codec->power_lock); - --codec->power_count; + codec->power_count += delta; trace_hda_power_count(codec); - if (!codec->power_on || codec->power_count || codec->power_transition) { - spin_unlock(&codec->power_lock); - return; - } - if (power_save(codec)) { - codec->power_transition = -1; /* avoid reentrance */ - queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); - } + if (delta > 0) + __snd_hda_power_up(codec, d3wait); + else + __snd_hda_power_down(codec); spin_unlock(&codec->power_lock); } -EXPORT_SYMBOL_HDA(snd_hda_power_down); +EXPORT_SYMBOL_HDA(snd_hda_power_save); /** * snd_hda_check_amp_list_power - Check the amp list and update the power diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index d772c2521cd9..1d4dfce631a0 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1062,16 +1062,64 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_SND_HDA_POWER_SAVE -void snd_hda_power_up(struct hda_codec *codec); -void snd_hda_power_up_d3wait(struct hda_codec *codec); -void snd_hda_power_down(struct hda_codec *codec); +void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_power_up(struct hda_codec *codec) {} -static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} -static inline void snd_hda_power_down(struct hda_codec *codec) {} +static inline void snd_hda_power_save(struct hda_codec *codec, int delta, + bool d3wait) {} #endif +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +static inline void snd_hda_power_up(struct hda_codec *codec) +{ + snd_hda_power_save(codec, 1, false); +} + +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + snd_hda_power_save(codec, 1, true); +} + +/** + * snd_hda_power_down - Power-down the codec + * @codec: HD-audio codec + * + * Decrement the power-up counter and schedules the power-off work if + * the counter rearches to zero. + */ +static inline void snd_hda_power_down(struct hda_codec *codec) +{ + snd_hda_power_save(codec, -1, false); +} + +/** + * snd_hda_power_sync - Synchronize the power-save status + * @codec: HD-audio codec + * + * Synchronize the actual power state with the power account; + * called when power_save parameter is changed + */ +static inline void snd_hda_power_sync(struct hda_codec *codec) +{ + snd_hda_power_save(codec, 0, false); +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware -- cgit v1.2.3 From 65fcd41d371f4464c3f5a0433a8056cc3d739397 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Aug 2012 17:13:32 +0200 Subject: ALSA: hda - Check the power state when power_save option is changed ... by calling the newly introduced snd_hda_power_sync(). I had to reimplement a wheel for adding the trigger at changing the parameter -- the parameter set ops is overwritten to pass the integer parameter, then trigger the power-state sync. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 58 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 57 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 464ea16bdd9e..f81489a2b134 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -110,8 +110,15 @@ MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " #endif #ifdef CONFIG_SND_HDA_POWER_SAVE +static int param_set_xint(const char *val, const struct kernel_param *kp); +static struct kernel_param_ops param_ops_xint = { + .set = param_set_xint, + .get = param_get_int, +}; +#define param_check_xint param_check_int + static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -module_param(power_save, int, 0644); +module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -502,6 +509,9 @@ struct azx { /* reboot notifier (for mysterious hangup problem at power-down) */ struct notifier_block reboot_notifier; + + /* card list (for power_save trigger) */ + struct list_head list; }; /* driver types */ @@ -2407,6 +2417,48 @@ static void azx_power_notify(struct hda_bus *bus) !bus->power_keep_link_on) azx_stop_chip(chip); } + +static DEFINE_MUTEX(card_list_lock); +static LIST_HEAD(card_list); + +static void azx_add_card_list(struct azx *chip) +{ + mutex_lock(&card_list_lock); + list_add(&chip->list, &card_list); + mutex_unlock(&card_list_lock); +} + +static void azx_del_card_list(struct azx *chip) +{ + mutex_lock(&card_list_lock); + list_del_init(&chip->list); + mutex_unlock(&card_list_lock); +} + +/* trigger power-save check at writing parameter */ +static int param_set_xint(const char *val, const struct kernel_param *kp) +{ + struct azx *chip; + struct hda_codec *c; + int prev = power_save; + int ret = param_set_int(val, kp); + + if (ret || prev == power_save) + return ret; + + mutex_lock(&card_list_lock); + list_for_each_entry(chip, &card_list, list) { + if (!chip->bus || chip->disabled) + continue; + list_for_each_entry(c, &chip->bus->codec_list, list) + snd_hda_power_sync(c); + } + mutex_unlock(&card_list_lock); + return 0; +} +#else +#define azx_add_card_list(chip) /* NOP */ +#define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_SND_HDA_POWER_SAVE */ #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) @@ -2604,6 +2656,8 @@ static int azx_free(struct azx *chip) { int i; + azx_del_card_list(chip); + azx_notifier_unregister(chip); if (use_vga_switcheroo(chip)) { @@ -2911,6 +2965,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); init_vga_switcheroo(chip); chip->position_fix[0] = chip->position_fix[1] = @@ -3288,6 +3343,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); + azx_add_card_list(chip); return 0; -- cgit v1.2.3 From 8dea9d382a7aca9ebb48a80a34d7e1cf4ac8dcb2 Mon Sep 17 00:00:00 2001 From: Andi Kleen Date: Sat, 18 Aug 2012 19:56:22 -0700 Subject: ALSA: lto, sound: Fix export symbols for !CONFIG_MODULES The new LTO EXPORT_SYMBOL references symbols even without CONFIG_MODULES. Since these functions are macros in this case this doesn't work. Add a ifdef to fix the build. Signed-off-by: Andi Kleen Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 5cf8d65ed5ef..60e8fc1b3440 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -569,5 +569,7 @@ EXPORT_SYMBOL(snd_seq_device_load_drivers); EXPORT_SYMBOL(snd_seq_device_new); EXPORT_SYMBOL(snd_seq_device_register_driver); EXPORT_SYMBOL(snd_seq_device_unregister_driver); +#ifdef CONFIG_MODULES EXPORT_SYMBOL(snd_seq_autoload_lock); EXPORT_SYMBOL(snd_seq_autoload_unlock); +#endif -- cgit v1.2.3 From 099d53c308f50e8ee5b6638ec3f40f0104b3ee31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Aug 2012 18:04:40 +0200 Subject: ALSA: hda - Add missing ifdef CONFIG_SND_HDA_POWER_SAVE to tracepoints Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_trace.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index 9a34b867d8c2..d42fe911abae 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -58,6 +58,7 @@ TRACE_EVENT(hda_bus_reset, TP_printk("[%d]", __entry->card) ); +#ifdef CONFIG_SND_HDA_POWER_SAVE DECLARE_EVENT_CLASS(hda_power, TP_PROTO(struct hda_codec *codec), @@ -110,6 +111,7 @@ TRACE_EVENT(hda_power_count, __entry->card, __entry->addr, __entry->power_count, __entry->power_on, __entry->power_transition) ); +#endif /* CONFIG_SND_HDA_POWER_SAVE */ TRACE_EVENT(hda_unsol_event, -- cgit v1.2.3 From 1245b7005de02d5bfa0c321df925f5b6c83c99e1 Mon Sep 17 00:00:00 2001 From: Namarta Kohli Date: Thu, 16 Aug 2012 17:10:41 +0530 Subject: ASoC: add compress stream support This patch adds the support to parse the compress dai's and then also adds the soc-compress.c file while handles the compress stream operations, mostly analogus to what is done in the soc-pcm.c and aditional handling of the compress opertaions Signed-off-by: Namarta Kohli Signed-off-by: Ramesh Babu K V Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/soc-compress.c | 295 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 57 +++++---- 4 files changed, 331 insertions(+), 24 deletions(-) create mode 100644 sound/soc/soc-compress.c (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index c5de0a84566f..c24de902f5f5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,6 +9,7 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER + select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 00a555a743b6..c1264007b4ee 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c new file mode 100644 index 000000000000..88d85badd932 --- /dev/null +++ b/sound/soc/soc-compress.c @@ -0,0 +1,295 @@ +/* + * soc-compress.c -- ALSA SoC Compress + * + * Copyright (C) 2012 Intel Corp. + * + * Authors: Namarta Kohli + * Ramesh Babu K V + * Vinod Koul + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +static int soc_compr_open(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->open) { + ret = platform->driver->compr_ops->open(cstream); + if (ret < 0) { + pr_err("compress asoc: can't open platform %s\n", platform->name); + goto out; + } + } + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->startup) { + ret = rtd->dai_link->compr_ops->startup(cstream); + if (ret < 0) { + pr_err("compress asoc: %s startup failed\n", rtd->dai_link->name); + goto machine_err; + } + } + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + + cpu_dai->active++; + codec_dai->active++; + rtd->codec->active++; + + return 0; + +machine_err: + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); +out: + return ret; +} + +static int soc_compr_free(struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + snd_soc_dai_digital_mute(codec_dai, 1); + + cpu_dai->active--; + codec_dai->active--; + codec->active--; + + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; + + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->shutdown) + rtd->dai_link->compr_ops->shutdown(cstream); + + if (platform->driver->compr_ops && platform->driver->compr_ops->free) + platform->driver->compr_ops->free(cstream); + cpu_dai->runtime = NULL; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) { + if (!rtd->pmdown_time || codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time) { + snd_soc_dapm_stream_event(rtd, + SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_STOP); + } else + codec_dai->pop_wait = 1; + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); + } else { + /* capture streams can be powered down now */ + snd_soc_dapm_stream_event(rtd, + SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_STOP); + } + + return 0; +} + +static int soc_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->trigger) { + ret = platform->driver->compr_ops->trigger(cstream, cmd); + if (ret < 0) + return ret; + } + + if (cmd == SNDRV_PCM_TRIGGER_START) + snd_soc_dai_digital_mute(codec_dai, 0); + else if (cmd == SNDRV_PCM_TRIGGER_STOP) + snd_soc_dai_digital_mute(codec_dai, 1); + + return ret; +} + +static int soc_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret = 0; + + /* first we call set_params for the platform driver + * this should configure the soc side + * if the machine has compressed ops then we call that as well + * expectation is that platform and machine will configure everything + * for this compress path, like configuring pcm port for codec + */ + if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { + ret = platform->driver->compr_ops->set_params(cstream, params); + if (ret < 0) + return ret; + } + + if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { + ret = rtd->dai_link->compr_ops->set_params(cstream); + if (ret < 0) + return ret; + } + + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + + return ret; +} + +static int soc_compr_get_params(struct snd_compr_stream *cstream, + struct snd_codec *params) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_params) + ret = platform->driver->compr_ops->get_params(cstream, params); + + return ret; +} + +static int soc_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_caps) + ret = platform->driver->compr_ops->get_caps(cstream, caps); + + return ret; +} + +static int soc_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_codec_caps) + ret = platform->driver->compr_ops->get_codec_caps(cstream, codec); + + return ret; +} + +static int soc_compr_ack(struct snd_compr_stream *cstream, size_t bytes) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->ack) + ret = platform->driver->compr_ops->ack(cstream, bytes); + + return ret; +} + +static int soc_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + platform->driver->compr_ops->pointer(cstream, tstamp); + + return 0; +} + +/* ASoC Compress operations */ +static struct snd_compr_ops soc_compr_ops = { + .open = soc_compr_open, + .free = soc_compr_free, + .set_params = soc_compr_set_params, + .get_params = soc_compr_get_params, + .trigger = soc_compr_trigger, + .pointer = soc_compr_pointer, + .ack = soc_compr_ack, + .get_caps = soc_compr_get_caps, + .get_codec_caps = soc_compr_get_codec_caps +}; + +/* create a new compress */ +int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_compr *compr; + char new_name[64]; + int ret = 0, direction = 0; + + /* check client and interface hw capabilities */ + snprintf(new_name, sizeof(new_name), "%s %s-%d", + rtd->dai_link->stream_name, codec_dai->name, num); + direction = SND_COMPRESS_PLAYBACK; + compr = kzalloc(sizeof(*compr), GFP_KERNEL); + if (compr == NULL) { + snd_printk(KERN_ERR "Cannot allocate compr\n"); + return -ENOMEM; + } + + compr->ops = &soc_compr_ops; + mutex_init(&compr->lock); + ret = snd_compress_new(rtd->card->snd_card, num, direction, compr); + if (ret < 0) { + pr_err("compress asoc: can't create compress for codec %s\n", + codec->name); + kfree(compr); + return ret; + } + + rtd->compr = compr; + compr->private_data = rtd; + + printk(KERN_INFO "compress asoc: %s <-> %s mapping ok\n", codec_dai->name, + cpu_dai->name); + return ret; +} diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f585d0239551..c7a00fd8cc66 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1388,37 +1388,48 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (ret < 0) pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); - if (!dai_link->params) { - /* create the pcm */ - ret = soc_new_pcm(rtd, num); + if (cpu_dai->driver->compress_dai) { + /*create compress_device"*/ + ret = soc_new_compress(rtd, num); if (ret < 0) { - pr_err("asoc: can't create pcm %s :%d\n", - dai_link->stream_name, ret); + pr_err("asoc: can't create compress %s\n", + dai_link->stream_name); return ret; } } else { - /* link the DAI widgets */ - play_w = codec_dai->playback_widget; - capture_w = cpu_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); + + if (!dai_link->params) { + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); return ret; } - } + } else { + /* link the DAI widgets */ + play_w = codec_dai->playback_widget; + capture_w = cpu_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, + capture_w, play_w); + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } + } - play_w = cpu_dai->playback_widget; - capture_w = codec_dai->capture_widget; - if (play_w && capture_w) { - ret = snd_soc_dapm_new_pcm(card, dai_link->params, + play_w = cpu_dai->playback_widget; + capture_w = codec_dai->capture_widget; + if (play_w && capture_w) { + ret = snd_soc_dapm_new_pcm(card, dai_link->params, capture_w, play_w); - if (ret != 0) { - dev_err(card->dev, "Can't link %s to %s: %d\n", - play_w->name, capture_w->name, ret); - return ret; + if (ret != 0) { + dev_err(card->dev, "Can't link %s to %s: %d\n", + play_w->name, capture_w->name, ret); + return ret; + } } } } -- cgit v1.2.3 From c514a9119a982a6c7a9fd29ee62c0ba8a8e4c7d1 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 16 Aug 2012 17:10:42 +0530 Subject: ASoC: mid-x86 - add support for compressed streams Signed-off-by: Namarta Kohli Signed-off-by: Ramesh Babu K V Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 9 ++ sound/soc/mid-x86/sst_dsp.h | 134 +++++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.c | 204 ++++++++++++++++++++++++++++++++++++++- sound/soc/mid-x86/sst_platform.h | 26 ++++- 4 files changed, 371 insertions(+), 2 deletions(-) create mode 100644 sound/soc/mid-x86/sst_dsp.h (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 2937e54da49e..2cc7782714b5 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -318,6 +318,15 @@ static struct snd_soc_dai_link mfld_msic_dailink[] = { .platform_name = "sst-platform", .init = NULL, }, + { + .name = "Medfield Compress", + .stream_name = "Speaker", + .cpu_dai_name = "Compress-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, }; /* SoC card */ diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h new file mode 100644 index 000000000000..0fce1de284ff --- /dev/null +++ b/sound/soc/mid-x86/sst_dsp.h @@ -0,0 +1,134 @@ +#ifndef __SST_DSP_H__ +#define __SST_DSP_H__ +/* + * sst_dsp.h - Intel SST Driver for audio engine + * + * Copyright (C) 2008-12 Intel Corporation + * Authors: Vinod Koul + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +enum sst_codec_types { + /* AUDIO/MUSIC CODEC Type Definitions */ + SST_CODEC_TYPE_UNKNOWN = 0, + SST_CODEC_TYPE_PCM, /* Pass through Audio codec */ + SST_CODEC_TYPE_MP3, + SST_CODEC_TYPE_MP24, + SST_CODEC_TYPE_AAC, + SST_CODEC_TYPE_AACP, + SST_CODEC_TYPE_eAACP, +}; + +enum stream_type { + SST_STREAM_TYPE_NONE = 0, + SST_STREAM_TYPE_MUSIC = 1, +}; + +struct snd_pcm_params { + u16 codec; /* codec type */ + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 reserved; /* Bitrate in bits per second */ + u32 sfreq; /* Sampling rate in Hz */ + u8 use_offload_path; + u8 reserved2; + u16 reserved3; + u8 channel_map[8]; +} __packed; + +/* MP3 Music Parameters Message */ +struct snd_mp3_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 crc_check; /* crc_check - disable (0) or enable (1) */ + u8 reserved1; /* unused*/ + u16 reserved2; /* Unused */ +} __packed; + +#define AAC_BIT_STREAM_ADTS 0 +#define AAC_BIT_STREAM_ADIF 1 +#define AAC_BIT_STREAM_RAW 2 + +/* AAC Music Parameters Message */ +struct snd_aac_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo*/ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */ + u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */ + u16 reser2; + u32 externalsr; /*sampling rate of basic AAC raw bit stream*/ + u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/ + u8 reser1; + u16 reser3; +} __packed; + +/* WMA Music Parameters Message */ +struct snd_wma_params { + u16 codec; + u8 num_chan; /* 1=Mono, 2=Stereo */ + u8 pcm_wd_sz; /* 16/24 - bit*/ + u32 brate; /* Use the hard coded value. */ + u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ + u32 channel_mask; /* Channel Mask */ + u16 format_tag; /* Format Tag */ + u16 block_align; /* packet size */ + u16 wma_encode_opt;/* Encoder option */ + u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */ + u8 reserved; /* reserved */ +} __packed; + +/* Codec params struture */ +union snd_sst_codec_params { + struct snd_pcm_params pcm_params; + struct snd_mp3_params mp3_params; + struct snd_aac_params aac_params; + struct snd_wma_params wma_params; +} __packed; + +/* Address and size info of a frame buffer */ +struct sst_address_info { + u32 addr; /* Address at IA */ + u32 size; /* Size of the buffer */ +}; + +struct snd_sst_alloc_params_ext { + struct sst_address_info ring_buf_info[8]; + u8 sg_count; + u8 reserved; + u16 reserved2; + u32 frag_size; /*Number of samples after which period elapsed + message is sent valid only if path = 0*/ +} __packed; + +struct snd_sst_stream_params { + union snd_sst_codec_params uc; +} __packed; + +struct snd_sst_params { + u32 stream_id; + u8 codec; + u8 ops; + u8 stream_type; + u8 device_type; + struct snd_sst_stream_params sparams; + struct snd_sst_alloc_params_ext aparams; +}; + +#endif /* __SST_DSP_H__ */ diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index d34563b12c3b..a263cbed8624 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -1,7 +1,7 @@ /* * sst_platform.c - Intel MID Platform driver * - * Copyright (C) 2010 Intel Corp + * Copyright (C) 2010-2012 Intel Corp * Author: Vinod Koul * Author: Harsha Priya * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -32,6 +32,7 @@ #include #include #include +#include #include "sst_platform.h" static struct sst_device *sst; @@ -152,6 +153,16 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE, }, }, +{ + .name = "Compress-cpu-dai", + .compress_dai = 1, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, }; /* helper functions */ @@ -463,8 +474,199 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) } return retval; } + +/* compress stream operations */ +static void sst_compr_fragment_elapsed(void *arg) +{ + struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg; + + pr_debug("fragment elapsed by driver\n"); + if (cstream) + snd_compr_fragment_elapsed(cstream); +} + +static int sst_platform_compr_open(struct snd_compr_stream *cstream) +{ + + int ret_val = 0; + struct snd_compr_runtime *runtime = cstream->runtime; + struct sst_runtime_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + + /* get the sst ops */ + if (!sst || !try_module_get(sst->dev->driver->owner)) { + pr_err("no device available to run\n"); + ret_val = -ENODEV; + goto out_ops; + } + stream->compr_ops = sst->compr_ops; + + stream->id = 0; + sst_set_stream_status(stream, SST_PLATFORM_INIT); + runtime->private_data = stream; + return 0; +out_ops: + kfree(stream); + return ret_val; +} + +static int sst_platform_compr_free(struct snd_compr_stream *cstream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + stream = cstream->runtime->private_data; + /*need to check*/ + str_id = stream->id; + if (str_id) + ret_val = stream->compr_ops->close(str_id); + module_put(sst->dev->driver->owner); + kfree(stream); + pr_debug("%s: %d\n", __func__, ret_val); + return 0; +} + +static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params) +{ + struct sst_runtime_stream *stream; + int retval; + struct snd_sst_params str_params; + struct sst_compress_cb cb; + + stream = cstream->runtime->private_data; + /* construct fw structure for this*/ + memset(&str_params, 0, sizeof(str_params)); + + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.stream_type = SST_STREAM_TYPE_MUSIC; + str_params.device_type = SND_SST_DEVICE_COMPRESS; + + switch (params->codec.id) { + case SND_AUDIOCODEC_MP3: { + str_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3; + str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.mp3_params.pcm_wd_sz = 16; + break; + } + + case SND_AUDIOCODEC_AAC: { + str_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC; + str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in; + str_params.sparams.uc.aac_params.pcm_wd_sz = 16; + if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_ADTS; + else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW) + str_params.sparams.uc.aac_params.bs_format = + AAC_BIT_STREAM_RAW; + else { + pr_err("Undefined format%d\n", params->codec.format); + return -EINVAL; + } + str_params.sparams.uc.aac_params.externalsr = + params->codec.sample_rate; + break; + } + + default: + pr_err("codec not supported, id =%d\n", params->codec.id); + return -EINVAL; + } + + str_params.aparams.ring_buf_info[0].addr = + virt_to_phys(cstream->runtime->buffer); + str_params.aparams.ring_buf_info[0].size = + cstream->runtime->buffer_size; + str_params.aparams.sg_count = 1; + str_params.aparams.frag_size = cstream->runtime->fragment_size; + + cb.param = cstream; + cb.compr_cb = sst_compr_fragment_elapsed; + + retval = stream->compr_ops->open(&str_params, &cb); + if (retval < 0) { + pr_err("stream allocation failed %d\n", retval); + return retval; + } + + stream->id = retval; + return 0; +} + +static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->control(cmd, stream->id); +} + +static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->tstamp(stream->id, tstamp); + tstamp->byte_offset = tstamp->copied_total % + (u32)cstream->runtime->buffer_size; + pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); + return 0; +} + +static int sst_platform_compr_ack(struct snd_compr_stream *cstream, + size_t bytes) +{ + struct sst_runtime_stream *stream; + + stream = cstream->runtime->private_data; + stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->bytes_written += bytes; + + return 0; +} + +static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_caps(caps); +} + +static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->get_codec_caps(codec); +} + +static struct snd_compr_ops sst_platform_compr_ops = { + + .open = sst_platform_compr_open, + .free = sst_platform_compr_free, + .set_params = sst_platform_compr_set_params, + .trigger = sst_platform_compr_trigger, + .pointer = sst_platform_compr_pointer, + .ack = sst_platform_compr_ack, + .get_caps = sst_platform_compr_get_caps, + .get_codec_caps = sst_platform_compr_get_codec_caps, +}; + static struct snd_soc_platform_driver sst_soc_platform_drv = { .ops = &sst_platform_ops, + .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, .pcm_free = sst_pcm_free, }; diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index f04f4f72daa0..d61c5d514ffa 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -27,6 +27,8 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ +#include "sst_dsp.h" + #define SST_MONO 1 #define SST_STEREO 2 #define SST_MAX_CAP 5 @@ -42,7 +44,6 @@ #define SST_MIN_PERIODS 2 #define SST_MAX_PERIODS (1024*2) #define SST_FIFO_SIZE 0 -#define SST_CODEC_TYPE_PCM 1 struct pcm_stream_info { int str_id; @@ -83,6 +84,7 @@ enum sst_audio_device_type { SND_SST_DEVICE_VIBRA, SND_SST_DEVICE_HAPTIC, SND_SST_DEVICE_CAPTURE, + SND_SST_DEVICE_COMPRESS, }; /* PCM Parameters */ @@ -107,6 +109,24 @@ struct sst_stream_params { struct sst_pcm_params sparams; }; +struct sst_compress_cb { + void *param; + void (*compr_cb)(void *param); +}; + +struct compress_sst_ops { + const char *name; + int (*open) (struct snd_sst_params *str_params, + struct sst_compress_cb *cb); + int (*control) (unsigned int cmd, unsigned int str_id); + int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); + int (*ack) (unsigned int str_id, unsigned long bytes); + int (*close) (unsigned int str_id); + int (*get_caps) (struct snd_compr_caps *caps); + int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + +}; + struct sst_ops { int (*open) (struct sst_stream_params *str_param); int (*device_control) (int cmd, void *arg); @@ -115,8 +135,11 @@ struct sst_ops { struct sst_runtime_stream { int stream_status; + unsigned int id; + size_t bytes_written; struct pcm_stream_info stream_info; struct sst_ops *ops; + struct compress_sst_ops *compr_ops; spinlock_t status_lock; }; @@ -124,6 +147,7 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct compress_sst_ops *compr_ops; }; int sst_register_dsp(struct sst_device *sst); -- cgit v1.2.3 From f9933487468c760b8cd9b4e9f7ec4e494f711a0a Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 20 Aug 2012 22:39:51 +0200 Subject: ALSA: introduce snd-cmi8328: C-Media CMI8328 driver Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). A separate driver is needed because CMI8328 is completely different chip to CMI8329/CMI8330. It's configured by magic registers (there's no PnP). Sound is provided by a real WSS codec (CS4231A) and the SB part is just a SB Pro emulation (for DOS games, useless for Linux). When SB is enabled, the CMI8328 chip disables access to the WSS codec, emulates SoundBlaster on one side and outputs sound data to the codec - so SB and WSS can't work together with this card. The WSS codec can do full duplex by itself so there's no need for crazy things like snd-cmi8330 does (combining SB and WSS parts into one driver). Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 12 ++ sound/isa/Makefile | 2 + sound/isa/cmi8328.c | 482 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 496 insertions(+) create mode 100644 sound/isa/cmi8328.c (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 52064cfa91f3..a38d9643e9d8 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -117,6 +117,18 @@ config SND_AZT2320 To compile this driver as a module, choose M here: the module will be called snd-azt2320. +config SND_CMI8328 + tristate "C-Media CMI8328" + select SND_WSS_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for soundcards based on the + C-Media CMI8328 chip. + + To compile this driver as a module, choose M here: the module + will be called snd-cmi8328. + config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 8d781e419e2e..9a15f1497b10 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -6,6 +6,7 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o +snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o @@ -16,6 +17,7 @@ snd-sscape-objs := sscape.o obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o +obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c new file mode 100644 index 000000000000..aefafff50756 --- /dev/null +++ b/sound/isa/cmi8328.c @@ -0,0 +1,482 @@ +/* + * Driver for C-Media CMI8328-based soundcards, such as AudioExcel AV500 + * Copyright (c) 2012 Ondrej Zary + * + * AudioExcel AV500 card consists of: + * - CMI8328 - main chip (SB Pro emulation, gameport, OPL3, MPU401, CD-ROM) + * - CS4231A - WSS codec + * - Dream SAM9233+GMS950400+RAM+ROM: Wavetable MIDI, connected to MPU401 + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#define SNDRV_LEGACY_FIND_FREE_IOPORT +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include + +MODULE_AUTHOR("Ondrej Zary "); +MODULE_DESCRIPTION("C-Media CMI8328"); +MODULE_LICENSE("GPL"); + +#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#define SUPPORT_JOYSTICK 1 +#endif + +/* I/O port is configured by jumpers on the card to one of these */ +static int cmi8328_ports[] = { 0x530, 0xe80, 0xf40, 0x604 }; +#define CMI8328_MAX ARRAY_SIZE(cmi8328_ports) + +static int index[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = -1}; +static char *id[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = NULL}; +static long port[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int irq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +static int dma1[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static int dma2[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static long mpuport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int mpuirq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +#ifdef SUPPORT_JOYSTICK +static bool gameport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = true}; +#endif + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); + +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); +#ifdef SUPPORT_JOYSTICK +module_param_array(gameport, bool, NULL, 0444); +MODULE_PARM_DESC(gameport, "Enable gameport."); +#endif + +struct snd_cmi8328 { + u16 port; + u8 cfg[3]; + u8 wss_cfg; + struct snd_card *card; + struct snd_wss *wss; +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif +}; + +/* CMI8328 configuration registers */ +#define CFG1 0x61 +#define CFG1_SB_DISABLE (1 << 0) +#define CFG1_GAMEPORT (1 << 1) +/* + * bit 0: SB: 0=enabled, 1=disabled + * bit 1: gameport: 0=disabled, 1=enabled + * bits 2-4: SB IRQ: 001=3, 010=5, 011=7, 100=9, 101=10, 110=11 + * bits 5-6: SB DMA: 00=disabled (when SB disabled), 01=DMA0, 10=DMA1, 11=DMA3 + * bit 7: SB port: 0=0x220, 1=0x240 + */ +#define CFG2 0x62 +#define CFG2_MPU_ENABLE (1 << 2) +/* + * bits 0-1: CD-ROM mode: 00=disabled, 01=Panasonic, 10=Sony/Mitsumi/Wearnes, + 11=IDE + * bit 2: MPU401: 0=disabled, 1=enabled + * bits 3-4: MPU401 IRQ: 00=3, 01=5, 10=7, 11=9, + * bits 5-7: MPU401 port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x332, + 101=0x334, 110=0x336 + */ +#define CFG3 0x63 +/* + * bits 0-2: CD-ROM IRQ: 000=disabled, 001=3, 010=5, 011=7, 100=9, 101=10, + 110=11 + * bits 3-4: CD-ROM DMA: 00=disabled, 01=DMA0, 10=DMA1, 11=DMA3 + * bits 5-7: CD-ROM port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x340, + 101=0x350, 110=0x360, 111=0x370 + */ + +static u8 snd_cmi8328_cfg_read(u16 port, u8 reg) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + return inb(port); +} + +static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + outb(val, port + 3); /* yes, value goes to the same port as index */ +} + +static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) +{ + cfg[0] = snd_cmi8328_cfg_read(port, CFG1); + cfg[1] = snd_cmi8328_cfg_read(port, CFG2); + cfg[2] = snd_cmi8328_cfg_read(port, CFG3); +} + +static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) +{ + snd_cmi8328_cfg_write(port, CFG1, cfg[0]); + snd_cmi8328_cfg_write(port, CFG2, cfg[1]); + snd_cmi8328_cfg_write(port, CFG3, cfg[2]); +} + +static int __devinit snd_cmi8328_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + struct snd_ctl_elem_id id1, id2; + int err; + + card = chip->card; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* rename AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX0 volume to CD */ + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 switch to Synth */ + strcpy(id1.name, "Aux Playback Switch"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 volume to Synth */ + strcpy(id1.name, "Aux Playback Volume"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + + return 0; +} + +/* find index of an item in "-1"-ended array */ +int array_find(int array[], int item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} +/* the same for long */ +int array_find_l(long array[], long item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} + +static int __devinit snd_cmi8328_probe(struct device *pdev, unsigned int ndev) +{ + struct snd_card *card; + struct snd_opl3 *opl3; + struct snd_cmi8328 *cmi; +#ifdef SUPPORT_JOYSTICK + struct resource *res; +#endif + int err, pos; + static long mpu_ports[] = { 0x330, 0x300, 0x310, 0x320, 0x332, 0x334, + 0x336, -1 }; + static u8 mpu_port_bits[] = { 3, 0, 1, 2, 4, 5, 6 }; + static int mpu_irqs[] = { 9, 7, 5, 3, -1 }; + static u8 mpu_irq_bits[] = { 3, 2, 1, 0 }; + static int irqs[] = { 9, 10, 11, 7, -1 }; + static u8 irq_bits[] = { 2, 3, 4, 1 }; + static int dma1s[] = { 3, 1, 0, -1 }; + static u8 dma_bits[] = { 3, 2, 1 }; + static int dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, {0, -1} }; + u16 port = cmi8328_ports[ndev]; + u8 val; + + /* 0xff is invalid configuration (but settable - hope it isn't set) */ + if (snd_cmi8328_cfg_read(port, CFG1) == 0xff) + return -ENODEV; + /* the SB disable bit must NEVER EVER be cleared or the WSS dies */ + snd_cmi8328_cfg_write(port, CFG1, CFG1_SB_DISABLE); + if (snd_cmi8328_cfg_read(port, CFG1) != CFG1_SB_DISABLE) + return -ENODEV; + /* disable everything first */ + snd_cmi8328_cfg_write(port, CFG2, 0); /* disable CDROM and MPU401 */ + snd_cmi8328_cfg_write(port, CFG3, 0); /* disable CDROM IRQ and DMA */ + + if (irq[ndev] == SNDRV_AUTO_IRQ) { + irq[ndev] = snd_legacy_find_free_irq(irqs); + if (irq[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (dma1[ndev] == SNDRV_AUTO_DMA) { + dma1[ndev] = snd_legacy_find_free_dma(dma1s); + if (dma1[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2[ndev] == SNDRV_AUTO_DMA) { + dma2[ndev] = snd_legacy_find_free_dma(dma2s[dma1[ndev] % 4]); + if (dma2[ndev] < 0) { + snd_printk(KERN_WARNING "unable to find a free DMA2, full-duplex will not work\n"); + dma2[ndev] = -1; + } + } + /* configure WSS IRQ... */ + pos = array_find(irqs, irq[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid IRQ %d\n", irq[ndev]); + return -EINVAL; + } + val = irq_bits[pos] << 3; + /* ...and DMA... */ + pos = array_find(dma1s, dma1[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA1 %d\n", dma1[ndev]); + return -EINVAL; + } + val |= dma_bits[pos]; + /* ...and DMA2 */ + if (dma2[ndev] >= 0 && dma1[ndev] != dma2[ndev]) { + pos = array_find(dma2s[dma1[ndev]], dma2[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA2 %d\n", dma2[ndev]); + return -EINVAL; + } + val |= 0x04; /* enable separate capture DMA */ + } + outb(val, port); + + err = snd_card_create(index[ndev], id[ndev], THIS_MODULE, + sizeof(struct snd_cmi8328), &card); + if (err < 0) + return err; + cmi = card->private_data; + cmi->card = card; + cmi->port = port; + cmi->wss_cfg = val; + snd_card_set_dev(card, pdev); + + err = snd_wss_create(card, port + 4, -1, irq[ndev], dma1[ndev], + dma2[ndev], WSS_HW_DETECT, 0, &cmi->wss); + if (err < 0) + goto error; + + err = snd_wss_pcm(cmi->wss, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(cmi->wss); + if (err < 0) + goto error; + err = snd_cmi8328_mixer(cmi->wss); + if (err < 0) + goto error; + + if (snd_wss_timer(cmi->wss, 0, NULL) < 0) + snd_printk(KERN_WARNING "error initializing WSS timer\n"); + + if (mpuport[ndev] == SNDRV_AUTO_PORT) { + mpuport[ndev] = snd_legacy_find_free_ioport(mpu_ports, 2); + if (mpuport[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + } + if (mpuirq[ndev] == SNDRV_AUTO_IRQ) { + mpuirq[ndev] = snd_legacy_find_free_irq(mpu_irqs); + if (mpuirq[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + } + /* enable and configure MPU401 */ + if (mpuport[ndev] > 0 && mpuirq[ndev] > 0) { + val = CFG2_MPU_ENABLE; + pos = array_find_l(mpu_ports, mpuport[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 port 0x%lx\n", + mpuport[ndev]); + else { + val |= mpu_port_bits[pos] << 5; + pos = array_find(mpu_irqs, mpuirq[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 IRQ %d\n", + mpuirq[ndev]); + else { + val |= mpu_irq_bits[pos] << 3; + snd_cmi8328_cfg_write(port, CFG2, val); + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, mpuport[ndev], + 0, mpuirq[ndev], NULL) < 0) + snd_printk(KERN_ERR "error initializing MPU401\n"); + } + } + } + /* OPL3 is hardwired to 0x388 and cannot be disabled */ + if (snd_opl3_create(card, 0x388, 0x38a, OPL3_HW_AUTO, 0, &opl3) < 0) + snd_printk(KERN_ERR "error initializing OPL3\n"); + else + if (snd_opl3_hwdep_new(opl3, 0, 1, NULL) < 0) + snd_printk(KERN_WARNING "error initializing OPL3 hwdep\n"); + + strcpy(card->driver, "CMI8328"); + strcpy(card->shortname, "C-Media CMI8328"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d,%d", + card->shortname, cmi->wss->port, irq[ndev], dma1[ndev], + (dma2[ndev] >= 0) ? dma2[ndev] : dma1[ndev]); + + dev_set_drvdata(pdev, card); + err = snd_card_register(card); + if (err < 0) + goto error; +#ifdef SUPPORT_JOYSTICK + if (!gameport[ndev]) + return 0; + /* gameport is hardwired to 0x200 */ + res = request_region(0x200, 8, "CMI8328 gameport"); + if (!res) + snd_printk(KERN_WARNING "unable to allocate gameport I/O port\n"); + else { + struct gameport *gp = cmi->gameport = gameport_allocate_port(); + if (!cmi->gameport) + release_and_free_resource(res); + else { + gameport_set_name(gp, "CMI8328 Gameport"); + gameport_set_phys(gp, "%s/gameport0", dev_name(pdev)); + gameport_set_dev_parent(gp, pdev); + gp->io = 0x200; + gameport_set_port_data(gp, res); + /* Enable gameport */ + snd_cmi8328_cfg_write(port, CFG1, + CFG1_SB_DISABLE | CFG1_GAMEPORT); + gameport_register_port(gp); + } + } +#endif + return 0; +error: + snd_card_free(card); + + return err; +} + +static int __devexit snd_cmi8328_remove(struct device *pdev, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(pdev); +#ifdef SUPPORT_JOYSTICK + struct snd_cmi8328 *cmi = card->private_data; + if (cmi->gameport) { + struct resource *res = gameport_get_port_data(cmi->gameport); + gameport_unregister_port(cmi->gameport); + release_and_free_resource(res); + } +#endif + /* disable everything */ + snd_cmi8328_cfg_write(cmi->port, CFG1, CFG1_SB_DISABLE); + snd_cmi8328_cfg_write(cmi->port, CFG2, 0); + snd_cmi8328_cfg_write(cmi->port, CFG3, 0); + snd_card_free(card); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#ifdef CONFIG_PM +static int snd_cmi8328_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_save(cmi->port, cmi->cfg); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(cmi->wss->pcm); + cmi->wss->suspend(cmi->wss); + + return 0; +} + +static int snd_cmi8328_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_restore(cmi->port, cmi->cfg); + outb(cmi->wss_cfg, cmi->port); + cmi->wss->resume(cmi->wss); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif + +static struct isa_driver snd_cmi8328_driver = { + .probe = snd_cmi8328_probe, + .remove = __devexit_p(snd_cmi8328_remove), +#ifdef CONFIG_PM + .suspend = snd_cmi8328_suspend, + .resume = snd_cmi8328_resume, +#endif + .driver = { + .name = "cmi8328" + }, +}; + +static int __init alsa_card_cmi8328_init(void) +{ + return isa_register_driver(&snd_cmi8328_driver, CMI8328_MAX); +} + +static void __exit alsa_card_cmi8328_exit(void) +{ + isa_unregister_driver(&snd_cmi8328_driver); +} + +module_init(alsa_card_cmi8328_init) +module_exit(alsa_card_cmi8328_exit) -- cgit v1.2.3 From 9d2667a910b8d889a8307bd2534032eb6ed0ea36 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 21 Aug 2012 12:15:02 +0530 Subject: ASoC: compress - fix code alignment Reported-by: Fengguang Wu Signed-off-by: Namarta Kohli Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 88d85badd932..cc0562d28df2 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -104,7 +104,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->free) platform->driver->compr_ops->free(cstream); - cpu_dai->runtime = NULL; + cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (!rtd->pmdown_time || codec->ignore_pmdown_time || -- cgit v1.2.3 From 1be437fa53609da9a796a4d213d3fc0a17fc03e0 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 21 Aug 2012 12:20:22 +0530 Subject: ASoC: soc-compress: Remove unused variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit codec_dai is not used in the function. sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: sound/soc/soc-compress.c:156:22: warning: unused variable ‘codec_dai’ [-Wunused-variable] Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index cc0562d28df2..967d0e173e1b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -153,7 +153,6 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; /* first we call set_params for the platform driver -- cgit v1.2.3 From 8a5354140a86b6d4057793a9ed28d29ac8ce6ba6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Aug 2012 16:40:24 +0200 Subject: ALSA: hda - Call snd_hda_jack_report_sync() generically in hda_codec.c Instead of calling the jack sync in the init callback of each codec, call it generically at initialization and resume. By calling it at the last of resume sequence, a possible race between the jack sync and the unsol event enablement in the current code will be closed, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_cirrus.c | 2 -- sound/pci/hda/patch_conexant.c | 1 - sound/pci/hda/patch_hdmi.c | 2 -- sound/pci/hda/patch_realtek.c | 2 -- sound/pci/hda/patch_sigmatel.c | 2 -- sound/pci/hda/patch_via.c | 1 - 7 files changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a6c34dc6dac1..4efd2711acd9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3618,6 +3618,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); } + snd_hda_jack_report_sync(codec); snd_hda_power_down(codec); /* flag down before returning */ } #endif /* CONFIG_PM */ @@ -3663,6 +3664,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) err = codec->patch_ops.build_controls(codec); if (err < 0) return err; + snd_hda_jack_report_sync(codec); /* call at the last init point */ return 0; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0c4c1a61b378..0bddb3ee1101 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1193,7 +1193,6 @@ static int cs_init(struct hda_codec *codec) init_output(codec); init_input(codec); init_digital(codec); - snd_hda_jack_report_sync(codec); return 0; } @@ -1643,7 +1642,6 @@ static int cs421x_init(struct hda_codec *codec) init_output(codec); init_input(codec); init_cs421x_digital(codec); - snd_hda_jack_report_sync(codec); return 0; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f43d2e..172895a8add4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4061,7 +4061,6 @@ static int cx_auto_init(struct hda_codec *codec) cx_auto_init_output(codec); cx_auto_init_input(codec); cx_auto_init_digital(codec); - snd_hda_jack_report_sync(codec); snd_hda_sync_vmaster_hook(&spec->vmaster_mute); return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8f23374fa642..d9439c59bb8b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1311,7 +1311,6 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } - snd_hda_jack_report_sync(codec); return 0; } @@ -1428,7 +1427,6 @@ static int simple_playback_init(struct hda_codec *codec) snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); snd_hda_jack_detect_enable(codec, pin, pin); - snd_hda_jack_report_sync(codec); return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f81dd44c837..ce99cc955326 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2053,8 +2053,6 @@ static int alc_init(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); - snd_hda_jack_report_sync(codec); - hda_call_check_power_status(codec, 0x01); return 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea5775a1a7db..435295402efa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4418,8 +4418,6 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 0); } - snd_hda_jack_report_sync(codec); - /* sync mute LED */ if (spec->gpio_led) { if (spec->vmaster_mute.hook) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 430771776915..4b0796b30f78 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2815,7 +2815,6 @@ static int via_init(struct hda_codec *codec) via_hp_automute(codec); vt1708_update_hp_work(spec); - snd_hda_jack_report_sync(codec); return 0; } -- cgit v1.2.3 From 3eadd88a37c330a83bfdee35b3e5837b7c2f7214 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Aug 2012 17:30:13 +0100 Subject: ASoC: wm9712: Provide TLV information for capture boost controls Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index c9c696ca76f7..6aa1bf8c6897 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -132,8 +132,9 @@ SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), -SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), +SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv), +SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1, + boost_tlv), SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), -- cgit v1.2.3 From 363947d7d999f74dfd710fb7b7ccad965590f098 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Aug 2012 19:54:24 +0100 Subject: ASoC: wm_hubs: Use explicit casts for converting to signed Should be no behaviour change. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 05a02e1b7e92..b340552efe4e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -297,12 +297,12 @@ static void enable_dc_servo(struct snd_soc_codec *codec) hubs->dcs_codes_l, hubs->dcs_codes_r); /* HPOUT1R */ - offset = reg_r; + offset = (s8)reg_r; offset += hubs->dcs_codes_r; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1L */ - offset = reg_l; + offset = (s8)reg_l; offset += hubs->dcs_codes_l; dcs_cfg |= (u8)offset; -- cgit v1.2.3 From 20bac1f3f470e2d5c87af7b41b10e088e47989bb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 20 Aug 2012 20:01:51 +0100 Subject: ASoC: wm_hubs: Add trace showing semantics of the DCS update Aids diagnostics. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index b340552efe4e..b2e939a8970e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -298,11 +298,15 @@ static void enable_dc_servo(struct snd_soc_codec *codec) /* HPOUT1R */ offset = (s8)reg_r; + dev_dbg(codec->dev, "DCS right %d->%d\n", offset, + offset + hubs->dcs_codes_r); offset += hubs->dcs_codes_r; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1L */ offset = (s8)reg_l; + dev_dbg(codec->dev, "DCS left %d->%d\n", offset, + offset + hubs->dcs_codes_l); offset += hubs->dcs_codes_l; dcs_cfg |= (u8)offset; -- cgit v1.2.3 From 02e79476998ba7e62842d20dca898c403ad55c7e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Aug 2012 17:54:52 +0100 Subject: ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdata Sometimes the analogue circuitry connected to the microphone needs some time to settle after power up. Allow systems to configure this delay in the platform data, the driver will then insert the required delay during power up of paths that involve the microphone. Signed-off-by: Mark Brown --- include/linux/mfd/wm8994/pdata.h | 4 ++++ include/sound/wm8993.h | 4 ++++ sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 35 +++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_hubs.h | 4 ++++ 6 files changed, 47 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index f0361c031927..fc87be4fdc25 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -164,6 +164,10 @@ struct wm8994_pdata { int num_micd_rates; struct wm8958_micd_rate *micd_rates; + /* Power up delays to add after microphone bias power up (ms) */ + int micb1_delay; + int micb2_delay; + /* LINEOUT can be differential or single ended */ unsigned int lineout1_diff:1; unsigned int lineout2_diff:1; diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h index eee19f63c0d8..8016fd826f5a 100644 --- a/include/sound/wm8993.h +++ b/include/sound/wm8993.h @@ -32,6 +32,10 @@ struct wm8993_platform_data { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* Delay to add for microphones to stabalise after power up */ + int micbias1_delay; + int micbias2_delay; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9fd80d688979..94737a30716b 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1520,6 +1520,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.lineout2fb, wm8993->pdata.jd_scthr, wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_delay, + wm8993->pdata.micbias2_delay, wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 353612eec8bf..b74df52d2820 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3145,6 +3145,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) pdata->lineout2fb, pdata->jd_scthr, pdata->jd_thr, + pdata->micb1_delay, + pdata->micb2_delay, pdata->micbias1_lvl, pdata->micbias2_lvl); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index b2e939a8970e..7a773a835b8e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -644,6 +644,28 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + switch (w->shift) { + case WM8993_MICB1_ENA_SHIFT: + if (hubs->micb1_delay) + msleep(hubs->micb1_delay); + break; + case WM8993_MICB2_ENA_SHIFT: + if (hubs->micb2_delay) + msleep(hubs->micb2_delay); + break; + default: + return -EINVAL; + } + + return 0; +} + void wm_hubs_update_class_w(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); @@ -834,8 +856,10 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), @@ -1170,13 +1194,16 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, - int jd_scthr, int jd_thr, int micbias1_lvl, - int micbias2_lvl) + int jd_scthr, int jd_thr, + int micbias1_delay, int micbias2_delay, + int micbias1_lvl, int micbias2_lvl) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); hubs->lineout1_se = !lineout1_diff; hubs->lineout2_se = !lineout2_diff; + hubs->micb1_delay = micbias1_delay; + hubs->micb2_delay = micbias2_delay; if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index a5a09e6f87d5..24c763df21f9 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -36,6 +36,9 @@ struct wm_hubs_data { struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); + int micb1_delay; + int micb2_delay; + bool lineout1_se; bool lineout1n_ena; bool lineout1p_ena; @@ -56,6 +59,7 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, int jd_scthr, int jd_thr, + int micbias1_dly, int micbias2_dly, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); -- cgit v1.2.3 From f199131a8f1dde3309c8520b663aa7a3daf5995e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:00 +0300 Subject: ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver Move the McBSP CLKS re-parenting code to ASoC driver from arch/arm/mach-omap2. The call fort the re-parenting has been already limited to OMAP2+ SoC in the ASoC driver. There is no longer need to have callback function for it. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Tony Lindgren Signed-off-by: Mark Brown --- arch/arm/mach-omap2/mcbsp.c | 40 --------------------------------- arch/arm/plat-omap/include/plat/mcbsp.h | 1 - sound/soc/omap/mcbsp.c | 31 ++++++++++++++++++++----- 3 files changed, 26 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c index 577cb77db26c..ebc801ea5be9 100644 --- a/arch/arm/mach-omap2/mcbsp.c +++ b/arch/arm/mach-omap2/mcbsp.c @@ -101,45 +101,6 @@ static int omap4_mcbsp4_mux_rx_clk(struct device *dev, const char *signal, return 0; } -/* McBSP CLKS source switching function */ -static int omap2_mcbsp_set_clk_src(struct device *dev, struct clk *clk, - const char *src) -{ - struct clk *fck_src; - char *fck_src_name; - int r; - - if (!strcmp(src, "clks_ext")) - fck_src_name = "pad_fck"; - else if (!strcmp(src, "clks_fclk")) - fck_src_name = "prcm_fck"; - else - return -EINVAL; - - fck_src = clk_get(dev, fck_src_name); - if (IS_ERR_OR_NULL(fck_src)) { - pr_err("omap-mcbsp: %s: could not clk_get() %s\n", "clks", - fck_src_name); - return -EINVAL; - } - - pm_runtime_put_sync(dev); - - r = clk_set_parent(clk, fck_src); - if (IS_ERR_VALUE(r)) { - pr_err("omap-mcbsp: %s: could not clk_set_parent() to %s\n", - "clks", fck_src_name); - clk_put(fck_src); - return -EINVAL; - } - - pm_runtime_get_sync(dev); - - clk_put(fck_src); - - return 0; -} - static int omap3_enable_st_clock(unsigned int id, bool enable) { unsigned int w; @@ -181,7 +142,6 @@ static int __init omap_init_mcbsp(struct omap_hwmod *oh, void *unused) pdata->reg_size = 4; pdata->has_ccr = true; } - pdata->set_clk_src = omap2_mcbsp_set_clk_src; /* On OMAP2/3 the McBSP1 port has 6 pin configuration */ if (id == 1 && oh->class->rev < MCBSP_CONFIG_TYPE4) diff --git a/arch/arm/plat-omap/include/plat/mcbsp.h b/arch/arm/plat-omap/include/plat/mcbsp.h index 18814127809a..0a7d5ca471e0 100644 --- a/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/arch/arm/plat-omap/include/plat/mcbsp.h @@ -47,7 +47,6 @@ struct omap_mcbsp_platform_data { bool has_wakeup; /* Wakeup capability */ bool has_ccr; /* Transceiver has configuration control registers */ int (*enable_st_clock)(unsigned int, bool); - int (*set_clk_src)(struct device *dev, struct clk *clk, const char *src); int (*mux_signal)(struct device *dev, const char *signal, const char *src); }; diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 34835e8a9160..6afbc26cef70 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -24,6 +24,7 @@ #include #include #include +#include #include @@ -726,19 +727,39 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) { + struct clk *fck_src; const char *src; + int r; if (fck_src_id == MCBSP_CLKS_PAD_SRC) - src = "clks_ext"; + src = "pad_fck"; else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) - src = "clks_fclk"; + src = "prcm_fck"; else return -EINVAL; - if (mcbsp->pdata->set_clk_src) - return mcbsp->pdata->set_clk_src(mcbsp->dev, mcbsp->fclk, src); - else + fck_src = clk_get(mcbsp->dev, src); + if (IS_ERR(fck_src)) { + dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src); return -EINVAL; + } + + pm_runtime_put_sync(mcbsp->dev); + + r = clk_set_parent(mcbsp->fclk, fck_src); + if (r) { + dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n", + src); + clk_put(fck_src); + return r; + } + + pm_runtime_get_sync(mcbsp->dev); + + clk_put(fck_src); + + return 0; + } int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -- cgit v1.2.3 From fca04aea36a82bc451ddc33dee6f9a48bb0f8a2a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:03 +0300 Subject: ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing The muxing is done at board level, no need to do it in the ASoC machine driver. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 20 ++------------------ 1 file changed, 2 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 009533ab8d18..a997988af147 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -47,26 +47,10 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, /* Set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret < 0) { + if (ret < 0) printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n"); - return ret; - } - snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, - SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); - return ret; - } - - return 0; + return ret; } static struct snd_soc_ops am3517evm_ops = { -- cgit v1.2.3 From 8fef6263ea68f6160637f370a5864d0a455c620d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:04 +0300 Subject: ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code Remove the feature to configure the CLKR/FSR mux on McBSP port with 6pin configuration. When moving to devicetree these callback can no longer be used in a clean way anymore. If a board require to change the 6pin port to work in 4pin setup it needs to set up the mux in the board file. For OMAP2/3: u32 devconf0; /* McBSP1 CLKR/FSR signal to be connected to CLKX/FSX pin */ devconf0 = omap_ctrl_readl(OMAP2_CONTROL_DEVCONF0); devconf0 |= OMAP2_MCBSP1_CLKR_MASK | OMAP2_MCBSP1_FSR_MASK; omap_ctrl_writel(devconf0, OMAP2_CONTROL_DEVCONF0); For OMAP4: u32 mcbsp_pad; /* McBSP4 CLKR/FSR signal to be connected to CLKX/FSX pin */ mcbsp_pad = omap4_ctrl_pad_readl(OMAP2_CONTROL_DEVCONF0); mcbsp_pad |= ((1 << 31) | (1 << 30)); omap4_ctrl_pad_writel(mcbsp_pad, OMAP2_CONTROL_DEVCONF0); In case when the kernel is booted with DT blob the pinctrl-single will be provided as soon as it is enabled on the platform. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Tony Lindgren Signed-off-by: Mark Brown --- arch/arm/mach-omap2/mcbsp.c | 77 --------------------------------- arch/arm/plat-omap/include/plat/mcbsp.h | 1 - sound/soc/omap/mcbsp.h | 3 -- sound/soc/omap/omap-mcbsp.c | 32 ++------------ sound/soc/omap/omap-mcbsp.h | 4 -- 5 files changed, 3 insertions(+), 114 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c index 6e046e1111b1..660e00b3ef82 100644 --- a/arch/arm/mach-omap2/mcbsp.c +++ b/arch/arm/mach-omap2/mcbsp.c @@ -25,8 +25,6 @@ #include #include -#include "control.h" - /* * FIXME: Find a mechanism to enable/disable runtime the McBSP ICLK autoidle. * Sidetone needs non-gated ICLK and sidetone autoidle is broken. @@ -34,73 +32,6 @@ #include "cm2xxx_3xxx.h" #include "cm-regbits-34xx.h" -/* McBSP1 internal signal muxing function for OMAP2/3 */ -static int omap2_mcbsp1_mux_rx_clk(struct device *dev, const char *signal, - const char *src) -{ - u32 v; - - v = omap_ctrl_readl(OMAP2_CONTROL_DEVCONF0); - - if (!strcmp(signal, "clkr")) { - if (!strcmp(src, "clkr")) - v &= ~OMAP2_MCBSP1_CLKR_MASK; - else if (!strcmp(src, "clkx")) - v |= OMAP2_MCBSP1_CLKR_MASK; - else - return -EINVAL; - } else if (!strcmp(signal, "fsr")) { - if (!strcmp(src, "fsr")) - v &= ~OMAP2_MCBSP1_FSR_MASK; - else if (!strcmp(src, "fsx")) - v |= OMAP2_MCBSP1_FSR_MASK; - else - return -EINVAL; - } else { - return -EINVAL; - } - - omap_ctrl_writel(v, OMAP2_CONTROL_DEVCONF0); - - return 0; -} - -/* McBSP4 internal signal muxing function for OMAP4 */ -#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX (1 << 31) -#define OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX (1 << 30) -static int omap4_mcbsp4_mux_rx_clk(struct device *dev, const char *signal, - const char *src) -{ - u32 v; - - /* - * In CONTROL_MCBSPLP register only bit 30 (CLKR mux), and bit 31 (FSR - * mux) is used */ - v = omap4_ctrl_pad_readl(OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP); - - if (!strcmp(signal, "clkr")) { - if (!strcmp(src, "clkr")) - v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX; - else if (!strcmp(src, "clkx")) - v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_CLKX; - else - return -EINVAL; - } else if (!strcmp(signal, "fsr")) { - if (!strcmp(src, "fsr")) - v &= ~OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX; - else if (!strcmp(src, "fsx")) - v |= OMAP4_CONTROL_MCBSPLP_ALBCTRLRX_FSX; - else - return -EINVAL; - } else { - return -EINVAL; - } - - omap4_ctrl_pad_writel(v, OMAP4_CTRL_MODULE_PAD_CORE_CONTROL_MCBSPLP); - - return 0; -} - static int omap3_enable_st_clock(unsigned int id, bool enable) { unsigned int w; @@ -143,14 +74,6 @@ static int __init omap_init_mcbsp(struct omap_hwmod *oh, void *unused) pdata->has_ccr = true; } - /* On OMAP2/3 the McBSP1 port has 6 pin configuration */ - if (id == 1 && oh->class->rev < MCBSP_CONFIG_TYPE4) - pdata->mux_signal = omap2_mcbsp1_mux_rx_clk; - - /* On OMAP4 the McBSP4 port has 6 pin configuration */ - if (id == 4 && oh->class->rev == MCBSP_CONFIG_TYPE4) - pdata->mux_signal = omap4_mcbsp4_mux_rx_clk; - if (oh->class->rev == MCBSP_CONFIG_TYPE2) { /* The FIFO has 128 locations */ pdata->buffer_size = 0x80; diff --git a/arch/arm/plat-omap/include/plat/mcbsp.h b/arch/arm/plat-omap/include/plat/mcbsp.h index 0a7d5ca471e0..c78d90b28b19 100644 --- a/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/arch/arm/plat-omap/include/plat/mcbsp.h @@ -47,7 +47,6 @@ struct omap_mcbsp_platform_data { bool has_wakeup; /* Wakeup capability */ bool has_ccr; /* Transceiver has configuration control registers */ int (*enable_st_clock)(unsigned int, bool); - int (*mux_signal)(struct device *dev, const char *signal, const char *src); }; /** diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 262a6152111f..49a67259ce5a 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -334,9 +334,6 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); /* McBSP functional clock source changing function */ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); -/* McBSP signal muxing API */ -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux); - /* Sidetone specific API */ int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1046083e90a0..506159493a9d 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,21 +516,9 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EBUSY; } - if (clk_id == OMAP_MCBSP_SYSCLK_CLK || - clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK || - clk_id == OMAP_MCBSP_SYSCLK_CLKS_EXT || - clk_id == OMAP_MCBSP_SYSCLK_CLKX_EXT || - clk_id == OMAP_MCBSP_SYSCLK_CLKR_EXT) { - mcbsp->in_freq = freq; - regs->srgr2 &= ~CLKSM; - regs->pcr0 &= ~SCLKME; - } else if (cpu_class_is_omap1()) { - /* - * McBSP CLKR/FSR signal muxing functions are only available on - * OMAP2 or newer versions - */ - return -EINVAL; - } + mcbsp->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: @@ -558,20 +546,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; break; - - - case OMAP_MCBSP_CLKR_SRC_CLKR: - err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR); - break; - case OMAP_MCBSP_CLKR_SRC_CLKX: - err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX); - break; - case OMAP_MCBSP_FSR_SRC_FSR: - err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR); - break; - case OMAP_MCBSP_FSR_SRC_FSX: - err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX); - break; default: err = -ENODEV; } diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index f877b16f19c9..baebcee30e30 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -32,10 +32,6 @@ enum omap_mcbsp_clksrg_clk { OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */ OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */ OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */ - OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */ - OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */ - OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */ - OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */ }; /* McBSP dividers */ -- cgit v1.2.3 From 8d3c09096500aa20702395b87c81a62ac9cbe3be Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:05 +0300 Subject: ASoC: omap-mcbsp: Remove unused defines NUM_LINKS is no longer in use by the code. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.h | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index baebcee30e30..ba8386a0d8dc 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -39,22 +39,6 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -#if defined(CONFIG_SOC_OMAP2420) -#define NUM_LINKS 2 -#endif -#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) -#undef NUM_LINKS -#define NUM_LINKS 3 -#endif -#if defined(CONFIG_ARCH_OMAP4) -#undef NUM_LINKS -#define NUM_LINKS 4 -#endif -#if defined(CONFIG_ARCH_OMAP3) || defined(CONFIG_SOC_OMAP2430) -#undef NUM_LINKS -#define NUM_LINKS 5 -#endif - int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd); #endif -- cgit v1.2.3 From dc26df52455348e06e4c34a2af2910d291369fe8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:06 +0300 Subject: ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code We can use the has_ccr flag to replace the cpu_is_omap* checks. This provides future proof implementation and we do not need to update the code if new OMAP revision starts to use the McBSP driver. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 506159493a9d..b9770eea28a0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -398,12 +398,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; regs->spcr1 |= RINTM(3); - /* RFIG and XFIG are not defined in 34xx */ - if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) { + /* RFIG and XFIG are not defined in 2430 and on OMAP3+ */ + if (!mcbsp->pdata->has_ccr) { regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; } - if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) { + + /* Configure XCCR/RCCR only for revisions which have ccr registers */ + if (mcbsp->pdata->has_ccr) { regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE; regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE; } -- cgit v1.2.3 From 11dd586421b3091007e6f084a9211f3baa66f9fc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Aug 2012 16:41:08 +0300 Subject: ASoC: omap-mcbsp: Add device tree bindings Device tree support for McBSP modules on OMAP2+ SoC. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-mcbsp.txt | 45 +++++++++++++++ sound/soc/omap/omap-mcbsp.c | 66 +++++++++++++++++++++- 2 files changed, 110 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt new file mode 100644 index 000000000000..447cb131e909 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -0,0 +1,45 @@ +* Texas Instruments OMAP2+ McBSP module + +Required properties: +- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420 + "ti,omap2430-mcbsp" for McBSP on OMAP2430 + "ti,omap3-mcbsp" for McBSP on OMAP3 + "ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC +- reg: Register location and size, for OMAP4+ as an array: + , + ; +- interrupts: Interrupt numbers for the McBSP port, as an array in case the + McBSP IP have more interrupt lines: + , + , + ; +- interrupt-parent: The parent interrupt controller +- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) +- ti,hwmods: Name of the hwmod associated to the McBSP port + +Sidetone support for OMAP3 McBSP2 and 3 ports: +- sidetone { }: Within this section the following parameters are required: +- reg: Register location and size for the ST block +- interrupts: The interrupt number for the ST block +- interrupt-parent: The parent interrupt controller for the ST block + +Example: + +mcbsp2: mcbsp@49022000 { + compatible = "ti,omap3-mcbsp"; + #address-cells = <1>; + #size-cells = <1>; + reg = <0x49022000 0xff>; + interrupts = <0 17 0x4>, /* OCP compliant interrup */ + <0 62 0x4>, /* TX interrup */ + <0 63 0x4>; /* RX interrup */ + interrupt-parent = <&intc>; + ti,buffer-size = <1280>; + ti,hwmods = "mcbsp2"; + + sidetone { + reg = <0x49028000 0xff>; + interrupts = <0 4 0x4>; + interrupt-parent = <&intc>; + }; +}; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index b9770eea28a0..2e1750e2ab31 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include #include @@ -737,13 +739,74 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); +static struct omap_mcbsp_platform_data omap2420_pdata = { + .reg_step = 4, + .reg_size = 2, +}; + +static struct omap_mcbsp_platform_data omap2430_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, +}; + +static struct omap_mcbsp_platform_data omap3_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, + .has_wakeup = true, +}; + +static struct omap_mcbsp_platform_data omap4_pdata = { + .reg_step = 4, + .reg_size = 4, + .has_ccr = true, + .has_wakeup = true, +}; + +static const struct of_device_id omap_mcbsp_of_match[] = { + { + .compatible = "ti,omap2420-mcbsp", + .data = &omap2420_pdata, + }, + { + .compatible = "ti,omap2430-mcbsp", + .data = &omap2430_pdata, + }, + { + .compatible = "ti,omap3-mcbsp", + .data = &omap3_pdata, + }, + { + .compatible = "ti,omap4-mcbsp", + .data = &omap4_pdata, + }, + { }, +}; +MODULE_DEVICE_TABLE(of, omap_mcbsp_of_match); + static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) { struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev); struct omap_mcbsp *mcbsp; + const struct of_device_id *match; int ret; - if (!pdata) { + match = of_match_device(omap_mcbsp_of_match, &pdev->dev); + if (match) { + struct device_node *node = pdev->dev.of_node; + int buffer_size; + + pdata = devm_kzalloc(&pdev->dev, + sizeof(struct omap_mcbsp_platform_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + memcpy(pdata, match->data, sizeof(*pdata)); + if (!of_property_read_u32(node, "ti,buffer-size", &buffer_size)) + pdata->buffer_size = buffer_size; + } else if (!pdata) { dev_err(&pdev->dev, "missing platform data.\n"); return -EINVAL; } @@ -785,6 +848,7 @@ static struct platform_driver asoc_mcbsp_driver = { .driver = { .name = "omap-mcbsp", .owner = THIS_MODULE, + .of_match_table = omap_mcbsp_of_match, }, .probe = asoc_mcbsp_probe, -- cgit v1.2.3 From b8dfc4624162c0547d7f36a9df48da2d9b4bd58a Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 23 Aug 2012 17:32:30 +0800 Subject: ALSA: hda - add runtime PM support Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gerneral sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled: -- If a codec supports 'stop-clock' in D3, it will request suspending the controller after it enters D3 and request resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. -- User IO operations and HW wakeup signal can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. More about 'stop-clock' feature: If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power State. [Fixed CONFIG_PM_RUNTIME dependency in hda_intel.c by tiwai] Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 +++++++++-- sound/pci/hda/hda_codec.h | 5 ++- sound/pci/hda/hda_intel.c | 83 +++++++++++++++++++++++++++++++++++++---------- 3 files changed, 87 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4efd2711acd9..0de1f76733d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3535,6 +3544,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3557,6 +3568,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4408,7 +4423,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4466,7 +4481,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 92ecb2b2a05c..13c834f20440 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -616,7 +616,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -875,6 +875,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 209bea435442..726f4208bd05 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include #include #include +#include #ifdef CONFIG_X86 /* for snoop control */ #include @@ -1032,7 +1033,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -1288,6 +1289,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; +#ifdef CONFIG_PM_RUNTIME + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; +#endif + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2400,23 +2406,17 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - struct hda_codec *c; - int power_on = 0; - list_for_each_entry(c, &bus->codec_list, list) { - if (c->power_on) { - power_on = 1; - break; - } - } - if (power_on) - azx_init_chip(chip, 1); - else if (chip->running && power_save_controller && - !bus->power_keep_link_on) - azx_stop_chip(chip); + if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + return; + + if (codec->power_on) + pm_runtime_get_sync(&chip->pci->dev); + else + pm_runtime_put_sync(&chip->pci->dev); } static DEFINE_MUTEX(card_list_lock); @@ -2520,11 +2520,43 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); +#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ + +#ifdef CONFIG_PM_RUNTIME +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + if (!power_save_controller) + return -EAGAIN; + + azx_stop_chip(chip); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + azx_init_chip(chip, 1); + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + +#ifdef CONFIG_PM +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define AZX_PM_OPS NULL -#endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ +#endif /* CONFIG_PM */ /* @@ -3239,6 +3271,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) } #endif +static void rpm_get_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &chip->bus->codec_list, list) { + pm_runtime_get_noresume(&chip->pci->dev); + } +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3290,6 +3331,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3342,6 +3386,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3356,6 +3401,10 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- cgit v1.2.3 From 52ca1138fa55bf6f46a5e02a2c1088756a5c8f2e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Aug 2012 15:50:45 +0100 Subject: ASoC: wm8994: Update for new WM1811 variants There are some new WM1811 variants distinguished by both revision and cust_id which need slightly different handling. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b74df52d2820..890b582b40f3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3823,14 +3823,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_cache_dac_hp_direct = true; wm8994->fll_byp = true; - switch (wm8994->revision) { + switch (control->cust_id) { case 0: - case 1: case 2: - case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -7; break; + case 1: + case 3: + wm8994->hubs.dcs_codes_l = -8; + wm8994->hubs.dcs_codes_r = -7; + break; default: break; } @@ -3919,7 +3922,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM1811: - if (wm8994->revision > 1) { + if (control->cust_id > 1 || wm8994->revision > 1) { ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm1811_jackdet_irq, "JACKDET", -- cgit v1.2.3 From e3523e01869da20fdd12ffd19ae1df7bf492650e Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 23 Aug 2012 15:59:56 +0100 Subject: ASoC: wm0010: Add initial wm0010 DSP driver The WM0010 is a compact digital signal processor that has been highly optimised for low-power audio applications. Extensive memory resources and core optimisation allow the device to manage all audio processing algorithms efficiently and autonomously, while the host processor sleeps or performs other tasks. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- include/sound/wm0010.h | 27 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm0010.c | 930 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 963 insertions(+) create mode 100644 include/sound/wm0010.h create mode 100644 sound/soc/codecs/wm0010.c (limited to 'sound') diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h new file mode 100644 index 000000000000..3261e90815af --- /dev/null +++ b/include/sound/wm0010.h @@ -0,0 +1,27 @@ +/* + * wm0010.h -- Platform data for WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Author: Dimitris Papastamos + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM0010_PDATA_H +#define WM0010_PDATA_H + +struct wm0010_pdata { + int gpio_reset; + + /* Set if there is an inverter between the GPIO controlling + * the reset signal and the device. + */ + int reset_active_high; + int irq_flags; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f8e8594aeb9..3684255e5fba 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -70,6 +70,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE + select SND_SOC_WM0010 if SPI_MASTER select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C @@ -326,6 +327,9 @@ config SND_SOC_UDA1380 config SND_SOC_WL1273 tristate +config SND_SOC_WM0010 + tristate + config SND_SOC_WM1250_EV1 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 34148bb59c68..ca508b251df7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -61,6 +61,7 @@ snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o +snd-soc-wm0010-objs := wm0010.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o +obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c new file mode 100644 index 000000000000..8e0b6d6bffaf --- /dev/null +++ b/sound/soc/codecs/wm0010.c @@ -0,0 +1,930 @@ +/* + * wm0010.c -- WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Authors: Mark Brown + * Dimitris Papastamos + * Scott Ling + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#define DEVICE_ID_WM0010 10 + +enum dfw_cmd { + DFW_CMD_FUSE = 0x01, + DFW_CMD_CODE_HDR, + DFW_CMD_CODE_DATA, + DFW_CMD_PLL, + DFW_CMD_INFO = 0xff +}; + +struct dfw_binrec { + u8 command; + u32 length:24; + u32 address; + uint8_t data[0]; +} __packed; + +struct dfw_pllrec { + u8 command; + u32 length:24; + u32 address; + u32 clkctrl1; + u32 clkctrl2; + u32 clkctrl3; + u32 ldetctrl; + u32 uart_div; + u32 spi_div; +} __packed; + +static struct pll_clock_map { + int max_sysclk; + int max_pll_spi_speed; + u32 pll_clkctrl1; +} pll_clock_map[] = { /* Dividers */ + { 22000000, 26000000, 0x00201f11 }, /* 2,32,2 */ + { 18000000, 26000000, 0x00203f21 }, /* 2,64,4 */ + { 14000000, 26000000, 0x00202620 }, /* 1,39,4 */ + { 10000000, 22000000, 0x00203120 }, /* 1,50,4 */ + { 6500000, 22000000, 0x00204520 }, /* 1,70,4 */ + { 5500000, 22000000, 0x00103f10 }, /* 1,64,2 */ +}; + +enum wm0010_state { + WM0010_POWER_OFF, + WM0010_OUT_OF_RESET, + WM0010_BOOTROM, + WM0010_STAGE2, + WM0010_FIRMWARE, +}; + +struct wm0010_priv { + struct snd_soc_codec *codec; + + struct mutex lock; + struct device *dev; + + struct wm0010_pdata pdata; + + int gpio_reset; + int gpio_reset_value; + + struct regulator_bulk_data core_supplies[2]; + struct regulator *dbvdd; + + int sysclk; + + enum wm0010_state state; + bool boot_failed; + int boot_done; + bool ready; + bool pll_running; + int max_spi_freq; + int board_max_spi_speed; + u32 pll_clkctrl1; + + spinlock_t irq_lock; + int irq; + + struct completion boot_completion; +}; + +struct wm0010_spi_msg { + struct spi_message m; + struct spi_transfer t; + u8 *tx_buf; + u8 *rx_buf; + size_t len; +}; + +static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { + { "SDI2 Playback", NULL, "SDI1 Playback" }, +}; + +static const char *wm0010_state_to_str(enum wm0010_state state) +{ + const char *state_to_str[] = { + "Power off", + "Out of reset", + "Bootrom", + "Stage2", + "Firmware" + }; + + if (state < 0 || state >= ARRAY_SIZE(state_to_str)) + return "null"; + return state_to_str[state]; +} + +/* Called with wm0010->lock held */ +static void wm0010_halt(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + enum wm0010_state state; + + /* Fetch the wm0010 state */ + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + switch (state) { + case WM0010_POWER_OFF: + /* If there's nothing to do, bail out */ + return; + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + case WM0010_FIRMWARE: + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + /* Disable the regulators */ + regulator_disable(wm0010->dbvdd); + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + break; + } + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); +} + +struct wm0010_boot_xfer { + struct list_head list; + struct snd_soc_codec *codec; + struct completion *done; + struct spi_message m; + struct spi_transfer t; +}; + +/* Called with wm0010->lock held */ +static void wm0010_mark_boot_failure(struct wm0010_priv *wm0010) +{ + enum wm0010_state state; + unsigned long flags; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_err(wm0010->dev, "Failed to transition from `%s' state to `%s' state\n", + wm0010_state_to_str(state), wm0010_state_to_str(state + 1)); + + wm0010->boot_failed = true; +} + +static void wm0010_boot_xfer_complete(void *data) +{ + struct wm0010_boot_xfer *xfer = data; + struct snd_soc_codec *codec = xfer->codec; + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + u32 *out32 = xfer->t.rx_buf; + int i; + + if (xfer->m.status != 0) { + dev_err(codec->dev, "SPI transfer failed: %d\n", + xfer->m.status); + wm0010_mark_boot_failure(wm0010); + if (xfer->done) + complete(xfer->done); + return; + } + + for (i = 0; i < xfer->t.len / 4; i++) { + dev_dbg(codec->dev, "%d: %04x\n", i, out32[i]); + + switch (be32_to_cpu(out32[i])) { + case 0xe0e0e0e0: + dev_err(codec->dev, + "%d: ROM error reported in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x55555555: + if (wm0010->boot_done == 0) + break; + dev_err(codec->dev, + "%d: ROM bootloader running in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0000: + dev_dbg(codec->dev, "Stage2 loader running\n"); + break; + + case 0x0fed0007: + dev_dbg(codec->dev, "CODE_HDR packet received\n"); + break; + + case 0x0fed0008: + dev_dbg(codec->dev, "CODE_DATA packet received\n"); + break; + + case 0x0fed0009: + dev_dbg(codec->dev, "Download complete\n"); + break; + + case 0x0fed000c: + dev_dbg(codec->dev, "Application start\n"); + break; + + case 0x0fed000e: + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + + case 0x0fed0025: + dev_err(codec->dev, "Device reports image too long\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed002c: + dev_err(codec->dev, "Device reports bad SPI packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0031: + dev_err(codec->dev, "Device reports SPI read overflow\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0032: + dev_err(codec->dev, "Device reports SPI underclock\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0033: + dev_err(codec->dev, "Device reports bad header packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0034: + dev_err(codec->dev, "Device reports invalid packet type\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0035: + dev_err(codec->dev, "Device reports data before header error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0038: + dev_err(codec->dev, "Device reports invalid PLL packet\n"); + break; + + case 0x0fed003a: + dev_err(codec->dev, "Device reports packet alignment error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + default: + dev_err(codec->dev, "Unrecognised return 0x%x\n", + be32_to_cpu(out32[i])); + wm0010_mark_boot_failure(wm0010); + break; + } + + if (wm0010->boot_failed) + break; + } + + wm0010->boot_done++; + if (xfer->done) + complete(xfer->done); +} + +static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) +{ + int i; + + for (i = 0; i < len / 8; i++) + data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); +} + +static int wm0010_boot(struct snd_soc_codec *codec) +{ + struct spi_device *spi = to_spi_device(codec->dev); + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + struct list_head xfer_list; + struct wm0010_boot_xfer *xfer; + int ret; + struct completion done; + const struct firmware *fw; + const struct dfw_binrec *rec; + struct spi_message m; + struct spi_transfer t; + struct dfw_pllrec pll_rec; + u32 *img, *p; + u64 *img_swap; + u8 *out; + u32 len, offset; + int i; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + if (wm0010->state != WM0010_POWER_OFF) + dev_warn(wm0010->dev, "DSP already powered up!\n"); + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + if (wm0010->sysclk > 26000000) { + dev_err(codec->dev, "Max DSP clock frequency is 26MHz\n"); + ret = -ECANCELED; + goto err; + } + + INIT_LIST_HEAD(&xfer_list); + + mutex_lock(&wm0010->lock); + wm0010->pll_running = false; + + dev_dbg(codec->dev, "max_spi_freq: %d\n", wm0010->max_spi_freq); + + ret = regulator_bulk_enable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable core supplies: %d\n", + ret); + mutex_unlock(&wm0010->lock); + goto err; + } + + ret = regulator_enable(wm0010->dbvdd); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable DBVDD: %d\n", ret); + goto err_core; + } + + /* Release reset */ + gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value); + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_OUT_OF_RESET; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* First the bootloader */ + ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request stage2 loader: %d\n", + ret); + goto abort; + } + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_BOOTROM; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + + /* Copy to local buffer first as vmalloc causes problems for dma */ + img = kzalloc(fw->size, GFP_KERNEL); + if (!img) { + dev_err(codec->dev, "Failed to allocate image buffer\n"); + goto abort; + } + + out = kzalloc(fw->size, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, "Failed to allocate output buffer\n"); + goto abort; + } + + memcpy(img, &fw->data[0], fw->size); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img; + t.len = fw->size; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 10; + spi_message_add_tail(&t, &m); + + dev_dbg(codec->dev, "Starting initial download at %dHz\n", + t.speed_hz); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Initial download failed: %d\n", ret); + goto abort; + } + + /* Look for errors from the boot ROM */ + for (i = 0; i < fw->size; i++) { + if (out[i] != 0x55) { + ret = -EBUSY; + dev_err(codec->dev, "Boot ROM error: %x in %d\n", + out[i], i); + wm0010_mark_boot_failure(wm0010); + goto abort; + } + } + + release_firmware(fw); + kfree(img); + kfree(out); + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP loader.\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_STAGE2; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* Only initialise PLL if max_spi_freq initialised */ + if (wm0010->max_spi_freq) { + + /* Initialise a PLL record */ + memset(&pll_rec, 0, sizeof(pll_rec)); + pll_rec.command = DFW_CMD_PLL; + pll_rec.length = (sizeof(pll_rec) - 8); + + /* On wm0010 only the CLKCTRL1 value is used */ + pll_rec.clkctrl1 = wm0010->pll_clkctrl1; + + len = pll_rec.length + 8; + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&pll_rec, img_swap, len); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img_swap; + t.len = len; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 6; + spi_message_add_tail(&t, &m); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "First PLL write failed: %d\n", ret); + goto abort; + } + + /* Use a second send of the message to get the return status */ + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Second PLL write failed: %d\n", ret); + goto abort; + } + + p = (u32 *)out; + + /* Look for PLL active code from the DSP */ + for (i = 0; i < len / 4; i++) { + if (*p == 0x0e00ed0f) { + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + } + p++; + } + + kfree(img_swap); + kfree(out); + } else + dev_dbg(codec->dev, "Not enabling DSP PLL."); + + ret = request_firmware(&fw, "wm0010.dfw", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request application: %d\n", + ret); + goto abort; + } + + rec = (const struct dfw_binrec *)fw->data; + offset = 0; + wm0010->boot_done = 0; + wm0010->boot_failed = false; + BUG_ON(!list_empty(&xfer_list)); + init_completion(&done); + + /* First record should be INFO */ + if (rec->command != DFW_CMD_INFO) { + dev_err(codec->dev, "First record not INFO\r\n"); + goto abort; + } + + /* Check it's a 0010 file */ + if (rec->data[0] != DEVICE_ID_WM0010) { + dev_err(codec->dev, "Not a WM0010 firmware file.\r\n"); + goto abort; + } + + /* Skip the info record as we don't need to send it */ + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + while (offset < fw->size) { + dev_dbg(codec->dev, + "Packet: command %d, data length = 0x%x\r\n", + rec->command, rec->length); + len = rec->length + 8; + + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&rec->command, img_swap, len); + + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + + spi_message_init(&xfer->m); + xfer->m.complete = wm0010_boot_xfer_complete; + xfer->m.context = xfer; + xfer->t.tx_buf = img_swap; + xfer->t.rx_buf = out; + xfer->t.len = len; + xfer->t.bits_per_word = 8; + + if (!wm0010->pll_running) { + xfer->t.speed_hz = wm0010->sysclk / 6; + } else { + xfer->t.speed_hz = wm0010->max_spi_freq; + + if (wm0010->board_max_spi_speed && + (wm0010->board_max_spi_speed < wm0010->max_spi_freq)) + xfer->t.speed_hz = wm0010->board_max_spi_speed; + } + + /* Store max usable spi frequency for later use */ + wm0010->max_spi_freq = xfer->t.speed_hz; + + spi_message_add_tail(&xfer->t, &xfer->m); + + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + if (offset >= fw->size) { + dev_dbg(codec->dev, "All transfers scheduled\n"); + xfer->done = &done; + } + + ret = spi_async(spi, &xfer->m); + if (ret != 0) { + dev_err(codec->dev, "Write failed: %d\n", ret); + goto abort; + } + + if (wm0010->boot_failed) + goto abort; + } + + wait_for_completion(&done); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_FIRMWARE; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + mutex_unlock(&wm0010->lock); + + release_firmware(fw); + + while (!list_empty(&xfer_list)) { + xfer = list_first_entry(&xfer_list, struct wm0010_boot_xfer, + list); + kfree(xfer->t.rx_buf); + kfree(xfer->t.tx_buf); + list_del(&xfer->list); + kfree(xfer); + } + + return 0; + +abort: + /* Put the chip back into reset */ + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + return ret; +err_core: + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); +err: + return ret; +} + +static int wm0010_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) + wm0010_boot(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + mutex_lock(&wm0010->lock); + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source, + int clk_id, unsigned int freq, int dir) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + wm0010->sysclk = freq; + + if (freq < pll_clock_map[ARRAY_SIZE(pll_clock_map)-1].max_sysclk) { + wm0010->max_spi_freq = 0; + } else { + for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++) + if (freq >= pll_clock_map[i].max_sysclk) + break; + + wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed; + wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1; + } + + return 0; +} + +static int wm0010_probe(struct snd_soc_codec *codec); + +static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { + .probe = wm0010_probe, + .set_bias_level = wm0010_set_bias_level, + .set_sysclk = wm0010_set_sysclk, + + .dapm_routes = wm0010_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), +}; + +#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm0010_dai[] = { + { + .name = "wm0010-sdi1", + .playback = { + .stream_name = "SDI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, + { + .name = "wm0010-sdi2", + .playback = { + .stream_name = "SDI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, +}; + +static irqreturn_t wm0010_irq(int irq, void *data) +{ + struct wm0010_priv *wm0010 = data; + + switch (wm0010->state) { + case WM0010_POWER_OFF: + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + spin_lock(&wm0010->irq_lock); + complete(&wm0010->boot_completion); + spin_unlock(&wm0010->irq_lock); + return IRQ_HANDLED; + default: + return IRQ_NONE; + } + + return IRQ_NONE; +} + +static int wm0010_probe(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + struct spi_device *spi = to_spi_device(wm0010->dev); + unsigned long flags; + unsigned long gpio_flags; + int ret; + int trigger; + int irq; + + wm0010->codec = codec; + + init_completion(&wm0010->boot_completion); + + wm0010->core_supplies[0].supply = "AVDD"; + wm0010->core_supplies[1].supply = "DCVDD"; + ret = devm_regulator_bulk_get(wm0010->dev, ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(wm0010->dev, "Failed to obtain core supplies: %d\n", + ret); + return ret; + } + + wm0010->dbvdd = devm_regulator_get(wm0010->dev, "DBVDD"); + if (IS_ERR(wm0010->dbvdd)) { + ret = PTR_ERR(wm0010->dbvdd); + dev_err(wm0010->dev, "Failed to obtain DBVDD: %d\n", ret); + return ret; + } + + if (wm0010->pdata.gpio_reset) { + wm0010->gpio_reset = wm0010->pdata.gpio_reset; + + if (wm0010->pdata.reset_active_high) + wm0010->gpio_reset_value = 1; + else + wm0010->gpio_reset_value = 0; + + if (wm0010->gpio_reset_value) + gpio_flags = GPIOF_OUT_INIT_HIGH; + else + gpio_flags = GPIOF_OUT_INIT_LOW; + + ret = devm_gpio_request_one(wm0010->dev, wm0010->gpio_reset, + gpio_flags, "wm0010 reset"); + if (ret < 0) { + dev_err(wm0010->dev, + "Failed to request GPIO for DSP reset: %d\n", + ret); + return ret; + } + } else { + dev_err(wm0010->dev, "No reset GPIO configured\n"); + return ret; + } + + irq = spi->irq; + if (wm0010->pdata.irq_flags) + trigger = wm0010->pdata.irq_flags; + else + trigger = IRQF_TRIGGER_FALLING; + trigger |= IRQF_ONESHOT; + + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, + "wm0010", wm0010); + if (ret) + dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", + irq, ret); + wm0010->irq = irq; + + if (spi->max_speed_hz) + wm0010->board_max_spi_speed = spi->max_speed_hz; + else + wm0010->board_max_spi_speed = 0; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ + struct wm0010_priv *wm0010; + int ret; + + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm0010, wm0010_dai, + ARRAY_SIZE(wm0010_dai)); + if (ret < 0) + return ret; + + return 0; +} + +static int __devexit wm0010_spi_remove(struct spi_device *spi) +{ + struct wm0010_priv *wm0010 = spi_get_drvdata(spi); + + snd_soc_unregister_codec(&spi->dev); + + if (wm0010->gpio_reset) { + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_free(wm0010->gpio_reset); + } + + if (wm0010->irq) + free_irq(wm0010->irq, wm0010); + + return 0; +} + +static struct spi_driver wm0010_spi_driver = { + .driver = { + .name = "wm0010", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm0010_spi_probe, + .remove = __devexit_p(wm0010_spi_remove), +}; + +module_spi_driver(wm0010_spi_driver); + +MODULE_DESCRIPTION("ASoC WM0010 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 5d6147f10113c240845ea9e8e4e744bbadce5bcb Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 24 Aug 2012 12:06:30 +0800 Subject: ALSA: hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 ++++++ sound/pci/hda/hda_intel.c | 2 ++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0de1f76733d2..4a2f35ccbf05 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1320,11 +1320,13 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_CLKSTOP); if (!codec->d3_stop_clk) bus->power_keep_link_on = 1; +#endif /* power-up all before initialization */ hda_set_power_state(codec, @@ -3544,7 +3546,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk_ok = 0; +#endif if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); @@ -3569,9 +3573,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } +#ifdef CONFIG_SND_HDA_POWER_SAVE if ((power_state == AC_PWRST_D3) && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; +#endif } #ifdef CONFIG_SND_HDA_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 726f4208bd05..1c9c779dda55 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2528,8 +2528,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- cgit v1.2.3 From 56244d0868f48f233d1735541e7da3fd33434b3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Aug 2012 07:52:03 +0200 Subject: ALSA: cmi8328: Fix build error with CONFIG_GAMEPORT=n sound/isa/cmi8328.c: In function 'snd_cmi8328_remove': sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function) sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/isa/cmi8328.o] Error 1 Reported-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/isa/cmi8328.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index aefafff50756..bde60139bb95 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -404,8 +404,9 @@ error: static int __devexit snd_cmi8328_remove(struct device *pdev, unsigned int dev) { struct snd_card *card = dev_get_drvdata(pdev); -#ifdef SUPPORT_JOYSTICK struct snd_cmi8328 *cmi = card->private_data; + +#ifdef SUPPORT_JOYSTICK if (cmi->gameport) { struct resource *res = gameport_get_port_data(cmi->gameport); gameport_unregister_port(cmi->gameport); -- cgit v1.2.3 From 1549c34bfdf3dc29b769c803f6cfdc53dfc67f93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Aug 2012 17:50:52 +0100 Subject: ASoC: wm0010: Fix passthrough routing Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8e0b6d6bffaf..e1315e0e7856 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -119,7 +119,8 @@ struct wm0010_spi_msg { }; static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { - { "SDI2 Playback", NULL, "SDI1 Playback" }, + { "SDI2 Capture", NULL, "SDI1 Playback" }, + { "SDI1 Capture", NULL, "SDI2 Playback" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) -- cgit v1.2.3 From f9372c9c06166dc24a17cf25d325d83a9a06a02d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Aug 2012 17:05:48 +0100 Subject: ASoC: samsung: Add hookup of WM0010 on Speyside The Speyside platform by default has a WM0010 fitted. Now that we have a public driver hook it up in the machine integration. Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/speyside.c | 42 +++++++++++++++++++++++++++++++++++++----- 2 files changed, 38 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fb5600083612..f17dd25e0f4a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -191,6 +191,7 @@ config SND_SOC_SPEYSIDE select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 + select SND_SOC_WM0010 select SND_SOC_WM1250_EV1 config SND_SOC_TOBERMORY diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index a4a9fc7e8c76..c7e1c28528a4 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -25,7 +25,7 @@ static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; int ret; if (dapm->dev != codec_dai->dev) @@ -57,7 +57,7 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; int ret; if (dapm->dev != codec_dai->dev) @@ -126,6 +126,18 @@ static void speyside_set_polarity(struct snd_soc_codec *codec, snd_soc_dapm_sync(&codec->dapm); } +static int speyside_wm0010_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(dai, 0, MCLK_AUDIO_RATE, 0); + if (ret < 0) + return ret; + + return 0; +} + static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->codec_dai; @@ -172,17 +184,37 @@ static int speyside_late_probe(struct snd_soc_card *card) return 0; } +static const struct snd_soc_pcm_stream dsp_codec_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + static struct snd_soc_dai_link speyside_dai[] = { { - .name = "CPU", - .stream_name = "CPU", + .name = "CPU-DSP", + .stream_name = "CPU-DSP", .cpu_dai_name = "samsung-i2s.0", - .codec_dai_name = "wm8996-aif1", + .codec_dai_name = "wm0010-sdi1", .platform_name = "samsung-audio", + .codec_name = "spi0.0", + .init = speyside_wm0010_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "DSP-CODEC", + .stream_name = "DSP-CODEC", + .cpu_dai_name = "wm0010-sdi2", + .codec_dai_name = "wm8996-aif1", .codec_name = "wm8996.1-001a", .init = speyside_wm8996_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, + .params = &dsp_codec_params, + .ignore_suspend = 1, }, { .name = "Baseband", -- cgit v1.2.3 From 6df3198635e2ad961952566a05994bc592abe774 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 14:05:35 +0100 Subject: ASoC: wm0010: Enable 44.1kHz support With appropriate clocking configuration the WM0010 driver supports 44.1kHz audio; enable that. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index e1315e0e7856..f820b5725163 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -722,7 +722,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3 From 1470bfacb675ab0e25c30c97772a764ca16e0e52 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 09:17:42 -0700 Subject: ASoC: wm0010: Add dummy widget for CLKIN Make it easier to integrate the management of the clock supplying the WM0010 with DAPM by providing a dummy supply widget which supplies the interface widgets, this can be connected to clock outputs by the machines. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f820b5725163..b6f7097d7953 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -118,9 +118,18 @@ struct wm0010_spi_msg { size_t len; }; +static const struct snd_soc_dapm_widget wm0010_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("CLKIN", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { { "SDI2 Capture", NULL, "SDI1 Playback" }, { "SDI1 Capture", NULL, "SDI2 Playback" }, + + { "SDI1 Capture", NULL, "CLKIN" }, + { "SDI2 Capture", NULL, "CLKIN" }, + { "SDI1 Playback", NULL, "CLKIN" }, + { "SDI2 Playback", NULL, "CLKIN" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) @@ -718,6 +727,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .set_bias_level = wm0010_set_bias_level, .set_sysclk = wm0010_set_sysclk, + .dapm_widgets = wm0010_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm0010_dapm_widgets), .dapm_routes = wm0010_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -- cgit v1.2.3 From d3fd716e82ed643d804c49ca9ca554079c429a5c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 09:23:40 -0700 Subject: ASoC: wm0010: Set idle_bias_off Doesn't make any practical difference given that _SUSPEND and _OFF are equivalent for the driver but it's what we're really doing. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index b6f7097d7953..61ec7633fe5a 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -726,6 +726,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .probe = wm0010_probe, .set_bias_level = wm0010_set_bias_level, .set_sysclk = wm0010_set_sysclk, + .idle_bias_off = true, .dapm_widgets = wm0010_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm0010_dapm_widgets), -- cgit v1.2.3 From 4f3c3c1b32bb8fddcd7f626c2fb09d55f469f976 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 09:24:16 -0700 Subject: ASoC: wm0010: Don't double free reset GPIO We are using devm_ to allocate the GPIO so it will be freed automatically. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 61ec7633fe5a..6c4a2fa90fbe 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -917,7 +917,6 @@ static int __devexit wm0010_spi_remove(struct spi_device *spi) if (wm0010->gpio_reset) { /* Remember to put chip back into reset */ gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); - gpio_free(wm0010->gpio_reset); } if (wm0010->irq) -- cgit v1.2.3 From bf9d323722845c8643287dca436e04e34cb21bb8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 13:01:15 -0700 Subject: ASoC: wm0010: Tweak diagnostic output Make it scan better by writing ROM with capitals. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 6c4a2fa90fbe..30ec0bd85307 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -137,7 +137,7 @@ static const char *wm0010_state_to_str(enum wm0010_state state) const char *state_to_str[] = { "Power off", "Out of reset", - "Bootrom", + "Boot ROM", "Stage2", "Firmware" }; -- cgit v1.2.3 From 32c50a31aad77e8faf2718d149da13f2136c1b46 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 25 Aug 2012 13:04:04 -0700 Subject: ASoC: wm0010: Move resource acquisition to device probe This is more idimatic for modern drivers. Also fix a couple of return codes while we're at it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 54 +++++++++++++++++++++++------------------------ 1 file changed, 27 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 30ec0bd85307..5f99148447e1 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -799,14 +799,35 @@ static irqreturn_t wm0010_irq(int irq, void *data) static int wm0010_probe(struct snd_soc_codec *codec) { struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); - struct spi_device *spi = to_spi_device(wm0010->dev); + + wm0010->codec = codec; + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ unsigned long flags; unsigned long gpio_flags; int ret; int trigger; int irq; + struct wm0010_priv *wm0010; - wm0010->codec = codec; + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); init_completion(&wm0010->boot_completion); @@ -850,7 +871,7 @@ static int wm0010_probe(struct snd_soc_codec *codec) } } else { dev_err(wm0010->dev, "No reset GPIO configured\n"); - return ret; + return -EINVAL; } irq = spi->irq; @@ -862,9 +883,11 @@ static int wm0010_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, "wm0010", wm0010); - if (ret) + if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", irq, ret); + return ret; + } wm0010->irq = irq; if (spi->max_speed_hz) @@ -876,29 +899,6 @@ static int wm0010_probe(struct snd_soc_codec *codec) wm0010->state = WM0010_POWER_OFF; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - return 0; -} - -static int __devinit wm0010_spi_probe(struct spi_device *spi) -{ - struct wm0010_priv *wm0010; - int ret; - - wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), - GFP_KERNEL); - if (!wm0010) - return -ENOMEM; - - mutex_init(&wm0010->lock); - spin_lock_init(&wm0010->irq_lock); - - spi_set_drvdata(spi, wm0010); - wm0010->dev = &spi->dev; - - if (dev_get_platdata(&spi->dev)) - memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), - sizeof(wm0010->pdata)); - ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm0010, wm0010_dai, ARRAY_SIZE(wm0010_dai)); -- cgit v1.2.3 From 28739dfcffaad629b28cbab947193b259f745ea9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 22 Aug 2012 13:11:40 +0300 Subject: ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls In ddevice tree booted kernel all device have unique name and their device id is set to 0. Use the mcbsp->id for checking to decide which control set we should add for McBSP sidetone handling. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2e1750e2ab31..20d30c9ae573 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -722,7 +722,7 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) if (!mcbsp->st_data) return -ENODEV; - switch (cpu_dai->id) { + switch (mcbsp->id) { case 2: /* McBSP 2 */ return snd_soc_add_dai_controls(cpu_dai, omap_mcbsp2_st_controls, -- cgit v1.2.3 From 8a88df4cda5eaed97e027f9c9e76012a7113bf9a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 22 Aug 2012 13:11:41 +0300 Subject: ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port When asked to add the ST controls warn only if the st_data is missing. In this way we do not block the otherwise functional card to probe. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 20d30c9ae573..c1f466e1b8b6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -719,8 +719,10 @@ int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - if (!mcbsp->st_data) - return -ENODEV; + if (!mcbsp->st_data) { + dev_warn(mcbsp->dev, "No sidetone data for port\n"); + return 0; + } switch (mcbsp->id) { case 2: /* McBSP 2 */ -- cgit v1.2.3 From 8996a31c58374742dfb555675a8569617572d82c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 22 Aug 2012 13:11:42 +0300 Subject: ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls To remove duplicated code from the driver. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index c1f466e1b8b6..1b9f5ebba44f 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -688,31 +688,22 @@ static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = { - SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0, - omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), - OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume", - -32768, 32767, - omap_mcbsp_get_st_ch0_volume, - omap_mcbsp_set_st_ch0_volume), - OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume", - -32768, 32767, - omap_mcbsp_get_st_ch1_volume, - omap_mcbsp_set_st_ch1_volume), -}; +#define OMAP_MCBSP_ST_CONTROLS(port) \ +static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \ +SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0, \ + omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch0_volume, \ + omap_mcbsp_set_st_ch0_volume), \ +OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \ + -32768, 32767, \ + omap_mcbsp_get_st_ch1_volume, \ + omap_mcbsp_set_st_ch1_volume), \ +} -static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { - SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0, - omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), - OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume", - -32768, 32767, - omap_mcbsp_get_st_ch0_volume, - omap_mcbsp_set_st_ch0_volume), - OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume", - -32768, 32767, - omap_mcbsp_get_st_ch1_volume, - omap_mcbsp_set_st_ch1_volume), -}; +OMAP_MCBSP_ST_CONTROLS(2); +OMAP_MCBSP_ST_CONTROLS(3); int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) { -- cgit v1.2.3 From db61550931957ee6c7dba751662919424b4344f3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 22 Aug 2012 13:11:43 +0300 Subject: ASoC: omap-mcbsp: Single macro for st channel volume set/get Since we always need to have set and get callbacks for McBSP sidetone it makes sense to combine the two macro to create the two callbacks. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1b9f5ebba44f..9dff177cd4e2 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -619,9 +619,9 @@ static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, return 0; } -#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(channel) \ +#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel) \ static int \ -omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ @@ -637,11 +637,10 @@ omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ \ /* OMAP McBSP implementation uses index values 0..4 */ \ return omap_st_set_chgain(mcbsp, channel, val); \ -} - -#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(channel) \ +} \ + \ static int \ -omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ @@ -655,10 +654,8 @@ omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ return 0; \ } -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(1) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(0) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(1) +OMAP_MCBSP_ST_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_CHANNEL_VOLUME(1) static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 8c3f1b1cbc32d1fa899e3bd9edf427f0f2e7a843 Mon Sep 17 00:00:00 2001 From: Markus Bollinger Date: Fri, 24 Aug 2012 14:54:57 +0200 Subject: ALSA: pcxhr: Add 8 new sound cards add new sound cards VX442HR VX442e PCX442HR PCX442e VX822HR VX822e PCX822HR and PCX822e Signed-off-by: Markus Bollinger Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 24 ++++++++++++++++++++++++ sound/pci/pcxhr/pcxhr_hwdep.c | 6 +++--- 2 files changed, 27 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index e3ac1f768ff6..be4f1456009a 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -91,6 +91,14 @@ enum { PCI_ID_PCX924E, PCI_ID_PCX924HRMIC, PCI_ID_PCX924E_MIC, + PCI_ID_VX442HR, + PCI_ID_PCX442HR, + PCI_ID_VX442E, + PCI_ID_PCX442E, + PCI_ID_VX822HR, + PCI_ID_PCX822HR, + PCI_ID_VX822E, + PCI_ID_PCX822E, PCI_ID_LAST }; @@ -121,6 +129,14 @@ static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { { 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, }, { 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, }, { 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, }, + { 0x10b5, 0x9656, 0x1369, 0xd001, 0, 0, PCI_ID_VX442HR, }, + { 0x10b5, 0x9656, 0x1369, 0xd101, 0, 0, PCI_ID_PCX442HR, }, + { 0x10b5, 0x9056, 0x1369, 0xd021, 0, 0, PCI_ID_VX442E, }, + { 0x10b5, 0x9056, 0x1369, 0xd121, 0, 0, PCI_ID_PCX442E, }, + { 0x10b5, 0x9656, 0x1369, 0xd201, 0, 0, PCI_ID_VX822HR, }, + { 0x10b5, 0x9656, 0x1369, 0xd301, 0, 0, PCI_ID_PCX822HR, }, + { 0x10b5, 0x9056, 0x1369, 0xd221, 0, 0, PCI_ID_VX822E, }, + { 0x10b5, 0x9056, 0x1369, 0xd321, 0, 0, PCI_ID_PCX822E, }, { 0, } }; @@ -160,6 +176,14 @@ static struct board_parameters pcxhr_board_params[] = { [PCI_ID_PCX924E] = { "PCX924e", 1, 1, 5, 44 }, [PCI_ID_PCX924HRMIC] = { "PCX924HR-Mic", 1, 1, 5, 44 }, [PCI_ID_PCX924E_MIC] = { "PCX924e-Mic", 1, 1, 5, 44 }, +[PCI_ID_VX442HR] = { "VX442HR", 2, 2, 0, 41 }, +[PCI_ID_PCX442HR] = { "PCX442HR", 2, 2, 0, 41 }, +[PCI_ID_VX442E] = { "VX442e", 2, 2, 1, 41 }, +[PCI_ID_PCX442E] = { "PCX442e", 2, 2, 1, 41 }, +[PCI_ID_VX822HR] = { "VX822HR", 4, 1, 2, 42 }, +[PCI_ID_PCX822HR] = { "PCX822HR", 4, 1, 2, 42 }, +[PCI_ID_VX822E] = { "VX822e", 4, 1, 3, 42 }, +[PCI_ID_PCX822E] = { "PCX822e", 4, 1, 3, 42 }, }; /* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */ diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index ec1587cddb0c..bf207e317f71 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -66,10 +66,10 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - /* test 8 or 12 phys out */ - if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2) + /* test 4, 8 or 12 phys out */ + if ((rmh.stat[0] & MASK_FIRST_FIELD) < mgr->playback_chips * 2) return -EINVAL; - /* test 8 or 2 phys in */ + /* test 4, 8 or 2 phys in */ if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) < mgr->capture_chips * 2) return -EINVAL; -- cgit v1.2.3 From 9bb280a2eb9f4bc951549dceb1d4e0b0a316eb95 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 27 Aug 2012 17:00:26 +0530 Subject: ASoC: tlv320aic26: Use module_spi_driver module_spi_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 85944e953578..b1f6982c7c9c 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -444,14 +444,4 @@ static struct spi_driver aic26_spi = { .remove = aic26_spi_remove, }; -static int __init aic26_init(void) -{ - return spi_register_driver(&aic26_spi); -} -module_init(aic26_init); - -static void __exit aic26_exit(void) -{ - spi_unregister_driver(&aic26_spi); -} -module_exit(aic26_exit); +module_spi_driver(aic26_spi); -- cgit v1.2.3 From a5c8878017dd3b51f6f97a36d90c405f8061fe83 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 27 Aug 2012 17:00:27 +0530 Subject: ASoC: wm8770: Use module_spi_driver module_spi_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/wm8770.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index a5127b4ff9e1..c7c0034d3966 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -724,24 +724,7 @@ static struct spi_driver wm8770_spi_driver = { .remove = __devexit_p(wm8770_spi_remove) }; -static int __init wm8770_modinit(void) -{ - int ret = 0; - - ret = spi_register_driver(&wm8770_spi_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8770 SPI driver: %d\n", - ret); - } - return ret; -} -module_init(wm8770_modinit); - -static void __exit wm8770_exit(void) -{ - spi_unregister_driver(&wm8770_spi_driver); -} -module_exit(wm8770_exit); +module_spi_driver(wm8770_spi_driver); MODULE_DESCRIPTION("ASoC WM8770 driver"); MODULE_AUTHOR("Dimitris Papastamos "); -- cgit v1.2.3 From 2a9a9c876fd6486978a24cd8bc72bd1aeb228b7b Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 27 Aug 2012 17:00:28 +0530 Subject: ASoC: ad1836: Use module_spi_driver module_spi_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index c67b50d8b317..ae1eb51bc9da 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -379,17 +379,7 @@ static struct spi_driver ad1836_spi_driver = { .id_table = ad1836_ids, }; -static int __init ad1836_init(void) -{ - return spi_register_driver(&ad1836_spi_driver); -} -module_init(ad1836_init); - -static void __exit ad1836_exit(void) -{ - spi_unregister_driver(&ad1836_spi_driver); -} -module_exit(ad1836_exit); +module_spi_driver(ad1836_spi_driver); MODULE_DESCRIPTION("ASoC ad1836 driver"); MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); -- cgit v1.2.3 From c24fdc886fde9ce7bda8115b9c2b338818796c65 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:44 +0530 Subject: ASoC: tlv320aic3x: Add device tree bindings Device tree support for tlv320aic3x CODEC driver. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic3x.txt | 20 ++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 31 ++++++++++++++++++++++ 2 files changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt new file mode 100644 index 000000000000..e7b98f41fa5f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -0,0 +1,20 @@ +Texas Instruments - tlv320aic3x Codec module + +The tlv320aic3x serial control bus communicates through I2C protocols + +Required properties: +- compatible - "string" - "ti,tlv320aic3x" +- reg - - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai3x-gpio-func - - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + +Example: + +tlv320aic3x: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; +}; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 01485bd51404..5708a973a776 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include @@ -1457,6 +1458,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, { struct aic3x_pdata *pdata = i2c->dev.platform_data; struct aic3x_priv *aic3x; + struct aic3x_setup_data *ai3x_setup; + struct device_node *np = i2c->dev.of_node; int ret; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1471,6 +1474,25 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + } else if (np) { + ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), + GFP_KERNEL); + if (ai3x_setup == NULL) { + dev_err(&i2c->dev, "failed to create private data\n"); + return -ENOMEM; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret >= 0) + aic3x->gpio_reset = ret; + else + aic3x->gpio_reset = -1; + + if (of_property_read_u32_array(np, "ai3x-gpio-func", + ai3x_setup->gpio_func, 2) >= 0) { + aic3x->setup = ai3x_setup; + } + } else { aic3x->gpio_reset = -1; } @@ -1488,11 +1510,20 @@ static int aic3x_i2c_remove(struct i2c_client *client) return 0; } +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic3x_of_match[] = { + { .compatible = "ti,tlv320aic3x", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); +#endif + /* machine i2c codec control layer */ static struct i2c_driver aic3x_i2c_driver = { .driver = { .name = "tlv320aic3x-codec", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic3x_of_match), }, .probe = aic3x_i2c_probe, .remove = aic3x_i2c_remove, -- cgit v1.2.3 From f08095a408bf6489b4a710d794ae6d5475a007ef Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:39 +0530 Subject: ASoC: davinci: davinci-pcm does not need to be a plaform_driver Same as the commit 518de86 (ASoC: tegra: register 'platform' from DAIs, get rid of pdev). It makes davinci-pcm not a platform_driver but helper to register "platform", so that the platform_device for davinci-pcm can be saved completely. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 15 ++++++++------- sound/soc/davinci/davinci-i2s.c | 10 ++++++++++ sound/soc/davinci/davinci-mcasp.c | 10 ++++++++++ sound/soc/davinci/davinci-pcm.c | 23 ++++++----------------- sound/soc/davinci/davinci-pcm.h | 3 +++ sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/davinci/davinci-vcif.c | 8 ++++++++ 7 files changed, 46 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index c80c20a89b11..4b37e2ac4685 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -159,7 +159,7 @@ static struct snd_soc_dai_link dm6446_evm_dai = { .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcbsp", .init = evm_aic3x_init, .ops = &evm_ops, }; @@ -170,7 +170,7 @@ static struct snd_soc_dai_link dm355_evm_dai = { .cpu_dai_name = "davinci-mcbsp.1", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-001b", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcbsp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; @@ -184,14 +184,15 @@ static struct snd_soc_dai_link dm365_evm_dai = { .init = evm_aic3x_init, .codec_name = "tlv320aic3x-codec.1-0018", .ops = &evm_ops, + .platform_name = "davinci-mcbsp", #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", .stream_name = "CQ93", .cpu_dai_name = "davinci-vcif", .codec_dai_name = "cq93vc-hifi", .codec_name = "cq93vc-codec", + .platform_name = "avinci-vcif", #endif - .platform_name = "davinci-pcm-audio", }; static struct snd_soc_dai_link dm6467_evm_dai[] = { @@ -200,7 +201,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", - .platform_name ="davinci-pcm-audio", + .platform_name = "davinci-mcasp.0", .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, @@ -211,7 +212,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .cpu_dai_name= "davinci-mcasp.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif_dit", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcasp.1", .ops = &evm_spdif_ops, }, }; @@ -222,7 +223,7 @@ static struct snd_soc_dai_link da830_evm_dai = { .cpu_dai_name = "davinci-mcasp.1", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcasp.1", .init = evm_aic3x_init, .ops = &evm_ops, }; @@ -233,7 +234,7 @@ static struct snd_soc_dai_link da850_evm_dai = { .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic3x-hifi", .codec_name = "tlv320aic3x-codec.1-0018", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcasp.0", .init = evm_aic3x_init, .ops = &evm_ops, }; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 0a74b9587a2c..407df7233d6d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -732,8 +732,16 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; + ret = davinci_soc_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto err_unregister_dai; + } + return 0; +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); @@ -745,6 +753,8 @@ static int davinci_i2s_remove(struct platform_device *pdev) struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); + davinci_soc_platform_unregister(&pdev->dev); + clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d919fb8de7a3..8f3c5a4cf537 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -945,8 +945,17 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; + + ret = davinci_soc_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + goto err_unregister_dai; + } + return 0; +err_unregister_dai: + snd_soc_unregister_dai(&pdev->dev); err_release_clk: pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -957,6 +966,7 @@ static int davinci_mcasp_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&pdev->dev); + davinci_soc_platform_unregister(&pdev->dev); pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 97d77b298968..4b70828beedb 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -864,28 +864,17 @@ static struct snd_soc_platform_driver davinci_soc_platform = { .pcm_free = davinci_pcm_free, }; -static int __devinit davinci_soc_platform_probe(struct platform_device *pdev) +int davinci_soc_platform_register(struct device *dev) { - return snd_soc_register_platform(&pdev->dev, &davinci_soc_platform); + return snd_soc_register_platform(dev, &davinci_soc_platform); } +EXPORT_SYMBOL_GPL(davinci_soc_platform_register); -static int __devexit davinci_soc_platform_remove(struct platform_device *pdev) +void davinci_soc_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(&pdev->dev); - return 0; + snd_soc_unregister_platform(dev); } - -static struct platform_driver davinci_pcm_driver = { - .driver = { - .name = "davinci-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = davinci_soc_platform_probe, - .remove = __devexit_p(davinci_soc_platform_remove), -}; - -module_platform_driver(davinci_pcm_driver); +EXPORT_SYMBOL_GPL(davinci_soc_platform_unregister); MODULE_AUTHOR("Vladimir Barinov"); MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index c0d6c9be4b4d..5e5516460461 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -28,4 +28,7 @@ struct davinci_pcm_dma_params { unsigned int fifo_level; }; +int davinci_soc_platform_register(struct device *dev); +void davinci_soc_platform_unregister(struct device *dev); + #endif diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index f71175b29e38..5be65aae7e0e 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link sffsdr_dai = { .cpu_dai_name = "davinci-mcbsp", .codec_dai_name = "pcm3008-hifi", .codec_name = "pcm3008-codec", - .platform_name = "davinci-pcm-audio", + .platform_name = "davinci-mcbsp", .ops = &sffsdr_ops, }; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index da030ff883d5..07bde2e6f84e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -240,12 +240,20 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } + ret = davinci_soc_platform_register(&pdev->dev); + if (ret) { + dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + snd_soc_unregister_dai(&pdev->dev); + return ret; + } + return 0; } static int davinci_vcif_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&pdev->dev); + davinci_soc_platform_unregister(&pdev->dev); return 0; } -- cgit v1.2.3 From 896f66b7de293644e65cf62600e4933af954dcf2 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:41 +0530 Subject: ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific Davinci McASP header & driver are shared by few OMAP platforms (like TI81xx, AM335x). Splitting asp header into Davinci platform specific header and Audio specific header helps to share them across platforms. Audio specific defines is moved to to common so that the header can be accessed by all related platforms. While here, correct the header usage (remove multiple header re-definitions and unused headers) and remove platform names from structures comments and enum. Also some some coding style errors. Signed-off-by: Hebbar, Gururaja Acked-by: Vaibhav Bedia Signed-off-by: Mark Brown --- arch/arm/mach-davinci/asp.h | 49 +++++++++++ arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 1 + arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/asp.h | 137 ----------------------------- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- include/linux/platform_data/davinci_asp.h | 104 ++++++++++++++++++++++ sound/soc/davinci/davinci-evm.c | 3 - sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.h | 3 +- sound/soc/davinci/davinci-pcm.c | 1 - sound/soc/davinci/davinci-pcm.h | 3 +- 15 files changed, 165 insertions(+), 152 deletions(-) create mode 100644 arch/arm/mach-davinci/asp.h delete mode 100644 arch/arm/mach-davinci/include/mach/asp.h create mode 100644 include/linux/platform_data/davinci_asp.h (limited to 'sound') diff --git a/arch/arm/mach-davinci/asp.h b/arch/arm/mach-davinci/asp.h new file mode 100644 index 000000000000..d9b2acd12393 --- /dev/null +++ b/arch/arm/mach-davinci/asp.h @@ -0,0 +1,49 @@ +/* + * TI DaVinci Audio definitions + */ +#ifndef __ASM_ARCH_DAVINCI_ASP_H +#define __ASM_ARCH_DAVINCI_ASP_H + +/* Bases of dm644x and dm355 register banks */ +#define DAVINCI_ASP0_BASE 0x01E02000 +#define DAVINCI_ASP1_BASE 0x01E04000 + +/* Bases of dm365 register banks */ +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + +/* Bases of da830 McASP1 register banks */ +#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 + +/* EDMA channels of dm644x and dm355 */ +#define DAVINCI_DMA_ASP0_TX 2 +#define DAVINCI_DMA_ASP0_RX 3 +#define DAVINCI_DMA_ASP1_TX 8 +#define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 + +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + +/* EDMA channels of da830 McASP1 */ +#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 +#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 + +/* Interrupts */ +#define DAVINCI_ASP0_RX_INT IRQ_MBRINT +#define DAVINCI_ASP0_TX_INT IRQ_MBXINT +#define DAVINCI_ASP1_RX_INT IRQ_MBRINT +#define DAVINCI_ASP1_TX_INT IRQ_MBXINT + +#endif /* __ASM_ARCH_DAVINCI_ASP_H */ diff --git a/arch/arm/mach-davinci/davinci.h b/arch/arm/mach-davinci/davinci.h index 8db0fc6809dd..8661b2013527 100644 --- a/arch/arm/mach-davinci/davinci.h +++ b/arch/arm/mach-davinci/davinci.h @@ -22,10 +22,11 @@ #include #include #include +#include -#include #include #include +#include #include #include diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 4735d64fd6f2..bd2f72b414bc 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -24,6 +24,7 @@ #include #include "clock.h" +#include "asp.h" #define DA8XX_TPCC_BASE 0x01c00000 #define DA8XX_TPTC0_BASE 0x01c08000 diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 678cd99b7336..e47a3f0e8ac4 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -26,13 +26,13 @@ #include #include #include -#include #include #include #include "davinci.h" #include "clock.h" #include "mux.h" +#include "asp.h" #define DM355_UART2_BASE (IO_PHYS + 0x206000) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index a50d49de1883..f473745d6e3c 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include #include @@ -37,6 +36,7 @@ #include "davinci.h" #include "clock.h" #include "mux.h" +#include "asp.h" #define DM365_REF_FREQ 24000000 /* 24 MHz on the DM365 EVM */ diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index c8b866657fcb..0755d466221a 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -23,12 +23,12 @@ #include #include #include -#include #include #include "davinci.h" #include "clock.h" #include "mux.h" +#include "asp.h" /* * Device specific clocks diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index 9eb87c1d1edd..97c0f8e555bd 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -24,12 +24,12 @@ #include #include #include -#include #include #include "davinci.h" #include "clock.h" #include "mux.h" +#include "asp.h" #define DAVINCI_VPIF_BASE (0x01C12000) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h deleted file mode 100644 index 9aa240909a2c..000000000000 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ /dev/null @@ -1,137 +0,0 @@ -/* - * - DaVinci Audio Serial Port support - */ -#ifndef __ASM_ARCH_DAVINCI_ASP_H -#define __ASM_ARCH_DAVINCI_ASP_H - -#include -#include - -/* Bases of dm644x and dm355 register banks */ -#define DAVINCI_ASP0_BASE 0x01E02000 -#define DAVINCI_ASP1_BASE 0x01E04000 - -/* Bases of dm365 register banks */ -#define DAVINCI_DM365_ASP0_BASE 0x01D02000 - -/* Bases of dm646x register banks */ -#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 -#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 - -/* Bases of da850/da830 McASP0 register banks */ -#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 - -/* Bases of da830 McASP1 register banks */ -#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 - -/* EDMA channels of dm644x and dm355 */ -#define DAVINCI_DMA_ASP0_TX 2 -#define DAVINCI_DMA_ASP0_RX 3 -#define DAVINCI_DMA_ASP1_TX 8 -#define DAVINCI_DMA_ASP1_RX 9 - -/* EDMA channels of dm646x */ -#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 -#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 -#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 - -/* EDMA channels of da850/da830 McASP0 */ -#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 -#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 - -/* EDMA channels of da830 McASP1 */ -#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 -#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 - -/* Interrupts */ -#define DAVINCI_ASP0_RX_INT IRQ_MBRINT -#define DAVINCI_ASP0_TX_INT IRQ_MBXINT -#define DAVINCI_ASP1_RX_INT IRQ_MBRINT -#define DAVINCI_ASP1_TX_INT IRQ_MBXINT - -struct snd_platform_data { - u32 tx_dma_offset; - u32 rx_dma_offset; - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned int codec_fmt; - /* - * Allowing this is more efficient and eliminates left and right swaps - * caused by underruns, but will swap the left and right channels - * when compared to previous behavior. - */ - unsigned enable_channel_combine:1; - unsigned sram_size_playback; - unsigned sram_size_capture; - - /* - * If McBSP peripheral gets the clock from an external pin, - * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR - * and MCBSP_CLKS. - * Depending on different hardware connections it is possible - * to use this setting to change the behaviour of McBSP - * driver. The dm365_clk_input_pin enum is available for dm365 - */ - int clk_input_pin; - - /* - * This flag works when both clock and FS are outputs for the cpu - * and makes clock more accurate (FS is not symmetrical and the - * clock is very fast. - * The clock becoming faster is named - * i2s continuous serial clock (I2S_SCK) and it is an externally - * visible bit clock. - * - * first line : WordSelect - * second line : ContinuousSerialClock - * third line: SerialData - * - * SYMMETRICAL APPROACH: - * _______________________ LEFT - * _| RIGHT |______________________| - * _ _ _ _ _ _ _ _ - * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ - * _ _ _ _ _ _ _ _ - * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ - * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ - * - * ACCURATE CLOCK APPROACH: - * ______________ LEFT - * _| RIGHT |_______________________________| - * _ _ _ _ _ _ _ _ _ - * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | - * _ _ _ _ dummy cycles - * _/ \_ ... _/ \_/ \_ ... _/ \__________________ - * \_/ \_/ \_/ \_/ - * - */ - bool i2s_accurate_sck; - - /* McASP specific fields */ - int tdm_slots; - u8 op_mode; - u8 num_serializer; - u8 *serial_dir; - u8 version; - u8 txnumevt; - u8 rxnumevt; -}; - -enum { - MCASP_VERSION_1 = 0, /* DM646x */ - MCASP_VERSION_2, /* DA8xx/OMAPL1x */ -}; - -enum dm365_clk_input_pin { - MCBSP_CLKR = 0, /* DM365 */ - MCBSP_CLKS, -}; - -#define INACTIVE_MODE 0 -#define TX_MODE 1 -#define RX_MODE 2 - -#define DAVINCI_MCASP_IIS_MODE 0 -#define DAVINCI_MCASP_DIT_MODE 1 - -#endif /* __ASM_ARCH_DAVINCI_ASP_H */ diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h index a2f1f274f189..c74a6abef187 100644 --- a/arch/arm/mach-davinci/include/mach/da8xx.h +++ b/arch/arm/mach-davinci/include/mach/da8xx.h @@ -16,11 +16,11 @@ #include #include #include +#include #include #include #include -#include #include #include #include diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h new file mode 100644 index 000000000000..79c26aa11db6 --- /dev/null +++ b/include/linux/platform_data/davinci_asp.h @@ -0,0 +1,104 @@ +/* + * TI DaVinci Audio Serial Port support + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __DAVINCI_ASP_H +#define __DAVINCI_ASP_H + +struct snd_platform_data { + u32 tx_dma_offset; + u32 rx_dma_offset; + int asp_chan_q; /* event queue number for ASP channel */ + int ram_chan_q; /* event queue number for RAM channel */ + unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned enable_channel_combine:1; + unsigned sram_size_playback; + unsigned sram_size_capture; + + /* + * If McBSP peripheral gets the clock from an external pin, + * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR + * and MCBSP_CLKS. + * Depending on different hardware connections it is possible + * to use this setting to change the behaviour of McBSP + * driver. + */ + int clk_input_pin; + + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clock (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ +}; + +enum mcbsp_clk_input_pin { + MCBSP_CLKR = 0, /* as in DM365 */ + MCBSP_CLKS, +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + +#endif diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4b37e2ac4685..ab0ad4591b08 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -22,9 +22,6 @@ #include #include -#include -#include - #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 407df7233d6d..821831207180 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -23,8 +24,6 @@ #include #include -#include - #include "davinci-pcm.h" #include "davinci-i2s.h" diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 51479f9ee909..0de9ed6ce038 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -19,7 +19,8 @@ #define DAVINCI_MCASP_H #include -#include +#include + #include "davinci-pcm.h" #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 4b70828beedb..93ea3bf567e1 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -23,7 +23,6 @@ #include #include -#include #include #include "davinci-pcm.h" diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 5e5516460461..fc4d01cdd8c9 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,9 +12,8 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include #include -#include - struct davinci_pcm_dma_params { int channel; /* sync dma channel ID */ -- cgit v1.2.3 From 3e3b8c3415b15adb5a7ffcbfbeb360e7c9f5f4f7 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 27 Aug 2012 18:56:42 +0530 Subject: ASoC: Davinci: McASP: add device tree support for McASP Add device tree probe for McASP driver. Note: DMA parameters are not populated from DT and will be done later. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 44 ++++++++ sound/soc/davinci/davinci-mcasp.c | 124 ++++++++++++++++++++- 2 files changed, 167 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt new file mode 100644 index 000000000000..e6148eca2942 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -0,0 +1,44 @@ +Texas Instruments McASP controller + +Required properties: +- compatible : + "ti,dm646x-mcasp-audio" : for DM646x platforms + "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + +- reg : Should contain McASP registers offset and length +- interrupts : Interrupt number for McASP +- op-mode : I2S/DIT ops mode. +- tdm-slots : Slots for TDM operation. +- num-serializer : Serializers used by McASP. +- serial-dir : A list of serializer pin mode. The list number should be equal + to "num-serializer" parameter. Each entry is a number indication + serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) + + +Optional properties: + +- ti,hwmods : Must be "mcasp", n is controller instance starting 0 +- tx-num-evt : FIFO levels. +- rx-num-evt : FIFO levels. +- sram-size-playback : size of sram to be allocated during playback +- sram-size-capture : size of sram to be allocated during capture + +Example: + +mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + #address-cells = <1>; + #size-cells = <0>; + reg = <0x100000 0x3000>; + interrupts = <82 83>; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + num-serializer = <16>; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; +}; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8f3c5a4cf537..7ecf19dfb07c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -22,6 +22,9 @@ #include #include #include +#include +#include +#include #include #include @@ -856,6 +859,114 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { }; +static const struct of_device_id mcasp_dt_ids[] = { + { + .compatible = "ti,dm646x-mcasp-audio", + .data = (void *)MCASP_VERSION_1, + }, + { + .compatible = "ti,da830-mcasp-audio", + .data = (void *)MCASP_VERSION_2, + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mcasp_dt_ids); + +static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( + struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_platform_data *pdata = NULL; + const struct of_device_id *match = + of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + + const u32 *of_serial_dir32; + u8 *of_serial_dir; + u32 val; + int i, ret = 0; + + if (pdev->dev.platform_data) { + pdata = pdev->dev.platform_data; + return pdata; + } else if (match) { + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) { + ret = -ENOMEM; + goto nodata; + } + } else { + /* control shouldn't reach here. something is wrong */ + ret = -EINVAL; + goto nodata; + } + + if (match->data) + pdata->version = (u8)((int)match->data); + + ret = of_property_read_u32(np, "op-mode", &val); + if (ret >= 0) + pdata->op_mode = val; + + ret = of_property_read_u32(np, "tdm-slots", &val); + if (ret >= 0) + pdata->tdm_slots = val; + + ret = of_property_read_u32(np, "num-serializer", &val); + if (ret >= 0) + pdata->num_serializer = val; + + of_serial_dir32 = of_get_property(np, "serial-dir", &val); + val /= sizeof(u32); + if (val != pdata->num_serializer) { + dev_err(&pdev->dev, + "num-serializer(%d) != serial-dir size(%d)\n", + pdata->num_serializer, val); + ret = -EINVAL; + goto nodata; + } + + if (of_serial_dir32) { + of_serial_dir = devm_kzalloc(&pdev->dev, + (sizeof(*of_serial_dir) * val), + GFP_KERNEL); + if (!of_serial_dir) { + ret = -ENOMEM; + goto nodata; + } + + for (i = 0; i < pdata->num_serializer; i++) + of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]); + + pdata->serial_dir = of_serial_dir; + } + + ret = of_property_read_u32(np, "tx-num-evt", &val); + if (ret >= 0) + pdata->txnumevt = val; + + ret = of_property_read_u32(np, "rx-num-evt", &val); + if (ret >= 0) + pdata->rxnumevt = val; + + ret = of_property_read_u32(np, "sram-size-playback", &val); + if (ret >= 0) + pdata->sram_size_playback = val; + + ret = of_property_read_u32(np, "sram-size-capture", &val); + if (ret >= 0) + pdata->sram_size_capture = val; + + return pdata; + +nodata: + if (ret < 0) { + dev_err(&pdev->dev, "Error populating platform data, err %d\n", + ret); + pdata = NULL; + } + return pdata; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct davinci_pcm_dma_params *dma_data; @@ -864,11 +975,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct davinci_audio_dev *dev; int ret; + if (!pdev->dev.platform_data && !pdev->dev.of_node) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_audio_dev), GFP_KERNEL); if (!dev) return -ENOMEM; + pdata = davinci_mcasp_set_pdata_from_of(pdev); + if (!pdata) { + dev_err(&pdev->dev, "no platform data\n"); + return -EINVAL; + } + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "no mem resource?\n"); @@ -882,7 +1004,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EBUSY; } - pdata = pdev->dev.platform_data; pm_runtime_enable(&pdev->dev); ret = pm_runtime_get_sync(&pdev->dev); @@ -980,6 +1101,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mcasp_dt_ids), }, }; -- cgit v1.2.3 From 5d908ab941e90291b5c31d990bdb8e87fa3f7146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Aug 2012 18:40:10 +0200 Subject: ALSA: hda - Make clear built-in driver optimization Use unsigned int to make clear that the codes required only for modules will be reduced by the compiler optimization. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a2f35ccbf05..5f9da87d8762 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -808,7 +808,7 @@ find_codec_preset(struct hda_codec *codec) { struct hda_codec_preset_list *tbl; const struct hda_codec_preset *preset; - int mod_requested = 0; + unsigned int mod_requested = 0; if (is_generic_config(codec)) return NULL; /* use the generic parser */ -- cgit v1.2.3 From 3d721a34e66e8133404bda6852897b818df69b4c Mon Sep 17 00:00:00 2001 From: Sangsu Park Date: Tue, 28 Aug 2012 17:55:49 +0900 Subject: ASoC: SAMSUNG: Change Kconfig to support all SAMSUNG ASoC All SAMSUNG ASoC needs SND_SOC_SAMSUNG configuration. This patch change Kconfig to support all SAMSUNG ASoC. Signed-off-by: Sangsu Park Acked-by: Sangbeom Kim Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f17dd25e0f4a..e7b83179aca2 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on PLAT_SAMSUNG select S3C64XX_DMA if ARCH_S3C64XX select S3C2410_DMA if ARCH_S3C24XX help -- cgit v1.2.3 From 0ed275eff31029c39355828cb48c46c0a006e2f8 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Tue, 21 Aug 2012 19:21:51 +0400 Subject: ASoC: Rename ep93xx soc directory to cirrus This patch is to rename the directory "ep93xx" in "cirrus". Name more accurately reflects the manufacturer and allows to add drivers not only for architecture ep93xx in this directory. Patch not contain any functional changes. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/cirrus/Kconfig | 42 ++++ sound/soc/cirrus/Makefile | 17 ++ sound/soc/cirrus/edb93xx.c | 128 ++++++++++++ sound/soc/cirrus/ep93xx-ac97.c | 435 +++++++++++++++++++++++++++++++++++++++ sound/soc/cirrus/ep93xx-i2s.c | 451 +++++++++++++++++++++++++++++++++++++++++ sound/soc/cirrus/ep93xx-pcm.c | 242 ++++++++++++++++++++++ sound/soc/cirrus/ep93xx-pcm.h | 20 ++ sound/soc/cirrus/simone.c | 90 ++++++++ sound/soc/cirrus/snappercl15.c | 146 +++++++++++++ sound/soc/ep93xx/Kconfig | 42 ---- sound/soc/ep93xx/Makefile | 17 -- sound/soc/ep93xx/edb93xx.c | 128 ------------ sound/soc/ep93xx/ep93xx-ac97.c | 435 --------------------------------------- sound/soc/ep93xx/ep93xx-i2s.c | 451 ----------------------------------------- sound/soc/ep93xx/ep93xx-pcm.c | 242 ---------------------- sound/soc/ep93xx/ep93xx-pcm.h | 20 -- sound/soc/ep93xx/simone.c | 90 -------- sound/soc/ep93xx/snappercl15.c | 146 ------------- 20 files changed, 1573 insertions(+), 1573 deletions(-) create mode 100644 sound/soc/cirrus/Kconfig create mode 100644 sound/soc/cirrus/Makefile create mode 100644 sound/soc/cirrus/edb93xx.c create mode 100644 sound/soc/cirrus/ep93xx-ac97.c create mode 100644 sound/soc/cirrus/ep93xx-i2s.c create mode 100644 sound/soc/cirrus/ep93xx-pcm.c create mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/cirrus/simone.c create mode 100644 sound/soc/cirrus/snappercl15.c delete mode 100644 sound/soc/ep93xx/Kconfig delete mode 100644 sound/soc/ep93xx/Makefile delete mode 100644 sound/soc/ep93xx/edb93xx.c delete mode 100644 sound/soc/ep93xx/ep93xx-ac97.c delete mode 100644 sound/soc/ep93xx/ep93xx-i2s.c delete mode 100644 sound/soc/ep93xx/ep93xx-pcm.c delete mode 100644 sound/soc/ep93xx/ep93xx-pcm.h delete mode 100644 sound/soc/ep93xx/simone.c delete mode 100644 sound/soc/ep93xx/snappercl15.c (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index c24de902f5f5..5da8ca7aee05 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,9 +33,9 @@ config SND_SOC_DMAENGINE_PCM source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" +source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" -source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index c1264007b4ee..bcbf1d00aa85 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,9 +10,9 @@ obj-$(CONFIG_SND_SOC) += generic/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ +obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ -obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ obj-$(CONFIG_SND_SOC) += mid-x86/ diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig new file mode 100644 index 000000000000..88143db7e753 --- /dev/null +++ b/sound/soc/cirrus/Kconfig @@ -0,0 +1,42 @@ +config SND_EP93XX_SOC + tristate "SoC Audio support for the Cirrus Logic EP93xx series" + depends on ARCH_EP93XX && SND_SOC + select SND_SOC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to + the EP93xx I2S or AC97 interfaces. + +config SND_EP93XX_SOC_I2S + tristate + +config SND_EP93XX_SOC_AC97 + tristate + select AC97_BUS + select SND_SOC_AC97_BUS + +config SND_EP93XX_SOC_SNAPPERCL15 + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23 + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. + +config SND_EP93XX_SOC_SIMONE + tristate "SoC Audio support for Simplemachines Sim.One board" + depends on SND_EP93XX_SOC && MACH_SIM_ONE + select SND_EP93XX_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y or M here if you want to add support for AC97 audio on the + Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/cirrus/Makefile b/sound/soc/cirrus/Makefile new file mode 100644 index 000000000000..5514146cbdf0 --- /dev/null +++ b/sound/soc/cirrus/Makefile @@ -0,0 +1,17 @@ +# EP93xx Platform Support +snd-soc-ep93xx-objs := ep93xx-pcm.o +snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o +snd-soc-ep93xx-ac97-objs := ep93xx-ac97.o + +obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o +obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o +obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o + +# EP93XX Machine Support +snd-soc-snappercl15-objs := snappercl15.o +snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o + +obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o +obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c new file mode 100644 index 000000000000..e01cb02abd3a --- /dev/null +++ b/sound/soc/cirrus/edb93xx.c @@ -0,0 +1,128 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "ep93xx-pcm.h" + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int mclk_rate; + unsigned int rate = params_rate(params); + + /* + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates + */ + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .owner = THIS_MODULE, + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static int __devinit edb93xx_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_edb93xx; + int ret; + + ret = ep93xx_i2s_acquire(); + if (ret) + return ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); + } + + return ret; +} + +static int __devexit edb93xx_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + ep93xx_i2s_release(); + + return 0; +} + +static struct platform_driver edb93xx_driver = { + .driver = { + .name = "edb93xx-audio", + .owner = THIS_MODULE, + }, + .probe = edb93xx_probe, + .remove = __devexit_p(edb93xx_remove), +}; + +module_platform_driver(edb93xx_driver); + +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:edb93xx-audio"); diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c new file mode 100644 index 000000000000..bdffab33e160 --- /dev/null +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -0,0 +1,435 @@ +/* + * ASoC driver for Cirrus Logic EP93xx AC97 controller. + * + * Copyright (c) 2010 Mika Westerberg + * + * Based on s3c-ac97 ASoC driver by Jaswinder Singh. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include +#include "ep93xx-pcm.h" + +/* + * Per channel (1-4) registers. + */ +#define AC97CH(n) (((n) - 1) * 0x20) + +#define AC97DR(n) (AC97CH(n) + 0x0000) + +#define AC97RXCR(n) (AC97CH(n) + 0x0004) +#define AC97RXCR_REN BIT(0) +#define AC97RXCR_RX3 BIT(3) +#define AC97RXCR_RX4 BIT(4) +#define AC97RXCR_CM BIT(15) + +#define AC97TXCR(n) (AC97CH(n) + 0x0008) +#define AC97TXCR_TEN BIT(0) +#define AC97TXCR_TX3 BIT(3) +#define AC97TXCR_TX4 BIT(4) +#define AC97TXCR_CM BIT(15) + +#define AC97SR(n) (AC97CH(n) + 0x000c) +#define AC97SR_TXFE BIT(1) +#define AC97SR_TXUE BIT(6) + +#define AC97RISR(n) (AC97CH(n) + 0x0010) +#define AC97ISR(n) (AC97CH(n) + 0x0014) +#define AC97IE(n) (AC97CH(n) + 0x0018) + +/* + * Global AC97 controller registers. + */ +#define AC97S1DATA 0x0080 +#define AC97S2DATA 0x0084 +#define AC97S12DATA 0x0088 + +#define AC97RGIS 0x008c +#define AC97GIS 0x0090 +#define AC97IM 0x0094 +/* + * Common bits for RGIS, GIS and IM registers. + */ +#define AC97_SLOT2RXVALID BIT(1) +#define AC97_CODECREADY BIT(5) +#define AC97_SLOT2TXCOMPLETE BIT(6) + +#define AC97EOI 0x0098 +#define AC97EOI_WINT BIT(0) +#define AC97EOI_CODECREADY BIT(1) + +#define AC97GCR 0x009c +#define AC97GCR_AC97IFE BIT(0) + +#define AC97RESET 0x00a0 +#define AC97RESET_TIMEDRESET BIT(0) + +#define AC97SYNC 0x00a4 +#define AC97SYNC_TIMEDSYNC BIT(0) + +#define AC97_TIMEOUT msecs_to_jiffies(5) + +/** + * struct ep93xx_ac97_info - EP93xx AC97 controller info structure + * @lock: mutex serializing access to the bus (slot 1 & 2 ops) + * @dev: pointer to the platform device dev structure + * @regs: mapped AC97 controller registers + * @done: bus ops wait here for an interrupt + */ +struct ep93xx_ac97_info { + struct mutex lock; + struct device *dev; + void __iomem *regs; + struct completion done; +}; + +/* currently ALSA only supports a single AC97 device */ +static struct ep93xx_ac97_info *ep93xx_ac97_info; + +static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_out = { + .name = "ac97-pcm-out", + .dma_port = EP93XX_DMA_AAC1, +}; + +static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_in = { + .name = "ac97-pcm-in", + .dma_port = EP93XX_DMA_AAC1, +}; + +static inline unsigned ep93xx_ac97_read_reg(struct ep93xx_ac97_info *info, + unsigned reg) +{ + return __raw_readl(info->regs + reg); +} + +static inline void ep93xx_ac97_write_reg(struct ep93xx_ac97_info *info, + unsigned reg, unsigned val) +{ + __raw_writel(val, info->regs + reg); +} + +static unsigned short ep93xx_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct ep93xx_ac97_info *info = ep93xx_ac97_info; + unsigned short val; + + mutex_lock(&info->lock); + + ep93xx_ac97_write_reg(info, AC97S1DATA, reg); + ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2RXVALID); + if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) { + dev_warn(info->dev, "timeout reading register %x\n", reg); + mutex_unlock(&info->lock); + return -ETIMEDOUT; + } + val = (unsigned short)ep93xx_ac97_read_reg(info, AC97S2DATA); + + mutex_unlock(&info->lock); + return val; +} + +static void ep93xx_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, + unsigned short val) +{ + struct ep93xx_ac97_info *info = ep93xx_ac97_info; + + mutex_lock(&info->lock); + + /* + * Writes to the codec need to be done so that slot 2 is filled in + * before slot 1. + */ + ep93xx_ac97_write_reg(info, AC97S2DATA, val); + ep93xx_ac97_write_reg(info, AC97S1DATA, reg); + + ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2TXCOMPLETE); + if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) + dev_warn(info->dev, "timeout writing register %x\n", reg); + + mutex_unlock(&info->lock); +} + +static void ep93xx_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct ep93xx_ac97_info *info = ep93xx_ac97_info; + + mutex_lock(&info->lock); + + /* + * We are assuming that before this functions gets called, the codec + * BIT_CLK is stopped by forcing the codec into powerdown mode. We can + * control the SYNC signal directly via AC97SYNC register. Using + * TIMEDSYNC the controller will keep the SYNC high > 1us. + */ + ep93xx_ac97_write_reg(info, AC97SYNC, AC97SYNC_TIMEDSYNC); + ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); + if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) + dev_warn(info->dev, "codec warm reset timeout\n"); + + mutex_unlock(&info->lock); +} + +static void ep93xx_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct ep93xx_ac97_info *info = ep93xx_ac97_info; + + mutex_lock(&info->lock); + + /* + * For doing cold reset, we disable the AC97 controller interface, clear + * WINT and CODECREADY bits, and finally enable the interface again. + */ + ep93xx_ac97_write_reg(info, AC97GCR, 0); + ep93xx_ac97_write_reg(info, AC97EOI, AC97EOI_CODECREADY | AC97EOI_WINT); + ep93xx_ac97_write_reg(info, AC97GCR, AC97GCR_AC97IFE); + + /* + * Now, assert the reset and wait for the codec to become ready. + */ + ep93xx_ac97_write_reg(info, AC97RESET, AC97RESET_TIMEDRESET); + ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); + if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) + dev_warn(info->dev, "codec cold reset timeout\n"); + + /* + * Give the codec some time to come fully out from the reset. This way + * we ensure that the subsequent reads/writes will work. + */ + usleep_range(15000, 20000); + + mutex_unlock(&info->lock); +} + +static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) +{ + struct ep93xx_ac97_info *info = dev_id; + unsigned status, mask; + + /* + * Just mask out the interrupt and wake up the waiting thread. + * Interrupts are cleared via reading/writing to slot 1 & 2 registers by + * the waiting thread. + */ + status = ep93xx_ac97_read_reg(info, AC97GIS); + mask = ep93xx_ac97_read_reg(info, AC97IM); + mask &= ~status; + ep93xx_ac97_write_reg(info, AC97IM, mask); + + complete(&info->done); + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = ep93xx_ac97_read, + .write = ep93xx_ac97_write, + .reset = ep93xx_ac97_cold_reset, + .warm_reset = ep93xx_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); + unsigned v = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* + * Enable compact mode, TX slots 3 & 4, and the TX FIFO + * itself. + */ + v |= AC97TXCR_CM; + v |= AC97TXCR_TX3 | AC97TXCR_TX4; + v |= AC97TXCR_TEN; + ep93xx_ac97_write_reg(info, AC97TXCR(1), v); + } else { + /* + * Enable compact mode, RX slots 3 & 4, and the RX FIFO + * itself. + */ + v |= AC97RXCR_CM; + v |= AC97RXCR_RX3 | AC97RXCR_RX4; + v |= AC97RXCR_REN; + ep93xx_ac97_write_reg(info, AC97RXCR(1), v); + } + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* + * As per Cirrus EP93xx errata described below: + * + * http://www.cirrus.com/en/pubs/errata/ER667E2B.pdf + * + * we will wait for the TX FIFO to be empty before + * clearing the TEN bit. + */ + unsigned long timeout = jiffies + AC97_TIMEOUT; + + do { + v = ep93xx_ac97_read_reg(info, AC97SR(1)); + if (time_after(jiffies, timeout)) { + dev_warn(info->dev, "TX timeout\n"); + break; + } + } while (!(v & (AC97SR_TXFE | AC97SR_TXUE))); + + /* disable the TX FIFO */ + ep93xx_ac97_write_reg(info, AC97TXCR(1), 0); + } else { + /* disable the RX FIFO */ + ep93xx_ac97_write_reg(info, AC97RXCR(1), 0); + } + break; + + default: + dev_warn(info->dev, "unknown command %d\n", cmd); + return -EINVAL; + } + + return 0; +} + +static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ep93xx_pcm_dma_params *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &ep93xx_ac97_pcm_out; + else + dma_data = &ep93xx_ac97_pcm_in; + + snd_soc_dai_set_dma_data(dai, substream, dma_data); + return 0; +} + +static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { + .startup = ep93xx_ac97_startup, + .trigger = ep93xx_ac97_trigger, +}; + +static struct snd_soc_dai_driver ep93xx_ac97_dai = { + .name = "ep93xx-ac97", + .id = 0, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ep93xx_ac97_dai_ops, +}; + +static int __devinit ep93xx_ac97_probe(struct platform_device *pdev) +{ + struct ep93xx_ac97_info *info; + struct resource *res; + unsigned int irq; + int ret; + + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENODEV; + + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; + + irq = platform_get_irq(pdev, 0); + if (!irq) + return -ENODEV; + + ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, + IRQF_TRIGGER_HIGH, pdev->name, info); + if (ret) + goto fail; + + dev_set_drvdata(&pdev->dev, info); + + mutex_init(&info->lock); + init_completion(&info->done); + info->dev = &pdev->dev; + + ep93xx_ac97_info = info; + platform_set_drvdata(pdev, info); + + ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); + if (ret) + goto fail; + + return 0; + +fail: + platform_set_drvdata(pdev, NULL); + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); + return ret; +} + +static int __devexit ep93xx_ac97_remove(struct platform_device *pdev) +{ + struct ep93xx_ac97_info *info = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + + /* disable the AC97 controller */ + ep93xx_ac97_write_reg(info, AC97GCR, 0); + + platform_set_drvdata(pdev, NULL); + ep93xx_ac97_info = NULL; + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static struct platform_driver ep93xx_ac97_driver = { + .probe = ep93xx_ac97_probe, + .remove = __devexit_p(ep93xx_ac97_remove), + .driver = { + .name = "ep93xx-ac97", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(ep93xx_ac97_driver); + +MODULE_DESCRIPTION("EP93xx AC97 ASoC Driver"); +MODULE_AUTHOR("Mika Westerberg "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ep93xx-ac97"); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c new file mode 100644 index 000000000000..8df8f6dc474f --- /dev/null +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -0,0 +1,451 @@ +/* + * linux/sound/soc/ep93xx-i2s.c + * EP93xx I2S driver + * + * Copyright (C) 2010 Ryan Mallon + * + * Based on the original driver by: + * Copyright (C) 2007 Chase Douglas + * Copyright (C) 2006 Lennert Buytenhek + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "ep93xx-pcm.h" + +#define EP93XX_I2S_TXCLKCFG 0x00 +#define EP93XX_I2S_RXCLKCFG 0x04 +#define EP93XX_I2S_GLCTRL 0x0C + +#define EP93XX_I2S_TXLINCTRLDATA 0x28 +#define EP93XX_I2S_TXCTRL 0x2C +#define EP93XX_I2S_TXWRDLEN 0x30 +#define EP93XX_I2S_TX0EN 0x34 + +#define EP93XX_I2S_RXLINCTRLDATA 0x58 +#define EP93XX_I2S_RXCTRL 0x5C +#define EP93XX_I2S_RXWRDLEN 0x60 +#define EP93XX_I2S_RX0EN 0x64 + +#define EP93XX_I2S_WRDLEN_16 (0 << 0) +#define EP93XX_I2S_WRDLEN_24 (1 << 0) +#define EP93XX_I2S_WRDLEN_32 (2 << 0) + +#define EP93XX_I2S_LINCTRLDATA_R_JUST (1 << 2) /* Right justify */ + +#define EP93XX_I2S_CLKCFG_LRS (1 << 0) /* lrclk polarity */ +#define EP93XX_I2S_CLKCFG_CKP (1 << 1) /* Bit clock polarity */ +#define EP93XX_I2S_CLKCFG_REL (1 << 2) /* First bit transition */ +#define EP93XX_I2S_CLKCFG_MASTER (1 << 3) /* Master mode */ +#define EP93XX_I2S_CLKCFG_NBCG (1 << 4) /* Not bit clock gating */ + +struct ep93xx_i2s_info { + struct clk *mclk; + struct clk *sclk; + struct clk *lrclk; + struct ep93xx_pcm_dma_params *dma_params; + void __iomem *regs; +}; + +struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = { + [SNDRV_PCM_STREAM_PLAYBACK] = { + .name = "i2s-pcm-out", + .dma_port = EP93XX_DMA_I2S1, + }, + [SNDRV_PCM_STREAM_CAPTURE] = { + .name = "i2s-pcm-in", + .dma_port = EP93XX_DMA_I2S1, + }, +}; + +static inline void ep93xx_i2s_write_reg(struct ep93xx_i2s_info *info, + unsigned reg, unsigned val) +{ + __raw_writel(val, info->regs + reg); +} + +static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info, + unsigned reg) +{ + return __raw_readl(info->regs + reg); +} + +static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) +{ + unsigned base_reg; + int i; + + if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && + (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { + /* Enable clocks */ + clk_enable(info->mclk); + clk_enable(info->sclk); + clk_enable(info->lrclk); + + /* Enable i2s */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1); + } + + /* Enable fifos */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + base_reg = EP93XX_I2S_TX0EN; + else + base_reg = EP93XX_I2S_RX0EN; + for (i = 0; i < 3; i++) + ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1); +} + +static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) +{ + unsigned base_reg; + int i; + + /* Disable fifos */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + base_reg = EP93XX_I2S_TX0EN; + else + base_reg = EP93XX_I2S_RX0EN; + for (i = 0; i < 3; i++) + ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0); + + if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && + (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { + /* Disable i2s */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0); + + /* Disable clocks */ + clk_disable(info->lrclk); + clk_disable(info->sclk); + clk_disable(info->mclk); + } +} + +static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &info->dma_params[substream->stream]); + return 0; +} + +static void ep93xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + + ep93xx_i2s_disable(info, substream->stream); +} + +static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(cpu_dai); + unsigned int clk_cfg, lin_ctrl; + + clk_cfg = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXCLKCFG); + lin_ctrl = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXLINCTRLDATA); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + clk_cfg |= EP93XX_I2S_CLKCFG_REL; + lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + case SND_SOC_DAIFMT_LEFT_J: + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + case SND_SOC_DAIFMT_RIGHT_J: + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + lin_ctrl |= EP93XX_I2S_LINCTRLDATA_R_JUST; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* CPU is master */ + clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec is master */ + clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Negative bit clock, lrclk low on left word */ + clk_cfg &= ~(EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL); + break; + + case SND_SOC_DAIFMT_NB_IF: + /* Negative bit clock, lrclk low on right word */ + clk_cfg &= ~EP93XX_I2S_CLKCFG_CKP; + clk_cfg |= EP93XX_I2S_CLKCFG_REL; + break; + + case SND_SOC_DAIFMT_IB_NF: + /* Positive bit clock, lrclk low on left word */ + clk_cfg |= EP93XX_I2S_CLKCFG_CKP; + clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; + break; + + case SND_SOC_DAIFMT_IB_IF: + /* Positive bit clock, lrclk low on right word */ + clk_cfg |= EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL; + break; + } + + /* Write new register values */ + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXCLKCFG, clk_cfg); + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCLKCFG, clk_cfg); + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXLINCTRLDATA, lin_ctrl); + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXLINCTRLDATA, lin_ctrl); + return 0; +} + +static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + unsigned word_len, div, sdiv, lrdiv; + int err; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = EP93XX_I2S_WRDLEN_16; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + word_len = EP93XX_I2S_WRDLEN_24; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + word_len = EP93XX_I2S_WRDLEN_32; + break; + + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len); + else + ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); + + /* + * EP93xx I2S module can be setup so SCLK / LRCLK value can be + * 32, 64, 128. MCLK / SCLK value can be 2 and 4. + * We set LRCLK equal to `rate' and minimum SCLK / LRCLK + * value is 64, because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + */ + div = clk_get_rate(info->mclk) / params_rate(params); + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } + + err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); + if (err) + return err; + + err = clk_set_rate(info->lrclk, clk_get_rate(info->sclk) / lrdiv); + if (err) + return err; + + ep93xx_i2s_enable(info, substream->stream); + return 0; +} + +static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(cpu_dai); + + if (dir == SND_SOC_CLOCK_IN || clk_id != 0) + return -EINVAL; + + return clk_set_rate(info->mclk, freq); +} + +#ifdef CONFIG_PM +static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + + if (!dai->active) + return 0; + + ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); + ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; +} + +static int ep93xx_i2s_resume(struct snd_soc_dai *dai) +{ + struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); + + if (!dai->active) + return 0; + + ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); + ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; +} +#else +#define ep93xx_i2s_suspend NULL +#define ep93xx_i2s_resume NULL +#endif + +static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { + .startup = ep93xx_i2s_startup, + .shutdown = ep93xx_i2s_shutdown, + .hw_params = ep93xx_i2s_hw_params, + .set_sysclk = ep93xx_i2s_set_sysclk, + .set_fmt = ep93xx_i2s_set_dai_fmt, +}; + +#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver ep93xx_i2s_dai = { + .symmetric_rates= 1, + .suspend = ep93xx_i2s_suspend, + .resume = ep93xx_i2s_resume, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = EP93XX_I2S_FORMATS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = EP93XX_I2S_FORMATS, + }, + .ops = &ep93xx_i2s_dai_ops, +}; + +static int ep93xx_i2s_probe(struct platform_device *pdev) +{ + struct ep93xx_i2s_info *info; + struct resource *res; + int err; + + info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENODEV; + + info->regs = devm_request_and_ioremap(&pdev->dev, res); + if (!info->regs) + return -ENXIO; + + info->mclk = clk_get(&pdev->dev, "mclk"); + if (IS_ERR(info->mclk)) { + err = PTR_ERR(info->mclk); + goto fail; + } + + info->sclk = clk_get(&pdev->dev, "sclk"); + if (IS_ERR(info->sclk)) { + err = PTR_ERR(info->sclk); + goto fail_put_mclk; + } + + info->lrclk = clk_get(&pdev->dev, "lrclk"); + if (IS_ERR(info->lrclk)) { + err = PTR_ERR(info->lrclk); + goto fail_put_sclk; + } + + dev_set_drvdata(&pdev->dev, info); + info->dma_params = ep93xx_i2s_dma_params; + + err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); + if (err) + goto fail_put_lrclk; + + return 0; + +fail_put_lrclk: + dev_set_drvdata(&pdev->dev, NULL); + clk_put(info->lrclk); +fail_put_sclk: + clk_put(info->sclk); +fail_put_mclk: + clk_put(info->mclk); +fail: + return err; +} + +static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) +{ + struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + clk_put(info->lrclk); + clk_put(info->sclk); + clk_put(info->mclk); + return 0; +} + +static struct platform_driver ep93xx_i2s_driver = { + .probe = ep93xx_i2s_probe, + .remove = __devexit_p(ep93xx_i2s_remove), + .driver = { + .name = "ep93xx-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(ep93xx_i2s_driver); + +MODULE_ALIAS("platform:ep93xx-i2s"); +MODULE_AUTHOR("Ryan Mallon"); +MODULE_DESCRIPTION("EP93XX I2S driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c new file mode 100644 index 000000000000..4eea98b42bc8 --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -0,0 +1,242 @@ +/* + * linux/sound/arm/ep93xx-pcm.c - EP93xx ALSA PCM interface + * + * Copyright (C) 2006 Lennert Buytenhek + * Copyright (C) 2006 Applied Data Systems + * + * Rewritten for the SoC audio subsystem (Based on PXA2xx code): + * Copyright (c) 2008 Ryan Mallon + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "ep93xx-pcm.h" + +static const struct snd_pcm_hardware ep93xx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = SNDRV_PCM_RATE_8000, + .rate_max = SNDRV_PCM_RATE_192000, + + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + + .buffer_bytes_max = 131072, + .period_bytes_min = 32, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 32, +}; + +static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) +{ + struct ep93xx_dma_data *data = filter_param; + + if (data->direction == ep93xx_dma_chan_direction(chan)) { + chan->private = data; + return true; + } + + return false; +} + +static int ep93xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct ep93xx_pcm_dma_params *dma_params; + struct ep93xx_dma_data *dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); + + dma_data = kmalloc(sizeof(*dma_data), GFP_KERNEL); + if (!dma_data) + return -ENOMEM; + + dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream); + dma_data->port = dma_params->dma_port; + dma_data->name = dma_params->name; + dma_data->direction = snd_pcm_substream_to_dma_direction(substream); + + ret = snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, dma_data); + if (ret) { + kfree(dma_data); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int ep93xx_pcm_close(struct snd_pcm_substream *substream) +{ + struct dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + + snd_dmaengine_pcm_close(substream); + kfree(dma_data); + return 0; +} + +static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops ep93xx_pcm_ops = { + .open = ep93xx_pcm_open, + .close = ep93xx_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ep93xx_pcm_hw_params, + .hw_free = ep93xx_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer_no_residue, + .mmap = ep93xx_pcm_mmap, +}; + +static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ep93xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + buf->bytes = size; + + return (buf->area == NULL) ? -ENOMEM : 0; +} + +static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, + buf->addr); + buf->area = NULL; + } +} + +static u64 ep93xx_pcm_dmamask = DMA_BIT_MASK(32); + +static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &ep93xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = ep93xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = ep93xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +static struct snd_soc_platform_driver ep93xx_soc_platform = { + .ops = &ep93xx_pcm_ops, + .pcm_new = &ep93xx_pcm_new, + .pcm_free = &ep93xx_pcm_free_dma_buffers, +}; + +static int __devinit ep93xx_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &ep93xx_soc_platform); +} + +static int __devexit ep93xx_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver ep93xx_pcm_driver = { + .driver = { + .name = "ep93xx-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = ep93xx_soc_platform_probe, + .remove = __devexit_p(ep93xx_soc_platform_remove), +}; + +module_platform_driver(ep93xx_pcm_driver); + +MODULE_AUTHOR("Ryan Mallon"); +MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/cirrus/ep93xx-pcm.h b/sound/soc/cirrus/ep93xx-pcm.h new file mode 100644 index 000000000000..111e1121ecb8 --- /dev/null +++ b/sound/soc/cirrus/ep93xx-pcm.h @@ -0,0 +1,20 @@ +/* + * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface + * + * Copyright (C) 2006 Lennert Buytenhek + * Copyright (C) 2006 Applied Data Systems + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _EP93XX_SND_SOC_PCM_H +#define _EP93XX_SND_SOC_PCM_H + +struct ep93xx_pcm_dma_params { + char *name; + int dma_port; +}; + +#endif /* _EP93XX_SND_SOC_PCM_H */ diff --git a/sound/soc/cirrus/simone.c b/sound/soc/cirrus/simone.c new file mode 100644 index 000000000000..dd997094eb30 --- /dev/null +++ b/sound/soc/cirrus/simone.c @@ -0,0 +1,90 @@ +/* + * simone.c -- ASoC audio for Simplemachines Sim.One board + * + * Copyright (c) 2010 Mika Westerberg + * + * Based on snappercl15 machine driver by Ryan Mallon. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include +#include +#include + +#include +#include + +#include "ep93xx-pcm.h" + +static struct snd_soc_dai_link simone_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai_name = "ep93xx-ac97", + .codec_dai_name = "ac97-hifi", + .codec_name = "ac97-codec", + .platform_name = "ep93xx-pcm-audio", +}; + +static struct snd_soc_card snd_soc_simone = { + .name = "Sim.One", + .owner = THIS_MODULE, + .dai_link = &simone_dai, + .num_links = 1, +}; + +static struct platform_device *simone_snd_ac97_device; + +static int __devinit simone_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_simone; + int ret; + + simone_snd_ac97_device = platform_device_register_simple("ac97-codec", + -1, NULL, 0); + if (IS_ERR(simone_snd_ac97_device)) + return PTR_ERR(simone_snd_ac97_device); + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + platform_device_unregister(simone_snd_ac97_device); + } + + return ret; +} + +static int __devexit simone_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + platform_device_unregister(simone_snd_ac97_device); + + return 0; +} + +static struct platform_driver simone_driver = { + .driver = { + .name = "simone-audio", + .owner = THIS_MODULE, + }, + .probe = simone_probe, + .remove = __devexit_p(simone_remove), +}; + +module_platform_driver(simone_driver); + +MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); +MODULE_AUTHOR("Mika Westerberg "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:simone-audio"); diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c new file mode 100644 index 000000000000..a193cea3cf3c --- /dev/null +++ b/sound/soc/cirrus/snappercl15.c @@ -0,0 +1,146 @@ +/* + * snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module + * + * Copyright (C) 2008 Bluewater Systems Ltd + * Author: Ryan Mallon + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include + +#include +#include + +#include "../codecs/tlv320aic23.h" +#include "ep93xx-pcm.h" + +#define CODEC_CLOCK 5644800 + +static int snappercl15_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, + SND_SOC_CLOCK_IN); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (err) + return err; + + return 0; +} + +static struct snd_soc_ops snappercl15_ops = { + .hw_params = snappercl15_hw_params, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + return 0; +} + +static struct snd_soc_dai_link snappercl15_dai = { + .name = "tlv320aic23", + .stream_name = "AIC23", + .cpu_dai_name = "ep93xx-i2s", + .codec_dai_name = "tlv320aic23-hifi", + .codec_name = "tlv320aic23-codec.0-001a", + .platform_name = "ep93xx-pcm-audio", + .init = snappercl15_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &snappercl15_ops, +}; + +static struct snd_soc_card snd_soc_snappercl15 = { + .name = "Snapper CL15", + .owner = THIS_MODULE, + .dai_link = &snappercl15_dai, + .num_links = 1, +}; + +static int __devinit snappercl15_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_snappercl15; + int ret; + + ret = ep93xx_i2s_acquire(); + if (ret) + return ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + ep93xx_i2s_release(); + } + + return ret; +} + +static int __devexit snappercl15_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + ep93xx_i2s_release(); + + return 0; +} + +static struct platform_driver snappercl15_driver = { + .driver = { + .name = "snappercl15-audio", + .owner = THIS_MODULE, + }, + .probe = snappercl15_probe, + .remove = __devexit_p(snappercl15_remove), +}; + +module_platform_driver(snappercl15_driver); + +MODULE_AUTHOR("Ryan Mallon"); +MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:snappercl15-audio"); diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig deleted file mode 100644 index 88143db7e753..000000000000 --- a/sound/soc/ep93xx/Kconfig +++ /dev/null @@ -1,42 +0,0 @@ -config SND_EP93XX_SOC - tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on ARCH_EP93XX && SND_SOC - select SND_SOC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to - the EP93xx I2S or AC97 interfaces. - -config SND_EP93XX_SOC_I2S - tristate - -config SND_EP93XX_SOC_AC97 - tristate - select AC97_BUS - select SND_SOC_AC97_BUS - -config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23 - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. - -config SND_EP93XX_SOC_SIMONE - tristate "SoC Audio support for Simplemachines Sim.One board" - depends on SND_EP93XX_SOC && MACH_SIM_ONE - select SND_EP93XX_SOC_AC97 - select SND_SOC_AC97_CODEC - help - Say Y or M here if you want to add support for AC97 audio on the - Simplemachines Sim.One board. - -config SND_EP93XX_SOC_EDB93XX - tristate "SoC Audio support for Cirrus Logic EDB93xx boards" - depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) - select SND_EP93XX_SOC_I2S - select SND_SOC_CS4271 - help - Say Y or M here if you want to add support for I2S audio on the - Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile deleted file mode 100644 index 5514146cbdf0..000000000000 --- a/sound/soc/ep93xx/Makefile +++ /dev/null @@ -1,17 +0,0 @@ -# EP93xx Platform Support -snd-soc-ep93xx-objs := ep93xx-pcm.o -snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o -snd-soc-ep93xx-ac97-objs := ep93xx-ac97.o - -obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o -obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o -obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o - -# EP93XX Machine Support -snd-soc-snappercl15-objs := snappercl15.o -snd-soc-simone-objs := simone.o -snd-soc-edb93xx-objs := edb93xx.o - -obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o -obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o -obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c deleted file mode 100644 index e01cb02abd3a..000000000000 --- a/sound/soc/ep93xx/edb93xx.c +++ /dev/null @@ -1,128 +0,0 @@ -/* - * SoC audio for EDB93xx - * - * Copyright (c) 2010 Alexander Sverdlin - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * as published by the Free Software Foundation; either version 2 - * of the License, or (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * This driver support CS4271 codec being master or slave, working - * in control port mode, connected either via SPI or I2C. - * The data format accepted is I2S or left-justified. - * DAPM support not implemented. - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "ep93xx-pcm.h" - -static int edb93xx_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int err; - unsigned int mclk_rate; - unsigned int rate = params_rate(params); - - /* - * According to CS4271 datasheet we use MCLK/LRCK=256 for - * rates below 50kHz and 128 for higher sample rates - */ - if (rate < 50000) - mclk_rate = rate * 64 * 4; - else - mclk_rate = rate * 64 * 2; - - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, - SND_SOC_CLOCK_IN); - if (err) - return err; - - return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, - SND_SOC_CLOCK_OUT); -} - -static struct snd_soc_ops edb93xx_ops = { - .hw_params = edb93xx_hw_params, -}; - -static struct snd_soc_dai_link edb93xx_dai = { - .name = "CS4271", - .stream_name = "CS4271 HiFi", - .platform_name = "ep93xx-pcm-audio", - .cpu_dai_name = "ep93xx-i2s", - .codec_name = "spi0.0", - .codec_dai_name = "cs4271-hifi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &edb93xx_ops, -}; - -static struct snd_soc_card snd_soc_edb93xx = { - .name = "EDB93XX", - .owner = THIS_MODULE, - .dai_link = &edb93xx_dai, - .num_links = 1, -}; - -static int __devinit edb93xx_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_edb93xx; - int ret; - - ret = ep93xx_i2s_acquire(); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - ep93xx_i2s_release(); - } - - return ret; -} - -static int __devexit edb93xx_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - ep93xx_i2s_release(); - - return 0; -} - -static struct platform_driver edb93xx_driver = { - .driver = { - .name = "edb93xx-audio", - .owner = THIS_MODULE, - }, - .probe = edb93xx_probe, - .remove = __devexit_p(edb93xx_remove), -}; - -module_platform_driver(edb93xx_driver); - -MODULE_AUTHOR("Alexander Sverdlin "); -MODULE_DESCRIPTION("ALSA SoC EDB93xx"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:edb93xx-audio"); diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c deleted file mode 100644 index bdffab33e160..000000000000 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ /dev/null @@ -1,435 +0,0 @@ -/* - * ASoC driver for Cirrus Logic EP93xx AC97 controller. - * - * Copyright (c) 2010 Mika Westerberg - * - * Based on s3c-ac97 ASoC driver by Jaswinder Singh. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include - -#include -#include "ep93xx-pcm.h" - -/* - * Per channel (1-4) registers. - */ -#define AC97CH(n) (((n) - 1) * 0x20) - -#define AC97DR(n) (AC97CH(n) + 0x0000) - -#define AC97RXCR(n) (AC97CH(n) + 0x0004) -#define AC97RXCR_REN BIT(0) -#define AC97RXCR_RX3 BIT(3) -#define AC97RXCR_RX4 BIT(4) -#define AC97RXCR_CM BIT(15) - -#define AC97TXCR(n) (AC97CH(n) + 0x0008) -#define AC97TXCR_TEN BIT(0) -#define AC97TXCR_TX3 BIT(3) -#define AC97TXCR_TX4 BIT(4) -#define AC97TXCR_CM BIT(15) - -#define AC97SR(n) (AC97CH(n) + 0x000c) -#define AC97SR_TXFE BIT(1) -#define AC97SR_TXUE BIT(6) - -#define AC97RISR(n) (AC97CH(n) + 0x0010) -#define AC97ISR(n) (AC97CH(n) + 0x0014) -#define AC97IE(n) (AC97CH(n) + 0x0018) - -/* - * Global AC97 controller registers. - */ -#define AC97S1DATA 0x0080 -#define AC97S2DATA 0x0084 -#define AC97S12DATA 0x0088 - -#define AC97RGIS 0x008c -#define AC97GIS 0x0090 -#define AC97IM 0x0094 -/* - * Common bits for RGIS, GIS and IM registers. - */ -#define AC97_SLOT2RXVALID BIT(1) -#define AC97_CODECREADY BIT(5) -#define AC97_SLOT2TXCOMPLETE BIT(6) - -#define AC97EOI 0x0098 -#define AC97EOI_WINT BIT(0) -#define AC97EOI_CODECREADY BIT(1) - -#define AC97GCR 0x009c -#define AC97GCR_AC97IFE BIT(0) - -#define AC97RESET 0x00a0 -#define AC97RESET_TIMEDRESET BIT(0) - -#define AC97SYNC 0x00a4 -#define AC97SYNC_TIMEDSYNC BIT(0) - -#define AC97_TIMEOUT msecs_to_jiffies(5) - -/** - * struct ep93xx_ac97_info - EP93xx AC97 controller info structure - * @lock: mutex serializing access to the bus (slot 1 & 2 ops) - * @dev: pointer to the platform device dev structure - * @regs: mapped AC97 controller registers - * @done: bus ops wait here for an interrupt - */ -struct ep93xx_ac97_info { - struct mutex lock; - struct device *dev; - void __iomem *regs; - struct completion done; -}; - -/* currently ALSA only supports a single AC97 device */ -static struct ep93xx_ac97_info *ep93xx_ac97_info; - -static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_out = { - .name = "ac97-pcm-out", - .dma_port = EP93XX_DMA_AAC1, -}; - -static struct ep93xx_pcm_dma_params ep93xx_ac97_pcm_in = { - .name = "ac97-pcm-in", - .dma_port = EP93XX_DMA_AAC1, -}; - -static inline unsigned ep93xx_ac97_read_reg(struct ep93xx_ac97_info *info, - unsigned reg) -{ - return __raw_readl(info->regs + reg); -} - -static inline void ep93xx_ac97_write_reg(struct ep93xx_ac97_info *info, - unsigned reg, unsigned val) -{ - __raw_writel(val, info->regs + reg); -} - -static unsigned short ep93xx_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - unsigned short val; - - mutex_lock(&info->lock); - - ep93xx_ac97_write_reg(info, AC97S1DATA, reg); - ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2RXVALID); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) { - dev_warn(info->dev, "timeout reading register %x\n", reg); - mutex_unlock(&info->lock); - return -ETIMEDOUT; - } - val = (unsigned short)ep93xx_ac97_read_reg(info, AC97S2DATA); - - mutex_unlock(&info->lock); - return val; -} - -static void ep93xx_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, - unsigned short val) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * Writes to the codec need to be done so that slot 2 is filled in - * before slot 1. - */ - ep93xx_ac97_write_reg(info, AC97S2DATA, val); - ep93xx_ac97_write_reg(info, AC97S1DATA, reg); - - ep93xx_ac97_write_reg(info, AC97IM, AC97_SLOT2TXCOMPLETE); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "timeout writing register %x\n", reg); - - mutex_unlock(&info->lock); -} - -static void ep93xx_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * We are assuming that before this functions gets called, the codec - * BIT_CLK is stopped by forcing the codec into powerdown mode. We can - * control the SYNC signal directly via AC97SYNC register. Using - * TIMEDSYNC the controller will keep the SYNC high > 1us. - */ - ep93xx_ac97_write_reg(info, AC97SYNC, AC97SYNC_TIMEDSYNC); - ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "codec warm reset timeout\n"); - - mutex_unlock(&info->lock); -} - -static void ep93xx_ac97_cold_reset(struct snd_ac97 *ac97) -{ - struct ep93xx_ac97_info *info = ep93xx_ac97_info; - - mutex_lock(&info->lock); - - /* - * For doing cold reset, we disable the AC97 controller interface, clear - * WINT and CODECREADY bits, and finally enable the interface again. - */ - ep93xx_ac97_write_reg(info, AC97GCR, 0); - ep93xx_ac97_write_reg(info, AC97EOI, AC97EOI_CODECREADY | AC97EOI_WINT); - ep93xx_ac97_write_reg(info, AC97GCR, AC97GCR_AC97IFE); - - /* - * Now, assert the reset and wait for the codec to become ready. - */ - ep93xx_ac97_write_reg(info, AC97RESET, AC97RESET_TIMEDRESET); - ep93xx_ac97_write_reg(info, AC97IM, AC97_CODECREADY); - if (!wait_for_completion_timeout(&info->done, AC97_TIMEOUT)) - dev_warn(info->dev, "codec cold reset timeout\n"); - - /* - * Give the codec some time to come fully out from the reset. This way - * we ensure that the subsequent reads/writes will work. - */ - usleep_range(15000, 20000); - - mutex_unlock(&info->lock); -} - -static irqreturn_t ep93xx_ac97_interrupt(int irq, void *dev_id) -{ - struct ep93xx_ac97_info *info = dev_id; - unsigned status, mask; - - /* - * Just mask out the interrupt and wake up the waiting thread. - * Interrupts are cleared via reading/writing to slot 1 & 2 registers by - * the waiting thread. - */ - status = ep93xx_ac97_read_reg(info, AC97GIS); - mask = ep93xx_ac97_read_reg(info, AC97IM); - mask &= ~status; - ep93xx_ac97_write_reg(info, AC97IM, mask); - - complete(&info->done); - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = ep93xx_ac97_read, - .write = ep93xx_ac97_write, - .reset = ep93xx_ac97_cold_reset, - .warm_reset = ep93xx_ac97_warm_reset, -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); - unsigned v = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* - * Enable compact mode, TX slots 3 & 4, and the TX FIFO - * itself. - */ - v |= AC97TXCR_CM; - v |= AC97TXCR_TX3 | AC97TXCR_TX4; - v |= AC97TXCR_TEN; - ep93xx_ac97_write_reg(info, AC97TXCR(1), v); - } else { - /* - * Enable compact mode, RX slots 3 & 4, and the RX FIFO - * itself. - */ - v |= AC97RXCR_CM; - v |= AC97RXCR_RX3 | AC97RXCR_RX4; - v |= AC97RXCR_REN; - ep93xx_ac97_write_reg(info, AC97RXCR(1), v); - } - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* - * As per Cirrus EP93xx errata described below: - * - * http://www.cirrus.com/en/pubs/errata/ER667E2B.pdf - * - * we will wait for the TX FIFO to be empty before - * clearing the TEN bit. - */ - unsigned long timeout = jiffies + AC97_TIMEOUT; - - do { - v = ep93xx_ac97_read_reg(info, AC97SR(1)); - if (time_after(jiffies, timeout)) { - dev_warn(info->dev, "TX timeout\n"); - break; - } - } while (!(v & (AC97SR_TXFE | AC97SR_TXUE))); - - /* disable the TX FIFO */ - ep93xx_ac97_write_reg(info, AC97TXCR(1), 0); - } else { - /* disable the RX FIFO */ - ep93xx_ac97_write_reg(info, AC97RXCR(1), 0); - } - break; - - default: - dev_warn(info->dev, "unknown command %d\n", cmd); - return -EINVAL; - } - - return 0; -} - -static int ep93xx_ac97_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct ep93xx_pcm_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &ep93xx_ac97_pcm_out; - else - dma_data = &ep93xx_ac97_pcm_in; - - snd_soc_dai_set_dma_data(dai, substream, dma_data); - return 0; -} - -static const struct snd_soc_dai_ops ep93xx_ac97_dai_ops = { - .startup = ep93xx_ac97_startup, - .trigger = ep93xx_ac97_trigger, -}; - -static struct snd_soc_dai_driver ep93xx_ac97_dai = { - .name = "ep93xx-ac97", - .id = 0, - .ac97_control = 1, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &ep93xx_ac97_dai_ops, -}; - -static int __devinit ep93xx_ac97_probe(struct platform_device *pdev) -{ - struct ep93xx_ac97_info *info; - struct resource *res; - unsigned int irq; - int ret; - - info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); - if (!info) - return -ENOMEM; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; - - irq = platform_get_irq(pdev, 0); - if (!irq) - return -ENODEV; - - ret = devm_request_irq(&pdev->dev, irq, ep93xx_ac97_interrupt, - IRQF_TRIGGER_HIGH, pdev->name, info); - if (ret) - goto fail; - - dev_set_drvdata(&pdev->dev, info); - - mutex_init(&info->lock); - init_completion(&info->done); - info->dev = &pdev->dev; - - ep93xx_ac97_info = info; - platform_set_drvdata(pdev, info); - - ret = snd_soc_register_dai(&pdev->dev, &ep93xx_ac97_dai); - if (ret) - goto fail; - - return 0; - -fail: - platform_set_drvdata(pdev, NULL); - ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); - return ret; -} - -static int __devexit ep93xx_ac97_remove(struct platform_device *pdev) -{ - struct ep93xx_ac97_info *info = platform_get_drvdata(pdev); - - snd_soc_unregister_dai(&pdev->dev); - - /* disable the AC97 controller */ - ep93xx_ac97_write_reg(info, AC97GCR, 0); - - platform_set_drvdata(pdev, NULL); - ep93xx_ac97_info = NULL; - dev_set_drvdata(&pdev->dev, NULL); - - return 0; -} - -static struct platform_driver ep93xx_ac97_driver = { - .probe = ep93xx_ac97_probe, - .remove = __devexit_p(ep93xx_ac97_remove), - .driver = { - .name = "ep93xx-ac97", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(ep93xx_ac97_driver); - -MODULE_DESCRIPTION("EP93xx AC97 ASoC Driver"); -MODULE_AUTHOR("Mika Westerberg "); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-ac97"); diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c deleted file mode 100644 index 8df8f6dc474f..000000000000 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ /dev/null @@ -1,451 +0,0 @@ -/* - * linux/sound/soc/ep93xx-i2s.c - * EP93xx I2S driver - * - * Copyright (C) 2010 Ryan Mallon - * - * Based on the original driver by: - * Copyright (C) 2007 Chase Douglas - * Copyright (C) 2006 Lennert Buytenhek - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ - -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include - -#include "ep93xx-pcm.h" - -#define EP93XX_I2S_TXCLKCFG 0x00 -#define EP93XX_I2S_RXCLKCFG 0x04 -#define EP93XX_I2S_GLCTRL 0x0C - -#define EP93XX_I2S_TXLINCTRLDATA 0x28 -#define EP93XX_I2S_TXCTRL 0x2C -#define EP93XX_I2S_TXWRDLEN 0x30 -#define EP93XX_I2S_TX0EN 0x34 - -#define EP93XX_I2S_RXLINCTRLDATA 0x58 -#define EP93XX_I2S_RXCTRL 0x5C -#define EP93XX_I2S_RXWRDLEN 0x60 -#define EP93XX_I2S_RX0EN 0x64 - -#define EP93XX_I2S_WRDLEN_16 (0 << 0) -#define EP93XX_I2S_WRDLEN_24 (1 << 0) -#define EP93XX_I2S_WRDLEN_32 (2 << 0) - -#define EP93XX_I2S_LINCTRLDATA_R_JUST (1 << 2) /* Right justify */ - -#define EP93XX_I2S_CLKCFG_LRS (1 << 0) /* lrclk polarity */ -#define EP93XX_I2S_CLKCFG_CKP (1 << 1) /* Bit clock polarity */ -#define EP93XX_I2S_CLKCFG_REL (1 << 2) /* First bit transition */ -#define EP93XX_I2S_CLKCFG_MASTER (1 << 3) /* Master mode */ -#define EP93XX_I2S_CLKCFG_NBCG (1 << 4) /* Not bit clock gating */ - -struct ep93xx_i2s_info { - struct clk *mclk; - struct clk *sclk; - struct clk *lrclk; - struct ep93xx_pcm_dma_params *dma_params; - void __iomem *regs; -}; - -struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = { - [SNDRV_PCM_STREAM_PLAYBACK] = { - .name = "i2s-pcm-out", - .dma_port = EP93XX_DMA_I2S1, - }, - [SNDRV_PCM_STREAM_CAPTURE] = { - .name = "i2s-pcm-in", - .dma_port = EP93XX_DMA_I2S1, - }, -}; - -static inline void ep93xx_i2s_write_reg(struct ep93xx_i2s_info *info, - unsigned reg, unsigned val) -{ - __raw_writel(val, info->regs + reg); -} - -static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info, - unsigned reg) -{ - return __raw_readl(info->regs + reg); -} - -static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) -{ - unsigned base_reg; - int i; - - if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && - (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { - /* Enable clocks */ - clk_enable(info->mclk); - clk_enable(info->sclk); - clk_enable(info->lrclk); - - /* Enable i2s */ - ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1); - } - - /* Enable fifos */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - base_reg = EP93XX_I2S_TX0EN; - else - base_reg = EP93XX_I2S_RX0EN; - for (i = 0; i < 3; i++) - ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1); -} - -static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) -{ - unsigned base_reg; - int i; - - /* Disable fifos */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - base_reg = EP93XX_I2S_TX0EN; - else - base_reg = EP93XX_I2S_RX0EN; - for (i = 0; i < 3; i++) - ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0); - - if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 && - (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) { - /* Disable i2s */ - ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0); - - /* Disable clocks */ - clk_disable(info->lrclk); - clk_disable(info->sclk); - clk_disable(info->mclk); - } -} - -static int ep93xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - snd_soc_dai_set_dma_data(cpu_dai, substream, - &info->dma_params[substream->stream]); - return 0; -} - -static void ep93xx_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - - ep93xx_i2s_disable(info, substream->stream); -} - -static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(cpu_dai); - unsigned int clk_cfg, lin_ctrl; - - clk_cfg = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXCLKCFG); - lin_ctrl = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXLINCTRLDATA); - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - clk_cfg |= EP93XX_I2S_CLKCFG_REL; - lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; - break; - - case SND_SOC_DAIFMT_LEFT_J: - clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; - lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST; - break; - - case SND_SOC_DAIFMT_RIGHT_J: - clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; - lin_ctrl |= EP93XX_I2S_LINCTRLDATA_R_JUST; - break; - - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* CPU is master */ - clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; - break; - - case SND_SOC_DAIFMT_CBM_CFM: - /* Codec is master */ - clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; - break; - - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Negative bit clock, lrclk low on left word */ - clk_cfg &= ~(EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL); - break; - - case SND_SOC_DAIFMT_NB_IF: - /* Negative bit clock, lrclk low on right word */ - clk_cfg &= ~EP93XX_I2S_CLKCFG_CKP; - clk_cfg |= EP93XX_I2S_CLKCFG_REL; - break; - - case SND_SOC_DAIFMT_IB_NF: - /* Positive bit clock, lrclk low on left word */ - clk_cfg |= EP93XX_I2S_CLKCFG_CKP; - clk_cfg &= ~EP93XX_I2S_CLKCFG_REL; - break; - - case SND_SOC_DAIFMT_IB_IF: - /* Positive bit clock, lrclk low on right word */ - clk_cfg |= EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL; - break; - } - - /* Write new register values */ - ep93xx_i2s_write_reg(info, EP93XX_I2S_RXCLKCFG, clk_cfg); - ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCLKCFG, clk_cfg); - ep93xx_i2s_write_reg(info, EP93XX_I2S_RXLINCTRLDATA, lin_ctrl); - ep93xx_i2s_write_reg(info, EP93XX_I2S_TXLINCTRLDATA, lin_ctrl); - return 0; -} - -static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - unsigned word_len, div, sdiv, lrdiv; - int err; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_len = EP93XX_I2S_WRDLEN_16; - break; - - case SNDRV_PCM_FORMAT_S24_LE: - word_len = EP93XX_I2S_WRDLEN_24; - break; - - case SNDRV_PCM_FORMAT_S32_LE: - word_len = EP93XX_I2S_WRDLEN_32; - break; - - default: - return -EINVAL; - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len); - else - ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len); - - /* - * EP93xx I2S module can be setup so SCLK / LRCLK value can be - * 32, 64, 128. MCLK / SCLK value can be 2 and 4. - * We set LRCLK equal to `rate' and minimum SCLK / LRCLK - * value is 64, because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - */ - div = clk_get_rate(info->mclk) / params_rate(params); - sdiv = 4; - if (div > (256 + 512) / 2) { - lrdiv = 128; - } else { - lrdiv = 64; - if (div < (128 + 256) / 2) - sdiv = 2; - } - - err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); - if (err) - return err; - - err = clk_set_rate(info->lrclk, clk_get_rate(info->sclk) / lrdiv); - if (err) - return err; - - ep93xx_i2s_enable(info, substream->stream); - return 0; -} - -static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, - unsigned int freq, int dir) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(cpu_dai); - - if (dir == SND_SOC_CLOCK_IN || clk_id != 0) - return -EINVAL; - - return clk_set_rate(info->mclk, freq); -} - -#ifdef CONFIG_PM -static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - - if (!dai->active) - return 0; - - ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); - ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); - - return 0; -} - -static int ep93xx_i2s_resume(struct snd_soc_dai *dai) -{ - struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); - - if (!dai->active) - return 0; - - ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); - ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); - - return 0; -} -#else -#define ep93xx_i2s_suspend NULL -#define ep93xx_i2s_resume NULL -#endif - -static const struct snd_soc_dai_ops ep93xx_i2s_dai_ops = { - .startup = ep93xx_i2s_startup, - .shutdown = ep93xx_i2s_shutdown, - .hw_params = ep93xx_i2s_hw_params, - .set_sysclk = ep93xx_i2s_set_sysclk, - .set_fmt = ep93xx_i2s_set_dai_fmt, -}; - -#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) - -static struct snd_soc_dai_driver ep93xx_i2s_dai = { - .symmetric_rates= 1, - .suspend = ep93xx_i2s_suspend, - .resume = ep93xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = EP93XX_I2S_FORMATS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = EP93XX_I2S_FORMATS, - }, - .ops = &ep93xx_i2s_dai_ops, -}; - -static int ep93xx_i2s_probe(struct platform_device *pdev) -{ - struct ep93xx_i2s_info *info; - struct resource *res; - int err; - - info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); - if (!info) - return -ENOMEM; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) - return -ENODEV; - - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; - - info->mclk = clk_get(&pdev->dev, "mclk"); - if (IS_ERR(info->mclk)) { - err = PTR_ERR(info->mclk); - goto fail; - } - - info->sclk = clk_get(&pdev->dev, "sclk"); - if (IS_ERR(info->sclk)) { - err = PTR_ERR(info->sclk); - goto fail_put_mclk; - } - - info->lrclk = clk_get(&pdev->dev, "lrclk"); - if (IS_ERR(info->lrclk)) { - err = PTR_ERR(info->lrclk); - goto fail_put_sclk; - } - - dev_set_drvdata(&pdev->dev, info); - info->dma_params = ep93xx_i2s_dma_params; - - err = snd_soc_register_dai(&pdev->dev, &ep93xx_i2s_dai); - if (err) - goto fail_put_lrclk; - - return 0; - -fail_put_lrclk: - dev_set_drvdata(&pdev->dev, NULL); - clk_put(info->lrclk); -fail_put_sclk: - clk_put(info->sclk); -fail_put_mclk: - clk_put(info->mclk); -fail: - return err; -} - -static int __devexit ep93xx_i2s_remove(struct platform_device *pdev) -{ - struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); - - snd_soc_unregister_dai(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - clk_put(info->lrclk); - clk_put(info->sclk); - clk_put(info->mclk); - return 0; -} - -static struct platform_driver ep93xx_i2s_driver = { - .probe = ep93xx_i2s_probe, - .remove = __devexit_p(ep93xx_i2s_remove), - .driver = { - .name = "ep93xx-i2s", - .owner = THIS_MODULE, - }, -}; - -module_platform_driver(ep93xx_i2s_driver); - -MODULE_ALIAS("platform:ep93xx-i2s"); -MODULE_AUTHOR("Ryan Mallon"); -MODULE_DESCRIPTION("EP93XX I2S driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c deleted file mode 100644 index 4eea98b42bc8..000000000000 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ /dev/null @@ -1,242 +0,0 @@ -/* - * linux/sound/arm/ep93xx-pcm.c - EP93xx ALSA PCM interface - * - * Copyright (C) 2006 Lennert Buytenhek - * Copyright (C) 2006 Applied Data Systems - * - * Rewritten for the SoC audio subsystem (Based on PXA2xx code): - * Copyright (c) 2008 Ryan Mallon - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include - -#include -#include -#include - -#include "ep93xx-pcm.h" - -static const struct snd_pcm_hardware ep93xx_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER), - - .rates = SNDRV_PCM_RATE_8000_192000, - .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_192000, - - .formats = (SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE), - - .buffer_bytes_max = 131072, - .period_bytes_min = 32, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 32, - .fifo_size = 32, -}; - -static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) -{ - struct ep93xx_dma_data *data = filter_param; - - if (data->direction == ep93xx_dma_chan_direction(chan)) { - chan->private = data; - return true; - } - - return false; -} - -static int ep93xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct ep93xx_pcm_dma_params *dma_params; - struct ep93xx_dma_data *dma_data; - int ret; - - snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); - - dma_data = kmalloc(sizeof(*dma_data), GFP_KERNEL); - if (!dma_data) - return -ENOMEM; - - dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream); - dma_data->port = dma_params->dma_port; - dma_data->name = dma_params->name; - dma_data->direction = snd_pcm_substream_to_dma_direction(substream); - - ret = snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, dma_data); - if (ret) { - kfree(dma_data); - return ret; - } - - snd_dmaengine_pcm_set_data(substream, dma_data); - - return 0; -} - -static int ep93xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - - snd_dmaengine_pcm_close(substream); - kfree(dma_data); - return 0; -} - -static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - return 0; -} - -static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops ep93xx_pcm_ops = { - .open = ep93xx_pcm_open, - .close = ep93xx_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = ep93xx_pcm_hw_params, - .hw_free = ep93xx_pcm_hw_free, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = ep93xx_pcm_mmap, -}; - -static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ep93xx_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - buf->bytes = size; - - return (buf->area == NULL) ? -ENOMEM : 0; -} - -static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, - buf->addr); - buf->area = NULL; - } -} - -static u64 ep93xx_pcm_dmamask = DMA_BIT_MASK(32); - -static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &ep93xx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = ep93xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver ep93xx_soc_platform = { - .ops = &ep93xx_pcm_ops, - .pcm_new = &ep93xx_pcm_new, - .pcm_free = &ep93xx_pcm_free_dma_buffers, -}; - -static int __devinit ep93xx_soc_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &ep93xx_soc_platform); -} - -static int __devexit ep93xx_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver ep93xx_pcm_driver = { - .driver = { - .name = "ep93xx-pcm-audio", - .owner = THIS_MODULE, - }, - - .probe = ep93xx_soc_platform_probe, - .remove = __devexit_p(ep93xx_soc_platform_remove), -}; - -module_platform_driver(ep93xx_pcm_driver); - -MODULE_AUTHOR("Ryan Mallon"); -MODULE_DESCRIPTION("EP93xx ALSA PCM interface"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:ep93xx-pcm-audio"); diff --git a/sound/soc/ep93xx/ep93xx-pcm.h b/sound/soc/ep93xx/ep93xx-pcm.h deleted file mode 100644 index 111e1121ecb8..000000000000 --- a/sound/soc/ep93xx/ep93xx-pcm.h +++ /dev/null @@ -1,20 +0,0 @@ -/* - * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface - * - * Copyright (C) 2006 Lennert Buytenhek - * Copyright (C) 2006 Applied Data Systems - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _EP93XX_SND_SOC_PCM_H -#define _EP93XX_SND_SOC_PCM_H - -struct ep93xx_pcm_dma_params { - char *name; - int dma_port; -}; - -#endif /* _EP93XX_SND_SOC_PCM_H */ diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c deleted file mode 100644 index dd997094eb30..000000000000 --- a/sound/soc/ep93xx/simone.c +++ /dev/null @@ -1,90 +0,0 @@ -/* - * simone.c -- ASoC audio for Simplemachines Sim.One board - * - * Copyright (c) 2010 Mika Westerberg - * - * Based on snappercl15 machine driver by Ryan Mallon. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include - -#include -#include -#include - -#include -#include - -#include "ep93xx-pcm.h" - -static struct snd_soc_dai_link simone_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "ep93xx-ac97", - .codec_dai_name = "ac97-hifi", - .codec_name = "ac97-codec", - .platform_name = "ep93xx-pcm-audio", -}; - -static struct snd_soc_card snd_soc_simone = { - .name = "Sim.One", - .owner = THIS_MODULE, - .dai_link = &simone_dai, - .num_links = 1, -}; - -static struct platform_device *simone_snd_ac97_device; - -static int __devinit simone_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_simone; - int ret; - - simone_snd_ac97_device = platform_device_register_simple("ac97-codec", - -1, NULL, 0); - if (IS_ERR(simone_snd_ac97_device)) - return PTR_ERR(simone_snd_ac97_device); - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - platform_device_unregister(simone_snd_ac97_device); - } - - return ret; -} - -static int __devexit simone_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - platform_device_unregister(simone_snd_ac97_device); - - return 0; -} - -static struct platform_driver simone_driver = { - .driver = { - .name = "simone-audio", - .owner = THIS_MODULE, - }, - .probe = simone_probe, - .remove = __devexit_p(simone_remove), -}; - -module_platform_driver(simone_driver); - -MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One"); -MODULE_AUTHOR("Mika Westerberg "); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:simone-audio"); diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c deleted file mode 100644 index a193cea3cf3c..000000000000 --- a/sound/soc/ep93xx/snappercl15.c +++ /dev/null @@ -1,146 +0,0 @@ -/* - * snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module - * - * Copyright (C) 2008 Bluewater Systems Ltd - * Author: Ryan Mallon - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include -#include -#include -#include -#include - -#include -#include - -#include "../codecs/tlv320aic23.h" -#include "ep93xx-pcm.h" - -#define CODEC_CLOCK 5644800 - -static int snappercl15_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int err; - - err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, - SND_SOC_CLOCK_IN); - if (err) - return err; - - err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK, - SND_SOC_CLOCK_OUT); - if (err) - return err; - - return 0; -} - -static struct snd_soc_ops snappercl15_ops = { - .hw_params = snappercl15_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - -static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - -static struct snd_soc_dai_link snappercl15_dai = { - .name = "tlv320aic23", - .stream_name = "AIC23", - .cpu_dai_name = "ep93xx-i2s", - .codec_dai_name = "tlv320aic23-hifi", - .codec_name = "tlv320aic23-codec.0-001a", - .platform_name = "ep93xx-pcm-audio", - .init = snappercl15_tlv320aic23_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBS_CFS, - .ops = &snappercl15_ops, -}; - -static struct snd_soc_card snd_soc_snappercl15 = { - .name = "Snapper CL15", - .owner = THIS_MODULE, - .dai_link = &snappercl15_dai, - .num_links = 1, -}; - -static int __devinit snappercl15_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_snappercl15; - int ret; - - ret = ep93xx_i2s_acquire(); - if (ret) - return ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - ep93xx_i2s_release(); - } - - return ret; -} - -static int __devexit snappercl15_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - ep93xx_i2s_release(); - - return 0; -} - -static struct platform_driver snappercl15_driver = { - .driver = { - .name = "snappercl15-audio", - .owner = THIS_MODULE, - }, - .probe = snappercl15_probe, - .remove = __devexit_p(snappercl15_remove), -}; - -module_platform_driver(snappercl15_driver); - -MODULE_AUTHOR("Ryan Mallon"); -MODULE_DESCRIPTION("ALSA SoC Snapper CL15"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:snappercl15-audio"); -- cgit v1.2.3 From 1338fc97d07a04e74a7b75ff28b7751061f4cf38 Mon Sep 17 00:00:00 2001 From: David Flater Date: Mon, 27 Aug 2012 22:25:21 -0400 Subject: ALSA: emu8000: fix emu8000 DRAM sized 512 KiB too small v2: Fixed result still wrong in the case of 512 KiB DRAM. Oops. Applicable to 3.5.3 mainline. In emu8000.c, size_dram determines the amount of memory on the sound card by doing write/readback tests starting at 512 KiB and incrementing by 512 KiB. On success, detected_size is updated to the successful address and testing continues. On failure, the loop is immediately exited. The resulting detected_size is 512 KiB too small except in two special cases: 1. If there is no memory, the initial 0 value of detected_size is used, which is correct. 2. If the address space wraps around, detected_size is updated before the bailout, so the result is correct. The patch corrects all cases and was tested with an AWE64 Gold. Before: EMU8000 [0x620]: 3584 Kb on-board memory detected asfxload 4GMGSMT.SF2 (4174814 B) fails. After: EMU8000 [0x620]: 4096 Kb on-board memory detected asfxload 4GMGSMT.SF2 succeeds. I do not have a card with 512 KiB to test with, but by forcibly enabling the added conditional I verified on the AWE64 Gold that it detects 512 KiB (successfully reading from the first memory location) and does not hang the card. C.f. Bug 46451 https://bugzilla.kernel.org/show_bug.cgi?id=46451 Signed-off-by: David Flater Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 71887874679c..2aae6a0efbcd 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -417,9 +417,6 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) break; /* no memory at this address */ - - detected_size = size; - snd_emu8000_read_wait(emu); /* @@ -432,6 +429,18 @@ size_dram(struct snd_emu8000 *emu) if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) break; /* we must have wrapped around */ snd_emu8000_read_wait(emu); + + /* Otherwise, it's valid memory. */ + detected_size = size + 512 * 1024; + } + + /* Distinguish 512 KiB from 0. */ + if (detected_size == 0) { + snd_emu8000_read_wait(emu); + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); + EMU8000_SMLD_READ(emu); /* discard stale data */ + if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) + detected_size = 512 * 1024; } /* wait until FULL bit in SMAxW register is false */ -- cgit v1.2.3 From 48ee7cb8b4867093c75eb5102f2359c9799b3341 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 16:27:26 -0700 Subject: ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substream The two entries are duplicated in struct snd_usb_endpoint. Seems forgotten in the last clean-up. Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index 2b9fffff23b6..bcb5267a02e8 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -115,8 +115,6 @@ struct snd_usb_substream { unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ - unsigned long active_mask; /* bitmask of active urbs */ - unsigned long unlink_mask; /* bitmask of unlinked urbs */ /* data and sync endpoints for this stream */ unsigned int ep_num; /* the endpoint number */ -- cgit v1.2.3 From 68467f51c1b578ad98593bf5dee4337bd8d7798d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 09:14:29 -0700 Subject: ALSA: hda - Fix runtime PM leftover refcounts When the HD-audio is removed, it leaves the refcounts when codecs are powered up (usually yes) in the destructor. For fixing the unbalance, and cleaning up the code mess, this patch changes the following: - change pm_notify callback to take the explicit power on/off state, - check of D3 stop-clock and keep_link_on flags is moved to the caller side, - call pm_notify callback in snd_hda_codec_new() and snd_hda_codec_free() so that the refcounts are proprely updated. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 21 ++++++++++++++++----- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_intel.c | 19 +++---------------- 3 files changed, 20 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1b35115f7195..90b34e830415 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -98,9 +98,15 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) +static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) +{ + if (bus->ops.pm_notify) + bus->ops.pm_notify(bus, power_up); +} #else static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 +#define hda_call_pm_notify(bus, state) {} #endif /** @@ -1199,6 +1205,10 @@ static void snd_hda_codec_free(struct hda_codec *codec) codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (codec->power_on) + hda_call_pm_notify(codec->bus, false); +#endif module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); @@ -1271,6 +1281,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, * phase. */ hda_keep_power_on(codec); + hda_call_pm_notify(bus, true); #endif if (codec->bus->modelname) { @@ -3576,7 +3587,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, } #ifdef CONFIG_SND_HDA_POWER_SAVE - if ((power_state == AC_PWRST_D3) + if (!codec->bus->power_keep_link_on && power_state == AC_PWRST_D3 && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; #endif @@ -4430,8 +4441,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); hda_call_codec_suspend(codec); - if (bus->ops.pm_notify) - bus->ops.pm_notify(bus, codec); + if (codec->d3_stop_clk_ok) + hda_call_pm_notify(bus, false); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4488,8 +4499,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) codec->power_transition = 1; /* avoid reentrance */ spin_unlock(&codec->power_lock); - if (bus->ops.pm_notify) - bus->ops.pm_notify(bus, codec); + if (codec->d3_stop_clk_ok) /* flag set at suspend */ + hda_call_pm_notify(bus, true); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 72477ccb20f9..2e5a22fec0be 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -616,7 +616,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); + void (*pm_notify)(struct hda_bus *bus, bool power_up); #endif }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779dda55..1b6e856e7ab1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1033,7 +1033,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); +static void azx_power_notify(struct hda_bus *bus, bool power_up); #endif /* reset codec link */ @@ -2406,14 +2406,11 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) +static void azx_power_notify(struct hda_bus *bus, bool power_up) { struct azx *chip = bus->private_data; - if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) - return; - - if (codec->power_on) + if (power_up) pm_runtime_get_sync(&chip->pci->dev); else pm_runtime_put_sync(&chip->pci->dev); @@ -3273,15 +3270,6 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) } #endif -static void rpm_get_all_codecs(struct azx *chip) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &chip->bus->codec_list, list) { - pm_runtime_get_noresume(&chip->pci->dev); - } -} - static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3388,7 +3376,6 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); -- cgit v1.2.3 From 432c641e013d6e294e2ddf06d32a610eb1d4d856 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 09:59:20 -0700 Subject: ALSA: hda - Fix D3 clock stop check for codecs with own set_power_state op When a codec provides its own set_power_state op, the D3-clock-stop isn't checked correctly. And the recent changes for repeating the state-setting operation isn't applied to such a codec, too. This patch fixes these issues by moving the call of codec's own op to the place where the generic power-set operation is done, and move the power-state synchronization code out of snd_hda_set_power_state_to_all() so that it can be called always at the end of power-up/down sequence, and updates the D3 clock-stop flag properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 61 ++++++++++++++++++++++++++++------------------- 1 file changed, 37 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 90b34e830415..409f5ecd0a71 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3518,20 +3518,6 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, power_state); } - - if (power_state == AC_PWRST_D0) { - unsigned long end_time; - int state; - /* wait until the codec reachs to D0 */ - end_time = jiffies + msecs_to_jiffies(500); - do { - state = snd_hda_codec_read(codec, fg, 0, - AC_VERB_GET_POWER_STATE, 0); - if (state == power_state) - break; - msleep(1); - } while (time_after_eq(end_time, jiffies)); - } } EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); @@ -3551,6 +3537,32 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg return false; } +/* + * wait until the state is reached, returns the current state + */ +static unsigned int hda_sync_power_state(struct hda_codec *codec, + hda_nid_t fg, + unsigned int power_state) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(500); + unsigned int state, actual_state; + + for (;;) { + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (state & AC_PWRST_ERROR) + break; + actual_state = (state >> 4) & 0x0f; + if (actual_state == power_state) + break; + if (time_after_eq(jiffies, end_time)) + break; + /* wait until the codec reachs to the target state */ + msleep(1); + } + return state; +} + /* * set power state of the codec */ @@ -3564,11 +3576,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, codec->d3_stop_clk_ok = 0; #endif - if (codec->patch_ops.set_power_state) { - codec->patch_ops.set_power_state(codec, fg, power_state); - return; - } - /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ @@ -3577,11 +3584,17 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, /* repeat power states setting at most 10 times*/ for (count = 0; count < 10; count++) { - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); - state = snd_hda_codec_read(codec, fg, 0, - AC_VERB_GET_POWER_STATE, 0); + if (codec->patch_ops.set_power_state) + codec->patch_ops.set_power_state(codec, fg, + power_state); + else { + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, + true); + } + state = hda_sync_power_state(codec, fg, power_state); if (!(state & AC_PWRST_ERROR)) break; } -- cgit v1.2.3 From 83012a7ccbb90dee33c97a004b3e374f988612af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Aug 2012 18:38:08 +0200 Subject: ALSA: hda - Clean up CONFIG_SND_HDA_POWER_SAVE CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its behavior can be well controlled via the default value and module parameter. Let's just replace it with the standard CONFIG_PM. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 10 +--------- sound/pci/hda/hda_codec.c | 23 +++++++++-------------- sound/pci/hda/hda_codec.h | 10 ++++------ sound/pci/hda/hda_generic.c | 8 ++++---- sound/pci/hda/hda_hwdep.c | 4 ++-- sound/pci/hda/hda_intel.c | 16 +++++++--------- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/hda_trace.h | 4 ++-- sound/pci/hda/patch_analog.c | 36 +++++++++++++++++------------------- sound/pci/hda/patch_conexant.c | 6 +++--- sound/pci/hda/patch_hdmi.c | 2 +- sound/pci/hda/patch_realtek.c | 14 +++++++------- sound/pci/hda/patch_via.c | 4 +--- 13 files changed, 59 insertions(+), 80 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 194d625c1f83..7105c3de1bca 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -228,17 +228,9 @@ config SND_HDA_GENERIC Say Y here to enable the generic HD-audio codec parser in snd-hda-intel driver. -config SND_HDA_POWER_SAVE - bool "Aggressive power-saving on HD-audio" - depends on PM - help - Say Y here to enable more aggressive power-saving mode on - HD-audio driver. The power-saving timeout can be configured - via power_save option or over sysfs on-the-fly. - config SND_HDA_POWER_SAVE_DEFAULT int "Default time-out for HD-audio power-save mode" - depends on SND_HDA_POWER_SAVE + depends on PM default 0 help The default time-out value in seconds for HD-audio automatic diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 409f5ecd0a71..8a72f7b30a33 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -94,7 +94,7 @@ int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) } EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) @@ -1192,7 +1192,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) return; snd_hda_jack_tbl_clear(codec); restore_init_pincfgs(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); #endif @@ -1205,7 +1205,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM if (codec->power_on) hda_call_pm_notify(codec->bus, false); #endif @@ -1273,7 +1273,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64); snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spin_lock_init(&codec->power_lock); INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. @@ -1331,7 +1331,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_CLKSTOP); @@ -2353,7 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* OK, let it free */ -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM cancel_delayed_work_sync(&codec->power_work); codec->power_on = 0; codec->power_transition = 0; @@ -3572,7 +3572,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM codec->d3_stop_clk_ok = 0; #endif @@ -3599,7 +3599,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM if (!codec->bus->power_keep_link_on && power_state == AC_PWRST_D3 && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; @@ -3629,7 +3629,6 @@ static void hda_call_codec_suspend(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); -#ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); @@ -3638,7 +3637,6 @@ static void hda_call_codec_suspend(struct hda_codec *codec) codec->power_transition = 0; codec->power_jiffies = jiffies; spin_unlock(&codec->power_lock); -#endif } /* @@ -4434,7 +4432,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static void hda_power_work(struct work_struct *work) { struct hda_codec *codec = @@ -5107,9 +5105,6 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend); * @bus: the HDA bus * * Returns 0 if successful. - * - * This function is defined only when POWER_SAVE isn't set. - * In the power-save mode, the codec is resumed dynamically. */ int snd_hda_resume(struct hda_bus *bus) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2e5a22fec0be..7e1709cb8995 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,7 +614,7 @@ struct hda_bus_ops { struct hda_pcm *pcm); /* reset bus for retry verb */ void (*bus_reset)(struct hda_bus *bus); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM /* notify power-up/down from codec to controller */ void (*pm_notify)(struct hda_bus *bus, bool power_up); #endif @@ -712,8 +712,6 @@ struct hda_codec_ops { #ifdef CONFIG_PM int (*suspend)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); -#endif -#ifdef CONFIG_SND_HDA_POWER_SAVE int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); @@ -867,7 +865,7 @@ struct hda_codec { unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ @@ -1049,7 +1047,7 @@ int snd_hda_resume(struct hda_bus *bus); static inline int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) { -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM if (codec->patch_ops.check_power_status) return codec->patch_ops.check_power_status(codec, nid); #endif @@ -1066,7 +1064,7 @@ const char *snd_hda_get_jack_location(u32 cfg); /* * power saving */ -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); #else diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 431bf868711e..b81d3d0b952d 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -70,7 +70,7 @@ struct hda_gspec { struct list_head nid_list; /* list of widgets */ -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM #define MAX_LOOPBACK_AMPS 7 struct hda_loopback_check loopback; int num_loopbacks; @@ -654,7 +654,7 @@ static int parse_input(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static void add_input_loopback(struct hda_codec *codec, hda_nid_t nid, int dir, int idx) { @@ -1028,7 +1028,7 @@ static int build_generic_pcms(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int generic_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct hda_gspec *spec = codec->spec; @@ -1043,7 +1043,7 @@ static struct hda_codec_ops generic_patch_ops = { .build_controls = build_generic_controls, .build_pcms = build_generic_pcms, .free = snd_hda_generic_free, -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM .check_power_status = generic_check_power_status, #endif }; diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index b9a644ca03b3..1af86d40eb23 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -155,7 +155,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static ssize_t power_on_acct_show(struct device *dev, struct device_attribute *attr, char *buf) @@ -191,7 +191,7 @@ int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) hwdep->device, &power_attrs[i]); return 0; } -#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#endif /* CONFIG_PM */ #ifdef CONFIG_SND_HDA_RECONFIG diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b6e856e7ab1..6a19f6af9517 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -110,7 +110,7 @@ MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); static struct kernel_param_ops param_ops_xint = { .set = param_set_xint, @@ -130,7 +130,7 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); -#endif +#endif /* CONFIG_PM */ static int align_buffer_size = -1; module_param(align_buffer_size, bint, 0644); @@ -1032,7 +1032,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, return azx_rirb_get_response(bus, addr); } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static void azx_power_notify(struct hda_bus *bus, bool power_up); #endif @@ -1605,7 +1605,7 @@ static int DELAYED_INIT_MARK azx_codec_create(struct azx *chip, const char *mode bus_temp.ops.get_response = azx_get_response; bus_temp.ops.attach_pcm = azx_attach_pcm_stream; bus_temp.ops.bus_reset = azx_bus_reset; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM bus_temp.power_save = &power_save; bus_temp.ops.pm_notify = azx_power_notify; #endif @@ -2404,7 +2404,7 @@ static void azx_stop_chip(struct azx *chip) chip->initialized = 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM /* power-up/down the controller */ static void azx_power_notify(struct hda_bus *bus, bool power_up) { @@ -2457,7 +2457,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #else #define azx_add_card_list(chip) /* NOP */ #define azx_del_card_list(chip) /* NOP */ -#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#endif /* CONFIG_PM */ #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* @@ -2525,10 +2525,8 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; -#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; -#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); @@ -3232,7 +3230,7 @@ static int DELAYED_INIT_MARK azx_first_init(struct azx *chip) static void power_down_all_codecs(struct azx *chip) { -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM /* The codecs were powered up in snd_hda_codec_new(). * Now all initialization done, so turn them down if possible */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1b4c12941baa..09dbdc37f781 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -529,7 +529,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec); static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; } #endif -#if defined(CONFIG_SND_HDA_POWER_SAVE) && defined(CONFIG_SND_HDA_HWDEP) +#if defined(CONFIG_PM) && defined(CONFIG_SND_HDA_HWDEP) int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec); #else static inline int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index d42fe911abae..3a1c63161eb1 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -58,7 +58,7 @@ TRACE_EVENT(hda_bus_reset, TP_printk("[%d]", __entry->card) ); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM DECLARE_EVENT_CLASS(hda_power, TP_PROTO(struct hda_codec *codec), @@ -111,7 +111,7 @@ TRACE_EVENT(hda_power_count, __entry->card, __entry->addr, __entry->power_count, __entry->power_on, __entry->power_transition) ); -#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#endif /* CONFIG_PM */ TRACE_EVENT(hda_unsol_event, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 21218853366d..1a82cce4b551 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -85,7 +85,7 @@ struct ad198x_spec { unsigned int analog_beep: 1; /* analog beep input present */ unsigned int avoid_init_slave_vol:1; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM struct hda_loopback_check loopback; #endif /* for virtual master */ @@ -269,7 +269,7 @@ static int ad198x_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct ad198x_spec *spec = codec->spec; @@ -654,10 +654,8 @@ static const struct hda_codec_ops ad198x_patch_ops = { .build_pcms = ad198x_build_pcms, .init = ad198x_init, .free = ad198x_free, -#ifdef CONFIG_SND_HDA_POWER_SAVE - .check_power_status = ad198x_check_power_status, -#endif #ifdef CONFIG_PM + .check_power_status = ad198x_check_power_status, .suspend = ad198x_suspend, #endif .reboot_notify = ad198x_shutup, @@ -1231,7 +1229,7 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { {} }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1986a_loopbacks[] = { { 0x13, HDA_OUTPUT, 0 }, /* Mic */ { 0x14, HDA_OUTPUT, 0 }, /* Phone */ @@ -1278,7 +1276,7 @@ static int patch_ad1986a(struct hda_codec *codec) spec->mixers[0] = ad1986a_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1986a_init_verbs; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1986a_loopbacks; #endif spec->vmaster_nid = 0x1b; @@ -1537,7 +1535,7 @@ static const struct hda_verb ad1983_init_verbs[] = { { } /* end */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1983_loopbacks[] = { { 0x12, HDA_OUTPUT, 0 }, /* Mic */ { 0x13, HDA_OUTPUT, 0 }, /* Line */ @@ -1576,7 +1574,7 @@ static int patch_ad1983(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1983_init_verbs; spec->spdif_route = 0; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1983_loopbacks; #endif spec->vmaster_nid = 0x05; @@ -1704,7 +1702,7 @@ static const struct hda_verb ad1981_init_verbs[] = { { } /* end */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1981_loopbacks[] = { { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */ { 0x13, HDA_OUTPUT, 0 }, /* Line */ @@ -1982,7 +1980,7 @@ static int patch_ad1981(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1981_init_verbs; spec->spdif_route = 0; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1981_loopbacks; #endif spec->vmaster_nid = 0x05; @@ -2807,7 +2805,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_sequence_write(codec, ad1988_laptop_hp_off); } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1988_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Line */ @@ -3399,7 +3397,7 @@ static int patch_ad1988(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; break; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1988_loopbacks; #endif spec->vmaster_nid = 0x04; @@ -3555,7 +3553,7 @@ static const struct hda_verb ad1884_init_verbs[] = { { } /* end */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1884_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ @@ -3602,7 +3600,7 @@ static int patch_ad1884(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1884_init_verbs; spec->spdif_route = 0; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1884_loopbacks; #endif spec->vmaster_nid = 0x04; @@ -3994,7 +3992,7 @@ static const struct hda_verb ad1884a_init_verbs[] = { { } /* end */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1884a_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ @@ -4602,7 +4600,7 @@ static int patch_ad1884a(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1884a_init_verbs; spec->spdif_route = 0; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1884a_loopbacks; #endif codec->patch_ops = ad198x_patch_ops; @@ -4959,7 +4957,7 @@ static const struct hda_verb ad1882_3stack_automute_verbs[] = { { } /* end */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static const struct hda_amp_list ad1882_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ @@ -5022,7 +5020,7 @@ static int patch_ad1882(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1882_init_verbs; spec->spdif_route = 0; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->loopback.amplist = ad1882_loopbacks; #endif spec->vmaster_nid = 0x04; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 172895a8add4..7d70210d7623 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -553,7 +553,7 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int conexant_suspend(struct hda_codec *codec) { snd_hda_shutup_pins(codec); @@ -567,7 +567,7 @@ static const struct hda_codec_ops conexant_patch_ops = { .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM .suspend = conexant_suspend, #endif .reboot_notify = snd_hda_shutup_pins, @@ -4395,7 +4395,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = cx_auto_unsol_event, -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM .suspend = conexant_suspend, #endif .reboot_notify = snd_hda_shutup_pins, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d9439c59bb8b..afd6850670e7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1115,7 +1115,7 @@ static int hdmi_parse_codec(struct hda_codec *codec) * can be lost and presence sense verb will become inaccurate if the * HDA link is powered off at hot plug or hw initialization time. */ -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & AC_PWRST_EPSS)) codec->bus->power_keep_link_on = 1; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce99cc955326..613499932b2b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -174,7 +174,7 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM void (*power_hook)(struct hda_codec *codec); #endif void (*shutup)(struct hda_codec *codec); @@ -215,7 +215,7 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; struct hda_vmaster_mute_hook vmaster_mute; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM struct hda_loopback_check loopback; int num_loopbacks; struct hda_amp_list loopback_list[8]; @@ -2057,7 +2057,7 @@ static int alc_init(struct hda_codec *codec) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct alc_spec *spec = codec->spec; @@ -2435,7 +2435,7 @@ static void alc_free(struct hda_codec *codec) snd_hda_detach_beep_device(codec); } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static void alc_power_eapd(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); @@ -2475,7 +2475,7 @@ static const struct hda_codec_ops alc_patch_ops = { #ifdef CONFIG_PM .resume = alc_resume, #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif @@ -2631,7 +2631,7 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return channel_name[ch]; } -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM /* add the powersave loopback-list entry */ static void add_loopback_list(struct alc_spec *spec, hda_nid_t mix, int idx) { @@ -6501,7 +6501,7 @@ static int patch_alc861(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM spec->power_hook = alc_power_eapd; #endif diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 4b0796b30f78..d5f36a179eb5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1764,7 +1764,7 @@ static int via_suspend(struct hda_codec *codec) } #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE +#ifdef CONFIG_PM static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct via_spec *spec = codec->spec; @@ -1785,8 +1785,6 @@ static const struct hda_codec_ops via_patch_ops = { .unsol_event = via_unsol_event, #ifdef CONFIG_PM .suspend = via_suspend, -#endif -#ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif }; -- cgit v1.2.3 From 4b927345a38c8e9eb2482c091f96ac06884aff8e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Aug 2012 16:39:25 -0700 Subject: ALSA: hda - Optimize bitfield usage in struct hda_codec Move up a few bitfields to be packed into a single int. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7e1709cb8995..9ac477334d2d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -867,6 +867,8 @@ struct hda_codec { unsigned int epss:1; /* supporting EPSS? */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ @@ -874,9 +876,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - - unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ -- cgit v1.2.3 From fbaf6a5a35a830c2ae0449210efa31c165445735 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Aug 2012 07:57:38 -0700 Subject: ALSA: korg1212: Fix reverted min/max ADC sense range k1212MinADCSens and k1212MaxADCSens are defined wrongly. The max must be greater than the min by obvious reason. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46561 Signed-off-by: Takashi Iwai --- sound/pci/korg1212/korg1212.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index e69ce5f9c31e..8a67ce95f246 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -196,8 +196,8 @@ enum MonitorModeSelector { #define K1212_ADAT_BUF_SIZE (K1212_ADAT_CHANNELS * 2 * kPlayBufferFrames * kNumBuffers) #define K1212_MAX_BUF_SIZE (K1212_ANALOG_BUF_SIZE + K1212_ADAT_BUF_SIZE) -#define k1212MinADCSens 0x7f -#define k1212MaxADCSens 0x00 +#define k1212MinADCSens 0x00 +#define k1212MaxADCSens 0x7f #define k1212MaxVolume 0x7fff #define k1212MaxWaveVolume 0xffff #define k1212MinVolume 0x0000 -- cgit v1.2.3 From 5a798394c85f3bca963505d3be49180416fce132 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Aug 2012 13:21:00 -0700 Subject: ALSA: cs5530: Fix resource leak in error path Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=44741 Signed-off-by: Takashi Iwai --- sound/pci/cs5530.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index f1e4229993af..d1cca2831575 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -142,8 +142,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { - kfree(chip); - pci_disable_device(pci); + snd_cs5530_free(chip); return -EBUSY; } -- cgit v1.2.3 From 08fa20ae20eb378225e5519db4e07f663ce405fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Aug 2012 07:46:56 -0700 Subject: ALSA: hda - Yet another fix for D3 stop-clock refcounting The call of pm_notify callback in snd_hda_codec_free() should be with the check of the current state whether pm_notify(false) is called or not, instead of codec->power_on check. For improving the code readability and fixing this inconsistency, codec->d3_stop_clk_ok is renamed to codec->pm_down_notified, and this flag is set only when runtime PM down is called. The new name reflects to a more direct purpose of the flag. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++++------------------- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 23 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8a72f7b30a33..8e7dbb0921e6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1206,7 +1206,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (codec->patch_ops.free) codec->patch_ops.free(codec); #ifdef CONFIG_PM - if (codec->power_on) + if (!codec->pm_down_notified) /* cancel leftover refcounts */ hda_call_pm_notify(codec->bus, false); #endif module_put(codec->owner); @@ -1222,7 +1222,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); -static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, +static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); /** @@ -3564,18 +3564,14 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec, } /* - * set power state of the codec + * set power state of the codec, and return the power state */ -static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) +static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) { int count; unsigned int state; -#ifdef CONFIG_PM - codec->d3_stop_clk_ok = 0; -#endif - /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ @@ -3599,11 +3595,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } -#ifdef CONFIG_PM - if (!codec->bus->power_keep_link_on && power_state == AC_PWRST_D3 - && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) - codec->d3_stop_clk_ok = 1; -#endif + return state; } #ifdef CONFIG_SND_HDA_HWDEP @@ -3620,13 +3612,16 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #ifdef CONFIG_PM /* * call suspend and power-down; used both from PM and power-save + * this function returns the power state in the end */ -static void hda_call_codec_suspend(struct hda_codec *codec) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { + unsigned int state; + if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); - hda_set_power_state(codec, + state = hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); cancel_delayed_work(&codec->power_work); @@ -3637,6 +3632,7 @@ static void hda_call_codec_suspend(struct hda_codec *codec) codec->power_transition = 0; codec->power_jiffies = jiffies; spin_unlock(&codec->power_lock); + return state; } /* @@ -4438,6 +4434,7 @@ static void hda_power_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, power_work.work); struct hda_bus *bus = codec->bus; + unsigned int state; spin_lock(&codec->power_lock); if (codec->power_transition > 0) { /* during power-up sequence? */ @@ -4451,9 +4448,12 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); - hda_call_codec_suspend(codec); - if (codec->d3_stop_clk_ok) + state = hda_call_codec_suspend(codec); + codec->pm_down_notified = 0; + if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); + } } static void hda_keep_power_on(struct hda_codec *codec) @@ -4510,8 +4510,11 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) codec->power_transition = 1; /* avoid reentrance */ spin_unlock(&codec->power_lock); - if (codec->d3_stop_clk_ok) /* flag set at suspend */ + if (codec->pm_down_notified) { + codec->pm_down_notified = 0; hda_call_pm_notify(bus, true); + } + hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9ac477334d2d..8fd9f636cfe1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -868,7 +868,7 @@ struct hda_codec { #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ + unsigned int pm_down_notified:1; /* PM notified to controller */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- cgit v1.2.3 From d819387ef7ca97b4cb3494bfaba9c0cd510122a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 Aug 2012 07:54:38 -0700 Subject: ALSA: hda - Clean up redundant FG checks Just refactoring, no functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 30 ++++++++++++------------------ 1 file changed, 12 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8e7dbb0921e6..ff97cf3e8bad 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1222,7 +1222,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); -static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, +static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state); /** @@ -1239,6 +1239,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, { struct hda_codec *codec; char component[31]; + hda_nid_t fg; int err; if (snd_BUG_ON(!bus)) @@ -1315,7 +1316,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, goto error; } - err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg); + fg = codec->afg ? codec->afg : codec->mfg; + err = read_widget_caps(codec, fg); if (err < 0) { snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); goto error; @@ -1325,27 +1327,22 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, goto error; if (!codec->subsystem_id) { - hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; codec->subsystem_id = - snd_hda_codec_read(codec, nid, 0, + snd_hda_codec_read(codec, fg, 0, AC_VERB_GET_SUBSYSTEM_ID, 0); } #ifdef CONFIG_PM - codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, - codec->afg ? codec->afg : codec->mfg, + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_CLKSTOP); if (!codec->d3_stop_clk) bus->power_keep_link_on = 1; #endif - codec->epss = snd_hda_codec_get_supported_ps(codec, - codec->afg ? codec->afg : codec->mfg, + codec->epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); /* power-up all before initialization */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); + hda_set_power_state(codec, AC_PWRST_D0); snd_hda_codec_proc_new(codec); @@ -3566,9 +3563,10 @@ static unsigned int hda_sync_power_state(struct hda_codec *codec, /* * set power state of the codec, and return the power state */ -static unsigned int hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, +static unsigned int hda_set_power_state(struct hda_codec *codec, unsigned int power_state) { + hda_nid_t fg = codec->afg ? codec->afg : codec->mfg; int count; unsigned int state; @@ -3621,9 +3619,7 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec) if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); - state = hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D3); + state = hda_set_power_state(codec, AC_PWRST_D3); cancel_delayed_work(&codec->power_work); spin_lock(&codec->power_lock); snd_hda_update_power_acct(codec); @@ -3644,9 +3640,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) * in the resume / power-save sequence */ hda_keep_power_on(codec); - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); + hda_set_power_state(codec, AC_PWRST_D0); restore_pincfgs(codec); /* restore all current pin configs */ restore_shutup_pins(codec); hda_exec_init_verbs(codec); -- cgit v1.2.3 From ffb690d5aa36d38d7bed7579e3f07b84ff6b3a08 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Fri, 31 Aug 2012 18:20:59 +0530 Subject: ASoC: Davinci: evm: Fix typo in cpu dai name Fix typo caused by recent commit (cf53756 - ASoC: davinci: davinci-pcm does not need to be a plaform_driver) Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index ab0ad4591b08..6fac5af13298 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -188,7 +188,7 @@ static struct snd_soc_dai_link dm365_evm_dai = { .cpu_dai_name = "davinci-vcif", .codec_dai_name = "cq93vc-hifi", .codec_name = "cq93vc-codec", - .platform_name = "avinci-vcif", + .platform_name = "davinci-vcif", #endif }; -- cgit v1.2.3 From a03e4a66c71b2bb33dc795eb6451a3d2ad1398bb Mon Sep 17 00:00:00 2001 From: Josh Triplett Date: Sun, 2 Sep 2012 20:04:17 -0700 Subject: ALSA: Remove the last mention of SNDRV_MAIN_OBJECT_FILE SNDRV_MAIN_OBJECT_FILE hasn't done anything since the pre-git days, and the only remaining reference occurs as a #define in sound/last.c. Drop that last mention of it. Signed-off-by: Josh Triplett Signed-off-by: Takashi Iwai --- sound/last.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/last.c b/sound/last.c index 7ffc182e0844..43f222825038 100644 --- a/sound/last.c +++ b/sound/last.c @@ -19,7 +19,6 @@ * */ -#define SNDRV_MAIN_OBJECT_FILE #include #include -- cgit v1.2.3 From 1f3b14072bacc80d62597ed7798e3daf7615dcc6 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sun, 2 Sep 2012 22:10:27 +0800 Subject: ALSA: fix possible memory leak in snd_mixer_oss_build_input() uinfo has been allocated in this function and should be freed before leaving from the error handling cases. spatch with a semantic match is used to found this problem. (http://coccinelle.lip6.fr/) Signed-off-by: Wei Yongjun Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 18297f7f2c55..29f6ded02555 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1046,6 +1046,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix if (kctl->info(kctl, uinfo)) { up_read(&mixer->card->controls_rwsem); + kfree(uinfo); return 0; } strcpy(str, ptr->name); @@ -1061,6 +1062,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix uinfo->value.enumerated.item = slot.capture_item; if (kctl->info(kctl, uinfo)) { up_read(&mixer->card->controls_rwsem); + kfree(uinfo); return 0; } if (!strcmp(uinfo->value.enumerated.name, str)) { -- cgit v1.2.3 From c05fce586d4da2dfe0309bef3795a8586e967bc3 Mon Sep 17 00:00:00 2001 From: Marko Friedemann Date: Mon, 3 Sep 2012 10:12:40 +0200 Subject: ALSA: USB: Support for (original) Xbox Communicator Added support for Xbox Communicator to USB quirks. Signed-off-by: Marko Friedemann Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 53 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 79780fa57a43..d73ac9bc4272 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2781,6 +2781,59 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Microsoft XboxLive Headset/Xbox Communicator */ +{ + USB_DEVICE(0x045e, 0x0283), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Microsoft", + .product_name = "XboxLive Headset/Xbox Communicator", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + /* playback */ + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 1, + .iface = 0, + .altsetting = 0, + .altset_idx = 0, + .attributes = 0, + .endpoint = 0x04, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 22050, + .rate_max = 22050 + } + }, + { + /* capture */ + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 1, + .iface = 1, + .altsetting = 0, + .altset_idx = 0, + .attributes = 0, + .endpoint = 0x85, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 16000, + .rate_max = 16000 + } + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.2.3 From 2b58fd5b3193fd3af3d15114d95706087d25a7fe Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 4 Sep 2012 10:23:07 +0200 Subject: ALSA: snd-usb: Add quirks for Playback Designs devices Playback Designs' USB devices have some hardware limitations on their USB interface. In particular: - They need a 20ms delay after each class compliant request as the hardware ACKs the USB packets before the device is actually ready for the next command. Sending data immediately will result in buffer overflows in the hardware. - The devices send bogus feedback data at the start of each stream which confuse the feedback format auto-detection. This patch introduces a new quirks hook that is called after each control packet and which adds a delay for all devices that match Playback Designs' USB VID for now. In addition, it adds a counter to snd_usb_endpoint to drop received packets on the floor. Another new quirks function that is called once an endpoint is started initializes that counter for these devices on their sync endpoint. Signed-off-by: Daniel Mack Reported-and-tested-by: Andreas Koch Supported-by: Demian Martin Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 ++ sound/usb/endpoint.c | 8 ++++++++ sound/usb/helper.c | 5 +++++ sound/usb/quirks.c | 24 ++++++++++++++++++++++++ sound/usb/quirks.h | 10 ++++++++++ 5 files changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index bcb5267a02e8..23b6f23bd36a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -92,6 +92,8 @@ struct snd_usb_endpoint { unsigned char silence_value; unsigned int stride; int iface, alt_idx; + int skip_packets; /* quirks for devices to ignore the first n packets + in a stream */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c41181202688..94b08a6087a3 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -31,6 +31,7 @@ #include "card.h" #include "endpoint.h" #include "pcm.h" +#include "quirks.h" #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 @@ -170,6 +171,11 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, { struct urb *urb = urb_ctx->urb; + if (unlikely(ep->skip_packets > 0)) { + ep->skip_packets--; + return; + } + if (ep->sync_slave) snd_usb_handle_sync_urb(ep->sync_slave, ep, urb); @@ -825,6 +831,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->unlink_mask = 0; ep->phase = 0; + snd_usb_endpoint_start_quirk(ep); + /* * If this endpoint has a data endpoint as implicit feedback source, * don't start the urbs here. Instead, mark them all as available, diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 9eed8f40b179..c1db28f874c2 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -21,6 +21,7 @@ #include "usbaudio.h" #include "helper.h" +#include "quirks.h" /* * combine bytes and get an integer value @@ -97,6 +98,10 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, memcpy(data, buf, size); kfree(buf); } + + snd_usb_ctl_msg_quirk(dev, pipe, request, requesttype, + value, index, data, size); + return err; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 27817266867a..0f58b4b6d702 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -761,3 +761,27 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, } } +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) +{ + /* + * "Playback Design" products send bogus feedback data at the start + * of the stream. Ignore them. + */ + if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) && + ep->type == SND_USB_ENDPOINT_TYPE_SYNC) + ep->skip_packets = 4; +} + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size) +{ + /* + * "Playback Design" products need a 20ms delay after each + * class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); +} + diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 03e5e94098cd..0ca9e91067a6 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -1,6 +1,10 @@ #ifndef __USBAUDIO_QUIRKS_H #define __USBAUDIO_QUIRKS_H +struct audioformat; +struct snd_usb_endpoint; +struct snd_usb_substream; + int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, struct usb_driver *driver, @@ -20,4 +24,10 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp); +void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); + +void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, + __u16 index, void *data, __u16 size); + #endif /* __USBAUDIO_QUIRKS_H */ -- cgit v1.2.3 From 4266274836e81575ee82498d84f4bd08ab7a7378 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 4 Sep 2012 11:21:45 +0200 Subject: ALSA: remove the main version information Remove the main ALSA version number from the kernel ALSA driver. The ALSA driver package release diverges from the upstream. This may confuse users to see the same ALSA version for many kernel releases and this version lost it's original purpose and connection. The "ioctl" APIs have own version numbers, so the user space may check for specific API changes only. Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 3 --- sound/core/info.c | 7 +++---- sound/core/info_oss.c | 3 +-- sound/core/sound.c | 3 +-- 4 files changed, 5 insertions(+), 11 deletions(-) delete mode 100644 include/sound/version.h (limited to 'sound') diff --git a/include/sound/version.h b/include/sound/version.h deleted file mode 100644 index cc75024c1089..000000000000 --- a/include/sound/version.h +++ /dev/null @@ -1,3 +0,0 @@ -/* include/version.h */ -#define CONFIG_SND_VERSION "1.0.25" -#define CONFIG_SND_DATE "" diff --git a/sound/core/info.c b/sound/core/info.c index c1e611c65c8f..6b368d25073b 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include #include @@ -986,9 +986,8 @@ static struct snd_info_entry *snd_info_version_entry; static void snd_info_version_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { snd_iprintf(buffer, - "Advanced Linux Sound Architecture Driver Version " - CONFIG_SND_VERSION CONFIG_SND_DATE ".\n" - ); + "Advanced Linux Sound Architecture Driver Version k%s.\n", + init_utsname()->release); } static int __init snd_info_version_init(void) diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index cf42ab5080eb..83c29dbff9c0 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include @@ -94,7 +93,7 @@ static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int d static void snd_sndstat_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_iprintf(buffer, "Sound Driver:3.8.1a-980706 (ALSA v" CONFIG_SND_VERSION " emulation code)\n"); + snd_iprintf(buffer, "Sound Driver:3.8.1a-980706 (ALSA emulation code)\n"); snd_iprintf(buffer, "Kernel: %s %s %s %s %s\n", init_utsname()->sysname, init_utsname()->nodename, diff --git a/sound/core/sound.c b/sound/core/sound.c index 28f35593a750..643976000ce8 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include @@ -468,7 +467,7 @@ static int __init alsa_sound_init(void) } snd_info_minor_register(); #ifndef MODULE - printk(KERN_INFO "Advanced Linux Sound Architecture Driver Version " CONFIG_SND_VERSION CONFIG_SND_DATE ".\n"); + printk(KERN_INFO "Advanced Linux Sound Architecture Driver Initialized.\n"); #endif return 0; } -- cgit v1.2.3 From 3a4a7ef5678416b04927102f85998e38b52f7196 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 5 Sep 2012 14:33:21 +0800 Subject: ALSA: opl4: use list_move_tail instead of list_del/list_add_tail Using list_move_tail() instead of list_del() + list_add_tail(). Signed-off-by: Wei Yongjun Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/drivers/opl4/opl4_synth.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/drivers/opl4/opl4_synth.c b/sound/drivers/opl4/opl4_synth.c index 49b9e240915c..4b91adc0238c 100644 --- a/sound/drivers/opl4/opl4_synth.c +++ b/sound/drivers/opl4/opl4_synth.c @@ -504,8 +504,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha spin_lock_irqsave(&opl4->reg_lock, flags); for (i = 0; i < voices; i++) { voice[i] = snd_opl4_get_voice(opl4); - list_del(&voice[i]->list); - list_add_tail(&voice[i]->list, &opl4->on_voices); + list_move_tail(&voice[i]->list, &opl4->on_voices); voice[i]->chan = chan; voice[i]->note = note; voice[i]->velocity = vel & 0x7f; @@ -555,8 +554,7 @@ void snd_opl4_note_on(void *private_data, int note, int vel, struct snd_midi_cha static void snd_opl4_voice_off(struct snd_opl4 *opl4, struct opl4_voice *voice) { - list_del(&voice->list); - list_add_tail(&voice->list, &opl4->off_voices); + list_move_tail(&voice->list, &opl4->off_voices); voice->reg_misc &= ~OPL4_KEY_ON_BIT; snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); @@ -571,8 +569,7 @@ void snd_opl4_note_off(void *private_data, int note, int vel, struct snd_midi_ch static void snd_opl4_terminate_voice(struct snd_opl4 *opl4, struct opl4_voice *voice) { - list_del(&voice->list); - list_add_tail(&voice->list, &opl4->off_voices); + list_move_tail(&voice->list, &opl4->off_voices); voice->reg_misc = (voice->reg_misc & ~OPL4_KEY_ON_BIT) | OPL4_DAMP_BIT; snd_opl4_write(opl4, OPL4_REG_MISC + voice->number, voice->reg_misc); -- cgit v1.2.3 From 292f2b6254c9dbb98def6d3521b07a837545ead0 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 5 Sep 2012 15:00:15 +0800 Subject: ALSA: emu10k1: use list_move_tail instead of list_del/list_add_tail Using list_move_tail() instead of list_del() + list_add_tail(). Signed-off-by: Wei Yongjun Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 0a436626182b..ae709c1ab3a8 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -263,8 +263,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b spin_lock_irqsave(&emu->memblk_lock, flags); if (blk->mapped_page >= 0) { /* update order link */ - list_del(&blk->mapped_order_link); - list_add_tail(&blk->mapped_order_link, &emu->mapped_order_link_head); + list_move_tail(&blk->mapped_order_link, + &emu->mapped_order_link_head); spin_unlock_irqrestore(&emu->memblk_lock, flags); return 0; } -- cgit v1.2.3 From e93c7d1bc350189511d32cec2f0af79c30e7fa47 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 Aug 2012 17:06:15 +0300 Subject: ASoC: omap-mcbsp: Fix compilation error due to leftover code Part of commit (which patches sound/soc/omap/mcbsp.c file): 8fef626 ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code since the tree where it has been applied did not had the earlier patch: d0db84e ASoC: omap-mcbsp: Fix 6pin mux configuration which changed code around omap_mcbsp_6pin_src_mux(). Because of the missing part from 8fef626 the sound/soc/omap/mcbsp.c does not compile in linux-next. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 31 ------------------------------- 1 file changed, 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 935ccf633976..bc06175e6367 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -762,37 +762,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) } -int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *signal, *src; - - if (!mcbsp->pdata->mux_signal) - return -EINVAL; - - switch (mux) { - case CLKR_SRC_CLKR: - signal = "clkr"; - src = "clkr"; - break; - case CLKR_SRC_CLKX: - signal = "clkr"; - src = "clkx"; - break; - case FSR_SRC_FSR: - signal = "fsr"; - src = "fsr"; - break; - case FSR_SRC_FSX: - signal = "fsr"; - src = "fsx"; - break; - default: - return -EINVAL; - } - - return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); -} - #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) #define THRESHOLD_PROP_BUILDER(prop) \ -- cgit v1.2.3 From 58d468328646effa72ada4deaa33e80d678980d6 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Thu, 30 Aug 2012 08:16:52 -0700 Subject: ASoC: wm0010: Add missing IRQF_ONESHOT FYI, there are new coccinelle warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.7 head: e3523e01869da20fdd12ffd19ae1df7bf492650e commit: e3523e01869da20fdd12ffd19ae1df7bf492650e [95/95] ASoC: wm0010: Add initial wm0010 DSP driver All coccinelle warnings: + sound/soc/codecs/wm0010.c:850:7-27: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT -- + sound/soc/codecs/wm0010.c:660:1-7: preceding lock on line 359 vim +850 sound/soc/codecs/wm0010.c 847 trigger = IRQF_TRIGGER_FALLING; 848 trigger |= IRQF_ONESHOT; 849 > 850 ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, 851 "wm0010", wm0010); 852 if (ret) 853 dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", Please consider folding the attached diff :-) Signed-off-by: Fengguang Wu Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 5f99148447e1..a4c35119792e 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -881,7 +881,7 @@ static int __devinit wm0010_spi_probe(struct spi_device *spi) trigger = IRQF_TRIGGER_FALLING; trigger |= IRQF_ONESHOT; - ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger | IRQF_ONESHOT, "wm0010", wm0010); if (ret) { dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", -- cgit v1.2.3 From 4f3ad7956d91a5371a572f0420cc07f8c4f32c22 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 5 Sep 2012 15:29:46 +0300 Subject: ASoC: wm0010: unlock on error path We're holding the wm0010->lock mutex when we goto err_core. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index a4c35119792e..0274f04a940a 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -663,7 +663,9 @@ abort: wm0010_halt(codec); mutex_unlock(&wm0010->lock); return ret; + err_core: + mutex_unlock(&wm0010->lock); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), wm0010->core_supplies); err: -- cgit v1.2.3 From 5d86e25c70407cd97a5aa8f39cc3be390bcab116 Mon Sep 17 00:00:00 2001 From: Emil Goode Date: Wed, 5 Sep 2012 22:22:24 +0200 Subject: ASoC: wm0010: Fix warning, use format %zu for type size_t MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix warning by using format specifier %zu for type size_t Sparse warning: sound/soc/codecs/wm0010.c:411:2: warning: format ‘%d’ expects argument of type ‘int’, but argument 4 has type ‘size_t’ [-Wformat] Signed-off-by: Emil Goode Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 0274f04a940a..f8d6c31db870 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -408,7 +408,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) wm0010->state = WM0010_BOOTROM; spin_unlock_irqrestore(&wm0010->irq_lock, flags); - dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ img = kzalloc(fw->size, GFP_KERNEL); -- cgit v1.2.3 From 03f67433758a3eeb37b9c1559886c377da874ad2 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 5 Sep 2012 10:27:14 -0600 Subject: ASoC: tegra: move platform data header Move the Tegra+WM8903 ASoC platform data header out of arch/arm/mach-tegra, as a pre-requisite of single zImage. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- .../mach-tegra/include/mach/tegra_wm8903_pdata.h | 23 ------------------- include/sound/tegra_wm8903.h | 26 ++++++++++++++++++++++ sound/soc/tegra/tegra_wm8903.c | 3 +-- 3 files changed, 27 insertions(+), 25 deletions(-) delete mode 100644 arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h create mode 100644 include/sound/tegra_wm8903.h (limited to 'sound') diff --git a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h b/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h deleted file mode 100644 index 9d293344a7ff..000000000000 --- a/arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h +++ /dev/null @@ -1,23 +0,0 @@ -/* - * arch/arm/mach-tegra/include/mach/tegra_wm8903_pdata.h - * - * Copyright 2011 NVIDIA, Inc. - * - * This software is licensed under the terms of the GNU General Public - * License version 2, as published by the Free Software Foundation, and - * may be copied, distributed, and modified under those terms. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - */ - -struct tegra_wm8903_platform_data { - int gpio_spkr_en; - int gpio_hp_det; - int gpio_hp_mute; - int gpio_int_mic_en; - int gpio_ext_mic_en; -}; diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h new file mode 100644 index 000000000000..57b202ee97c3 --- /dev/null +++ b/include/sound/tegra_wm8903.h @@ -0,0 +1,26 @@ +/* + * Copyright 2011 NVIDIA, Inc. + * + * This software is licensed under the terms of the GNU General Public + * License version 2, as published by the Free Software Foundation, and + * may be copied, distributed, and modified under those terms. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_TEGRA_WM38903_H +#define __SOUND_TEGRA_WM38903_H + +struct tegra_wm8903_platform_data { + int gpio_spkr_en; + int gpio_hp_det; + int gpio_hp_mute; + int gpio_int_mic_en; + int gpio_ext_mic_en; +}; + +#endif diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index d4f14e492341..cee13b7bfb94 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -34,13 +34,12 @@ #include #include -#include - #include #include #include #include #include +#include #include "../codecs/wm8903.h" -- cgit v1.2.3 From e5ec69da24803c68f5c035662a68d367359a4132 Mon Sep 17 00:00:00 2001 From: "Hebbar, Gururaja" Date: Mon, 3 Sep 2012 13:40:40 +0530 Subject: ASoC: Davinci: McASP: add support new McASP IP Variant The OMAP2+ variant of McASP is different from Davinci variant w.r.to some register offset. Changes - Add new MCASP_VERSION_3 to identify new variant. New DT compatible "ti,omap2-mcasp-audio" to identify version 3 controller. - The register offsets are handled depending on the version. Note: DMA parameters (dma fifo offset) are not updated and will be done later. Signed-off-by: Hebbar, Gururaja Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcasp-audio.txt | 1 + include/linux/platform_data/davinci_asp.h | 1 + sound/soc/davinci/davinci-mcasp.c | 86 ++++++++++++++++++---- 3 files changed, 75 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index e6148eca2942..374e145c2ef1 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,6 +4,7 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms + "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - reg : Should contain McASP registers offset and length - interrupts : Interrupt number for McASP diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 79c26aa11db6..d0c5825876f8 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -87,6 +87,7 @@ struct snd_platform_data { enum { MCASP_VERSION_1 = 0, /* DM646x */ MCASP_VERSION_2, /* DA8xx/OMAPL1x */ + MCASP_VERSION_3, /* TI81xx/AM33xx */ }; enum mcbsp_clk_input_pin { diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c3eae1d8e077..714e51e5be5b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -111,6 +111,10 @@ #define DAVINCI_MCASP_WFIFOSTS (0x1014) #define DAVINCI_MCASP_RFIFOCTL (0x1018) #define DAVINCI_MCASP_RFIFOSTS (0x101C) +#define MCASP_VER3_WFIFOCTL (0x1000) +#define MCASP_VER3_WFIFOSTS (0x1004) +#define MCASP_VER3_RFIFOCTL (0x1008) +#define MCASP_VER3_RFIFOSTS (0x100C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -384,18 +388,36 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { if (dev->txnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } } mcasp_start_tx(dev); } else { if (dev->rxnumevt) { /* enable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_set_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + mcasp_set_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } mcasp_start_rx(dev); } @@ -416,14 +438,31 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (dev->txnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + if (dev->txnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_WFIFOCTL, FIFO_ENABLE); + break; + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_tx(dev); } else { - if (dev->rxnumevt) /* disable FIFO */ - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + if (dev->rxnumevt) { /* disable FIFO */ + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_clr_bits(dev->base + MCASP_VER3_RFIFOCTL, FIFO_ENABLE); + break; + + default: + mcasp_clr_bits(dev->base + + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } + } mcasp_stop_rx(dev); } } @@ -622,20 +661,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) if (dev->txnumevt * tx_ser > 64) dev->txnumevt = 1; - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, tx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + tx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + } } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { if (dev->rxnumevt * rx_ser > 64) dev->rxnumevt = 1; - - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + switch (dev->version) { + case MCASP_VERSION_3: + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, rx_ser, NUMDMA_MASK); - mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + mcasp_mod_bits(dev->base + MCASP_VER3_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + break; + default: + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + rx_ser, NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + } } } @@ -874,6 +930,10 @@ static const struct of_device_id mcasp_dt_ids[] = { .compatible = "ti,da830-mcasp-audio", .data = (void *)MCASP_VERSION_2, }, + { + .compatible = "ti,omap2-mcasp-audio", + .data = (void *)MCASP_VERSION_3, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, mcasp_dt_ids); -- cgit v1.2.3 From e2d32ff6ce4ee9958f3973a086f3fa5d009e6306 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 31 Aug 2012 17:38:32 -0700 Subject: ASoC: dapm: Ensure bypass paths are suspended and resumed Since bypass paths aren't part of DAPM streams and we may not have any DAPM streams there may not be anything that triggers a DAPM sync for them. Mark all input and output widgets as dirty and then sync to do so at the end of suspend and resume. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 8 ++++++++ sound/soc/soc-dapm.c | 22 ++++++++++++++++++++++ 3 files changed, 31 insertions(+) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0a1553748d65..07e2510619a1 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -412,6 +412,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b95d1fb388a1..ad65459da28e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -609,6 +609,10 @@ int snd_soc_suspend(struct device *dev) SND_SOC_DAPM_STREAM_SUSPEND); } + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); + /* suspend all CODECs */ list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with @@ -756,6 +760,10 @@ static void soc_resume_deferred(struct work_struct *work) /* userspace can access us now we are back as we were before */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); + + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); } /* powers up audio subsystem after a suspend */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dd7c49fafd75..f7999e949acb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -141,6 +141,28 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } EXPORT_SYMBOL_GPL(dapm_mark_dirty); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_card *card = dapm->card; + struct snd_soc_dapm_widget *w; + + mutex_lock(&card->dapm_mutex); + + list_for_each_entry(w, &card->widgets, list) { + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + dapm_mark_dirty(w, "Rechecking inputs and outputs"); + break; + default: + break; + } + } + + mutex_unlock(&card->dapm_mutex); +} +EXPORT_SYMBOL_GPL(dapm_mark_io_dirty); + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) -- cgit v1.2.3 From 104c229932f18605acef2351373f8fd8f6c6f8c6 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Wed, 5 Sep 2012 21:30:10 -0300 Subject: ASoC: Revert 'ASoC: imx-ssi: Remove mono support' Revert 0865a75(ASoC: imx-ssi: Remove mono support). The bug this patch is meant to solve doesn't occur in Visstrim_M10 boards. Furthermore, after applying this patch sound in Visstrim_M10 is played at slower rates. Signed-off-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/fsl/imx-ssi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 7074ae689984..3c520c46fa4a 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,14 +380,13 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- cgit v1.2.3 From 72357c78b75d39ee9e8d9fb4308957d9525aa674 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 6 Sep 2012 10:02:36 +0800 Subject: ALSA: HDMI - Fix channel_allocation array wrong order The array channel_allocations[] is an ordered list, add function to get correct order by ca_index. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index afd6850670e7..4fb8199d813a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -469,6 +469,17 @@ static void init_channel_allocations(void) } } +static int get_channel_allocation_order(int ca) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channel_allocations[i].ca_index == ca) + break; + } + return i; +} + /* * The transformation takes two steps: * @@ -541,9 +552,11 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, { int i; int err; + int order; + order = get_channel_allocation_order(ca); if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) + for (i = 0; i < channel_allocations[order].channels; i++) hdmi_channel_mapping[ca][i] = i | (i << 4); for (; i < 8; i++) hdmi_channel_mapping[ca][i] = 0xf | (i << 4); -- cgit v1.2.3 From 433968da4d93a194b79da552f4ca707f979ef33b Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 6 Sep 2012 10:02:37 +0800 Subject: ALSA: HDMI - Enable HBR feature on Intel chips HDMI channel remapping apparently effects HBR packets on Intel's chips. For compressed non-PCM audio, use "straight-through" channel mapping. For uncompressed multi-channel pcm audio, use normal channel mapping. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4fb8199d813a..a6835bd9d9b5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -34,6 +34,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_jack.h" @@ -60,6 +61,7 @@ struct hdmi_spec_per_cvt { u32 rates; u64 formats; unsigned int maxbps; + bool non_pcm; }; struct hdmi_spec_per_pin { @@ -548,13 +550,17 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, + hda_nid_t cvt_nid, + bool non_pcm, int ca) { int i; int err; int order; + int non_pcm_mapping[8]; order = get_channel_allocation_order(ca); + if (hdmi_channel_mapping[ca][1] == 0) { for (i = 0; i < channel_allocations[order].channels; i++) hdmi_channel_mapping[ca][i] = i | (i << 4); @@ -562,10 +568,17 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hdmi_channel_mapping[ca][i] = 0xf | (i << 4); } + if (non_pcm) { + for (i = 0; i < channel_allocations[order].channels; i++) + non_pcm_mapping[i] = i | (i << 4); + for (; i < 8; i++) + non_pcm_mapping[i] = 0xf | (i << 4); + } + for (i = 0; i < 8; i++) { err = snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); + non_pcm ? non_pcm_mapping[i] : hdmi_channel_mapping[ca][i]); if (err) { snd_printdd(KERN_NOTICE "HDMI: channel mapping failed\n"); @@ -699,15 +712,27 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - struct snd_pcm_substream *substream) + hda_nid_t cvt_nid, struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + struct hdmi_spec_per_cvt *per_cvt; + struct hda_spdif_out *spdif; hda_nid_t pin_nid = per_pin->pin_nid; int channels = substream->runtime->channels; struct hdmi_eld *eld; int ca; + int cvt_idx; union audio_infoframe ai; + bool non_pcm = false; + + cvt_idx = cvt_nid_to_cvt_index(spec, cvt_nid); + per_cvt = &spec->cvts[cvt_idx]; + + mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); + non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); + mutex_unlock(&codec->spdif_mutex); eld = &spec->pins[pin_idx].sink_eld; if (!eld->monitor_present) @@ -750,12 +775,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, ca); + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, non_pcm, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } + + per_cvt->non_pcm = non_pcm; } @@ -1077,6 +1104,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) per_cvt->cvt_nid = cvt_nid; per_cvt->channels_min = 2; + per_cvt->non_pcm = false; if (chans <= 16) per_cvt->channels_max = chans; @@ -1164,7 +1192,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- cgit v1.2.3 From 2d7e887cbba8272a46c42d228d3ffd9a2de04168 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 6 Sep 2012 10:02:38 +0800 Subject: ALSA: HDMI - Setup channel mapping for non_pcm audio For HBR stream test, use straight channel mapping way. when switched back to "speaker-test -c8", even the audio infoframe is up-to-date, there should be correct channel mapping setup. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a6835bd9d9b5..5361298be4d0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -780,6 +780,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); + } else { + /* For non-pcm audio switch, setup new channel mapping + * accordingly */ + if (per_cvt->non_pcm != non_pcm) + hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, non_pcm, ca); } per_cvt->non_pcm = non_pcm; -- cgit v1.2.3 From 298efee7f5a20a32c9ebfa0f7469d87d84998ba1 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 6 Sep 2012 11:17:58 +0200 Subject: ALSA: hda - fix control names for multiple speaker out on IDT/STAC For multiple speaker outs, the names were previously "Speaker,0", "Speaker,1", "Center"/"LFE", "Speaker,3". This is inconsistent, confusing, and is not picked up correctly by PulseAudio. Instead use "Front", "Surround", "Center"/"LFE", "Side" which is more standard. BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1046734 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 9db305633efe..c16d0f822b5b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3226,9 +3226,12 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - idx = i; - break; + if (num_outs <= 1) { + name = "Speaker"; + idx = i; + break; + } + /* Fall through in case of multi speaker outs */ default: name = chname[i]; idx = 0; -- cgit v1.2.3 From 6d97c09c64974ab41708e2d38394928ec0eeb2f0 Mon Sep 17 00:00:00 2001 From: Gaëtan Carlier Date: Thu, 6 Sep 2012 09:40:12 +0200 Subject: ASoC: imx-mc13783: use defines instead of numerical address of register MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This uses already defined name of registers and makes code more readable. Signed-off-by: Gaëtan Carlier Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 58 +++++++++++++++++++++++++--------------------- 1 file changed, 31 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8f726c063f42..bbfa5535cdd3 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -426,16 +426,16 @@ static int mc13783_set_tdm_slot_sync(struct snd_soc_dai *dai, } static const struct snd_kcontrol_new mc1l_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 7, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 7, 1, 0); static const struct snd_kcontrol_new mc1r_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 5, 1, 0); static const struct snd_kcontrol_new mc2_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 9, 1, 0); static const struct snd_kcontrol_new atx_amp_ctl = - SOC_DAPM_SINGLE("Switch", 38, 11, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_TX, 11, 1, 0); /* Virtual mux. The chip does the input selection automatically @@ -461,22 +461,22 @@ static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); static const struct snd_kcontrol_new samp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 3, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); static const struct snd_kcontrol_new lamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 5, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); static const struct snd_kcontrol_new hlamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 10, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 10, 1, 0); static const struct snd_kcontrol_new hramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 9, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 9, 1, 0); static const struct snd_kcontrol_new llamp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 16, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 16, 1, 0); static const struct snd_kcontrol_new lramp_ctl = - SOC_DAPM_SINGLE("Switch", 36, 15, 1, 0); + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 15, 1, 0); static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { /* Input */ @@ -487,13 +487,13 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RXINL"), SND_SOC_DAPM_INPUT("TXIN"), - SND_SOC_DAPM_SUPPLY("MC1 Bias", 38, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("MC2 Bias", 38, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC1 Bias", MC13783_AUDIO_TX, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MC2 Bias", MC13783_AUDIO_TX, 1, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("MC1L Amp", 38, 7, 0, &mc1l_amp_ctl), - SND_SOC_DAPM_SWITCH("MC1R Amp", 38, 5, 0, &mc1r_amp_ctl), - SND_SOC_DAPM_SWITCH("MC2 Amp", 38, 9, 0, &mc2_amp_ctl), - SND_SOC_DAPM_SWITCH("TXIN Amp", 38, 11, 0, &atx_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1L Amp", MC13783_AUDIO_TX, 7, 0, &mc1l_amp_ctl), + SND_SOC_DAPM_SWITCH("MC1R Amp", MC13783_AUDIO_TX, 5, 0, &mc1r_amp_ctl), + SND_SOC_DAPM_SWITCH("MC2 Amp", MC13783_AUDIO_TX, 9, 0, &mc2_amp_ctl), + SND_SOC_DAPM_SWITCH("TXIN Amp", MC13783_AUDIO_TX, 11, 0, &atx_amp_ctl), SND_SOC_DAPM_VIRT_MUX("PGA Left Input Mux", SND_SOC_NOPM, 0, 0, &left_input_mux), @@ -503,12 +503,12 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", 40, 11, 0), - SND_SOC_DAPM_SUPPLY("ADC_Reset", 40, 15, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), + SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), /* Output */ - SND_SOC_DAPM_SUPPLY("DAC_E", 41, 11, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC_Reset", 41, 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("RXOUTL"), SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), @@ -516,14 +516,18 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), - SND_SOC_DAPM_SWITCH("Speaker Amp", 36, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Left", 36, 10, 0, &hlamp_ctl), - SND_SOC_DAPM_SWITCH("Headset Amp Right", 36, 9, 0, &hramp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Left", 36, 16, 0, &llamp_ctl), - SND_SOC_DAPM_SWITCH("Line out Amp Right", 36, 15, 0, &lramp_ctl), - SND_SOC_DAPM_DAC("DAC", "Playback", 36, 22, 0), - SND_SOC_DAPM_PGA("DAC PGA", 37, 5, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, + &hlamp_ctl), + SND_SOC_DAPM_SWITCH("Headset Amp Right", MC13783_AUDIO_RX0, 9, 0, + &hramp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Left", MC13783_AUDIO_RX0, 16, 0, + &llamp_ctl), + SND_SOC_DAPM_SWITCH("Line out Amp Right", MC13783_AUDIO_RX0, 15, 0, + &lramp_ctl), + SND_SOC_DAPM_DAC("DAC", "Playback", MC13783_AUDIO_RX0, 22, 0), + SND_SOC_DAPM_PGA("DAC PGA", MC13783_AUDIO_RX1, 5, 0, NULL, 0), }; static struct snd_soc_dapm_route mc13783_routes[] = { -- cgit v1.2.3 From 1a6003b525b9815ccd9d23ae3b70372a17d0bce2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2012 17:42:08 +0200 Subject: ALSA: hda - Move non-PCM check to per_pin in patch_hdmi.c Recently the check for non-PCM stream state was added to the generic HDMI driver code. But this check should be done rather to each pin instead of each converter. Otherwise when a different converter is assigned at the next open, the audio infoframe can be inconsistent with the setup using the previous converter. For fixing this issue, this patch moves the state of the current non-PCM status from per_cvt to per_pin. (In addition an unused argument cvt_nid is stripped from hdmi_setup_channel_mapping()) Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 46 +++++++++++++++++++++++++--------------------- 1 file changed, 25 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5361298be4d0..333d5338c15c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -61,7 +61,6 @@ struct hdmi_spec_per_cvt { u32 rates; u64 formats; unsigned int maxbps; - bool non_pcm; }; struct hdmi_spec_per_pin { @@ -73,6 +72,7 @@ struct hdmi_spec_per_pin { struct hdmi_eld sink_eld; struct delayed_work work; int repoll_count; + bool non_pcm; }; struct hdmi_spec { @@ -550,7 +550,6 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, - hda_nid_t cvt_nid, bool non_pcm, int ca) { @@ -712,27 +711,16 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, } static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, - hda_nid_t cvt_nid, struct snd_pcm_substream *substream) + bool non_pcm, + struct snd_pcm_substream *substream) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; - struct hdmi_spec_per_cvt *per_cvt; - struct hda_spdif_out *spdif; hda_nid_t pin_nid = per_pin->pin_nid; int channels = substream->runtime->channels; struct hdmi_eld *eld; int ca; - int cvt_idx; union audio_infoframe ai; - bool non_pcm = false; - - cvt_idx = cvt_nid_to_cvt_index(spec, cvt_nid); - per_cvt = &spec->cvts[cvt_idx]; - - mutex_lock(&codec->spdif_mutex); - spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); - non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); - mutex_unlock(&codec->spdif_mutex); eld = &spec->pins[pin_idx].sink_eld; if (!eld->monitor_present) @@ -775,7 +763,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, non_pcm, ca); + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -783,11 +771,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, } else { /* For non-pcm audio switch, setup new channel mapping * accordingly */ - if (per_cvt->non_pcm != non_pcm) - hdmi_setup_channel_mapping(codec, pin_nid, cvt_nid, non_pcm, ca); + if (per_pin->non_pcm != non_pcm) + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca); } - per_cvt->non_pcm = non_pcm; + per_pin->non_pcm = non_pcm; } @@ -1080,6 +1068,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) per_pin = &spec->pins[pin_idx]; per_pin->pin_nid = pin_nid; + per_pin->non_pcm = false; err = hdmi_read_pin_conn(codec, pin_idx); if (err < 0) @@ -1109,7 +1098,6 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) per_cvt->cvt_nid = cvt_nid; per_cvt->channels_min = 2; - per_cvt->non_pcm = false; if (chans <= 16) per_cvt->channels_max = chans; @@ -1179,6 +1167,19 @@ static char *get_hdmi_pcm_name(int idx) return &names[idx][0]; } +static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) +{ + struct hda_spdif_out *spdif; + bool non_pcm; + + mutex_lock(&codec->spdif_mutex); + spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); + non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); + mutex_unlock(&codec->spdif_mutex); + return non_pcm; +} + + /* * HDMI callbacks */ @@ -1194,10 +1195,13 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; int pinctl; + bool non_pcm; + + non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); - hdmi_setup_audio_infoframe(codec, pin_idx, cvt_nid, substream); + hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); pinctl = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); -- cgit v1.2.3 From 2d3391ec0ecca37efb6bc995906292f47522b471 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jul 2012 18:27:00 +0200 Subject: ALSA: PCM: channel mapping API implementation This patch implements the basic data types for the standard channel mapping API handling. - The definitions of the channel positions and the new TLV types are added in sound/asound.h and sound/tlv.h, so that they can be referred from user-space. - Introduced a new helper function snd_pcm_add_chmap_ctls() to create control elements representing the channel maps for each PCM (sub)stream. - Some standard pre-defined channel maps are provided for convenience. Signed-off-by: Takashi Iwai --- include/sound/asound.h | 30 +++++++ include/sound/pcm.h | 48 +++++++++++ include/sound/tlv.h | 8 ++ sound/core/pcm.c | 4 + sound/core/pcm_lib.c | 215 +++++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 305 insertions(+) (limited to 'sound') diff --git a/include/sound/asound.h b/include/sound/asound.h index 0876a1e76aef..376e75632e07 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -472,6 +472,36 @@ enum { SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, }; +/* channel positions */ +enum { + SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_FL, /* front left */ + SNDRV_CHMAP_FC, /* front center */ + SNDRV_CHMAP_FR, /* front right */ + SNDRV_CHMAP_FLC, /* front left center */ + SNDRV_CHMAP_FRC, /* front right center */ + SNDRV_CHMAP_RL, /* rear left */ + SNDRV_CHMAP_RC, /* rear center */ + SNDRV_CHMAP_RR, /* rear right */ + SNDRV_CHMAP_RLC, /* rear left center */ + SNDRV_CHMAP_RRC, /* rear right center */ + SNDRV_CHMAP_SL, /* side left */ + SNDRV_CHMAP_SR, /* side right */ + SNDRV_CHMAP_LFE, /* LFE */ + SNDRV_CHMAP_FLW, /* front left wide */ + SNDRV_CHMAP_FRW, /* front right wide */ + SNDRV_CHMAP_FLH, /* front left high */ + SNDRV_CHMAP_FCH, /* front center high */ + SNDRV_CHMAP_FRH, /* front right high */ + SNDRV_CHMAP_TC, /* top center */ + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, +}; + +#define SNDRV_CHMAP_POSITION_MASK 0xffff +#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16) +#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16) + #define SNDRV_PCM_IOCTL_PVERSION _IOR('A', 0x00, int) #define SNDRV_PCM_IOCTL_INFO _IOR('A', 0x01, struct snd_pcm_info) #define SNDRV_PCM_IOCTL_TSTAMP _IOW('A', 0x02, int) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab1e711..669c85a7fb03 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -437,6 +437,7 @@ struct snd_pcm_str { struct snd_info_entry *proc_xrun_debug_entry; #endif #endif + struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */ }; struct snd_pcm { @@ -1086,4 +1087,51 @@ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream return "Capture"; } +/* + * PCM channel-mapping control API + */ +/* array element of channel maps */ +struct snd_pcm_chmap_elem { + unsigned char channels; + unsigned char map[15]; +}; + +/* channel map information; retrieved via snd_kcontrol_chip() */ +struct snd_pcm_chmap { + struct snd_pcm *pcm; /* assigned PCM instance */ + int stream; /* PLAYBACK or CAPTURE */ + struct snd_kcontrol *kctl; + const struct snd_pcm_chmap_elem *chmap; + unsigned int max_channels; + unsigned int channel_mask; /* optional: active channels bitmask */ + void *private_data; /* optional: private data pointer */ +}; + +/* get the PCM substream assigned to the given chmap info */ +static inline struct snd_pcm_substream * +snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx) +{ + struct snd_pcm_substream *s; + for (s = info->pcm->streams[info->stream].substream; s; s = s->next) + if (s->number == idx) + return s; + return NULL; +} + +/* ALSA-standard channel maps (RL/RR prior to C/LFE) */ +extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[]; +/* Other world's standard channel maps (C/LFE prior to RL/RR) */ +extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[]; + +/* bit masks to be passed to snd_pcm_chmap.channel_mask field */ +#define SND_PCM_CHMAP_MASK_24 ((1U << 2) | (1U << 4)) +#define SND_PCM_CHMAP_MASK_246 (SND_PCM_CHMAP_MASK_24 | (1U << 6)) +#define SND_PCM_CHMAP_MASK_2468 (SND_PCM_CHMAP_MASK_246 | (1U << 8)) + +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret); + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/tlv.h b/include/sound/tlv.h index a64d8fe3f855..28c65e1ada21 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -86,4 +86,12 @@ #define TLV_DB_GAIN_MUTE -9999999 +/* + * channel-mapping TLV items + * TLV length must match with num_channels + */ +#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */ +#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */ +#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */ + #endif /* __SOUND_TLV_H */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 1a3070b4e5b5..f2991940b271 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1105,6 +1105,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) break; } snd_unregister_device(devtype, pcm->card, pcm->device); + if (pcm->streams[cidx].chmap_kctl) { + snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); + pcm->streams[cidx].chmap_kctl = NULL; + } } unlock: mutex_unlock(®ister_mutex); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae671923393..565102705eda 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -2302,3 +2303,217 @@ snd_pcm_sframes_t snd_pcm_lib_readv(struct snd_pcm_substream *substream, } EXPORT_SYMBOL(snd_pcm_lib_readv); + +/* + * standard channel mapping helpers + */ + +/* default channel maps for multi-channel playbacks, up to 8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_FC } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_std_chmaps); + +/* alternative channel maps with CLFE <-> surround swapped for 6/8 channels */ +const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_FC } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 8, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; +EXPORT_SYMBOL_GPL(snd_pcm_alt_chmaps); + +static bool valid_chmap_channels(const struct snd_pcm_chmap *info, int ch) +{ + if (ch > info->max_channels) + return false; + return !info->channel_mask || (info->channel_mask & (1U << ch)); +} + +static int pcm_chmap_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 0; + uinfo->count = info->max_channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_CHMAP_LAST; + return 0; +} + +/* get callback for channel map ctl element + * stores the channel position firstly matching with the current channels + */ +static int pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct snd_pcm_substream *substream; + const struct snd_pcm_chmap_elem *map; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + substream = snd_pcm_chmap_substream(info, idx); + if (!substream) + return -ENODEV; + memset(ucontrol->value.integer.value, 0, + sizeof(ucontrol->value.integer.value)); + if (!substream->runtime) + return 0; /* no channels set */ + for (map = info->chmap; map->channels; map++) { + int i; + if (map->channels == substream->runtime->channels && + valid_chmap_channels(info, map->channels)) { + for (i = 0; i < map->channels; i++) + ucontrol->value.integer.value[i] = map->map[i]; + return 0; + } + } + return -EINVAL; +} + +/* tlv callback for channel map ctl element + * expands the pre-defined channel maps in a form of TLV + */ +static int pcm_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + const struct snd_pcm_chmap_elem *map; + unsigned int __user *dst; + int c, count = 0; + + if (snd_BUG_ON(!info->chmap)) + return -EINVAL; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CONTAINER, tlv)) + return -EFAULT; + size -= 8; + dst = tlv + 2; + for (map = info->chmap; map->channels; map++) { + int chs_bytes = map->channels * 4; + if (!valid_chmap_channels(info, map->channels)) + continue; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CHMAP_FIXED, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + size -= chs_bytes; + count += chs_bytes; + for (c = 0; c < map->channels; c++) { + if (put_user(map->map[c], dst)) + return -EFAULT; + dst++; + } + } + if (put_user(count, tlv + 1)) + return -EFAULT; + return 0; +} + +static void pcm_chmap_ctl_private_free(struct snd_kcontrol *kcontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + info->pcm->streams[info->stream].chmap_kctl = NULL; + kfree(info); +} + +/** + * snd_pcm_add_chmap_ctls - create channel-mapping control elements + * @pcm: the assigned PCM instance + * @stream: stream direction + * @chmap: channel map elements (for query) + * @max_channels: the max number of channels for the stream + * @private_value: the value passed to each kcontrol's private_value field + * @info_ret: store struct snd_pcm_chmap instance if non-NULL + * + * Create channel-mapping control elements assigned to the given PCM stream(s). + * Returns zero if succeed, or a negative error value. + */ +int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, + const struct snd_pcm_chmap_elem *chmap, + int max_channels, + unsigned long private_value, + struct snd_pcm_chmap **info_ret) +{ + struct snd_pcm_chmap *info; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE | /* no notification */ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = pcm_chmap_ctl_info, + .get = pcm_chmap_ctl_get, + .tlv.c = pcm_chmap_ctl_tlv, + }; + int err; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + info->pcm = pcm; + info->stream = stream; + info->chmap = chmap; + info->max_channels = max_channels; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + knew.name = "Playback Channel Map"; + else + knew.name = "Capture Channel Map"; + knew.device = pcm->device; + knew.count = pcm->streams[stream].substream_count; + knew.private_value = private_value; + info->kctl = snd_ctl_new1(&knew, info); + if (!info->kctl) { + kfree(info); + return -ENOMEM; + } + info->kctl->private_free = pcm_chmap_ctl_private_free; + err = snd_ctl_add(pcm->card, info->kctl); + if (err < 0) + return err; + pcm->streams[stream].chmap_kctl = info->kctl; + if (info_ret) + *info_ret = info; + return 0; +} +EXPORT_SYMBOL_GPL(snd_pcm_add_chmap_ctls); -- cgit v1.2.3 From a8d372f171db9b90a64778fbcd9237c9bc256e06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jul 2012 13:47:07 +0200 Subject: ALSA: control: Fix missing VOLATILE flag at creating controls The SNDRV_CTL_ELEM_ACCESS_VOLATILE bit flag wasn't properly inherited at creating control elements via snd_ctl_new1(). Signed-off-by: Takashi Iwai --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 2487a6bb1c54..7e86a5b9f3b5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -246,6 +246,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| + SNDRV_CTL_ELEM_ACCESS_VOLATILE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| -- cgit v1.2.3 From 9c9a5175e65b2667001e6a3b6dedddebeee82aa2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 31 Jul 2012 11:35:35 +0200 Subject: ALSA: hda - Add standard channel maps Although HD-audio allows pair-wise channel configurations, only the fixed channel positions are used in this version. In future, this can be changed and allow user to modify the channel positions. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 36 ++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_realtek.c | 2 ++ 3 files changed, 39 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ff97cf3e8bad..c4c62edca556 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3688,6 +3688,36 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) } EXPORT_SYMBOL_HDA(snd_hda_build_controls); +/* + * add standard channel maps if not specified + */ +static int add_std_chmaps(struct hda_codec *codec) +{ + int i, str, err; + + for (i = 0; i < codec->num_pcms; i++) { + for (str = 0; str < 2; str++) { + struct snd_pcm *pcm = codec->pcm_info[i].pcm; + struct hda_pcm_stream *hinfo = + &codec->pcm_info[i].stream[str]; + struct snd_pcm_chmap *chmap; + + if (codec->pcm_info[i].own_chmap) + continue; + if (!pcm || !hinfo->substreams) + continue; + err = snd_pcm_add_chmap_ctls(pcm, str, + snd_pcm_std_chmaps, + hinfo->channels_max, + 0, &chmap); + if (err < 0) + return err; + chmap->channel_mask = SND_PCM_CHMAP_MASK_2468; + } + } + return 0; +} + int snd_hda_codec_build_controls(struct hda_codec *codec) { int err = 0; @@ -3699,6 +3729,12 @@ int snd_hda_codec_build_controls(struct hda_codec *codec) err = codec->patch_ops.build_controls(codec); if (err < 0) return err; + + /* we create chmaps here instead of build_pcms */ + err = add_std_chmaps(codec); + if (err < 0) + return err; + snd_hda_jack_report_sync(codec); /* call at the last init point */ return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8fd9f636cfe1..ed3de8d151c8 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -776,6 +776,7 @@ struct hda_pcm { unsigned int pcm_type; /* HDA_PCM_TYPE_XXX */ int device; /* device number to assign */ struct snd_pcm *pcm; /* assigned PCM instance */ + bool own_chmap; /* codec driver provides own channel maps */ }; /* codec information */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 613499932b2b..1907ddad3840 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2287,6 +2287,8 @@ static int alc_build_pcms(struct hda_codec *codec) p = &alc_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.max_channels; } if (spec->adc_nids) { p = spec->stream_analog_capture; -- cgit v1.2.3 From d45e6889ee69456a4d5b1bbb32252f460cd48fa9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 31 Jul 2012 11:36:00 +0200 Subject: ALSA: hda - Provide the proper channel mapping for generic HDMI driver ... instead of the standard fixed channel maps. The generic HDMI is based on the audio infoframe, and its configuration can be selected via CA bits. Thus we need a translation between the CA index and the verbose channel map list. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 308 ++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 303 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 333d5338c15c..4a5d24fe8adf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -35,6 +35,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_jack.h" @@ -73,6 +74,8 @@ struct hdmi_spec_per_pin { struct delayed_work work; int repoll_count; bool non_pcm; + bool chmap_set; /* channel-map override by ALSA API? */ + unsigned char chmap[8]; /* ALSA API channel-map */ }; struct hdmi_spec { @@ -82,6 +85,7 @@ struct hdmi_spec { int num_pins; struct hdmi_spec_per_pin pins[MAX_HDMI_PINS]; struct hda_pcm pcm_rec[MAX_HDMI_PINS]; + unsigned int channels_max; /* max over all cvts */ /* * Non-generic ATI/NVIDIA specific @@ -548,7 +552,7 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, } -static void hdmi_setup_channel_mapping(struct hda_codec *codec, +static void hdmi_std_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, bool non_pcm, int ca) @@ -588,6 +592,136 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, hdmi_debug_channel_mapping(codec, pin_nid); } +struct channel_map_table { + unsigned char map; /* ALSA API channel map position */ + unsigned char cea_slot; /* CEA slot value */ + int spk_mask; /* speaker position bit mask */ +}; + +static struct channel_map_table map_tables[] = { + { SNDRV_CHMAP_FL, 0x00, FL }, + { SNDRV_CHMAP_FR, 0x01, FR }, + { SNDRV_CHMAP_RL, 0x04, RL }, + { SNDRV_CHMAP_RR, 0x05, RR }, + { SNDRV_CHMAP_LFE, 0x02, LFE }, + { SNDRV_CHMAP_FC, 0x03, FC }, + { SNDRV_CHMAP_RLC, 0x06, RLC }, + { SNDRV_CHMAP_RRC, 0x07, RRC }, + {} /* terminator */ +}; + +/* from ALSA API channel position to speaker bit mask */ +static int to_spk_mask(unsigned char c) +{ + struct channel_map_table *t = map_tables; + for (; t->map; t++) { + if (t->map == c) + return t->spk_mask; + } + return 0; +} + +/* from ALSA API channel position to CEA slot */ +static int to_cea_slot(unsigned char c) +{ + struct channel_map_table *t = map_tables; + for (; t->map; t++) { + if (t->map == c) + return t->cea_slot; + } + return 0x0f; +} + +/* from CEA slot to ALSA API channel position */ +static int from_cea_slot(unsigned char c) +{ + struct channel_map_table *t = map_tables; + for (; t->map; t++) { + if (t->cea_slot == c) + return t->map; + } + return 0; +} + +/* from speaker bit mask to ALSA API channel position */ +static int spk_to_chmap(int spk) +{ + struct channel_map_table *t = map_tables; + for (; t->map; t++) { + if (t->spk_mask == spk) + return t->map; + } + return 0; +} + +/* get the CA index corresponding to the given ALSA API channel map */ +static int hdmi_manual_channel_allocation(int chs, unsigned char *map) +{ + int i, spks = 0, spk_mask = 0; + + for (i = 0; i < chs; i++) { + int mask = to_spk_mask(map[i]); + if (mask) { + spk_mask |= mask; + spks++; + } + } + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if ((chs == channel_allocations[i].channels || + spks == channel_allocations[i].channels) && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) + return channel_allocations[i].ca_index; + } + return -1; +} + +/* set up the channel slots for the given ALSA API channel map */ +static int hdmi_manual_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, + int chs, unsigned char *map) +{ + int i; + for (i = 0; i < 8; i++) { + int val, err; + if (i < chs) + val = to_cea_slot(map[i]); + else + val = 0xf; + val |= (i << 4); + err = snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, val); + if (err) + return -EINVAL; + } + return 0; +} + +/* store ALSA API channel map from the current default map */ +static void hdmi_setup_fake_chmap(unsigned char *map, int ca) +{ + int i; + for (i = 0; i < 8; i++) { + if (i < channel_allocations[ca].channels) + map[i] = from_cea_slot((hdmi_channel_mapping[ca][i] >> 4) & 0x0f); + else + map[i] = 0; + } +} + +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, bool non_pcm, int ca, + int channels, unsigned char *map) +{ + if (!non_pcm && map) { + hdmi_manual_setup_channel_mapping(codec, pin_nid, + channels, map); + } else { + hdmi_std_setup_channel_mapping(codec, pin_nid, non_pcm, ca); + hdmi_setup_fake_chmap(map, ca); + } +} /* * Audio InfoFrame routines @@ -726,7 +860,12 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, if (!eld->monitor_present) return; - ca = hdmi_channel_allocation(eld, channels); + if (!non_pcm && per_pin->chmap_set) + ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); + else + ca = hdmi_channel_allocation(eld, channels); + if (ca < 0) + ca = 0; memset(&ai, 0, sizeof(ai)); if (eld->conn_type == 0) { /* HDMI */ @@ -763,7 +902,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca); + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -772,7 +912,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, /* For non-pcm audio switch, setup new channel mapping * accordingly */ if (per_pin->non_pcm != non_pcm) - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca); + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap); } per_pin->non_pcm = non_pcm; @@ -1098,8 +1239,11 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) per_cvt->cvt_nid = cvt_nid; per_cvt->channels_min = 2; - if (chans <= 16) + if (chans <= 16) { per_cvt->channels_max = chans; + if (chans > spec->channels_max) + spec->channels_max = chans; + } err = snd_hda_query_supported_pcm(codec, cvt_nid, &per_cvt->rates, @@ -1250,7 +1394,10 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl & ~PIN_OUT); snd_hda_spdif_ctls_unassign(codec, pin_idx); + per_pin->chmap_set = false; + memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); } + return 0; } @@ -1261,6 +1408,135 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +/* + * ALSA API channel-map control callbacks + */ +static int hdmi_chmap_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct hda_codec *codec = info->private_data; + struct hdmi_spec *spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = spec->channels_max; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_CHMAP_LAST; + return 0; +} + +static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct hda_codec *codec = info->private_data; + struct hdmi_spec *spec = codec->spec; + const unsigned int valid_mask = + FL | FR | RL | RR | LFE | FC | RLC | RRC; + unsigned int __user *dst; + int chs, count = 0; + + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CONTAINER, tlv)) + return -EFAULT; + size -= 8; + dst = tlv + 2; + for (chs = 2; chs <= spec->channels_max; chs += 2) { + int i, c; + struct cea_channel_speaker_allocation *cap; + cap = channel_allocations; + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { + int chs_bytes = chs * 4; + if (cap->channels != chs) + continue; + if (cap->spk_mask & ~valid_mask) + continue; + if (size < 8) + return -ENOMEM; + if (put_user(SNDRV_CTL_TLVT_CHMAP_VAR, dst) || + put_user(chs_bytes, dst + 1)) + return -EFAULT; + dst += 2; + size -= 8; + count += 8; + if (size < chs_bytes) + return -ENOMEM; + size -= chs_bytes; + count += chs_bytes; + for (c = 7; c >= 0; c--) { + int spk = cap->speakers[c]; + if (!spk) + continue; + if (put_user(spk_to_chmap(spk), dst)) + return -EFAULT; + dst++; + } + } + } + if (put_user(count, tlv + 1)) + return -EFAULT; + return 0; +} + +static int hdmi_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct hda_codec *codec = info->private_data; + struct hdmi_spec *spec = codec->spec; + int pin_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + int i; + + for (i = 0; i < ARRAY_SIZE(per_pin->chmap); i++) + ucontrol->value.integer.value[i] = per_pin->chmap[i]; + return 0; +} + +static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct hda_codec *codec = info->private_data; + struct hdmi_spec *spec = codec->spec; + int pin_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx]; + unsigned int ctl_idx; + struct snd_pcm_substream *substream; + unsigned char chmap[8]; + int i, ca, prepared = 0; + + ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + substream = snd_pcm_chmap_substream(info, ctl_idx); + if (!substream || !substream->runtime) + return -EBADFD; + switch (substream->runtime->status->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + break; + case SNDRV_PCM_STATE_PREPARED: + prepared = 1; + break; + default: + return -EBUSY; + } + memset(chmap, 0, sizeof(chmap)); + for (i = 0; i < ARRAY_SIZE(chmap); i++) + chmap[i] = ucontrol->value.integer.value[i]; + if (!memcmp(chmap, per_pin->chmap, sizeof(chmap))) + return 0; + ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); + if (ca < 0) + return -EINVAL; + per_pin->chmap_set = true; + memcpy(per_pin->chmap, chmap, sizeof(chmap)); + if (prepared) + hdmi_setup_audio_infoframe(codec, pin_idx, per_pin->non_pcm, + substream); + + return 0; +} + static int generic_hdmi_build_pcms(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1273,6 +1549,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) info = &spec->pcm_rec[pin_idx]; info->name = get_hdmi_pcm_name(pin_idx); info->pcm_type = HDA_PCM_TYPE_HDMI; + info->own_chmap = true; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; pstr->substreams = 1; @@ -1330,6 +1607,27 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) hdmi_present_sense(per_pin, 0); } + /* add channel maps */ + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct snd_pcm_chmap *chmap; + struct snd_kcontrol *kctl; + int i; + err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, + SNDRV_PCM_STREAM_PLAYBACK, + NULL, 0, pin_idx, &chmap); + if (err < 0) + return err; + /* override handlers */ + chmap->private_data = codec; + kctl = chmap->kctl; + for (i = 0; i < kctl->count; i++) + kctl->vd[i].access |= SNDRV_CTL_ELEM_ACCESS_WRITE; + kctl->info = hdmi_chmap_ctl_info; + kctl->get = hdmi_chmap_ctl_get; + kctl->put = hdmi_chmap_ctl_put; + kctl->tlv.c = hdmi_chmap_ctl_tlv; + } + return 0; } -- cgit v1.2.3 From 53775b0d0cb91ab217c9853efddc51597b58bbff Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Aug 2012 12:17:41 +0200 Subject: ALSA: hda - Fix channel maps for Nvidia 7x 8ch HDMI codecs Some old Nvidia HDMI codecs with 8ch support only 2/8 or 2/6/8 channels and with the fixed CLFE-first map. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4a5d24fe8adf..2c53ea889da9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2155,6 +2155,43 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) return 0; } +static int nvhdmi_7x_8ch_build_pcms(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int err = simple_playback_build_pcms(codec); + spec->pcm_rec[0].own_chmap = true; + return err; +} + +static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_chmap *chmap; + int err; + + err = simple_playback_build_controls(codec); + if (err < 0) + return err; + + /* add channel maps */ + err = snd_pcm_add_chmap_ctls(spec->pcm_rec[0].pcm, + SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, 8, 0, &chmap); + if (err < 0) + return err; + switch (codec->preset->id) { + case 0x10de0002: + case 0x10de0003: + case 0x10de0005: + case 0x10de0006: + chmap->channel_mask = (1U << 2) | (1U << 8); + break; + case 0x10de0007: + chmap->channel_mask = (1U << 2) | (1U << 6) | (1U << 8); + } + return 0; +} + static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -2165,6 +2202,8 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) spec->multiout.max_channels = 8; spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x; codec->patch_ops.init = nvhdmi_7x_init_8ch; + codec->patch_ops.build_pcms = nvhdmi_7x_8ch_build_pcms; + codec->patch_ops.build_controls = nvhdmi_7x_8ch_build_controls; /* Initialize the audio infoframe channel mask and checksum to something * valid */ -- cgit v1.2.3 From 833a493b7ed2eb8f9059338a0ebf06bebbb6ae93 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 17:59:36 +0200 Subject: ALSA: ac97: Implement channel map workaround for ALC650 ALC650 has a channel swap option between surround and CLFE channels, so we need to tweak the channel maps dynamically depending on the register bit. Now struct snd_ac97 can contain chmap pointers for playback and capture. The driver may store these and let ac97 driver changing the channel mapping dynamically. Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 3 +++ sound/pci/ac97/ac97_patch.c | 24 +++++++++++++++++++++++- 2 files changed, 26 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 02cbb50225bb..4458b87649ff 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -422,6 +422,7 @@ */ struct snd_ac97; +struct snd_pcm_chmap; struct snd_ac97_build_ops { int (*build_3d) (struct snd_ac97 *ac97); @@ -528,6 +529,8 @@ struct snd_ac97 { struct delayed_work power_work; #endif struct device dev; + + struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */ }; #define to_ac97_t(d) container_of(d, struct snd_ac97, dev) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a872d0a82976..66a3bc95fb84 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -2595,6 +2595,21 @@ static void alc650_update_jacks(struct snd_ac97 *ac97) shared ? 0 : 0x100); } +static int alc650_swap_surround_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); + struct snd_pcm_chmap *map = ac97->chmaps[SNDRV_PCM_STREAM_PLAYBACK]; + + if (map) { + if (ucontrol->value.integer.value[0]) + map->chmap = snd_pcm_std_chmaps; + else + map->chmap = snd_pcm_alt_chmaps; + } + return snd_ac97_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new snd_ac97_controls_alc650[] = { AC97_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0), AC97_SINGLE("Surround Down Mix", AC97_ALC650_MULTICH, 1, 1, 0), @@ -2608,7 +2623,14 @@ static const struct snd_kcontrol_new snd_ac97_controls_alc650[] = { /* 9: Line-In/Surround share */ /* 10: Mic/CLFE share */ /* 11-13: in IEC958 controls */ - AC97_SINGLE("Swap Surround Slot", AC97_ALC650_MULTICH, 14, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Swap Surround Slot", + .info = snd_ac97_info_volsw, + .get = snd_ac97_get_volsw, + .put = alc650_swap_surround_put, + .private_value = AC97_SINGLE_VALUE(AC97_ALC650_MULTICH, 14, 1, 0), + }, #if 0 /* always set in patch_alc650 */ AC97_SINGLE("IEC958 Input Clock Enable", AC97_ALC650_CLOCK, 0, 1, 0), AC97_SINGLE("IEC958 Input Pin Enable", AC97_ALC650_CLOCK, 1, 1, 0), -- cgit v1.2.3 From e36e3b86c78cee9c7435eb33e0ef8a788193e812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 18:01:40 +0200 Subject: ALSA: Implement channel maps for standard onboard AC97 drivers Just set the channel maps depending on the hardware availability. Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 9 +++++++++ sound/pci/fm801.c | 7 +++++++ sound/pci/intel8x0.c | 20 ++++++++++++++++++++ sound/pci/via82xx.c | 23 +++++++++++++++++++++++ 4 files changed, 59 insertions(+) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index c744df5bb1c6..368df8b0853e 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1250,6 +1250,7 @@ static struct atiixp_dma_ops snd_atiixp_spdif_dma_ops = { static int __devinit snd_atiixp_pcm_new(struct atiixp *chip) { struct snd_pcm *pcm; + struct snd_pcm_chmap *chmap; struct snd_ac97_bus *pbus = chip->ac97_bus; int err, i, num_pcms; @@ -1293,6 +1294,14 @@ static int __devinit snd_atiixp_pcm_new(struct atiixp *chip) snd_dma_pci_data(chip->pci), 64*1024, 128*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, chip->max_channels, 0, + &chmap); + if (err < 0) + return err; + chmap->channel_mask = SND_PCM_CHMAP_MASK_2468; + chip->ac97[0]->chmaps[SNDRV_PCM_STREAM_PLAYBACK] = chmap; + /* no SPDIF support on codec? */ if (chip->pcms[ATI_PCM_SPDIF] && ! chip->pcms[ATI_PCM_SPDIF]->rates) return 0; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ce3e548de41d..cc2e91d15538 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -711,6 +711,13 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc snd_dma_pci_data(chip->pci), chip->multichannel ? 128*1024 : 64*1024, 128*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, + chip->multichannel ? 6 : 2, 0, + NULL); + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; return 0; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 5c4115289a9a..848102e5d864 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1541,6 +1541,26 @@ static int __devinit snd_intel8x0_pcm1(struct intel8x0 *chip, int device, snd_dma_pci_data(chip->pci), rec->prealloc_size, rec->prealloc_max_size); + if (rec->ac97_idx == ICHD_PCMOUT && rec->playback_ops) { + struct snd_pcm_chmap *chmap; + int chs = 2; + if (rec->ac97_idx == ICHD_PCMOUT) { + if (chip->multi8) + chs = 8; + else if (chip->multi6) + chs = 6; + else if (chip->multi4) + chs = 4; + } + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, chs, 0, + &chmap); + if (err < 0) + return err; + chmap->channel_mask = SND_PCM_CHMAP_MASK_2468; + chip->ac97[0]->chmaps[SNDRV_PCM_STREAM_PLAYBACK] = chmap; + } + return 0; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index e3d32e2d574e..f0b4efdb483c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1440,6 +1440,7 @@ static void init_viadev(struct via82xx *chip, int idx, unsigned int reg_offset, static int __devinit snd_via8233_pcm_new(struct via82xx *chip) { struct snd_pcm *pcm; + struct snd_pcm_chmap *chmap; int i, err; chip->playback_devno = 0; /* x 4 */ @@ -1467,6 +1468,12 @@ static int __devinit snd_via8233_pcm_new(struct via82xx *chip) snd_dma_pci_data(chip->pci), 64*1024, VIA_MAX_BUFSIZE); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_std_chmaps, 2, 0, + &chmap); + if (err < 0) + return err; + /* PCM #1: multi-channel playback and 2nd capture */ err = snd_pcm_new(chip->card, chip->card->shortname, 1, 1, 1, &pcm); if (err < 0) @@ -1484,6 +1491,14 @@ static int __devinit snd_via8233_pcm_new(struct via82xx *chip) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), 64*1024, VIA_MAX_BUFSIZE); + + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, 6, 0, + &chmap); + if (err < 0) + return err; + chip->ac97->chmaps[SNDRV_PCM_STREAM_PLAYBACK] = chmap; + return 0; } @@ -1493,6 +1508,7 @@ static int __devinit snd_via8233_pcm_new(struct via82xx *chip) static int __devinit snd_via8233a_pcm_new(struct via82xx *chip) { struct snd_pcm *pcm; + struct snd_pcm_chmap *chmap; int err; chip->multi_devno = 0; @@ -1519,6 +1535,13 @@ static int __devinit snd_via8233a_pcm_new(struct via82xx *chip) snd_dma_pci_data(chip->pci), 64*1024, VIA_MAX_BUFSIZE); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, 6, 0, + &chmap); + if (err < 0) + return err; + chip->ac97->chmaps[SNDRV_PCM_STREAM_PLAYBACK] = chmap; + /* SPDIF supported? */ if (! ac97_can_spdif(chip->ac97)) return 0; -- cgit v1.2.3 From f49921b894ec338696e47d985acc09df3cddbfd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 18:02:38 +0200 Subject: ALSA: cmipci: Implement channel mapping Simply enable the channel map according to the h/w capability. Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 7c25906b5f88..22122ff26e34 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1962,6 +1962,12 @@ static int __devinit snd_cmipci_pcm_spdif_new(struct cmipci *cm, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(cm->pci), 64*1024, 128*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, cm->max_channels, 0, + NULL); + if (err < 0) + return err; + return 0; } -- cgit v1.2.3 From 8d50cdc1f52843d4fe593c2613528c3f6677949a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Aug 2012 18:03:17 +0200 Subject: ALSA: ctxfi: Implement channel maps Assign the multi-channel map to front PCM, and other channel map to each other channel PCM. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctpcm.c | 50 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index d021876901bb..6c191429b126 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -395,12 +395,38 @@ static struct snd_pcm_ops ct_pcm_capture_ops = { .page = snd_pcm_sgbuf_ops_page, }; +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_RC } }, + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +static const struct snd_pcm_chmap_elem clfe_map[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_FC } }, + { .channels = 2, + .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; + +static const struct snd_pcm_chmap_elem side_map[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_RC } }, + { .channels = 2, + .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; + /* Create ALSA pcm device */ int ct_alsa_pcm_create(struct ct_atc *atc, enum CTALSADEVS device, const char *device_name) { struct snd_pcm *pcm; + const struct snd_pcm_chmap_elem *map; + int chs; int err; int playback_count, capture_count; @@ -427,6 +453,30 @@ int ct_alsa_pcm_create(struct ct_atc *atc, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(atc->pci), 128*1024, 128*1024); + chs = 2; + switch (device) { + case FRONT: + chs = 8; + map = snd_pcm_std_chmaps; + break; + case SURROUND: + map = surround_map; + break; + case CLFE: + map = clfe_map; + break; + case SIDE: + map = side_map; + break; + default: + map = snd_pcm_std_chmaps; + break; + } + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, chs, + 0, NULL); + if (err < 0) + return err; + #ifdef CONFIG_PM_SLEEP atc->pcms[device] = pcm; #endif -- cgit v1.2.3 From 6e67683d718c627aa585a82ae0eb44a7058eae5a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Aug 2012 14:49:47 +0200 Subject: ALSA: Remove VOLATILE flag from chmap ctls The VOLATILE flag was added to control elements by snd_pcm_add_chmap_ctls() just because I didn't want to have a side-effect of "alsactl restore". But now the set operation doesn't allow to change the value unless the PCM stream is in PREAPRED state, there is no reason to keep this flag. Let's rip it off. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 565102705eda..0a750ec5c400 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2479,7 +2479,6 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, struct snd_kcontrol_new knew = { .iface = SNDRV_CTL_ELEM_IFACE_PCM, .access = SNDRV_CTL_ELEM_ACCESS_READ | - SNDRV_CTL_ELEM_ACCESS_VOLATILE | /* no notification */ SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, .info = pcm_chmap_ctl_info, -- cgit v1.2.3 From 5fe8e1e6717c0ed40abff8be3a441d0e2eb47169 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 7 Sep 2012 07:25:44 +0200 Subject: ALSA: hda - Remove ignore_misc_bit The purpose of this flag is unclear. If the problem is that some machines have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups. In addition, this causes jack detection functionality to be flawed on the M31EI, where there are two jacks without jack detection (which is properly marked as NO_PRESENCE), but due to ignore_misc_bit, these jacks are instead being reported as being present but always unplugged. BugLink: https://bugs.launchpad.net/bugs/939161 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 4 ---- sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_jack.c | 5 ++--- 3 files changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 4f7d2dfcef7b..9acd5a964eb6 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -141,7 +141,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, memset(sequences_hp, 0, sizeof(sequences_hp)); assoc_line_out = 0; - codec->ignore_misc_bit = true; end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); @@ -157,9 +156,6 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - codec->ignore_misc_bit = false; conn = get_defcfg_connect(def_conf); if (conn == AC_JACK_PORT_NONE) continue; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8fd9f636cfe1..ee4ae8e271e3 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -861,7 +861,6 @@ struct hda_codec { unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ - unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index aaccc0236bda..c9333c9688fb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -26,9 +26,8 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) return false; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) + if (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE) return false; if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) return false; -- cgit v1.2.3 From 822b4b8d63e09076a4487eb881d3b7a13b28121c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Sep 2012 10:54:32 +0800 Subject: ASoC: dapm: Add flags to regulator supplies This will be used to enable additional control of the regulators. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 5 +++-- sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 6 +++--- sound/soc/codecs/wm5102.c | 12 ++++++------ sound/soc/codecs/wm5110.c | 12 ++++++------ sound/soc/codecs/wm8996.c | 2 +- 7 files changed, 25 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 07e2510619a1..c96bf5ae80a6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -244,10 +244,11 @@ struct device; { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ .shift = wshift, .invert = winvert, .event = wevent, \ .event_flags = wflags} -#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \ +#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \ { .id = snd_soc_dapm_regulator_supply, .name = wname, \ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ - .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ + .invert = wflags} /* dapm kcontrol types */ diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b7836503dc69..3f46bffeb0c3 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -390,10 +390,10 @@ static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), /* Regulators */ - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), - SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0), /* Power */ SND_SOC_DAPM_SUPPLY("Audio Power", diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 71debd0a3822..efa93dbb0191 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1117,8 +1117,8 @@ SND_SOC_DAPM_SUPPLY("MICBIAS1", WM2200_MIC_BIAS_CTRL_1, WM2200_MICB1_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM2200_MIC_BIAS_CTRL_2, WM2200_MICB2_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20, 0), SND_SOC_DAPM_INPUT("IN1L"), SND_SOC_DAPM_INPUT("IN1R"), diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f4817292ef45..4da1b92b22c2 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -848,9 +848,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, NULL, 0), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e2fb07ee68a7..4a2db4e10885 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -305,12 +305,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 57c7d9c0aadb..bf47914234b3 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -310,12 +310,12 @@ SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), -SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), -SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0, 0), SND_SOC_DAPM_SIGGEN("TONE"), SND_SOC_DAPM_SIGGEN("NOISE"), diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f183dfa454..6dcb02c3666f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -931,7 +931,7 @@ SND_SOC_DAPM_INPUT("IN2RP"), SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), -SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), -- cgit v1.2.3 From d6e2dc150ba748c4de518532b0d71275e3c3d959 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 15:14:16 +0800 Subject: ASoC: wm8983: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 367388fdc486..57c33fc19681 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1085,7 +1085,7 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&spi->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1094,15 +1094,12 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); return ret; } static int __devexit wm8983_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1123,7 +1120,7 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, struct wm8983_priv *wm8983; int ret; - wm8983 = kzalloc(sizeof *wm8983, GFP_KERNEL); + wm8983 = devm_kzalloc(&i2c->dev, sizeof *wm8983, GFP_KERNEL); if (!wm8983) return -ENOMEM; @@ -1132,15 +1129,13 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8983, &wm8983_dai, 1); - if (ret < 0) - kfree(wm8983); + return ret; } static __devexit int wm8983_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 2ee01ac63b72a3101d6293b50d70d830959bbe8f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 15:28:19 +0800 Subject: ASoC: wm8983: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 151 ++++++++++++++++++++++++++-------------------- 1 file changed, 85 insertions(+), 66 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 57c33fc19681..d8879f262d27 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -27,61 +28,60 @@ #include "wm8983.h" -static const u16 wm8983_reg_defs[WM8983_MAX_REGISTER + 1] = { - [0x00] = 0x0000, /* R0 - Software Reset */ - [0x01] = 0x0000, /* R1 - Power management 1 */ - [0x02] = 0x0000, /* R2 - Power management 2 */ - [0x03] = 0x0000, /* R3 - Power management 3 */ - [0x04] = 0x0050, /* R4 - Audio Interface */ - [0x05] = 0x0000, /* R5 - Companding control */ - [0x06] = 0x0140, /* R6 - Clock Gen control */ - [0x07] = 0x0000, /* R7 - Additional control */ - [0x08] = 0x0000, /* R8 - GPIO Control */ - [0x09] = 0x0000, /* R9 - Jack Detect Control 1 */ - [0x0A] = 0x0000, /* R10 - DAC Control */ - [0x0B] = 0x00FF, /* R11 - Left DAC digital Vol */ - [0x0C] = 0x00FF, /* R12 - Right DAC digital vol */ - [0x0D] = 0x0000, /* R13 - Jack Detect Control 2 */ - [0x0E] = 0x0100, /* R14 - ADC Control */ - [0x0F] = 0x00FF, /* R15 - Left ADC Digital Vol */ - [0x10] = 0x00FF, /* R16 - Right ADC Digital Vol */ - [0x12] = 0x012C, /* R18 - EQ1 - low shelf */ - [0x13] = 0x002C, /* R19 - EQ2 - peak 1 */ - [0x14] = 0x002C, /* R20 - EQ3 - peak 2 */ - [0x15] = 0x002C, /* R21 - EQ4 - peak 3 */ - [0x16] = 0x002C, /* R22 - EQ5 - high shelf */ - [0x18] = 0x0032, /* R24 - DAC Limiter 1 */ - [0x19] = 0x0000, /* R25 - DAC Limiter 2 */ - [0x1B] = 0x0000, /* R27 - Notch Filter 1 */ - [0x1C] = 0x0000, /* R28 - Notch Filter 2 */ - [0x1D] = 0x0000, /* R29 - Notch Filter 3 */ - [0x1E] = 0x0000, /* R30 - Notch Filter 4 */ - [0x20] = 0x0038, /* R32 - ALC control 1 */ - [0x21] = 0x000B, /* R33 - ALC control 2 */ - [0x22] = 0x0032, /* R34 - ALC control 3 */ - [0x23] = 0x0000, /* R35 - Noise Gate */ - [0x24] = 0x0008, /* R36 - PLL N */ - [0x25] = 0x000C, /* R37 - PLL K 1 */ - [0x26] = 0x0093, /* R38 - PLL K 2 */ - [0x27] = 0x00E9, /* R39 - PLL K 3 */ - [0x29] = 0x0000, /* R41 - 3D control */ - [0x2A] = 0x0000, /* R42 - OUT4 to ADC */ - [0x2B] = 0x0000, /* R43 - Beep control */ - [0x2C] = 0x0033, /* R44 - Input ctrl */ - [0x2D] = 0x0010, /* R45 - Left INP PGA gain ctrl */ - [0x2E] = 0x0010, /* R46 - Right INP PGA gain ctrl */ - [0x2F] = 0x0100, /* R47 - Left ADC BOOST ctrl */ - [0x30] = 0x0100, /* R48 - Right ADC BOOST ctrl */ - [0x31] = 0x0002, /* R49 - Output ctrl */ - [0x32] = 0x0001, /* R50 - Left mixer ctrl */ - [0x33] = 0x0001, /* R51 - Right mixer ctrl */ - [0x34] = 0x0039, /* R52 - LOUT1 (HP) volume ctrl */ - [0x35] = 0x0039, /* R53 - ROUT1 (HP) volume ctrl */ - [0x36] = 0x0039, /* R54 - LOUT2 (SPK) volume ctrl */ - [0x37] = 0x0039, /* R55 - ROUT2 (SPK) volume ctrl */ - [0x38] = 0x0001, /* R56 - OUT3 mixer ctrl */ - [0x39] = 0x0001, /* R57 - OUT4 (MONO) mix ctrl */ - [0x3D] = 0x0000 /* R61 - BIAS CTRL */ +static const struct reg_default wm8983_defaults[] = { + { 0x01, 0x0000 }, /* R1 - Power management 1 */ + { 0x02, 0x0000 }, /* R2 - Power management 2 */ + { 0x03, 0x0000 }, /* R3 - Power management 3 */ + { 0x04, 0x0050 }, /* R4 - Audio Interface */ + { 0x05, 0x0000 }, /* R5 - Companding control */ + { 0x06, 0x0140 }, /* R6 - Clock Gen control */ + { 0x07, 0x0000 }, /* R7 - Additional control */ + { 0x08, 0x0000 }, /* R8 - GPIO Control */ + { 0x09, 0x0000 }, /* R9 - Jack Detect Control 1 */ + { 0x0A, 0x0000 }, /* R10 - DAC Control */ + { 0x0B, 0x00FF }, /* R11 - Left DAC digital Vol */ + { 0x0C, 0x00FF }, /* R12 - Right DAC digital vol */ + { 0x0D, 0x0000 }, /* R13 - Jack Detect Control 2 */ + { 0x0E, 0x0100 }, /* R14 - ADC Control */ + { 0x0F, 0x00FF }, /* R15 - Left ADC Digital Vol */ + { 0x10, 0x00FF }, /* R16 - Right ADC Digital Vol */ + { 0x12, 0x012C }, /* R18 - EQ1 - low shelf */ + { 0x13, 0x002C }, /* R19 - EQ2 - peak 1 */ + { 0x14, 0x002C }, /* R20 - EQ3 - peak 2 */ + { 0x15, 0x002C }, /* R21 - EQ4 - peak 3 */ + { 0x16, 0x002C }, /* R22 - EQ5 - high shelf */ + { 0x18, 0x0032 }, /* R24 - DAC Limiter 1 */ + { 0x19, 0x0000 }, /* R25 - DAC Limiter 2 */ + { 0x1B, 0x0000 }, /* R27 - Notch Filter 1 */ + { 0x1C, 0x0000 }, /* R28 - Notch Filter 2 */ + { 0x1D, 0x0000 }, /* R29 - Notch Filter 3 */ + { 0x1E, 0x0000 }, /* R30 - Notch Filter 4 */ + { 0x20, 0x0038 }, /* R32 - ALC control 1 */ + { 0x21, 0x000B }, /* R33 - ALC control 2 */ + { 0x22, 0x0032 }, /* R34 - ALC control 3 */ + { 0x23, 0x0000 }, /* R35 - Noise Gate */ + { 0x24, 0x0008 }, /* R36 - PLL N */ + { 0x25, 0x000C }, /* R37 - PLL K 1 */ + { 0x26, 0x0093 }, /* R38 - PLL K 2 */ + { 0x27, 0x00E9 }, /* R39 - PLL K 3 */ + { 0x29, 0x0000 }, /* R41 - 3D control */ + { 0x2A, 0x0000 }, /* R42 - OUT4 to ADC */ + { 0x2B, 0x0000 }, /* R43 - Beep control */ + { 0x2C, 0x0033 }, /* R44 - Input ctrl */ + { 0x2D, 0x0010 }, /* R45 - Left INP PGA gain ctrl */ + { 0x2E, 0x0010 }, /* R46 - Right INP PGA gain ctrl */ + { 0x2F, 0x0100 }, /* R47 - Left ADC BOOST ctrl */ + { 0x30, 0x0100 }, /* R48 - Right ADC BOOST ctrl */ + { 0x31, 0x0002 }, /* R49 - Output ctrl */ + { 0x32, 0x0001 }, /* R50 - Left mixer ctrl */ + { 0x33, 0x0001 }, /* R51 - Right mixer ctrl */ + { 0x34, 0x0039 }, /* R52 - LOUT1 (HP) volume ctrl */ + { 0x35, 0x0039 }, /* R53 - ROUT1 (HP) volume ctrl */ + { 0x36, 0x0039 }, /* R54 - LOUT2 (SPK) volume ctrl */ + { 0x37, 0x0039 }, /* R55 - ROUT2 (SPK) volume ctrl */ + { 0x38, 0x0001 }, /* R56 - OUT3 mixer ctrl */ + { 0x39, 0x0001 }, /* R57 - OUT4 (MONO) mix ctrl */ + { 0x3D, 0x0000 }, /* R61 - BIAS CTRL */ }; static const struct wm8983_reg_access { @@ -159,7 +159,7 @@ static const int vol_update_regs[] = { }; struct wm8983_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u32 sysclk; u32 bclk; }; @@ -610,7 +610,7 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8983_readable(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8983_readable(struct device *dev, unsigned int reg) { if (reg > WM8983_MAX_REGISTER) return 0; @@ -905,6 +905,7 @@ static int wm8983_set_sysclk(struct snd_soc_dai *dai, static int wm8983_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8983_priv *wm8983 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -917,7 +918,7 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8983->regmap); if (ret < 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); return ret; @@ -994,10 +995,9 @@ static int wm8983_remove(struct snd_soc_codec *codec) static int wm8983_probe(struct snd_soc_codec *codec) { int ret; - struct wm8983_priv *wm8983 = snd_soc_codec_get_drvdata(codec); int i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8983->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1067,16 +1067,23 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8983 = { .suspend = wm8983_suspend, .resume = wm8983_resume, .set_bias_level = wm8983_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8983_reg_defs), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8983_reg_defs, .controls = wm8983_snd_controls, .num_controls = ARRAY_SIZE(wm8983_snd_controls), .dapm_widgets = wm8983_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8983_dapm_widgets), .dapm_routes = wm8983_audio_map, .num_dapm_routes = ARRAY_SIZE(wm8983_audio_map), - .readable_register = wm8983_readable +}; + +static const struct regmap_config wm8983_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .reg_defaults = wm8983_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8983_defaults), + .cache_type = REGCACHE_RBTREE, + + .readable_reg = wm8983_readable, }; #if defined(CONFIG_SPI_MASTER) @@ -1089,7 +1096,13 @@ static int __devinit wm8983_spi_probe(struct spi_device *spi) if (!wm8983) return -ENOMEM; - wm8983->control_type = SND_SOC_SPI; + wm8983->regmap = devm_regmap_init_spi(spi, &wm8983_regmap); + if (IS_ERR(wm8983->regmap)) { + ret = PTR_ERR(wm8983->regmap); + dev_err(&spi->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8983); ret = snd_soc_register_codec(&spi->dev, @@ -1124,7 +1137,13 @@ static __devinit int wm8983_i2c_probe(struct i2c_client *i2c, if (!wm8983) return -ENOMEM; - wm8983->control_type = SND_SOC_I2C; + wm8983->regmap = devm_regmap_init_i2c(i2c, &wm8983_regmap); + if (IS_ERR(wm8983->regmap)) { + ret = PTR_ERR(wm8983->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8983); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From 7d014db8baf70bcc7e9cf1457350b11bc2affbbd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 16:18:56 +0800 Subject: ASoC: wm8523: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 1c3ffb290cdc..af2289ee6cd6 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -516,7 +516,8 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, struct wm8523_priv *wm8523; int ret; - wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL); + wm8523 = devm_kzalloc(&i2c->dev, sizeof(struct wm8523_priv), + GFP_KERNEL); if (wm8523 == NULL) return -ENOMEM; @@ -525,8 +526,7 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8523, &wm8523_dai, 1); - if (ret < 0) - kfree(wm8523); + return ret; } @@ -534,7 +534,6 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, static __devexit int wm8523_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 719b0c593cbb2a199e977b4dcca1d096a4a0d6a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 16:23:34 +0800 Subject: ASoC: wm8523: Move regulator acquisition to I2C probe() This is better style since we acquire all needed resources before we try to do the ASoC card probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index af2289ee6cd6..c4c64e225c42 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -402,7 +402,7 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = @@ -414,16 +414,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++) - wm8523->supplies[i].supply = wm8523_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8523->supplies), - wm8523->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); if (ret != 0) { @@ -471,7 +461,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); return ret; } @@ -481,7 +470,6 @@ static int wm8523_remove(struct snd_soc_codec *codec) struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); return 0; } @@ -514,13 +502,23 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8523_priv *wm8523; - int ret; + int ret, i; wm8523 = devm_kzalloc(&i2c->dev, sizeof(struct wm8523_priv), GFP_KERNEL); if (wm8523 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++) + wm8523->supplies[i].supply = wm8523_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8523); wm8523->control_type = SND_SOC_I2C; -- cgit v1.2.3 From b9288f49dc5ac8342cc34163093c9f7d096b6378 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 16:27:45 +0800 Subject: ASoC: wm8523: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 69 ++++++++++++++++++++++++++--------------------- 1 file changed, 38 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index c4c64e225c42..d7d5fe6866a6 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -39,33 +40,31 @@ static const char *wm8523_supply_names[WM8523_NUM_SUPPLIES] = { /* codec private data */ struct wm8523_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8523_NUM_SUPPLIES]; unsigned int sysclk; unsigned int rate_constraint_list[WM8523_NUM_RATES]; struct snd_pcm_hw_constraint_list rate_constraint; }; -static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { - 0x8523, /* R0 - DEVICE_ID */ - 0x0001, /* R1 - REVISION */ - 0x0000, /* R2 - PSCTRL1 */ - 0x1812, /* R3 - AIF_CTRL1 */ - 0x0000, /* R4 - AIF_CTRL2 */ - 0x0001, /* R5 - DAC_CTRL3 */ - 0x0190, /* R6 - DAC_GAINL */ - 0x0190, /* R7 - DAC_GAINR */ - 0x0000, /* R8 - ZERO_DETECT */ +static const struct reg_default wm8523_reg_defaults[] = { + { 2, 0x0000 }, /* R2 - PSCTRL1 */ + { 3, 0x1812 }, /* R3 - AIF_CTRL1 */ + { 4, 0x0000 }, /* R4 - AIF_CTRL2 */ + { 5, 0x0001 }, /* R5 - DAC_CTRL3 */ + { 6, 0x0190 }, /* R6 - DAC_GAINL */ + { 7, 0x0190 }, /* R7 - DAC_GAINR */ + { 8, 0x0000 }, /* R8 - ZERO_DETECT */ }; -static int wm8523_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8523_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8523_DEVICE_ID: case WM8523_REVISION: - return 1; + return true; default: - return 0; + return false; } } @@ -301,8 +300,7 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; - int ret, i; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -325,16 +323,13 @@ static int wm8523_set_bias_level(struct snd_soc_codec *codec, return ret; } + /* Sync back default/cached values */ + regcache_sync(wm8523->regmap); + /* Initial power up */ snd_soc_update_bits(codec, WM8523_PSCTRL1, WM8523_SYS_ENA_MASK, 1); - /* Sync back default/cached values */ - for (i = WM8523_AIF_CTRL1; - i < WM8523_MAX_REGISTER; i++) - snd_soc_write(codec, i, reg_cache[i]); - - msleep(100); } @@ -408,7 +403,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8523->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -426,7 +421,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to read ID register\n"); goto err_enable; } - if (ret != wm8523_reg[WM8523_DEVICE_ID]) { + if (ret != 0x8523) { dev_err(codec->dev, "Device is not a WM8523, ID is %x\n", ret); ret = -EINVAL; goto err_enable; @@ -467,8 +462,6 @@ err_get: static int wm8523_remove(struct snd_soc_codec *codec) { - struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -479,10 +472,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = { .suspend = wm8523_suspend, .resume = wm8523_resume, .set_bias_level = wm8523_set_bias_level, - .reg_cache_size = WM8523_REGISTER_COUNT, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8523_reg, - .volatile_register = wm8523_volatile_register, .controls = wm8523_controls, .num_controls = ARRAY_SIZE(wm8523_controls), @@ -497,6 +486,18 @@ static const struct of_device_id wm8523_of_match[] = { { }, }; +static const struct regmap_config wm8523_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8523_ZERO_DETECT, + + .reg_defaults = wm8523_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8523_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8523_volatile_register, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -509,6 +510,13 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, if (wm8523 == NULL) return -ENOMEM; + wm8523->regmap = devm_regmap_init_i2c(i2c, &wm8523_regmap); + if (IS_ERR(wm8523->regmap)) { + ret = PTR_ERR(wm8523->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++) wm8523->supplies[i].supply = wm8523_supply_names[i]; @@ -520,7 +528,6 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, wm8523); - wm8523->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8523, &wm8523_dai, 1); -- cgit v1.2.3 From 59ac2149aee9b69732dad602ea250ecb60b9e617 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 17:07:24 +0800 Subject: ASoC: wm8523: Move device ID verification and reset to I2C probe Ensure that we have confirmed that we've got the device in place before we register with ASoC. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 84 +++++++++++++++++++++-------------------------- 1 file changed, 38 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index d7d5fe6866a6..8d5c27673501 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -68,11 +68,6 @@ static bool wm8523_volatile_register(struct device *dev, unsigned int reg) } } -static int wm8523_reset(struct snd_soc_codec *codec) -{ - return snd_soc_write(codec, WM8523_DEVICE_ID, 0); -} - static const DECLARE_TLV_DB_SCALE(dac_tlv, -10000, 25, 0); static const char *wm8523_zd_count_text[] = { @@ -409,38 +404,6 @@ static int wm8523_probe(struct snd_soc_codec *codec) return ret; } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), - wm8523->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - ret = snd_soc_read(codec, WM8523_DEVICE_ID); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_enable; - } - if (ret != 0x8523) { - dev_err(codec->dev, "Device is not a WM8523, ID is %x\n", ret); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8523_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read revision register\n"); - goto err_enable; - } - dev_info(codec->dev, "revision %c\n", - (ret & WM8523_CHIP_REV_MASK) + 'A'); - - ret = wm8523_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8523_DAC_GAINR, WM8523_DACR_VU, WM8523_DACR_VU); @@ -448,16 +411,7 @@ static int wm8523_probe(struct snd_soc_codec *codec) wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Bias level configuration will have done an extra enable */ - regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); - return 0; - -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); -err_get: - - return ret; } static int wm8523_remove(struct snd_soc_codec *codec) @@ -503,6 +457,7 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8523_priv *wm8523; + unsigned int val; int ret, i; wm8523 = devm_kzalloc(&i2c->dev, sizeof(struct wm8523_priv), @@ -527,6 +482,40 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, return ret; } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(wm8523->regmap, WM8523_DEVICE_ID, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register\n"); + goto err_enable; + } + if (val != 0x8523) { + dev_err(&i2c->dev, "Device is not a WM8523, ID is %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = regmap_read(wm8523->regmap, WM8523_REVISION, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read revision register\n"); + goto err_enable; + } + dev_info(&i2c->dev, "revision %c\n", + (val & WM8523_CHIP_REV_MASK) + 'A'); + + ret = regmap_write(wm8523->regmap, WM8523_DEVICE_ID, 0x8523); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to reset device: %d\n", ret); + goto err_enable; + } + + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + i2c_set_clientdata(i2c, wm8523); ret = snd_soc_register_codec(&i2c->dev, @@ -534,6 +523,9 @@ static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, return ret; +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + return ret; } static __devexit int wm8523_i2c_remove(struct i2c_client *client) -- cgit v1.2.3 From aff041af948f5cdf51e2115f267957a89f28ac0f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 10:59:51 +0800 Subject: ASoC: sta32x: Move regulator acquisition to I2C probe This is better style as it ensures we don't try to do the ASoC probe without required resources. Also convert to devm_ while we're at it, saving a bit of code, and fix a leak of enable on error. Signed-off-by: Mark Brown Acked-by: Johannes Stezenbach --- sound/soc/codecs/sta32x.c | 32 +++++++++++++++----------------- 1 file changed, 15 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 51b7313a4c14..039597e66987 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -825,22 +825,11 @@ static int sta32x_probe(struct snd_soc_codec *codec) sta32x->codec = codec; sta32x->pdata = dev_get_platdata(codec->dev); - /* regulators */ - for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) - sta32x->supplies[i].supply = sta32x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sta32x->supplies), - sta32x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } - ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + return ret; } /* Tell ASoC what kind of I/O to use to read the registers. ASoC will @@ -849,7 +838,7 @@ static int sta32x_probe(struct snd_soc_codec *codec) ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; + goto err; } /* Chip documentation explicitly requires that the reset values @@ -915,9 +904,8 @@ static int sta32x_probe(struct snd_soc_codec *codec) return 0; -err_get: - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); err: + regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return ret; } @@ -928,7 +916,6 @@ static int sta32x_remove(struct snd_soc_codec *codec) sta32x_watchdog_stop(sta32x); sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; } @@ -966,13 +953,24 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sta32x_priv *sta32x; - int ret; + int ret, i; sta32x = devm_kzalloc(&i2c->dev, sizeof(struct sta32x_priv), GFP_KERNEL); if (!sta32x) return -ENOMEM; + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++) + sta32x->supplies[i].supply = sta32x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(sta32x->supplies), + sta32x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, sta32x); ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); -- cgit v1.2.3 From 29fdf4fbbe0891349d8444bde4c09f9cfaf744b6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 10:59:56 +0800 Subject: ASoC: sta32x: Convert to regmap Long term all drivers should be using regmap directly. This is more idiomatic and moves us towards the removal of the ASoC level cache code. The initialiasation of reserved register bits in probe() is slightly odd as the defaults being written don't appear to match the silicon defaults but the new code should have the same effect as the old code. The watchdog code will now unconditionally do a mute and unmute when resyncing but since we only sync when we are very sure there is something to sync this should have no impact. Signed-off-by: Mark Brown Acked-by: Johannes Stezenbach --- sound/soc/codecs/sta32x.c | 107 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 80 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 039597e66987..0935bfe62471 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -55,12 +56,50 @@ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) /* Power-up register defaults */ -static const u8 sta32x_regs[STA32X_REGISTER_COUNT] = { - 0x63, 0x80, 0xc2, 0x40, 0xc2, 0x5c, 0x10, 0xff, 0x60, 0x60, - 0x60, 0x80, 0x00, 0x00, 0x00, 0x40, 0x80, 0x77, 0x6a, 0x69, - 0x6a, 0x69, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2d, - 0xc0, 0xf3, 0x33, 0x00, 0x0c, +static const struct reg_default sta32x_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xc2 }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x10 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x80 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2d }, + { 0x28, 0xc0 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, }; /* regulator power supply names */ @@ -72,6 +111,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { + struct regmap *regmap; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_codec *codec; struct sta32x_platform_data *pdata; @@ -291,17 +331,15 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { + struct sta32x_priv *sta32x = codec->control_data; unsigned int mute; int rc; - if (!codec->cache_sync) - return 0; - /* mute during register sync */ mute = snd_soc_read(codec, STA32X_MMUTE); snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); sta32x_sync_coef_shadow(codec); - rc = snd_soc_cache_sync(codec); + rc = regcache_sync(sta32x->regmap); snd_soc_write(codec, STA32X_MMUTE, mute); return rc; } @@ -316,11 +354,11 @@ static void sta32x_watchdog(struct work_struct *work) /* check if sta32x has reset itself */ confa_cached = snd_soc_read(codec, STA32X_CONFA); - codec->cache_bypass = 1; + regcache_cache_bypass(sta32x->regmap, true); confa = snd_soc_read(codec, STA32X_CONFA); - codec->cache_bypass = 0; + regcache_cache_bypass(sta32x->regmap, false); if (confa != confa_cached) { - codec->cache_sync = 1; + regcache_mark_dirty(sta32x->regmap); sta32x_cache_sync(codec); } @@ -835,7 +873,8 @@ static int sta32x_probe(struct snd_soc_codec *codec) /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + codec->control_data = sta32x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); goto err; @@ -847,13 +886,15 @@ static int sta32x_probe(struct snd_soc_codec *codec) * so the write to the these registers are suppressed by the cache * restore code when it skips writes of default registers. */ - snd_soc_cache_write(codec, STA32X_CONFC, 0xc2); - snd_soc_cache_write(codec, STA32X_CONFE, 0xc2); - snd_soc_cache_write(codec, STA32X_CONFF, 0x5c); - snd_soc_cache_write(codec, STA32X_MMUTE, 0x10); - snd_soc_cache_write(codec, STA32X_AUTO1, 0x60); - snd_soc_cache_write(codec, STA32X_AUTO3, 0x00); - snd_soc_cache_write(codec, STA32X_C3CFG, 0x40); + regcache_cache_only(sta32x->regmap, true); + snd_soc_write(codec, STA32X_CONFC, 0xc2); + snd_soc_write(codec, STA32X_CONFE, 0xc2); + snd_soc_write(codec, STA32X_CONFF, 0x5c); + snd_soc_write(codec, STA32X_MMUTE, 0x10); + snd_soc_write(codec, STA32X_AUTO1, 0x60); + snd_soc_write(codec, STA32X_AUTO3, 0x00); + snd_soc_write(codec, STA32X_C3CFG, 0x40); + regcache_cache_only(sta32x->regmap, false); /* set thermal warning adjustment and recovery */ if (!(sta32x->pdata->thermal_conf & STA32X_THERMAL_ADJUSTMENT_ENABLE)) @@ -920,8 +961,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) return 0; } -static int sta32x_reg_is_volatile(struct snd_soc_codec *codec, - unsigned int reg) +static bool sta32x_reg_is_volatile(struct device *dev, unsigned int reg) { switch (reg) { case STA32X_CONFA ... STA32X_L2ATRT: @@ -936,10 +976,6 @@ static const struct snd_soc_codec_driver sta32x_codec = { .remove = sta32x_remove, .suspend = sta32x_suspend, .resume = sta32x_resume, - .reg_cache_size = STA32X_REGISTER_COUNT, - .reg_word_size = sizeof(u8), - .reg_cache_default = sta32x_regs, - .volatile_register = sta32x_reg_is_volatile, .set_bias_level = sta32x_set_bias_level, .controls = sta32x_snd_controls, .num_controls = ARRAY_SIZE(sta32x_snd_controls), @@ -949,6 +985,16 @@ static const struct snd_soc_codec_driver sta32x_codec = { .num_dapm_routes = ARRAY_SIZE(sta32x_dapm_routes), }; +static const struct regmap_config sta32x_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA32X_FDRC2, + .reg_defaults = sta32x_regs, + .num_reg_defaults = ARRAY_SIZE(sta32x_regs), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = sta32x_reg_is_volatile, +}; + static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -971,6 +1017,13 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, return ret; } + sta32x->regmap = devm_regmap_init_i2c(i2c, &sta32x_regmap); + if (IS_ERR(sta32x->regmap)) { + ret = PTR_ERR(sta32x->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, sta32x); ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); -- cgit v1.2.3 From c35e005f3115cd27d85625805645b90ba961f16f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Sep 2012 02:13:52 -0700 Subject: ASoC: fsi: fixup pm_runtime_disable() timing on fsi_probe() pm_runtime_disable() error handling timing on fsi_probe() was wrong. This patch fixes it up. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0540408a9fa9..8534989836aa 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1730,12 +1730,12 @@ exit_snd_soc: exit_free_irq: free_irq(irq, master); exit_fsib: + pm_runtime_disable(&pdev->dev); fsi_stream_remove(&master->fsib); exit_fsia: fsi_stream_remove(&master->fsia); exit_iounmap: iounmap(master->base); - pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; -- cgit v1.2.3 From dbd4e51cd164e7d94b00c0c0dd3ac5517364a8fb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Sep 2012 02:14:10 -0700 Subject: ASoC: fsi: tidyup: remove un-necessary operation from fsi_probe() struct fsi_master *master became member of struct fsi_priv from 71f6e0645be42f93c0f90dfcc93b9d2d277c2ee6 (ASoC: sh_fsi: avoid using global variable) So, master = NULL is not necessary on fsi_probe() now. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8534989836aa..a5ee2faa1073 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1738,7 +1738,6 @@ exit_iounmap: iounmap(master->base); exit_kfree: kfree(master); - master = NULL; exit: return ret; } -- cgit v1.2.3 From 6ac4262f367fd0d9b0219dfd014ffcca4a6cfa6a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 10 Sep 2012 02:14:31 -0700 Subject: ASoC: fsi: convert to devm_xxx() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a5ee2faa1073..5328ae5539f1 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1655,22 +1655,20 @@ static int fsi_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (!res || (int)irq <= 0) { dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); - ret = -ENODEV; - goto exit; + return -ENODEV; } - master = kzalloc(sizeof(*master), GFP_KERNEL); + master = devm_kzalloc(&pdev->dev, sizeof(*master), GFP_KERNEL); if (!master) { dev_err(&pdev->dev, "Could not allocate master\n"); - ret = -ENOMEM; - goto exit; + return -ENOMEM; } - master->base = ioremap_nocache(res->start, resource_size(res)); + master->base = devm_ioremap_nocache(&pdev->dev, + res->start, resource_size(res)); if (!master->base) { - ret = -ENXIO; dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); - goto exit_kfree; + return -ENXIO; } /* master setting */ @@ -1686,7 +1684,7 @@ static int fsi_probe(struct platform_device *pdev) ret = fsi_stream_probe(&master->fsia, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); - goto exit_iounmap; + return ret; } /* FSI B setting */ @@ -1734,11 +1732,7 @@ exit_fsib: fsi_stream_remove(&master->fsib); exit_fsia: fsi_stream_remove(&master->fsia); -exit_iounmap: - iounmap(master->base); -exit_kfree: - kfree(master); -exit: + return ret; } @@ -1757,9 +1751,6 @@ static int fsi_remove(struct platform_device *pdev) fsi_stream_remove(&master->fsia); fsi_stream_remove(&master->fsib); - iounmap(master->base); - kfree(master); - return 0; } -- cgit v1.2.3 From d9780550a354058bc47db6ac48d3b77f186882c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 17:52:59 +0800 Subject: ASoC: wm8741: Move regulator acquisition to I2C/SPI probe() Better style as we acquire resources before trying the ASoC card probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 37 ++++++++++++++++++++++--------------- 1 file changed, 22 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 35f3d23200e0..742744b4bba7 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -403,17 +403,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); int ret = 0; - int i; - - for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) - wm8741->supplies[i].supply = wm8741_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies), - wm8741->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; - } ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); @@ -450,7 +439,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err: return ret; } @@ -460,7 +448,6 @@ static int wm8741_remove(struct snd_soc_codec *codec) struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); - regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); return 0; } @@ -492,13 +479,23 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&i2c->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + i2c_set_clientdata(i2c, wm8741); wm8741->control_type = SND_SOC_I2C; @@ -536,13 +533,23 @@ static struct i2c_driver wm8741_i2c_driver = { static int __devinit wm8741_spi_probe(struct spi_device *spi) { struct wm8741_priv *wm8741; - int ret; + int ret, i; wm8741 = devm_kzalloc(&spi->dev, sizeof(struct wm8741_priv), GFP_KERNEL); if (wm8741 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) + wm8741->supplies[i].supply = wm8741_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + wm8741->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + wm8741->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8741); -- cgit v1.2.3 From fe98c0cf40883e7d12456e0abc269e4fa31bed69 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 18:00:21 +0800 Subject: ASoC: wm8741: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 86 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 62 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 742744b4bba7..4281a0802138 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -40,26 +41,43 @@ static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = { /* codec private data */ struct wm8741_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; -static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = { - 0x0000, /* R0 - DACLLSB Attenuation */ - 0x0000, /* R1 - DACLMSB Attenuation */ - 0x0000, /* R2 - DACRLSB Attenuation */ - 0x0000, /* R3 - DACRMSB Attenuation */ - 0x0000, /* R4 - Volume Control */ - 0x000A, /* R5 - Format Control */ - 0x0000, /* R6 - Filter Control */ - 0x0000, /* R7 - Mode Control 1 */ - 0x0002, /* R8 - Mode Control 2 */ - 0x0000, /* R9 - Reset */ - 0x0002, /* R32 - ADDITONAL_CONTROL_1 */ +static const struct reg_default wm8741_reg_defaults[] = { + { 0, 0x0000 }, /* R0 - DACLLSB Attenuation */ + { 1, 0x0000 }, /* R1 - DACLMSB Attenuation */ + { 2, 0x0000 }, /* R2 - DACRLSB Attenuation */ + { 3, 0x0000 }, /* R3 - DACRMSB Attenuation */ + { 4, 0x0000 }, /* R4 - Volume Control */ + { 5, 0x000A }, /* R5 - Format Control */ + { 6, 0x0000 }, /* R6 - Filter Control */ + { 7, 0x0000 }, /* R7 - Mode Control 1 */ + { 8, 0x0002 }, /* R8 - Mode Control 2 */ + { 32, 0x0002 }, /* R32 - ADDITONAL_CONTROL_1 */ }; +static bool wm8741_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8741_DACLLSB_ATTENUATION: + case WM8741_DACLMSB_ATTENUATION: + case WM8741_DACRLSB_ATTENUATION: + case WM8741_DACRMSB_ATTENUATION: + case WM8741_VOLUME_CONTROL: + case WM8741_FORMAT_CONTROL: + case WM8741_FILTER_CONTROL: + case WM8741_MODE_CONTROL_1: + case WM8741_MODE_CONTROL_2: + case WM8741_ADDITIONAL_CONTROL_1: + return true; + default: + return false; + } +} static int wm8741_reset(struct snd_soc_codec *codec) { @@ -411,7 +429,7 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_get; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); goto err_enable; @@ -439,7 +457,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies); err_get: -err: return ret; } @@ -456,9 +473,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8741 = { .probe = wm8741_probe, .remove = wm8741_remove, .resume = wm8741_resume, - .reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8741_reg_defaults, .controls = wm8741_snd_controls, .num_controls = ARRAY_SIZE(wm8741_snd_controls), @@ -474,6 +488,18 @@ static const struct of_device_id wm8741_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8741_of_match); +static const struct regmap_config wm8741_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8741_MAX_REGISTER, + + .reg_defaults = wm8741_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8741_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .readable_reg = wm8741_readable, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8741_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -492,12 +518,18 @@ static int wm8741_i2c_probe(struct i2c_client *i2c, ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), wm8741->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8741->regmap = regmap_init_i2c(i2c, &wm8741_regmap); + if (IS_ERR(wm8741->regmap)) { + ret = PTR_ERR(wm8741->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; } i2c_set_clientdata(i2c, wm8741); - wm8741->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8741, &wm8741_dai, 1); @@ -543,14 +575,20 @@ static int __devinit wm8741_spi_probe(struct spi_device *spi) for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++) wm8741->supplies[i].supply = wm8741_supply_names[i]; - ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies), + ret = devm_regulator_bulk_get(&spi->dev, ARRAY_SIZE(wm8741->supplies), wm8741->supplies); if (ret != 0) { dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); - goto err; + return ret; + } + + wm8741->regmap = regmap_init_spi(spi, &wm8741_regmap); + if (IS_ERR(wm8741->regmap)) { + ret = PTR_ERR(wm8741->regmap); + dev_err(&spi->dev, "Failed to init regmap: %d\n", ret); + return ret; } - wm8741->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8741); ret = snd_soc_register_codec(&spi->dev, -- cgit v1.2.3 From a044b75779201b1eb293ffcb887404357b21421e Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Mon, 10 Sep 2012 17:50:33 +0800 Subject: ASoC: wm8904: remove redundant code The core_intercon is added two times, remove the redundant one Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 0013afe48e66..2b9766d55384 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1185,8 +1185,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets, ARRAY_SIZE(wm8904_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, core_intercon, - ARRAY_SIZE(core_intercon)); snd_soc_dapm_add_routes(dapm, adc_intercon, ARRAY_SIZE(adc_intercon)); snd_soc_dapm_add_routes(dapm, dac_intercon, -- cgit v1.2.3 From 0ebe36c6c4fe8fcfeee3192f37f2ff8318a029bd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 19:23:57 +0800 Subject: ASoC: wm8960: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 107 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 84 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 804f4116912f..7cb0d07ca8a6 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -52,25 +52,72 @@ * We can't read the WM8960 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0000, 0x0000, - 0x0000, 0x0008, 0x0000, 0x000a, - 0x01c0, 0x0000, 0x00ff, 0x00ff, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x007b, 0x0100, 0x0032, - 0x0000, 0x00c3, 0x00c3, 0x01c0, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0100, 0x0100, 0x0050, 0x0050, - 0x0050, 0x0050, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0040, 0x0000, - 0x0000, 0x0050, 0x0050, 0x0000, - 0x0002, 0x0037, 0x004d, 0x0080, - 0x0008, 0x0031, 0x0026, 0x00e9, +static const struct reg_default wm8960_reg_defaults[] = { + { 0x0, 0x0097 }, + { 0x1, 0x0097 }, + { 0x2, 0x0000 }, + { 0x3, 0x0000 }, + { 0x4, 0x0000 }, + { 0x5, 0x0008 }, + { 0x6, 0x0000 }, + { 0x7, 0x000a }, + { 0x8, 0x01c0 }, + { 0x9, 0x0000 }, + { 0xa, 0x00ff }, + { 0xb, 0x00ff }, + + { 0x10, 0x0000 }, + { 0x11, 0x007b }, + { 0x12, 0x0100 }, + { 0x13, 0x0032 }, + { 0x14, 0x0000 }, + { 0x15, 0x00c3 }, + { 0x16, 0x00c3 }, + { 0x17, 0x01c0 }, + { 0x18, 0x0000 }, + { 0x19, 0x0000 }, + { 0x1a, 0x0000 }, + { 0x1b, 0x0000 }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + + { 0x20, 0x0100 }, + { 0x21, 0x0100 }, + { 0x22, 0x0050 }, + + { 0x25, 0x0050 }, + { 0x26, 0x0000 }, + { 0x27, 0x0000 }, + { 0x28, 0x0000 }, + { 0x29, 0x0000 }, + { 0x2a, 0x0040 }, + { 0x2b, 0x0000 }, + { 0x2c, 0x0000 }, + { 0x2d, 0x0050 }, + { 0x2e, 0x0050 }, + { 0x2f, 0x0000 }, + { 0x30, 0x0002 }, + { 0x31, 0x0037 }, + + { 0x33, 0x0080 }, + { 0x34, 0x0008 }, + { 0x35, 0x0031 }, + { 0x36, 0x0026 }, + { 0x37, 0x00e9 }, }; +static bool wm8960_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8960_RESET: + return true; + default: + return false; + } +} + struct wm8960_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); struct snd_soc_dapm_widget *lout1; @@ -555,6 +602,8 @@ static int wm8960_mute(struct snd_soc_dai *dai, int mute) static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -566,7 +615,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, @@ -667,7 +716,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - snd_soc_cache_sync(codec); + regcache_sync(wm8960->regmap); break; default: break; @@ -915,7 +964,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8960->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -963,9 +1012,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8960 = { .suspend = wm8960_suspend, .resume = wm8960_resume, .set_bias_level = wm8960_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8960_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8960_reg, +}; + +static const struct regmap_config wm8960_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8960_PLL4, + + .reg_defaults = wm8960_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8960_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8960_volatile, }; static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, @@ -979,8 +1037,11 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->regmap = regmap_init_i2c(i2c, &wm8960_regmap); + if (IS_ERR(wm8960->regmap)) + return PTR_ERR(wm8960->regmap); + i2c_set_clientdata(i2c, wm8960); - wm8960->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); -- cgit v1.2.3 From 498dab3aa7f8851a880ffe9d8e66516851950c59 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 10 Sep 2012 16:08:40 +0200 Subject: ALSA: hda - Allow 3/5/7 channel map for HDMI/DP Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2c53ea889da9..71555cc54db1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1441,7 +1441,7 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= spec->channels_max; chs += 2) { + for (chs = 2; chs <= spec->channels_max; chs++) { int i, c; struct cea_channel_speaker_allocation *cap; cap = channel_allocations; -- cgit v1.2.3 From 19ace0e97a605042c481c2ea3f7aeb59c0eb54ed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 12:48:11 +0800 Subject: ASoC: cs4270: Conver to data based control init Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/codecs/cs4270.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 44a176f74170..c6e5a73a9f17 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -521,14 +521,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) return ret; } - /* Add the non-DAPM controls */ - ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls, - ARRAY_SIZE(cs4270_snd_controls)); - if (ret < 0) { - dev_err(codec->dev, "failed to add controls\n"); - return ret; - } - /* get the power supply regulators */ for (i = 0; i < ARRAY_SIZE(supply_names); i++) cs4270->supplies[i].supply = supply_names[i]; @@ -634,6 +626,9 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .remove = cs4270_remove, .suspend = cs4270_soc_suspend, .resume = cs4270_soc_resume, + + .controls = cs4270_snd_controls, + .num_controls = ARRAY_SIZE(cs4270_snd_controls), .volatile_register = cs4270_reg_is_volatile, .readable_register = cs4270_reg_is_readable, .reg_cache_size = CS4270_LASTREG + 1, -- cgit v1.2.3 From b61d6d40323997d9da3b95ecce2570f0b782a07f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 12:53:12 +0800 Subject: ASoC: cs4270: Move regulator acquisition to I2C probe() This is better style since it has us obtaining all resources before we try the ASoC probe. This change also fixes a potential issue where we don't enable the regulators before trying to confirm the device ID which could cause a failure during probe in some system configurations. Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/codecs/cs4270.c | 46 +++++++++++++++++++--------------------------- 1 file changed, 19 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index c6e5a73a9f17..64c29b8a379f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -487,7 +487,7 @@ static struct snd_soc_dai_driver cs4270_dai = { static int cs4270_probe(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - int i, ret; + int ret; /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. @@ -521,25 +521,8 @@ static int cs4270_probe(struct snd_soc_codec *codec) return ret; } - /* get the power supply regulators */ - for (i = 0; i < ARRAY_SIZE(supply_names); i++) - cs4270->supplies[i].supply = supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); - if (ret < 0) - return ret; - ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); - if (ret < 0) - goto error_free_regulators; - - return 0; - -error_free_regulators: - regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); return ret; } @@ -555,7 +538,6 @@ static int cs4270_remove(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); - regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; }; @@ -658,7 +640,24 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, { struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; - int ret; + int ret, i; + + cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), + GFP_KERNEL); + if (!cs4270) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* get the power supply regulators */ + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4270->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c_client->dev, + ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + return ret; /* See if we have a way to bring the codec out of reset */ if (np) { @@ -694,13 +693,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, i2c_client->addr); dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); - cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), - GFP_KERNEL); - if (!cs4270) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); - return -ENOMEM; - } - i2c_set_clientdata(i2c_client, cs4270); cs4270->control_type = SND_SOC_I2C; -- cgit v1.2.3 From 1ca6517566b52662f1dea65e3c3d02997282cb01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Sep 2012 12:57:03 +0800 Subject: ASoC: cs4270: Convert to direct regmap API usage Signed-off-by: Mark Brown Acked-by: Timur Tabi --- sound/soc/codecs/cs4270.c | 55 ++++++++++++++++++++++++++++------------------- 1 file changed, 33 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 64c29b8a379f..815b53bc2d27 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -112,14 +112,15 @@ * This array contains the power-on default values of the registers, with the * exception of the "CHIPID" register (01h). The lower four bits of that * register contain the hardware revision, so it is treated as volatile. - * - * Also note that on the CS4270, the first readable register is 1, but ASoC - * assumes the first register is 0. Therfore, the array must have an entry for - * register 0, but we use cs4270_reg_is_readable() to tell ASoC that it can't - * be read. */ -static const u8 cs4270_default_reg_cache[CS4270_LASTREG + 1] = { - 0x00, 0x00, 0x00, 0x30, 0x00, 0x60, 0x20, 0x00, 0x00 +static const struct reg_default cs4270_reg_defaults[] = { + { 2, 0x00 }, + { 3, 0x30 }, + { 4, 0x00 }, + { 5, 0x60 }, + { 6, 0x20 }, + { 7, 0x00 }, + { 8, 0x00 }, }; static const char *supply_names[] = { @@ -128,7 +129,7 @@ static const char *supply_names[] = { /* Private data for the CS4270 */ struct cs4270_private { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; @@ -193,12 +194,12 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -static int cs4270_reg_is_readable(struct snd_soc_codec *codec, unsigned int reg) +static bool cs4270_reg_is_readable(struct device *dev, unsigned int reg) { return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG); } -static int cs4270_reg_is_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool cs4270_reg_is_volatile(struct device *dev, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) @@ -492,7 +493,7 @@ static int cs4270_probe(struct snd_soc_codec *codec) /* Tell ASoC what kind of I/O to use to read the registers. ASoC will * then do the I2C transactions itself. */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs4270->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); return ret; @@ -587,7 +588,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - snd_soc_cache_sync(codec); + regcache_sync(cs4270->regmap); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); @@ -611,11 +612,6 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), - .volatile_register = cs4270_reg_is_volatile, - .readable_register = cs4270_reg_is_readable, - .reg_cache_size = CS4270_LASTREG + 1, - .reg_word_size = sizeof(u8), - .reg_cache_default = cs4270_default_reg_cache, }; /* @@ -627,6 +623,18 @@ static const struct of_device_id cs4270_of_match[] = { }; MODULE_DEVICE_TABLE(of, cs4270_of_match); +static const struct regmap_config cs4270_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4270_LASTREG, + .reg_defaults = cs4270_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4270_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .readable_reg = cs4270_reg_is_readable, + .volatile_reg = cs4270_reg_is_volatile, +}; + /** * cs4270_i2c_probe - initialize the I2C interface of the CS4270 * @i2c_client: the I2C client object @@ -640,6 +648,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, { struct device_node *np = i2c_client->dev.of_node; struct cs4270_private *cs4270; + unsigned int val; int ret, i; cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private), @@ -674,16 +683,19 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, } } - /* Verify that we have a CS4270 */ + cs4270->regmap = devm_regmap_init_i2c(i2c_client, &cs4270_regmap); + if (IS_ERR(cs4270->regmap)) + return PTR_ERR(cs4270->regmap); - ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + /* Verify that we have a CS4270 */ + ret = regmap_read(cs4270->regmap, CS4270_CHIPID, &val); if (ret < 0) { dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n", i2c_client->addr); return ret; } /* The top four bits of the chip ID should be 1100. */ - if ((ret & 0xF0) != 0xC0) { + if ((val & 0xF0) != 0xC0) { dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n", i2c_client->addr); return -ENODEV; @@ -691,10 +703,9 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "found device at i2c address %X\n", i2c_client->addr); - dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); + dev_info(&i2c_client->dev, "hardware revision %X\n", val & 0xF); i2c_set_clientdata(i2c_client, cs4270); - cs4270->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs4270, &cs4270_dai, 1); -- cgit v1.2.3 From 1867b2cdd8b3ce377f7c415ee10ce0cf1e581316 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 10 Sep 2012 15:55:17 +0300 Subject: ASoC: am3517evm: Remove unused cpu_dai from hw_params cpu_dai is not in use in this function and just generates warning at compile time. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/am3517evm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index a997988af147..77a87012977c 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -41,7 +41,6 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; /* Set the codec system clock for DAC and ADC */ -- cgit v1.2.3 From 57d9a477f908cd20c1b4da06fdfe864722487d8b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Sep 2012 13:10:08 -0300 Subject: ASoC: Revert "ASoC: mc13783: Provide codec->control_data" Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap) , it is not necessary to provide codec->control_data anymore. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index bbfa5535cdd3..d89e343ff108 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -585,8 +585,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->mc13xxx; - mc13xxx_lock(priv->mc13xxx); /* these are the reset values */ -- cgit v1.2.3 From 4ac7903f1d2cc3dce289d15ce6a6929e935983f5 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Sep 2012 13:10:09 -0300 Subject: ASoC: Revert "ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements" Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap) , it is not necessary to provide codec->control_data anymore. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3f46bffeb0c3..2c1c2524ef8c 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2404,9 +2404,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); - /* Inform SoC Core that we have our own I/O arrangements. */ - codec->control_data = (void *)true; - /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- cgit v1.2.3 From dbad34eac26f3d31d168486ffb906b9f46657f63 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 10 Sep 2012 13:10:10 -0300 Subject: Revert "ASoC: AC97 doesn't use regmap by default" Since commit 98d3088e5 (SoC: core: Fix check before defaulting to regmap) , it is not necessary to provide codec->control_data anymore. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/stac9766.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - 4 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 11b1b714b8b5..8c39dddd7d00 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,7 +186,6 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 33c0f3d39c87..982e437799a8 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -340,7 +340,6 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); - codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1992a6295a16..4dd73ea08d0b 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -635,7 +635,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - codec->control_data = codec; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index d0b8a3287a85..3eb19fb71d17 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1196,7 +1196,6 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) if (wm9713 == NULL) return -ENOMEM; snd_soc_codec_set_drvdata(codec, wm9713); - codec->control_data = wm9713; /* we don't use regmap! */ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) -- cgit v1.2.3 From 6a58870df89b1941dc9a47e5ccb3c91bffad5b03 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Sep 2012 09:19:19 +0800 Subject: ASoC: wm8900: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 077c9628c70d..5f7a78ea518c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1226,7 +1226,8 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&spi->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1235,15 +1236,13 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static int __devexit wm8900_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1264,7 +1263,8 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, struct wm8900_priv *wm8900; int ret; - wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + wm8900 = devm_kzalloc(&i2c->dev, sizeof(struct wm8900_priv), + GFP_KERNEL); if (wm8900 == NULL) return -ENOMEM; @@ -1273,15 +1273,13 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); - if (ret < 0) - kfree(wm8900); + return ret; } static __devexit int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 499926246ec77d64b028a953f7a79e941e36b802 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Sep 2012 09:25:41 +0800 Subject: ASoC: wm8900: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 148 ++++++++++++++++++++++++++++------------------ 1 file changed, 90 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 5f7a78ea518c..5bc877b916eb 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -137,7 +138,7 @@ #define WM8900_LRC_MASK 0x03ff struct wm8900_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u32 fll_in; /* FLL input frequency */ u32 fll_out; /* FLL output frequency */ @@ -147,54 +148,77 @@ struct wm8900_priv { * wm8900 register cache. We can't read the entire register space and we * have slow control buses so we cache the registers. */ -static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { - 0x8900, 0x0000, - 0xc000, 0x0000, - 0x4050, 0x4000, - 0x0008, 0x0000, - 0x0040, 0x0040, - 0x1004, 0x00c0, - 0x00c0, 0x0000, - 0x0100, 0x00c0, - 0x00c0, 0x0000, - 0xb001, 0x0000, - 0x0000, 0x0044, - 0x004c, 0x004c, - 0x0044, 0x0044, - 0x0000, 0x0044, - 0x0000, 0x0000, - 0x0002, 0x0000, - 0x0000, 0x0000, - 0x0000, 0x0000, - 0x0008, 0x0000, - 0x0000, 0x0008, - 0x0097, 0x0100, - 0x0000, 0x0000, - 0x0050, 0x0050, - 0x0055, 0x0055, - 0x0055, 0x0000, - 0x0000, 0x0079, - 0x0079, 0x0079, - 0x0079, 0x0000, - /* Remaining registers all zero */ +static const struct reg_default wm8900_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0xc000 }, + { 3, 0x0000 }, + { 4, 0x4050 }, + { 5, 0x4000 }, + { 6, 0x0008 }, + { 7, 0x0000 }, + { 8, 0x0040 }, + { 9, 0x0040 }, + { 10, 0x1004 }, + { 11, 0x00c0 }, + { 12, 0x00c0 }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00c0 }, + { 16, 0x00c0 }, + { 17, 0x0000 }, + { 18, 0xb001 }, + { 19, 0x0000 }, + { 20, 0x0000 }, + { 21, 0x0044 }, + { 22, 0x004c }, + { 23, 0x004c }, + { 24, 0x0044 }, + { 25, 0x0044 }, + { 26, 0x0000 }, + { 27, 0x0044 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0002 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0000 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x0000 }, + { 38, 0x0000 }, + { 39, 0x0008 }, + { 40, 0x0097 }, + { 41, 0x0100 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0050 }, + { 45, 0x0050 }, + { 46, 0x0055 }, + { 47, 0x0055 }, + { 48, 0x0055 }, + { 49, 0x0000 }, + { 50, 0x0000 }, + { 51, 0x0079 }, + { 52, 0x0079 }, + { 53, 0x0079 }, + { 54, 0x0079 }, + { 55, 0x0000 }, }; -static int wm8900_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8900_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8900_REG_ID: - return 1; + return true; default: - return 0; + return false; } } static void wm8900_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8900_REG_RESET, 0); - - memcpy(codec->reg_cache, wm8900_reg_defaults, - sizeof(wm8900_reg_defaults)); } static int wm8900_hp_event(struct snd_soc_dapm_widget *w, @@ -1119,13 +1143,16 @@ static int wm8900_suspend(struct snd_soc_codec *codec) static int wm8900_resume(struct snd_soc_codec *codec) { struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); - u16 *cache; - int i, ret; - - cache = kmemdup(codec->reg_cache, sizeof(wm8900_reg_defaults), - GFP_KERNEL); + int ret; wm8900_reset(codec); + + ret = regcache_sync(wm8900->regmap); + if (ret != 0) { + dev_err(codec->dev, "Failed to restore cache: %d\n", ret); + return ret; + } + wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Restart the FLL? */ @@ -1139,27 +1166,18 @@ static int wm8900_resume(struct snd_soc_codec *codec) ret = wm8900_set_fll(codec, 0, fll_in, fll_out); if (ret != 0) { dev_err(codec->dev, "Failed to restart FLL\n"); - kfree(cache); return ret; } } - if (cache) { - for (i = 0; i < WM8900_MAXREG; i++) - snd_soc_write(codec, i, cache[i]); - kfree(cache); - } else - dev_err(codec->dev, "Unable to allocate register cache\n"); - return 0; } static int wm8900_probe(struct snd_soc_codec *codec) { - struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8900->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1207,10 +1225,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .suspend = wm8900_suspend, .resume = wm8900_resume, .set_bias_level = wm8900_set_bias_level, - .volatile_register = wm8900_volatile_register, - .reg_cache_size = ARRAY_SIZE(wm8900_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8900_reg_defaults, .controls = wm8900_snd_controls, .num_controls = ARRAY_SIZE(wm8900_snd_controls), @@ -1220,6 +1234,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8900 = { .num_dapm_routes = ARRAY_SIZE(wm8900_dapm_routes), }; +static const struct regmap_config wm8900_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8900_MAXREG, + + .reg_defaults = wm8900_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8900_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8900_volatile_register, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8900_spi_probe(struct spi_device *spi) { @@ -1231,7 +1257,10 @@ static int __devinit wm8900_spi_probe(struct spi_device *spi) if (wm8900 == NULL) return -ENOMEM; - wm8900->control_type = SND_SOC_SPI; + wm8900->regmap = devm_regmap_init_spi(spi, &wm8900_regmap); + if (IS_ERR(wm8900->regmap)) + return PTR_ERR(wm8900->regmap); + spi_set_drvdata(spi, wm8900); ret = snd_soc_register_codec(&spi->dev, @@ -1268,8 +1297,11 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, if (wm8900 == NULL) return -ENOMEM; + wm8900->regmap = devm_regmap_init_i2c(i2c, &wm8900_regmap); + if (IS_ERR(wm8900->regmap)) + return PTR_ERR(wm8900->regmap); + i2c_set_clientdata(i2c, wm8900); - wm8900->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8900, &wm8900_dai, 1); -- cgit v1.2.3 From 7e94ca4752991c3830515f6e17ee5a7334a7f590 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Sep 2012 09:26:09 +0800 Subject: ASoC: wm8900: Fix typo of name to wm9700 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 5bc877b916eb..e781f865e5d7 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -493,10 +493,10 @@ SOC_DAPM_SINGLE("RINPUT2 Switch", WM8900_REG_INCTL, 1, 1, 0), SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), }; -static const char *wm9700_lp_mux[] = { "Disabled", "Enabled" }; +static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" }; static const struct soc_enum wm8900_lineout2_lp_mux = -SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm9700_lp_mux); +SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux); static const struct snd_kcontrol_new wm8900_lineout2_lp = SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); -- cgit v1.2.3 From 2dc6fbf0078e8148b571f1bbec704ca5c8e2ec2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Sep 2012 14:24:43 +0200 Subject: ALSA: pcm - Use UNKNOWN chmap for mono streams In general, mono streams have no dedicated speaker assignment, thus they should be rather marked as UNKNOWN position. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0a750ec5c400..9647a22ed69a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2311,7 +2311,7 @@ EXPORT_SYMBOL(snd_pcm_lib_readv); /* default channel maps for multi-channel playbacks, up to 8 channels */ const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[] = { { .channels = 1, - .map = { SNDRV_CHMAP_FC } }, + .map = { SNDRV_CHMAP_UNKNOWN } }, { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, @@ -2333,7 +2333,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_std_chmaps); /* alternative channel maps with CLFE <-> surround swapped for 6/8 channels */ const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[] = { { .channels = 1, - .map = { SNDRV_CHMAP_FC } }, + .map = { SNDRV_CHMAP_UNKNOWN } }, { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, -- cgit v1.2.3 From 915bf29eb94beabed8bf577bfd62fc0ba9abbf37 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Sep 2012 15:19:10 +0200 Subject: ALSA: hda - Avoid BDL position workaround when no_period_wakeup is set Originally the bogus period at BDL head was introduced as a workaround for the mismatching position update at the period boundary, typically seen on dmix. However, for applications like PulseAudio that don't require period wake ups, this workaround is just superfluous. Thus better to disable it when no_period_wakeup is given in hw_params. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6a19f6af9517..76ec2421fae1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,7 @@ struct azx_dev { */ unsigned int insufficient :1; unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; }; /* CORB/RIRB */ @@ -1419,7 +1420,7 @@ static int azx_setup_periods(struct azx *chip, ofs = 0; azx_dev->frags = 0; pos_adj = bdl_pos_adj[chip->dev_index]; - if (pos_adj > 0) { + if (!azx_dev->no_period_wakeup && pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; int pos_align = pos_adj; pos_adj = (pos_adj * runtime->rate + 47999) / 48000; @@ -1435,8 +1436,7 @@ static int azx_setup_periods(struct azx *chip, pos_adj = 0; } else { ofs = setup_bdle(chip, substream, azx_dev, - &bdl, ofs, pos_adj, - !substream->runtime->no_period_wakeup); + &bdl, ofs, pos_adj, true); if (ofs < 0) goto error; } @@ -1449,7 +1449,7 @@ static int azx_setup_periods(struct azx *chip, else ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs, period_bytes, - !substream->runtime->no_period_wakeup); + !azx_dev->no_period_wakeup); if (ofs < 0) goto error; } @@ -1922,10 +1922,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || - format_val != azx_dev->format_val) { + format_val != azx_dev->format_val || + runtime->no_period_wakeup != azx_dev->no_period_wakeup) { azx_dev->bufsize = bufsize; azx_dev->period_bytes = period_bytes; azx_dev->format_val = format_val; + azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) return err; -- cgit v1.2.3 From a33b7b0a89a240a1416fa489f0b805488bfb9afd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Sep 2012 16:42:18 +0200 Subject: ALSA: hda - Check bit mask for codec SSID in snd_hda_pick_fixup() snd_hda_pick_fixup() didn't check the case where the device mask bits are set, typically used for SND_PCI_QUIRK_VENDOR() entries. Fix this. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 9acd5a964eb6..a98e25ebfd63 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -735,7 +735,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, for (q = quirk; q->subvendor; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); - if (vendorid == codec->subsystem_id) { + unsigned int mask = 0xffff0000 | q->subdevice_mask; + if ((codec->subsystem_id & mask) == (vendorid & mask)) { id = q->value; #ifdef CONFIG_SND_DEBUG_VERBOSE name = q->name; -- cgit v1.2.3 From b35aabd78ddae8300fda7e6a1ac6f0484ef3e804 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Aug 2012 19:09:45 +0200 Subject: ALSA: hda - Replace with the generic fixup codes in patch_cirrus.c ... to make easier to integrate into the common generic parser in near future. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 193 +++++++++++++++++++++++-------------------- 1 file changed, 103 insertions(+), 90 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0bddb3ee1101..6aca4a05a329 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -34,7 +34,8 @@ */ struct cs_spec { - int board_config; + struct hda_gen_spec gen; + struct auto_pin_cfg autocfg; struct hda_multi_out multiout; struct snd_kcontrol *vmaster_sw; @@ -80,16 +81,17 @@ enum { CS420X_MBP53, CS420X_MBP55, CS420X_IMAC27, - CS420X_IMAC27_122, - CS420X_APPLE, + CS420X_GPIO_13, + CS420X_GPIO_23, + CS420X_IMAC27_122 = CS420X_GPIO_23, + CS420X_APPLE = CS420X_GPIO_13, CS420X_AUTO, - CS420X_MODELS }; /* CS421x boards */ enum { CS421X_CDB4210, - CS421X_MODELS + CS421X_SENSE_B, }; /* Vendor-specific processing widget */ @@ -1278,38 +1280,30 @@ static int cs_parse_auto_config(struct hda_codec *codec) return 0; } -static const char * const cs420x_models[CS420X_MODELS] = { - [CS420X_MBP53] = "mbp53", - [CS420X_MBP55] = "mbp55", - [CS420X_IMAC27] = "imac27", - [CS420X_IMAC27_122] = "imac27_122", - [CS420X_APPLE] = "apple", - [CS420X_AUTO] = "auto", +static const struct hda_model_fixup cs420x_models[] = { + { .id = CS420X_MBP53, .name = "mbp53" }, + { .id = CS420X_MBP55, .name = "mbp55" }, + { .id = CS420X_IMAC27, .name = "imac27" }, + { .id = CS420X_IMAC27_122, .name = "imac27_122" }, + { .id = CS420X_APPLE, .name = "apple" }, + {} }; - -static const struct snd_pci_quirk cs420x_cfg_tbl[] = { +static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), /* this conflicts with too many other models */ /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ - {} /* terminator */ -}; -static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = { + /* codec SSID */ SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; -struct cs_pincfg { - hda_nid_t nid; - u32 val; -}; - -static const struct cs_pincfg mbp53_pincfgs[] = { +static const struct hda_pintbl mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1323,7 +1317,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static const struct hda_pintbl mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1337,7 +1331,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static const struct hda_pintbl imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1351,22 +1345,59 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { - [CS420X_MBP53] = mbp53_pincfgs, - [CS420X_MBP55] = mbp55_pincfgs, - [CS420X_IMAC27] = imac27_pincfgs, -}; +static void cs420x_fixup_gpio_13(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct cs_spec *spec = codec->spec; + spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */ + spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; + } +} -static void fix_pincfg(struct hda_codec *codec, int model, - const struct cs_pincfg **pin_configs) +static void cs420x_fixup_gpio_23(struct hda_codec *codec, + const struct hda_fixup *fix, int action) { - const struct cs_pincfg *cfg = pin_configs[model]; - if (!cfg) - return; - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct cs_spec *spec = codec->spec; + spec->gpio_eapd_hp = 4; /* GPIO2 = headphones */ + spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; + } } +static const struct hda_fixup cs420x_fixups[] = { + [CS420X_MBP53] = { + .type = HDA_FIXUP_PINS, + .v.pins = mbp53_pincfgs, + .chained = true, + .chain_id = CS420X_APPLE, + }, + [CS420X_MBP55] = { + .type = HDA_FIXUP_PINS, + .v.pins = mbp55_pincfgs, + .chained = true, + .chain_id = CS420X_GPIO_13, + }, + [CS420X_IMAC27] = { + .type = HDA_FIXUP_PINS, + .v.pins = imac27_pincfgs, + .chained = true, + .chain_id = CS420X_GPIO_13, + }, + [CS420X_GPIO_13] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs420x_fixup_gpio_13, + }, + [CS420X_GPIO_23] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs420x_fixup_gpio_23, + }, +}; + static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1379,33 +1410,9 @@ static int patch_cs420x(struct hda_codec *codec) spec->vendor_nid = CS420X_VENDOR_NID; - spec->board_config = - snd_hda_check_board_config(codec, CS420X_MODELS, - cs420x_models, cs420x_cfg_tbl); - if (spec->board_config < 0) - spec->board_config = - snd_hda_check_board_codec_sid_config(codec, - CS420X_MODELS, NULL, cs420x_codec_cfg_tbl); - if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config, cs_pincfgs); - - switch (spec->board_config) { - case CS420X_IMAC27: - case CS420X_MBP53: - case CS420X_MBP55: - case CS420X_APPLE: - spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */ - spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ - spec->gpio_mask = spec->gpio_dir = - spec->gpio_eapd_hp | spec->gpio_eapd_speaker; - break; - case CS420X_IMAC27_122: - spec->gpio_eapd_hp = 4; /* GPIO2 = headphones */ - spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ - spec->gpio_mask = spec->gpio_dir = - spec->gpio_eapd_hp | spec->gpio_eapd_speaker; - break; - } + snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, + cs420x_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); err = cs_parse_auto_config(codec); if (err < 0) @@ -1413,6 +1420,8 @@ static int patch_cs420x(struct hda_codec *codec) codec->patch_ops = cs_patch_ops; + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: @@ -1430,11 +1439,12 @@ static int patch_cs420x(struct hda_codec *codec) */ /* CS4210 board names */ -static const char *cs421x_models[CS421X_MODELS] = { - [CS421X_CDB4210] = "cdb4210", +static const struct hda_model_fixup cs421x_models[] = { + { .id = CS421X_CDB4210, .name = "cdb4210" }, + {} }; -static const struct snd_pci_quirk cs421x_cfg_tbl[] = { +static const struct snd_pci_quirk cs421x_fixup_tbl[] = { /* Test Intel board + CDB2410 */ SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), {} /* terminator */ @@ -1442,7 +1452,7 @@ static const struct snd_pci_quirk cs421x_cfg_tbl[] = { /* CS4210 board pinconfigs */ /* Default CS4210 (CDB4210)*/ -static const struct cs_pincfg cdb4210_pincfgs[] = { +static const struct hda_pintbl cdb4210_pincfgs[] = { { 0x05, 0x0321401f }, { 0x06, 0x90170010 }, { 0x07, 0x03813031 }, @@ -1452,8 +1462,26 @@ static const struct cs_pincfg cdb4210_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { - [CS421X_CDB4210] = cdb4210_pincfgs, +/* Setup GPIO/SENSE for each board (if used) */ +static void cs421x_fixup_sense_b(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct cs_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->sense_b = 1; +} + +static const struct hda_fixup cs421x_fixups[] = { + [CS421X_CDB4210] = { + .type = HDA_FIXUP_PINS, + .v.pins = cdb4210_pincfgs, + .chained = true, + .chain_id = CS421X_SENSE_B, + }, + [CS421X_SENSE_B] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs421x_fixup_sense_b, + } }; static const struct hda_verb cs421x_coef_init_verbs[] = { @@ -1935,26 +1963,9 @@ static int patch_cs4210(struct hda_codec *codec) spec->vendor_nid = CS4210_VENDOR_NID; - spec->board_config = - snd_hda_check_board_config(codec, CS421X_MODELS, - cs421x_models, cs421x_cfg_tbl); - if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config, cs421x_pincfgs); - /* - Setup GPIO/SENSE for each board (if used) - */ - switch (spec->board_config) { - case CS421X_CDB4210: - snd_printd("CS4210 board: %s\n", - cs421x_models[spec->board_config]); -/* spec->gpio_mask = 3; - spec->gpio_dir = 3; - spec->gpio_data = 3; -*/ - spec->sense_b = 1; - - break; - } + snd_hda_pick_fixup(codec, cs421x_models, cs421x_fixup_tbl, + cs421x_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); /* Update the GPIO/DMIC/SENSE_B pinmux before the configuration @@ -1969,6 +1980,8 @@ static int patch_cs4210(struct hda_codec *codec) codec->patch_ops = cs421x_patch_ops; + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: -- cgit v1.2.3 From ef596a57b4d7d8b258beb570ed309ef85bf24dd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Sep 2012 16:53:08 +0200 Subject: ALSA: hda - Add support for MacBook Pro 10,1 MacBook Pro 10,1 needs a few adjustments to make it working: - more COEF verbs - some pin config overrides to disable the unused pin (0x0d, 0x12), and fix the internal mic (0x0e) In addition, it uses GPIO 1 and 3 like other MacBooks. The internal digital mic on the machine is still problematic: it seems that only the right channel is used and the left is always static. This looks like a hardware design, so we need to cope in the software side somehow... The primary information and test were brought from Daniel J Blueman. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 6aca4a05a329..82253f137359 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -85,6 +85,8 @@ enum { CS420X_GPIO_23, CS420X_IMAC27_122 = CS420X_GPIO_23, CS420X_APPLE = CS420X_GPIO_13, + CS420X_MBP101, + CS420X_MBP101_COEF, CS420X_AUTO, }; @@ -1159,6 +1161,14 @@ static const struct hda_verb cs_errata_init_verbs[] = { {} /* terminator */ }; +static const struct hda_verb mbp101_init_verbs[] = { + {0x11, AC_VERB_SET_COEF_INDEX, 0x0002}, + {0x11, AC_VERB_SET_PROC_COEF, 0x100a}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0004}, + {0x11, AC_VERB_SET_PROC_COEF, 0x000f}, + {} +}; + /* SPDIF setup */ static void init_digital(struct hda_codec *codec) { @@ -1286,6 +1296,7 @@ static const struct hda_model_fixup cs420x_models[] = { { .id = CS420X_IMAC27, .name = "imac27" }, { .id = CS420X_IMAC27_122, .name = "imac27_122" }, { .id = CS420X_APPLE, .name = "apple" }, + { .id = CS420X_MBP101, .name = "mbp101" }, {} }; @@ -1299,6 +1310,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /* codec SSID */ SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), + SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -1345,6 +1357,13 @@ static const struct hda_pintbl imac27_pincfgs[] = { {} /* terminator */ }; +static const struct hda_pintbl mbp101_pincfgs[] = { + { 0x0d, 0x40ab90f0 }, + { 0x0e, 0x90a600f0 }, + { 0x12, 0x50a600f0 }, + {} /* terminator */ +}; + static void cs420x_fixup_gpio_13(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1396,6 +1415,18 @@ static const struct hda_fixup cs420x_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs420x_fixup_gpio_23, }, + [CS420X_MBP101] = { + .type = HDA_FIXUP_PINS, + .v.pins = mbp101_pincfgs, + .chained = true, + .chain_id = CS420X_MBP101_COEF, + }, + [CS420X_MBP101_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = mbp101_init_verbs, + .chained = true, + .chain_id = CS420X_GPIO_13, + }, }; static int patch_cs420x(struct hda_codec *codec) -- cgit v1.2.3 From 3217b0f5b6fd91440fd72cf24a8986b3f99d0d84 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:24:12 +0800 Subject: ASoC: wm8510: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 56a049555e2c..98f28a006faf 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -592,7 +592,8 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&spi->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -601,8 +602,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } @@ -630,7 +630,8 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, struct wm8510_priv *wm8510; int ret; - wm8510 = kzalloc(sizeof(struct wm8510_priv), GFP_KERNEL); + wm8510 = devm_kzalloc(&i2c->dev, sizeof(struct wm8510_priv), + GFP_KERNEL); if (wm8510 == NULL) return -ENOMEM; @@ -639,8 +640,7 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); - if (ret < 0) - kfree(wm8510); + return ret; } -- cgit v1.2.3 From 398c02f6c213c5d0a791ebf9517b6e7029dc5cf0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:25:59 +0800 Subject: ASoC: wm8580: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 7c68226376e4..cc198df15e6e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -899,7 +899,8 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, struct wm8580_priv *wm8580; int ret; - wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); + wm8580 = devm_kzalloc(&i2c->dev, sizeof(struct wm8580_priv), + GFP_KERNEL); if (wm8580 == NULL) return -ENOMEM; @@ -908,15 +909,13 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); - if (ret < 0) - kfree(wm8580); + return ret; } static int wm8580_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From e908ef40e4824a000889b0ab3f9eb9660bbe3f18 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:26:12 +0800 Subject: ASoC: wm8711: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 0b76d1dca5ea..1b1209a4c791 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -414,7 +414,8 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) struct wm8711_priv *wm8711; int ret; - wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + wm8711 = devm_kzalloc(&spi->dev, sizeof(struct wm8711_priv), + GFP_KERNEL); if (wm8711 == NULL) return -ENOMEM; @@ -423,15 +424,14 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); - if (ret < 0) - kfree(wm8711); + return ret; } static int __devexit wm8711_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + return 0; } @@ -453,7 +453,8 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, struct wm8711_priv *wm8711; int ret; - wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + wm8711 = devm_kzalloc(&client->dev, sizeof(struct wm8711_priv), + GFP_KERNEL); if (wm8711 == NULL) return -ENOMEM; @@ -462,15 +463,13 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); - if (ret < 0) - kfree(wm8711); + return ret; } static __devexit int wm8711_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 1a9585b0f7b3e21858f9a893d66a845f23d28ef1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:26:25 +0800 Subject: ASoC: wm8728: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 1467f97dce21..f274c250d492 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -280,7 +280,8 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) struct wm8728_priv *wm8728; int ret; - wm8728 = kzalloc(sizeof(struct wm8728_priv), GFP_KERNEL); + wm8728 = devm_kzalloc(&spi->dev, sizeof(struct wm8728_priv), + GFP_KERNEL); if (wm8728 == NULL) return -ENOMEM; @@ -289,15 +290,14 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); - if (ret < 0) - kfree(wm8728); + return ret; } static int __devexit wm8728_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + return 0; } @@ -319,7 +319,8 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, struct wm8728_priv *wm8728; int ret; - wm8728 = kzalloc(sizeof(struct wm8728_priv), GFP_KERNEL); + wm8728 = devm_kzalloc(&i2c->dev, sizeof(struct wm8728_priv), + GFP_KERNEL); if (wm8728 == NULL) return -ENOMEM; @@ -328,15 +329,13 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); - if (ret < 0) - kfree(wm8728); + return ret; } static __devexit int wm8728_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3 From 65fdd9bffa1367b75de0c331a105ce36de618794 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:26:38 +0800 Subject: ASoC: wm8737: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index d0520124616d..8c25442f9b47 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -647,7 +647,8 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, struct wm8737_priv *wm8737; int ret; - wm8737 = kzalloc(sizeof(struct wm8737_priv), GFP_KERNEL); + wm8737 = devm_kzalloc(&i2c->dev, sizeof(struct wm8737_priv), + GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; @@ -656,8 +657,7 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8737, &wm8737_dai, 1); - if (ret < 0) - kfree(wm8737); + return ret; } @@ -665,7 +665,7 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, static __devexit int wm8737_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } @@ -693,7 +693,8 @@ static int __devinit wm8737_spi_probe(struct spi_device *spi) struct wm8737_priv *wm8737; int ret; - wm8737 = kzalloc(sizeof(struct wm8737_priv), GFP_KERNEL); + wm8737 = devm_kzalloc(&spi->dev, sizeof(struct wm8737_priv), + GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; @@ -702,15 +703,14 @@ static int __devinit wm8737_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8737, &wm8737_dai, 1); - if (ret < 0) - kfree(wm8737); + return ret; } static int __devexit wm8737_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + return 0; } -- cgit v1.2.3 From 587cbbb36ef2657cd888b7705e02cfe96ab088b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:26:53 +0800 Subject: ASoC: wm8990: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index db63c97ddf51..c28c83e5395d 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1388,7 +1388,8 @@ static __devinit int wm8990_i2c_probe(struct i2c_client *i2c, struct wm8990_priv *wm8990; int ret; - wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL); + wm8990 = devm_kzalloc(&i2c->dev, sizeof(struct wm8990_priv), + GFP_KERNEL); if (wm8990 == NULL) return -ENOMEM; @@ -1396,15 +1397,14 @@ static __devinit int wm8990_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8990, &wm8990_dai, 1); - if (ret < 0) - kfree(wm8990); + return ret; } static __devexit int wm8990_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3 From 046d4f02e8835ff78f8ba5a09e358b2bc4832903 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 09:28:08 +0800 Subject: ASoC: wm8991: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index b9dbfebbda13..fe439f027e10 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1363,7 +1363,7 @@ static __devinit int wm8991_i2c_probe(struct i2c_client *i2c, struct wm8991_priv *wm8991; int ret; - wm8991 = kzalloc(sizeof *wm8991, GFP_KERNEL); + wm8991 = devm_kzalloc(&i2c->dev, sizeof(*wm8991), GFP_KERNEL); if (!wm8991) return -ENOMEM; @@ -1372,15 +1372,14 @@ static __devinit int wm8991_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8991, &wm8991_dai, 1); - if (ret < 0) - kfree(wm8991); + return ret; } static __devexit int wm8991_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3 From e643049d301142cda473bc4d7f4eba4992fe657c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 11:43:44 +0800 Subject: ASoC: wm8510: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8510.c | 117 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 92 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 98f28a006faf..c12a54e72e89 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -33,24 +34,75 @@ * We can't read the WM8510 register space when we are * using 2 wire for device control, so we cache them instead. */ -static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0050, 0x0000, 0x0140, 0x0000, - 0x0000, 0x0000, 0x0000, 0x00ff, - 0x0000, 0x0000, 0x0100, 0x00ff, - 0x0000, 0x0000, 0x012c, 0x002c, - 0x002c, 0x002c, 0x002c, 0x0000, - 0x0032, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0038, 0x000b, 0x0032, 0x0000, - 0x0008, 0x000c, 0x0093, 0x00e9, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0003, 0x0010, 0x0000, 0x0000, - 0x0000, 0x0002, 0x0001, 0x0000, - 0x0000, 0x0000, 0x0039, 0x0000, - 0x0001, +static const struct reg_default wm8510_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0x0000 }, + { 3, 0x0000 }, + { 4, 0x0050 }, + { 5, 0x0000 }, + { 6, 0x0140 }, + { 7, 0x0000 }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x0000 }, + { 11, 0x00ff }, + { 12, 0x0000 }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00ff }, + { 16, 0x0000 }, + { 17, 0x0000 }, + { 18, 0x012c }, + { 19, 0x002c }, + { 20, 0x002c }, + { 21, 0x002c }, + { 22, 0x002c }, + { 23, 0x0000 }, + { 24, 0x0032 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0038 }, + { 33, 0x000b }, + { 34, 0x0032 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x000c }, + { 38, 0x0093 }, + { 39, 0x00e9 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0003 }, + { 45, 0x0010 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0002 }, + { 50, 0x0001 }, + { 51, 0x0000 }, + { 52, 0x0000 }, + { 53, 0x0000 }, + { 54, 0x0039 }, + { 55, 0x0000 }, + { 56, 0x0001 }, }; +static bool wm8510_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8510_RESET: + return true; + default: + return false; + } +} + #define WM8510_POWER1_BIASEN 0x08 #define WM8510_POWER1_BUFIOEN 0x10 @@ -58,7 +110,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { /* codec private data */ struct wm8510_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; @@ -454,6 +506,7 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { @@ -467,7 +520,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8510->regmap); /* Initial cap charge at VMID 5k */ snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); @@ -536,10 +589,9 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - struct wm8510_priv *wm8510 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8510->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); return ret; @@ -569,9 +621,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = { .suspend = wm8510_suspend, .resume = wm8510_resume, .set_bias_level = wm8510_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8510_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default =wm8510_reg, .controls = wm8510_snd_controls, .num_controls = ARRAY_SIZE(wm8510_snd_controls), @@ -586,6 +635,18 @@ static const struct of_device_id wm8510_of_match[] = { { }, }; +static const struct regmap_config wm8510_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8510_MONOMIX, + + .reg_defaults = wm8510_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8510_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8510_volatile, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8510_spi_probe(struct spi_device *spi) { @@ -597,7 +658,10 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) if (wm8510 == NULL) return -ENOMEM; - wm8510->control_type = SND_SOC_SPI; + wm8510->regmap = devm_regmap_init_spi(spi, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + spi_set_drvdata(spi, wm8510); ret = snd_soc_register_codec(&spi->dev, @@ -635,8 +699,11 @@ static __devinit int wm8510_i2c_probe(struct i2c_client *i2c, if (wm8510 == NULL) return -ENOMEM; + wm8510->regmap = devm_regmap_init_i2c(i2c, &wm8510_regmap); + if (IS_ERR(wm8510->regmap)) + return PTR_ERR(wm8510->regmap); + i2c_set_clientdata(i2c, wm8510); - wm8510->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8510, &wm8510_dai, 1); -- cgit v1.2.3 From b689d9f9961befd9b322783f512195785fe82daa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 11:59:51 +0800 Subject: ASoC: wm8580: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 116 +++++++++++++++++++++++++++++++++++----------- 1 file changed, 90 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cc198df15e6e..02c75bec7d71 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008-11 Wolfson Microelectronics PLC. + * Copyright 2008-12 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -157,23 +158,72 @@ * We can't read the WM8580 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8580_reg[] = { - 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ - 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ - 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ - 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ - 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R27*/ - 0x01f0, 0x0040, 0x0000, 0x0000, /*R31(0x1F)*/ - 0x0000, 0x0000, 0x0031, 0x000b, /*R35*/ - 0x0039, 0x0000, 0x0010, 0x0032, /*R39*/ - 0x0054, 0x0076, 0x0098, 0x0000, /*R43(0x2B)*/ - 0x0000, 0x0000, 0x0000, 0x0000, /*R47*/ - 0x0000, 0x0000, 0x005e, 0x003e, /*R51(0x33)*/ - 0x0000, 0x0000 /*R53*/ +static const struct reg_default wm8580_reg_defaults[] = { + { 0, 0x0121 }, + { 1, 0x017e }, + { 2, 0x007d }, + { 3, 0x0014 }, + { 4, 0x0121 }, + { 5, 0x017e }, + { 6, 0x007d }, + { 7, 0x0194 }, + { 8, 0x0010 }, + { 9, 0x0002 }, + { 10, 0x0002 }, + { 11, 0x00c2 }, + { 12, 0x0182 }, + { 13, 0x0082 }, + { 14, 0x000a }, + { 15, 0x0024 }, + { 16, 0x0009 }, + { 17, 0x0000 }, + { 18, 0x00ff }, + { 19, 0x0000 }, + { 20, 0x00ff }, + { 21, 0x00ff }, + { 22, 0x00ff }, + { 23, 0x00ff }, + { 24, 0x00ff }, + { 25, 0x00ff }, + { 26, 0x00ff }, + { 27, 0x00ff }, + { 28, 0x01f0 }, + { 29, 0x0040 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0031 }, + { 35, 0x000b }, + { 36, 0x0039 }, + { 37, 0x0000 }, + { 38, 0x0010 }, + { 39, 0x0032 }, + { 40, 0x0054 }, + { 41, 0x0076 }, + { 42, 0x0098 }, + { 43, 0x0000 }, + { 44, 0x0000 }, + { 45, 0x0000 }, + { 46, 0x0000 }, + { 47, 0x0000 }, + { 48, 0x0000 }, + { 49, 0x0000 }, + { 50, 0x005e }, + { 51, 0x003e }, + { 52, 0x0000 }, }; +static bool wm8580_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8580_RESET: + return true; + default: + return false; + } +} + struct pll_state { unsigned int in; unsigned int out; @@ -188,7 +238,7 @@ static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { /* codec private data */ struct wm8580_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; struct pll_state a; struct pll_state b; @@ -203,14 +253,16 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; + struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); unsigned int reg = mc->reg; unsigned int reg2 = mc->rreg; int ret; - /* Clear the register cache so we write without VU set */ - reg_cache[reg] = 0; - reg_cache[reg2] = 0; + /* Clear the register cache VU so we write without VU set */ + regcache_cache_only(wm8580->regmap, true); + regmap_update_bits(wm8580->regmap, reg, 0x100, 0x000); + regmap_update_bits(wm8580->regmap, reg2, 0x100, 0x000); + regcache_cache_only(wm8580->regmap, false); ret = snd_soc_put_volsw(kcontrol, ucontrol); if (ret < 0) @@ -817,7 +869,7 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0,i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8580->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -875,9 +927,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = { .probe = wm8580_probe, .remove = wm8580_remove, .set_bias_level = wm8580_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8580_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8580_reg, .controls = wm8580_snd_controls, .num_controls = ARRAY_SIZE(wm8580_snd_controls), @@ -892,6 +941,18 @@ static const struct of_device_id wm8580_of_match[] = { { }, }; +static const struct regmap_config wm8580_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8580_MAX_REGISTER, + + .reg_defaults = wm8580_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8580_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8580_volatile, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -904,8 +965,11 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, if (wm8580 == NULL) return -ENOMEM; + wm8580->regmap = devm_regmap_init_i2c(i2c, &wm8580_regmap); + if (IS_ERR(wm8580->regmap)) + return PTR_ERR(wm8580->regmap); + i2c_set_clientdata(i2c, wm8580); - wm8580->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8580, wm8580_dai, ARRAY_SIZE(wm8580_dai)); -- cgit v1.2.3 From 18273b05def692d1883745d22e72fa275bc92c17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:03:51 +0800 Subject: ASoC: wm8580: Move regulator acquisition to I2C probe Better style as we get all the resources we need prior to starting the ASoC level probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 02c75bec7d71..5e9c40fa7eb2 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -867,7 +867,7 @@ static struct snd_soc_dai_driver wm8580_dai[] = { static int wm8580_probe(struct snd_soc_codec *codec) { struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); - int ret = 0,i; + int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { @@ -875,16 +875,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++) - wm8580->supplies[i].supply = wm8580_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8580->supplies), - wm8580->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); if (ret != 0) { @@ -906,7 +896,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) err_regulator_enable: regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); err_regulator_get: - regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); return ret; } @@ -918,7 +907,6 @@ static int wm8580_remove(struct snd_soc_codec *codec) wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); - regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); return 0; } @@ -958,7 +946,7 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8580_priv *wm8580; - int ret; + int ret, i; wm8580 = devm_kzalloc(&i2c->dev, sizeof(struct wm8580_priv), GFP_KERNEL); @@ -969,6 +957,16 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8580->regmap)) return PTR_ERR(wm8580->regmap); + for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++) + wm8580->supplies[i].supply = wm8580_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8580->supplies), + wm8580->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8580); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From 5aa5fa9fdbda20eee3351dd2492cd5c0d6fa7d1d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:15:20 +0800 Subject: ASoC: wm8711: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 50 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 38 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 1b1209a4c791..8b8bb70f1eb9 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -32,7 +33,7 @@ /* codec private data */ struct wm8711_priv { - enum snd_soc_control_type bus_type; + struct regmap *regmap; unsigned int sysclk; }; @@ -42,11 +43,21 @@ struct wm8711_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { - 0x0079, 0x0079, 0x000a, 0x0008, - 0x009f, 0x000a, 0x0000, 0x0000 +static const struct reg_default wm8711_reg_defaults[] = { + { 0, 0x0079 }, { 1, 0x0079 }, { 2, 0x000a }, { 3, 0x0008 }, + { 4, 0x009f }, { 5, 0x000a }, { 6, 0x0000 }, { 7, 0x0000 }, }; +static bool wm8711_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8711_RESET: + return true; + default: + return false; + } +} + #define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); @@ -289,6 +300,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8711_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; switch (level) { @@ -299,7 +311,7 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - snd_soc_cache_sync(codec); + regcache_sync(wm8711->regmap); snd_soc_write(codec, WM8711_PWR, reg | 0x0040); break; @@ -353,10 +365,9 @@ static int wm8711_resume(struct snd_soc_codec *codec) static int wm8711_probe(struct snd_soc_codec *codec) { - struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -391,9 +402,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .suspend = wm8711_suspend, .resume = wm8711_resume, .set_bias_level = wm8711_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8711_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8711_reg, .controls = wm8711_snd_controls, .num_controls = ARRAY_SIZE(wm8711_snd_controls), .dapm_widgets = wm8711_dapm_widgets, @@ -408,6 +416,18 @@ static const struct of_device_id wm8711_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8711_of_match); +static const struct regmap_config wm8711_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8711_RESET, + + .reg_defaults = wm8711_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8711_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8711_volatile, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8711_spi_probe(struct spi_device *spi) { @@ -419,8 +439,11 @@ static int __devinit wm8711_spi_probe(struct spi_device *spi) if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_spi(spi, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + spi_set_drvdata(spi, wm8711); - wm8711->bus_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); @@ -458,8 +481,11 @@ static __devinit int wm8711_i2c_probe(struct i2c_client *client, if (wm8711 == NULL) return -ENOMEM; + wm8711->regmap = devm_regmap_init_i2c(client, &wm8711_regmap); + if (IS_ERR(wm8711->regmap)) + return PTR_ERR(wm8711->regmap); + i2c_set_clientdata(client, wm8711); - wm8711->bus_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_wm8711, &wm8711_dai, 1); -- cgit v1.2.3 From d16383ef2a62fe53a929eac56b662d6def6bb8c7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:30:01 +0800 Subject: ASoC: wm8728: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 45 ++++++++++++++++++++++++++++----------------- 1 file changed, 28 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f274c250d492..00a12a0c3919 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -35,16 +36,16 @@ * the volume update bits, mute the output and enable infinite zero * detect. */ -static const u16 wm8728_reg_defaults[] = { - 0x1ff, - 0x1ff, - 0x001, - 0x100, +static const struct reg_default wm8728_reg_defaults[] = { + { 0, 0x1ff }, + { 1, 0x1ff }, + { 2, 0x001 }, + { 3, 0x100 }, }; /* codec private data */ struct wm8728_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); @@ -162,8 +163,8 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8728_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); u16 reg; - int i; switch (level) { case SND_SOC_BIAS_ON: @@ -175,9 +176,7 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ - for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - snd_soc_write(codec, i, - snd_soc_read(codec, i)); + regcache_sync(wm8728->regmap); } break; @@ -229,10 +228,9 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - struct wm8728_priv *wm8728 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8728->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", ret); @@ -257,9 +255,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .suspend = wm8728_suspend, .resume = wm8728_resume, .set_bias_level = wm8728_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8728_reg_defaults, .controls = wm8728_snd_controls, .num_controls = ARRAY_SIZE(wm8728_snd_controls), .dapm_widgets = wm8728_dapm_widgets, @@ -274,6 +269,16 @@ static const struct of_device_id wm8728_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8728_of_match); +static const struct regmap_config wm8728_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8728_IFCTL, + + .reg_defaults = wm8728_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8728_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8728_spi_probe(struct spi_device *spi) { @@ -285,7 +290,10 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) if (wm8728 == NULL) return -ENOMEM; - wm8728->control_type = SND_SOC_SPI; + wm8728->regmap = devm_regmap_init_spi(spi, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + spi_set_drvdata(spi, wm8728); ret = snd_soc_register_codec(&spi->dev, @@ -324,8 +332,11 @@ static __devinit int wm8728_i2c_probe(struct i2c_client *i2c, if (wm8728 == NULL) return -ENOMEM; + wm8728->regmap = devm_regmap_init_i2c(i2c, &wm8728_regmap); + if (IS_ERR(wm8728->regmap)) + return PTR_ERR(wm8728->regmap); + i2c_set_clientdata(i2c, wm8728); - wm8728->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8728, &wm8728_dai, 1); -- cgit v1.2.3 From 4f69bb31b8840713f82c6c476bae3f2356819ce2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:36:28 +0800 Subject: ASoC: wm8737: Move regulator acquisition to device registration This is better style as we acquire resources we will need before we go into the ASoC card probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 41 +++++++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 8c25442f9b47..3bf5bc73e2de 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -557,7 +557,7 @@ static int wm8737_resume(struct snd_soc_codec *codec) static int wm8737_probe(struct snd_soc_codec *codec) { struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8737->control_type); if (ret != 0) { @@ -565,16 +565,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) - wm8737->supplies[i].supply = wm8737_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8737->supplies), - wm8737->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { @@ -607,17 +597,12 @@ static int wm8737_probe(struct snd_soc_codec *codec) err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); err_get: - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - return ret; } static int wm8737_remove(struct snd_soc_codec *codec) { - struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); - wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); return 0; } @@ -645,13 +630,23 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&i2c->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8737); wm8737->control_type = SND_SOC_I2C; @@ -691,13 +686,23 @@ static struct i2c_driver wm8737_i2c_driver = { static int __devinit wm8737_spi_probe(struct spi_device *spi) { struct wm8737_priv *wm8737; - int ret; + int ret, i; wm8737 = devm_kzalloc(&spi->dev, sizeof(struct wm8737_priv), GFP_KERNEL); if (wm8737 == NULL) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(wm8737->supplies); i++) + wm8737->supplies[i].supply = wm8737_supply_names[i]; + + ret = devm_regulator_bulk_get(&spi->dev, ARRAY_SIZE(wm8737->supplies), + wm8737->supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + wm8737->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8737); -- cgit v1.2.3 From 3ef8ac0d7bd0532fbfb319f3fbf615538394119f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:48:48 +0800 Subject: ASoC: wm8737: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 75 +++++++++++++++++++++++++++++++---------------- 1 file changed, 50 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 3bf5bc73e2de..5c9634f4c1f0 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -40,29 +41,39 @@ static const char *wm8737_supply_names[WM8737_NUM_SUPPLIES] = { /* codec private data */ struct wm8737_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8737_NUM_SUPPLIES]; unsigned int mclk; }; -static const u16 wm8737_reg[WM8737_REGISTER_COUNT] = { - 0x00C3, /* R0 - Left PGA volume */ - 0x00C3, /* R1 - Right PGA volume */ - 0x0007, /* R2 - AUDIO path L */ - 0x0007, /* R3 - AUDIO path R */ - 0x0000, /* R4 - 3D Enhance */ - 0x0000, /* R5 - ADC Control */ - 0x0000, /* R6 - Power Management */ - 0x000A, /* R7 - Audio Format */ - 0x0000, /* R8 - Clocking */ - 0x000F, /* R9 - MIC Preamp Control */ - 0x0003, /* R10 - Misc Bias Control */ - 0x0000, /* R11 - Noise Gate */ - 0x007C, /* R12 - ALC1 */ - 0x0000, /* R13 - ALC2 */ - 0x0032, /* R14 - ALC3 */ +static const struct reg_default wm8737_reg_defaults[] = { + { 0, 0x00C3 }, /* R0 - Left PGA volume */ + { 1, 0x00C3 }, /* R1 - Right PGA volume */ + { 2, 0x0007 }, /* R2 - AUDIO path L */ + { 3, 0x0007 }, /* R3 - AUDIO path R */ + { 4, 0x0000 }, /* R4 - 3D Enhance */ + { 5, 0x0000 }, /* R5 - ADC Control */ + { 6, 0x0000 }, /* R6 - Power Management */ + { 7, 0x000A }, /* R7 - Audio Format */ + { 8, 0x0000 }, /* R8 - Clocking */ + { 9, 0x000F }, /* R9 - MIC Preamp Control */ + { 10, 0x0003 }, /* R10 - Misc Bias Control */ + { 11, 0x0000 }, /* R11 - Noise Gate */ + { 12, 0x007C }, /* R12 - ALC1 */ + { 13, 0x0000 }, /* R13 - ALC2 */ + { 14, 0x0032 }, /* R14 - ALC3 */ }; +static bool wm8737_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8737_RESET: + return true; + default: + return false; + } +} + static int wm8737_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8737_RESET, 0); @@ -479,7 +490,7 @@ static int wm8737_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + regcache_sync(wm8737->regmap); /* Fast VMID ramp at 2*2.5k */ snd_soc_update_bits(codec, WM8737_MISC_BIAS_CONTROL, @@ -559,7 +570,7 @@ static int wm8737_probe(struct snd_soc_codec *codec) struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8737->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -612,10 +623,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = { .suspend = wm8737_suspend, .resume = wm8737_resume, .set_bias_level = wm8737_set_bias_level, - - .reg_cache_size = WM8737_REGISTER_COUNT - 1, /* Skip reset */ - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8737_reg, }; static const struct of_device_id wm8737_of_match[] = { @@ -625,6 +632,18 @@ static const struct of_device_id wm8737_of_match[] = { MODULE_DEVICE_TABLE(of, wm8737_of_match); +static const struct regmap_config wm8737_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8737_MAX_REGISTER, + + .reg_defaults = wm8737_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8737_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8737_volatile, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -647,8 +666,11 @@ static __devinit int wm8737_i2c_probe(struct i2c_client *i2c, return ret; } + wm8737->regmap = devm_regmap_init_i2c(i2c, &wm8737_regmap); + if (IS_ERR(wm8737->regmap)) + return PTR_ERR(wm8737->regmap); + i2c_set_clientdata(i2c, wm8737); - wm8737->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8737, &wm8737_dai, 1); @@ -703,7 +725,10 @@ static int __devinit wm8737_spi_probe(struct spi_device *spi) return ret; } - wm8737->control_type = SND_SOC_SPI; + wm8737->regmap = devm_regmap_init_spi(spi, &wm8737_regmap); + if (IS_ERR(wm8737->regmap)) + return PTR_ERR(wm8737->regmap); + spi_set_drvdata(spi, wm8737); ret = snd_soc_register_codec(&spi->dev, -- cgit v1.2.3 From 7e8d613b536446139d93c956bb59f25aa934c520 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 15:19:20 +0200 Subject: ALSA: ctxfi: Fix mono channel map to UNKNOWN To follow the previous commit. Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctpcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 6c191429b126..d317107d98cc 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -397,7 +397,7 @@ static struct snd_pcm_ops ct_pcm_capture_ops = { static const struct snd_pcm_chmap_elem surround_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_RC } }, + .map = { SNDRV_CHMAP_UNKNOWN } }, { .channels = 2, .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, { } @@ -405,7 +405,7 @@ static const struct snd_pcm_chmap_elem surround_map[] = { static const struct snd_pcm_chmap_elem clfe_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_FC } }, + .map = { SNDRV_CHMAP_UNKNOWN } }, { .channels = 2, .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, { } @@ -413,7 +413,7 @@ static const struct snd_pcm_chmap_elem clfe_map[] = { static const struct snd_pcm_chmap_elem side_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_RC } }, + .map = { SNDRV_CHMAP_UNKNOWN } }, { .channels = 2, .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, { } -- cgit v1.2.3 From 1fe4d42e0e28c2c004b06dd590702604f47c2402 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 15:46:20 +0200 Subject: ALSA: ens1370: Reduce ifdefs ... just by defining CHIP_NAME and string concats. Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 38 ++++++++------------------------------ 1 file changed, 8 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 2ba58d365959..1e615c4d12f8 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -55,8 +55,10 @@ #ifdef CHIP1370 #define DRIVER_NAME "ENS1370" +#define CHIP_NAME "ES1370" /* it can be ENS but just to keep compatibility... */ #else #define DRIVER_NAME "ENS1371" +#define CHIP_NAME "ES1371" #endif @@ -1266,11 +1268,7 @@ static int __devinit snd_ensoniq_pcm(struct ensoniq * ensoniq, int device, if (rpcm) *rpcm = NULL; -#ifdef CHIP1370 - err = snd_pcm_new(ensoniq->card, "ES1370/1", device, 1, 1, &pcm); -#else - err = snd_pcm_new(ensoniq->card, "ES1371/1", device, 1, 1, &pcm); -#endif + err = snd_pcm_new(ensoniq->card, CHIP_NAME "/1", device, 1, 1, &pcm); if (err < 0) return err; @@ -1283,11 +1281,7 @@ static int __devinit snd_ensoniq_pcm(struct ensoniq * ensoniq, int device, pcm->private_data = ensoniq; pcm->info_flags = 0; -#ifdef CHIP1370 - strcpy(pcm->name, "ES1370 DAC2/ADC"); -#else - strcpy(pcm->name, "ES1371 DAC2/ADC"); -#endif + strcpy(pcm->name, CHIP_NAME " DAC2/ADC"); ensoniq->pcm1 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, @@ -1306,11 +1300,7 @@ static int __devinit snd_ensoniq_pcm2(struct ensoniq * ensoniq, int device, if (rpcm) *rpcm = NULL; -#ifdef CHIP1370 - err = snd_pcm_new(ensoniq->card, "ES1370/2", device, 1, 0, &pcm); -#else - err = snd_pcm_new(ensoniq->card, "ES1371/2", device, 1, 0, &pcm); -#endif + err = snd_pcm_new(ensoniq->card, CHIP_NAME "/2", device, 1, 0, &pcm); if (err < 0) return err; @@ -1321,11 +1311,7 @@ static int __devinit snd_ensoniq_pcm2(struct ensoniq * ensoniq, int device, #endif pcm->private_data = ensoniq; pcm->info_flags = 0; -#ifdef CHIP1370 - strcpy(pcm->name, "ES1370 DAC1"); -#else - strcpy(pcm->name, "ES1371 DAC1"); -#endif + strcpy(pcm->name, CHIP_NAME " DAC1"); ensoniq->pcm2 = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, @@ -1885,11 +1871,7 @@ static void snd_ensoniq_proc_read(struct snd_info_entry *entry, { struct ensoniq *ensoniq = entry->private_data; -#ifdef CHIP1370 - snd_iprintf(buffer, "Ensoniq AudioPCI ES1370\n\n"); -#else - snd_iprintf(buffer, "Ensoniq AudioPCI ES1371\n\n"); -#endif + snd_iprintf(buffer, "Ensoniq AudioPCI " CHIP_NAME "\n\n"); snd_iprintf(buffer, "Joystick enable : %s\n", ensoniq->ctrl & ES_JYSTK_EN ? "on" : "off"); #ifdef CHIP1370 @@ -2361,11 +2343,7 @@ static int __devinit snd_ensoniq_midi(struct ensoniq * ensoniq, int device, *rrawmidi = NULL; if ((err = snd_rawmidi_new(ensoniq->card, "ES1370/1", device, 1, 1, &rmidi)) < 0) return err; -#ifdef CHIP1370 - strcpy(rmidi->name, "ES1370"); -#else - strcpy(rmidi->name, "ES1371"); -#endif + strcpy(rmidi->name, CHIP_NAME); snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_ensoniq_midi_output); snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_ensoniq_midi_input); rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | -- cgit v1.2.3 From 21147f91f1be57c4cfc24e538e93c30636513ce8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 15:52:09 +0200 Subject: ALSA: ca0106: Define channel maps Provide channel maps for individual stereo streams of CA0106. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index fc6787699ba9..65c55910566b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1334,10 +1334,29 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +static const struct snd_pcm_chmap_elem clfe_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; + +static const struct snd_pcm_chmap_elem side_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; + static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device) { struct snd_pcm *pcm; struct snd_pcm_substream *substream; + const struct snd_pcm_chmap_elem *map = NULL; int err; err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm); @@ -1350,18 +1369,22 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device) case 0: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops); + map = snd_pcm_std_chmaps; break; case 1: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops); + map = surround_map; break; case 2: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops); + map = clfe_map; break; case 3: snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops); + map = side_map; break; } @@ -1388,6 +1411,11 @@ static int __devinit snd_ca0106_pcm(struct snd_ca0106 *emu, int device) return err; } + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2, + 1 << 2, NULL); + if (err < 0) + return err; + emu->pcm[device] = pcm; return 0; -- cgit v1.2.3 From 3adc497f98dd062bcf4d832c659a3d706ece5249 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 15:52:47 +0200 Subject: ALSA: emu10k1x: Define channel maps Provide channel maps for individual stereo streams of emu10k1x. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1x.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 5c8978b2c4d9..556fd6f456e3 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -830,9 +830,22 @@ static irqreturn_t snd_emu10k1x_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +static const struct snd_pcm_chmap_elem clfe_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; + static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; + const struct snd_pcm_chmap_elem *map = NULL; int err; int capture = 0; @@ -861,12 +874,15 @@ static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct s switch(device) { case 0: strcpy(pcm->name, "EMU10K1X Front"); + map = snd_pcm_std_chmaps; break; case 1: strcpy(pcm->name, "EMU10K1X Rear"); + map = surround_map; break; case 2: strcpy(pcm->name, "EMU10K1X Center/LFE"); + map = clfe_map; break; } emu->pcm = pcm; @@ -875,6 +891,11 @@ static int __devinit snd_emu10k1x_pcm(struct emu10k1x *emu, int device, struct s snd_dma_pci_data(emu->pci), 32*1024, 32*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2, + 1 << 2, NULL); + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; -- cgit v1.2.3 From 7fb6861d62b24fd0056a52275e5f74db0ef66f88 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 15:53:26 +0200 Subject: ALSA: ens1370: Define channel maps Provide channel maps for individual stereo streams of ENS1370 and ENS1371. Note that the configuration of ENS1370 uses the secondary PCM as the front unlike ENS1371. Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound') diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 1e615c4d12f8..9d432359d1cb 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1260,6 +1260,14 @@ static struct snd_pcm_ops snd_ensoniq_capture_ops = { .pointer = snd_ensoniq_capture_pointer, }; +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_UNKNOWN } }, + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + static int __devinit snd_ensoniq_pcm(struct ensoniq * ensoniq, int device, struct snd_pcm ** rpcm) { @@ -1287,6 +1295,16 @@ static int __devinit snd_ensoniq_pcm(struct ensoniq * ensoniq, int device, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024); +#ifdef CHIP1370 + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + surround_map, 2, 0, NULL); +#else + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_std_chmaps, 2, 0, NULL); +#endif + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; return 0; @@ -1317,6 +1335,16 @@ static int __devinit snd_ensoniq_pcm2(struct ensoniq * ensoniq, int device, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ensoniq->pci), 64*1024, 128*1024); +#ifdef CHIP1370 + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_std_chmaps, 2, 0, NULL); +#else + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + surround_map, 2, 0, NULL); +#endif + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; return 0; -- cgit v1.2.3 From 6b315958d330d3ebf46b7d45e0978a97be2c4ac0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 18:44:40 +0800 Subject: ASoC: arizona: Clarify logging for FLL lock status interrupt Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5e96a0a1669c..b79578e7e10c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -732,7 +732,7 @@ static irqreturn_t arizona_fll_lock(int irq, void *data) { struct arizona_fll *fll = data; - arizona_fll_dbg(fll, "Locked\n"); + arizona_fll_dbg(fll, "Lock status changed\n"); complete(&fll->lock); -- cgit v1.2.3 From 0afdb8f2869610b7c2eb99d75ba8b9003b8e88d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 16:14:37 +0200 Subject: ALSA: ymfpci: Define channel maps Provide channel maps for individual stereo streams of YMFPCI. Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index ee8b6366e48d..50b0804df904 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1166,6 +1166,11 @@ int __devinit snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_std_chmaps, 2, 0, NULL); + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; return 0; @@ -1257,6 +1262,14 @@ static struct snd_pcm_ops snd_ymfpci_playback_4ch_ops = { .pointer = snd_ymfpci_playback_pointer, }; +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 1, + .map = { SNDRV_CHMAP_UNKNOWN } }, + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + int __devinit snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd_pcm ** rpcm) { struct snd_pcm *pcm; @@ -1278,6 +1291,11 @@ int __devinit snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), 64*1024, 256*1024); + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, + surround_map, 2, 0, NULL); + if (err < 0) + return err; + if (rpcm) *rpcm = pcm; return 0; -- cgit v1.2.3 From 7b31d0095e87221dc32c95642a2a714ea08259aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Sep 2012 18:06:54 +0200 Subject: ALSA: Define more channel map positions For following the standard, define more channel map positions and shuffle the items a bit: - As both PulseAudio and gstreamer define MONO channel position explicitly, we should follow that, too. The mono streams point to this channel position unless they are explicitly assigned to certain channel positions. - Top-front-* and Top-rear-* positions are added, carried from PulseAudio's definitions. - Move NA and MONO definitions at the top of table right after UNKNOWN, since these are more abstract in comparison with other practical positions. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Channel-Mapping-API.txt | 13 ++++++++++--- include/sound/asound.h | 13 ++++++++++--- sound/pci/ctxfi/ctpcm.c | 6 +++--- 3 files changed, 23 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/Channel-Mapping-API.txt b/Documentation/sound/alsa/Channel-Mapping-API.txt index 4bbf12d5553f..3c43d1a4ca0e 100644 --- a/Documentation/sound/alsa/Channel-Mapping-API.txt +++ b/Documentation/sound/alsa/Channel-Mapping-API.txt @@ -76,8 +76,10 @@ here is a cut: /* channel positions */ enum { - /* this follows the alsa-lib mixer channel value + 1 */ SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ SNDRV_CHMAP_FL, /* front left */ SNDRV_CHMAP_FR, /* front right */ SNDRV_CHMAP_RL, /* rear left */ @@ -98,8 +100,13 @@ enum { SNDRV_CHMAP_FCH, /* front center high */ SNDRV_CHMAP_FRH, /* front right high */ SNDRV_CHMAP_TC, /* top center */ - SNDRV_CHMAP_NA, /* N/A, silent */ - SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, }; When a PCM stream can provide more than one channel map, you can diff --git a/include/sound/asound.h b/include/sound/asound.h index 27686da0f650..dfe7d441748c 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -474,8 +474,10 @@ enum { /* channel positions */ enum { - /* this follows the alsa-lib mixer channel value + 1 */ SNDRV_CHMAP_UNKNOWN = 0, + SNDRV_CHMAP_NA, /* N/A, silent */ + SNDRV_CHMAP_MONO, /* mono stream */ + /* this follows the alsa-lib mixer channel value + 3 */ SNDRV_CHMAP_FL, /* front left */ SNDRV_CHMAP_FR, /* front right */ SNDRV_CHMAP_RL, /* rear left */ @@ -496,8 +498,13 @@ enum { SNDRV_CHMAP_FCH, /* front center high */ SNDRV_CHMAP_FRH, /* front right high */ SNDRV_CHMAP_TC, /* top center */ - SNDRV_CHMAP_NA, /* N/A, silent */ - SNDRV_CHMAP_LAST = SNDRV_CHMAP_NA, + SNDRV_CHMAP_TFL, /* top front left */ + SNDRV_CHMAP_TFR, /* top front right */ + SNDRV_CHMAP_TFC, /* top front center */ + SNDRV_CHMAP_TRL, /* top rear left */ + SNDRV_CHMAP_TRR, /* top rear right */ + SNDRV_CHMAP_TRC, /* top rear center */ + SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC, }; #define SNDRV_CHMAP_POSITION_MASK 0xffff diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index d317107d98cc..e8a4feb1ed86 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -397,7 +397,7 @@ static struct snd_pcm_ops ct_pcm_capture_ops = { static const struct snd_pcm_chmap_elem surround_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, { } @@ -405,7 +405,7 @@ static const struct snd_pcm_chmap_elem surround_map[] = { static const struct snd_pcm_chmap_elem clfe_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, { } @@ -413,7 +413,7 @@ static const struct snd_pcm_chmap_elem clfe_map[] = { static const struct snd_pcm_chmap_elem side_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, { } -- cgit v1.2.3 From da8b8e0f15b375b44ed8ef4b0c5f5f60f19ccb37 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 12:21:52 +0800 Subject: ASoC: core: Mark regmap CODEC register maps as dirty when suspending The core has for a long time had support for marking the register maps of devices dirty when suspending so that they are resynced on resume. Also implement this feature for CODECs using regmap. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad65459da28e..2b418398ec18 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -635,6 +635,8 @@ int snd_soc_suspend(struct device *dev) codec->driver->suspend(codec); codec->suspended = 1; codec->cache_sync = 1; + if (codec->using_regmap) + regcache_mark_dirty(codec->control_data); break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- cgit v1.2.3 From ab7af5c8d456103ea57c7e9e6f5b03162965e665 Mon Sep 17 00:00:00 2001 From: Peter Senna Tschudin Date: Wed, 12 Sep 2012 17:06:46 +0200 Subject: ASoC: core: Remove useless kfree Remove useless kfree() and clean up code related to the removal. The semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @r exists@ position p1,p2; expression x; @@ if (x@p1 == NULL) { ... kfree@p2(x); ... return ...; } @unchanged exists@ position r.p1,r.p2; expression e <= r.x,x,e1; iterator I; statement S; @@ if (x@p1 == NULL) { ... when != I(x,...) S when != e = e1 when != e += e1 when != e -= e1 when != ++e when != --e when != e++ when != e-- when != &e kfree@p2(x); ... return ...; } @ok depends on unchanged exists@ position any r.p1; position r.p2; expression x; @@ ... when != true x@p1 == NULL kfree@p2(x); @depends on !ok && unchanged@ position r.p2; expression x; @@ *kfree@p2(x); // Signed-off-by: Peter Senna Tschudin Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2b418398ec18..e5b0713e6f3c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4054,8 +4054,6 @@ int snd_soc_register_codec(struct device *dev, return 0; fail: - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; kfree(codec->name); kfree(codec); return ret; -- cgit v1.2.3 From 03efce755cf1b4cf1c14007e118dd1671e39f862 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Sep 2012 09:56:57 +0200 Subject: ALSA: hda - Fix disordered enum definitions in patch_cirrus.c Due to the definitions of CS420X_IMAC27_122 and CS420X_APPLE as aliases, the rest enums are set to duplicated values unexpectedly. Move the alias definitions at the end so that the enum values are defined in the proper order. Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 82253f137359..fcfc9f0a056b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -83,11 +83,12 @@ enum { CS420X_IMAC27, CS420X_GPIO_13, CS420X_GPIO_23, - CS420X_IMAC27_122 = CS420X_GPIO_23, - CS420X_APPLE = CS420X_GPIO_13, CS420X_MBP101, CS420X_MBP101_COEF, CS420X_AUTO, + /* aliases */ + CS420X_IMAC27_122 = CS420X_GPIO_23, + CS420X_APPLE = CS420X_GPIO_13, }; /* CS421x boards */ -- cgit v1.2.3 From 040242ccfcd4ae878267b521d16539e7b3000527 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 13 Sep 2012 12:03:23 +0200 Subject: ASoC: ad193x: Use managed regmap init Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 50 ++++++++++------------------------------------- 1 file changed, 10 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 13e62be4f990..2f752660f678 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -381,40 +381,25 @@ static const struct regmap_config ad193x_spi_regmap_config = { static int __devinit ad193x_spi_probe(struct spi_device *spi) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); spi_set_drvdata(spi, ad193x); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_spi_remove(struct spi_device *spi) { - struct ad193x_priv *ad193x = spi_get_drvdata(spi); - snd_soc_unregister_codec(&spi->dev); - regmap_exit(ad193x->regmap); return 0; } @@ -449,40 +434,25 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct ad193x_priv *ad193x; - int ret; ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config); - if (IS_ERR(ad193x->regmap)) { - ret = PTR_ERR(ad193x->regmap); - goto err_out; - } + ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config); + if (IS_ERR(ad193x->regmap)) + return PTR_ERR(ad193x->regmap); i2c_set_clientdata(client, ad193x); - ret = snd_soc_register_codec(&client->dev, - &soc_codec_dev_ad193x, &ad193x_dai, 1); - if (ret < 0) - goto err_regmap_exit; - - return 0; - -err_regmap_exit: - regmap_exit(ad193x->regmap); -err_out: - return ret; + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } static int __devexit ad193x_i2c_remove(struct i2c_client *client) { - struct ad193x_priv *ad193x = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(ad193x->regmap); return 0; } -- cgit v1.2.3 From 7f22fd9c03c0b67ee6aa138bd10ae91bb0d22151 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 13 Sep 2012 12:04:51 +0200 Subject: ASoC: ad1836: Convert to direct regmap usage. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 76 +++++++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ae1eb51bc9da..dce6ebeef452 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -19,6 +19,8 @@ #include #include #include +#include + #include "ad1836.h" enum ad1836_type { @@ -30,6 +32,7 @@ enum ad1836_type { /* codec private data */ struct ad1836_priv { enum ad1836_type type; + struct regmap *regmap; }; /* @@ -161,8 +164,8 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(dai->codec); int word_len = 0; - struct snd_soc_codec *codec = dai->codec; /* bit size */ switch (params_format(params)) { @@ -178,10 +181,12 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len << AD1836_ADC_WORD_OFFSET); return 0; @@ -223,15 +228,17 @@ static struct snd_soc_dai_driver ad183x_dais[] = { #ifdef CONFIG_PM static int ad1836_suspend(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } static int ad1836_resume(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* restore clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, AD1836_ADC_AUX); } #else @@ -250,37 +257,30 @@ static int ad1836_probe(struct snd_soc_codec *codec) num_dacs = ad183x_dais[ad1836->type].playback.channels_max / 2; num_adcs = ad183x_dais[ad1836->type].capture.channels_max / 2; - ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - return ret; - } - /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); + regmap_write(ad1836->regmap, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL2, 0x180); /* volume */ for (i = 1; i <= num_dacs; ++i) { - snd_soc_write(codec, AD1836_DAC_L_VOL(i), 0x3FF); - snd_soc_write(codec, AD1836_DAC_R_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_L_VOL(i), 0x3FF); + regmap_write(ad1836->regmap, AD1836_DAC_R_VOL(i), 0x3FF); } if (ad1836->type == AD1836) { /* left/right diff:PGA/MUX */ - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x3A); ret = snd_soc_add_codec_controls(codec, ad1836_controls, ARRAY_SIZE(ad1836_controls)); if (ret) return ret; } else { - snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); + regmap_write(ad1836->regmap, AD1836_ADC_CTRL3, 0x00); } ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2); @@ -313,8 +313,9 @@ static int ad1836_probe(struct snd_soc_codec *codec) /* power down chip */ static int ad1836_remove(struct snd_soc_codec *codec) { + struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec); /* reset clock control mode */ - return snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + return regmap_update_bits(ad1836->regmap, AD1836_ADC_CTRL2, AD1836_ADC_SERFMT_MASK, 0); } @@ -323,8 +324,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .remove = ad1836_remove, .suspend = ad1836_suspend, .resume = ad1836_resume, - .reg_cache_size = AD1836_NUM_REGS, - .reg_word_size = sizeof(u16), .controls = ad183x_controls, .num_controls = ARRAY_SIZE(ad183x_controls), @@ -334,6 +333,33 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1836 = { .num_dapm_routes = ARRAY_SIZE(ad183x_dapm_routes), }; +static const struct reg_default ad1836_reg_defaults[] = { + { AD1836_DAC_CTRL1, 0x0000 }, + { AD1836_DAC_CTRL2, 0x0000 }, + { AD1836_DAC_L_VOL(0), 0x0000 }, + { AD1836_DAC_R_VOL(0), 0x0000 }, + { AD1836_DAC_L_VOL(1), 0x0000 }, + { AD1836_DAC_R_VOL(1), 0x0000 }, + { AD1836_DAC_L_VOL(2), 0x0000 }, + { AD1836_DAC_R_VOL(2), 0x0000 }, + { AD1836_DAC_L_VOL(3), 0x0000 }, + { AD1836_DAC_R_VOL(3), 0x0000 }, + { AD1836_ADC_CTRL1, 0x0000 }, + { AD1836_ADC_CTRL2, 0x0000 }, + { AD1836_ADC_CTRL3, 0x0000 }, +}; + +static const struct regmap_config ad1836_regmap_config = { + .val_bits = 12, + .reg_bits = 4, + .read_flag_mask = 0x08, + + .max_register = AD1836_ADC_CTRL3, + .reg_defaults = ad1836_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ad1836_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct ad1836_priv *ad1836; @@ -344,6 +370,10 @@ static int __devinit ad1836_spi_probe(struct spi_device *spi) if (ad1836 == NULL) return -ENOMEM; + ad1836->regmap = devm_regmap_init_spi(spi, &ad1836_regmap_config); + if (IS_ERR(ad1836->regmap)) + return PTR_ERR(ad1836->regmap); + ad1836->type = spi_get_device_id(spi)->driver_data; spi_set_drvdata(spi, ad1836); -- cgit v1.2.3 From 5efbc2610a7b2aac6c51f8fc943c019106568939 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Sep 2012 14:48:46 +0200 Subject: ALSA: Fix leftover chmap UNKNOWN -> MONO conversions A few files have been slipped from the previous commit to add MONO channel type. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 4 ++-- sound/pci/ens1370.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9647a22ed69a..f42c10a43315 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2311,7 +2311,7 @@ EXPORT_SYMBOL(snd_pcm_lib_readv); /* default channel maps for multi-channel playbacks, up to 8 channels */ const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, @@ -2333,7 +2333,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_std_chmaps); /* alternative channel maps with CLFE <-> surround swapped for 6/8 channels */ const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, { .channels = 4, diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 9d432359d1cb..5674cc316530 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1262,7 +1262,7 @@ static struct snd_pcm_ops snd_ensoniq_capture_ops = { static const struct snd_pcm_chmap_elem surround_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, { } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 50b0804df904..3a6f03f9b02f 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1264,7 +1264,7 @@ static struct snd_pcm_ops snd_ymfpci_playback_4ch_ops = { static const struct snd_pcm_chmap_elem surround_map[] = { { .channels = 1, - .map = { SNDRV_CHMAP_UNKNOWN } }, + .map = { SNDRV_CHMAP_MONO } }, { .channels = 2, .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, { } -- cgit v1.2.3 From 1dac6695c683c66d0cff10a84c6ed10dbbaabc18 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Sep 2012 14:59:47 +0200 Subject: ALSA: hda - Allow to pass position_fix=0 explicitly Set the default value of position_fix -1, and allow user passing position_fix=0 explicitly to set the "auto" position-fix mode. Otherwise the auto mode may be switched to others like COMBO of VIACOMBO when the controller prefers it, thus user can't set the auto mode any longer. Also updated the documentation appropriately, too. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 10 ++++++++-- sound/pci/hda/hda_intel.c | 5 +++-- 2 files changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4e4d0bc9816f..d90d8ec2853d 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -860,8 +860,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. [Multiple options for each card instance] model - force the model name - position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF, - 3 = VIACOMBO, 4 = COMBO) + position_fix - Fix DMA pointer + -1 = system default: choose appropriate one per controller + hardware + 0 = auto: falls back to LPIB when POSBUF doesn't work + 1 = use LPIB + 2 = POSBUF: use position buffer + 3 = VIACOMBO: VIA-specific workaround for capture + 4 = COMBO: use LPIB for playback, auto for capture stream probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) When the bit 8 (0x100) is set, the lower 8 bits are used as the "fixed" codec slots; i.e. the driver probes the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1a12c754de9..195d84726187 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -64,7 +64,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; -static int position_fix[SNDRV_CARDS]; +static int position_fix[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; @@ -88,7 +88,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -2813,6 +2813,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) const struct snd_pci_quirk *q; switch (fix) { + case POS_FIX_AUTO: case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: -- cgit v1.2.3 From fc488517cc0d50bcc9e4ffa90fee5755f9c914fc Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Thu, 12 Jul 2012 17:39:18 -0300 Subject: [media] snd_tea575x: Add support for tuning AM Add support for tuning AM (on devices with the necessary additional hardware components), and advertise the available bands using the new VIDIOC_ENUM_FREQ_BANDS ioctl. Signed-off-by: Hans de Goede CC: Ondrej Zary Signed-off-by: Mauro Carvalho Chehab --- drivers/media/radio/radio-shark.c | 1 + include/sound/tea575x-tuner.h | 3 + sound/i2c/other/tea575x-tuner.c | 197 +++++++++++++++++++++++++++++++------- 3 files changed, 165 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/drivers/media/radio/radio-shark.c b/drivers/media/radio/radio-shark.c index 72ded29728bb..e1970bf031a4 100644 --- a/drivers/media/radio/radio-shark.c +++ b/drivers/media/radio/radio-shark.c @@ -333,6 +333,7 @@ static int usb_shark_probe(struct usb_interface *intf, shark->tea.radio_nr = -1; shark->tea.ops = &shark_tea_ops; shark->tea.cannot_mute = true; + shark->tea.has_am = true; strlcpy(shark->tea.card, "Griffin radioSHARK", sizeof(shark->tea.card)); usb_make_path(shark->usbdev, shark->tea.bus_info, diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h index fe8590cac5c2..2a6953568554 100644 --- a/include/sound/tea575x-tuner.h +++ b/include/sound/tea575x-tuner.h @@ -28,6 +28,7 @@ #include #define TEA575X_FMIF 10700 +#define TEA575X_AMIF 450 #define TEA575X_DATA (1 << 0) #define TEA575X_CLK (1 << 1) @@ -52,12 +53,14 @@ struct snd_tea575x { struct video_device vd; /* video device */ int radio_nr; /* radio_nr */ bool tea5759; /* 5759 chip is present */ + bool has_am; /* Device can tune to AM freqs */ bool cannot_read_data; /* Device cannot read the data pin */ bool cannot_mute; /* Device cannot mute */ bool mute; /* Device is muted? */ bool stereo; /* receiving stereo */ bool tuned; /* tuned to a station */ unsigned int val; /* hw value */ + u32 band; /* 0: FM, 1: FM-Japan, 2: AM */ u32 freq; /* frequency */ struct mutex mutex; struct snd_tea575x_ops *ops; diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index d14edb7d6484..88bbd88c066e 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -37,9 +37,6 @@ MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); -#define FREQ_LO ((tea->tea5759 ? 760 : 875) * 1600U) -#define FREQ_HI ((tea->tea5759 ? 910 : 1080) * 1600U) - /* * definitions */ @@ -50,8 +47,8 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_BAND_MASK (3<<20) #define TEA575X_BIT_BAND_FM (0<<20) #define TEA575X_BIT_BAND_MW (1<<20) -#define TEA575X_BIT_BAND_LW (1<<21) -#define TEA575X_BIT_BAND_SW (1<<22) +#define TEA575X_BIT_BAND_LW (2<<20) +#define TEA575X_BIT_BAND_SW (3<<20) #define TEA575X_BIT_PORT_0 (1<<19) /* user bit */ #define TEA575X_BIT_PORT_1 (1<<18) /* user bit */ #define TEA575X_BIT_SEARCH_MASK (3<<16) /* search level */ @@ -62,6 +59,37 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff +enum { BAND_FM, BAND_FM_JAPAN, BAND_AM }; + +static const struct v4l2_frequency_band bands[] = { + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 87500 * 16, + .rangehigh = 108000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 0, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO | + V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 76000 * 16, + .rangehigh = 91000 * 16, + .modulation = V4L2_BAND_MODULATION_FM, + }, + { + .type = V4L2_TUNER_RADIO, + .index = 1, + .capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS, + .rangelow = 530 * 16, + .rangehigh = 1710 * 16, + .modulation = V4L2_BAND_MODULATION_AM, + }, +}; + /* * lowlevel part */ @@ -133,16 +161,29 @@ static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val) if (freq == 0) return freq; - /* freq *= 12.5 */ - freq *= 125; - freq /= 10; - /* crystal fixup */ - if (tea->tea5759) - freq += TEA575X_FMIF; - else + switch (tea->band) { + case BAND_FM: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ freq -= TEA575X_FMIF; + break; + case BAND_FM_JAPAN: + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + freq += TEA575X_FMIF; + break; + case BAND_AM: + /* crystal fixup */ + freq -= TEA575X_AMIF; + break; + } - return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ + return clamp(freq * 16, bands[tea->band].rangelow, + bands[tea->band].rangehigh); /* from kHz */ } static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) @@ -152,19 +193,35 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) static void snd_tea575x_set_freq(struct snd_tea575x *tea) { - u32 freq = tea->freq; + u32 freq = tea->freq / 16; /* to kHz */ + u32 band = 0; - freq /= 16; /* to kHz */ - /* crystal fixup */ - if (tea->tea5759) - freq -= TEA575X_FMIF; - else + switch (tea->band) { + case BAND_FM: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ freq += TEA575X_FMIF; - /* freq /= 12.5 */ - freq *= 10; - freq /= 125; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_FM_JAPAN: + band = TEA575X_BIT_BAND_FM; + /* crystal fixup */ + freq -= TEA575X_FMIF; + /* freq /= 12.5 */ + freq *= 10; + freq /= 125; + break; + case BAND_AM: + band = TEA575X_BIT_BAND_MW; + /* crystal fixup */ + freq += TEA575X_AMIF; + break; + } - tea->val &= ~TEA575X_BIT_FREQ_MASK; + tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK); + tea->val |= band; tea->val |= freq & TEA575X_BIT_FREQ_MASK; snd_tea575x_write(tea, tea->val); tea->freq = snd_tea575x_val_to_freq(tea, tea->val); @@ -190,23 +247,57 @@ static int vidioc_querycap(struct file *file, void *priv, return 0; } +static int vidioc_enum_freq_bands(struct file *file, void *priv, + struct v4l2_frequency_band *band) +{ + struct snd_tea575x *tea = video_drvdata(file); + int index; + + if (band->tuner != 0) + return -EINVAL; + + switch (band->index) { + case 0: + if (tea->tea5759) + index = BAND_FM_JAPAN; + else + index = BAND_FM; + break; + case 1: + if (tea->has_am) { + index = BAND_AM; + break; + } + /* Fall through */ + default: + return -EINVAL; + } + + *band = bands[index]; + if (!tea->cannot_read_data) + band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; + + return 0; +} + static int vidioc_g_tuner(struct file *file, void *priv, struct v4l2_tuner *v) { struct snd_tea575x *tea = video_drvdata(file); + struct v4l2_frequency_band band_fm = { 0, }; if (v->index > 0) return -EINVAL; snd_tea575x_read(tea); + vidioc_enum_freq_bands(file, priv, &band_fm); - strcpy(v->name, "FM"); + memset(v, 0, sizeof(*v)); + strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name)); v->type = V4L2_TUNER_RADIO; - v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO; - if (!tea->cannot_read_data) - v->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED; - v->rangelow = FREQ_LO; - v->rangehigh = FREQ_HI; + v->capability = band_fm.capability; + v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow; + v->rangehigh = band_fm.rangehigh; v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO; v->audmode = (tea->val & TEA575X_BIT_MONO) ? V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO; @@ -218,13 +309,17 @@ static int vidioc_s_tuner(struct file *file, void *priv, struct v4l2_tuner *v) { struct snd_tea575x *tea = video_drvdata(file); + u32 orig_val = tea->val; if (v->index) return -EINVAL; tea->val &= ~TEA575X_BIT_MONO; if (v->audmode == V4L2_TUNER_MODE_MONO) tea->val |= TEA575X_BIT_MONO; - snd_tea575x_write(tea, tea->val); + /* Only apply changes if currently tuning FM */ + if (tea->band != BAND_AM && tea->val != orig_val) + snd_tea575x_set_freq(tea); + return 0; } @@ -248,8 +343,15 @@ static int vidioc_s_frequency(struct file *file, void *priv, if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) return -EINVAL; - tea->val &= ~TEA575X_BIT_SEARCH; - tea->freq = clamp(f->frequency, FREQ_LO, FREQ_HI); + if (tea->has_am && f->frequency < (20000 * 16)) + tea->band = BAND_AM; + else if (tea->tea5759) + tea->band = BAND_FM_JAPAN; + else + tea->band = BAND_FM; + + tea->freq = clamp(f->frequency, bands[tea->band].rangelow, + bands[tea->band].rangehigh); snd_tea575x_set_freq(tea); return 0; } @@ -259,13 +361,35 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, { struct snd_tea575x *tea = video_drvdata(file); unsigned long timeout; - int i; + int i, spacing; if (tea->cannot_read_data) return -ENOTTY; if (a->tuner || a->wrap_around) return -EINVAL; + if (a->rangelow || a->rangehigh) { + for (i = 0; i < ARRAY_SIZE(bands); i++) { + if ((i == BAND_FM && tea->tea5759) || + (i == BAND_FM_JAPAN && !tea->tea5759) || + (i == BAND_AM && !tea->has_am)) + continue; + if (bands[i].rangelow == a->rangelow && + bands[i].rangehigh == a->rangehigh) + break; + } + if (i == ARRAY_SIZE(bands)) + return -EINVAL; /* No matching band found */ + if (i != tea->band) { + tea->band = i; + tea->freq = clamp(tea->freq, bands[i].rangelow, + bands[i].rangehigh); + snd_tea575x_set_freq(tea); + } + } + + spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */ + /* clear the frequency, HW will fill it in */ tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; @@ -297,10 +421,10 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, if (freq == 0) /* shouldn't happen */ break; /* - * if we moved by less than 50 kHz, or in the wrong - * direction, continue seeking + * if we moved by less than the spacing, or in the + * wrong direction, continue seeking */ - if (abs(tea->freq - freq) < 16 * 50 || + if (abs(tea->freq - freq) < 16 * spacing || (a->seek_upward && freq < tea->freq) || (!a->seek_upward && freq > tea->freq)) { snd_tea575x_write(tea, tea->val); @@ -344,6 +468,7 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { .vidioc_g_frequency = vidioc_g_frequency, .vidioc_s_frequency = vidioc_s_frequency, .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek, + .vidioc_enum_freq_bands = vidioc_enum_freq_bands, .vidioc_log_status = v4l2_ctrl_log_status, .vidioc_subscribe_event = v4l2_ctrl_subscribe_event, .vidioc_unsubscribe_event = v4l2_event_unsubscribe, -- cgit v1.2.3 From 559c2009003bb8092e4927a4bac99cbf75834979 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 11 Aug 2012 12:55:22 -0300 Subject: [media] radio-shark: Add support for suspend & resume Signed-off-by: Hans de Goede Signed-off-by: Mauro Carvalho Chehab --- drivers/media/radio/radio-shark.c | 43 ++++++++++++++++++++++++++++++++++++++- include/sound/tea575x-tuner.h | 1 + sound/i2c/other/tea575x-tuner.c | 3 ++- 3 files changed, 45 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/drivers/media/radio/radio-shark.c b/drivers/media/radio/radio-shark.c index e1970bf031a4..248bc3a30832 100644 --- a/drivers/media/radio/radio-shark.c +++ b/drivers/media/radio/radio-shark.c @@ -59,7 +59,8 @@ MODULE_LICENSE("GPL"); #define v4l2_dev_to_shark(d) container_of(d, struct shark_device, v4l2_dev) -enum { BLUE_LED, BLUE_PULSE_LED, RED_LED, NO_LEDS }; +/* Note BLUE_IS_PULSE comes after NO_LEDS as it is a status bit, not a LED */ +enum { BLUE_LED, BLUE_PULSE_LED, RED_LED, NO_LEDS, BLUE_IS_PULSE }; struct shark_device { struct usb_device *usbdev; @@ -190,6 +191,7 @@ static void shark_led_set_blue(struct led_classdev *led_cdev, atomic_set(&shark->brightness[BLUE_LED], value); set_bit(BLUE_LED, &shark->brightness_new); + clear_bit(BLUE_IS_PULSE, &shark->brightness_new); schedule_work(&shark->led_work); } @@ -201,6 +203,7 @@ static void shark_led_set_blue_pulse(struct led_classdev *led_cdev, atomic_set(&shark->brightness[BLUE_PULSE_LED], 256 - value); set_bit(BLUE_PULSE_LED, &shark->brightness_new); + set_bit(BLUE_IS_PULSE, &shark->brightness_new); schedule_work(&shark->led_work); } @@ -240,6 +243,7 @@ static int shark_register_leds(struct shark_device *shark, struct device *dev) { int i, retval; + atomic_set(&shark->brightness[BLUE_LED], 127); INIT_WORK(&shark->led_work, shark_led_work); for (i = 0; i < NO_LEDS; i++) { shark->leds[i] = shark_led_templates[i]; @@ -266,6 +270,16 @@ static void shark_unregister_leds(struct shark_device *shark) cancel_work_sync(&shark->led_work); } + +static void shark_resume_leds(struct shark_device *shark) +{ + if (test_bit(BLUE_IS_PULSE, &shark->brightness_new)) + set_bit(BLUE_PULSE_LED, &shark->brightness_new); + else + set_bit(BLUE_LED, &shark->brightness_new); + set_bit(RED_LED, &shark->brightness_new); + schedule_work(&shark->led_work); +} #else static int shark_register_leds(struct shark_device *shark, struct device *dev) { @@ -274,6 +288,7 @@ static int shark_register_leds(struct shark_device *shark, struct device *dev) return 0; } static inline void shark_unregister_leds(struct shark_device *shark) { } +static inline void shark_resume_leds(struct shark_device *shark) { } #endif static void usb_shark_disconnect(struct usb_interface *intf) @@ -359,6 +374,27 @@ err_alloc_buffer: return retval; } +#ifdef CONFIG_PM +int usb_shark_suspend(struct usb_interface *intf, pm_message_t message) +{ + return 0; +} + +int usb_shark_resume(struct usb_interface *intf) +{ + struct v4l2_device *v4l2_dev = usb_get_intfdata(intf); + struct shark_device *shark = v4l2_dev_to_shark(v4l2_dev); + + mutex_lock(&shark->tea.mutex); + snd_tea575x_set_freq(&shark->tea); + mutex_unlock(&shark->tea.mutex); + + shark_resume_leds(shark); + + return 0; +} +#endif + /* Specify the bcdDevice value, as the radioSHARK and radioSHARK2 share ids */ static struct usb_device_id usb_shark_device_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE_AND_VERSION | @@ -378,5 +414,10 @@ static struct usb_driver usb_shark_driver = { .probe = usb_shark_probe, .disconnect = usb_shark_disconnect, .id_table = usb_shark_device_table, +#ifdef CONFIG_PM + .suspend = usb_shark_suspend, + .resume = usb_shark_resume, + .reset_resume = usb_shark_resume, +#endif }; module_usb_driver(usb_shark_driver); diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h index 2a6953568554..098c4de44945 100644 --- a/include/sound/tea575x-tuner.h +++ b/include/sound/tea575x-tuner.h @@ -73,5 +73,6 @@ struct snd_tea575x { int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner); void snd_tea575x_exit(struct snd_tea575x *tea); +void snd_tea575x_set_freq(struct snd_tea575x *tea); #endif /* __SOUND_TEA575X_TUNER_H */ diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 88bbd88c066e..cd79ed590f9a 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -191,7 +191,7 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea) return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea)); } -static void snd_tea575x_set_freq(struct snd_tea575x *tea) +void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq / 16; /* to kHz */ u32 band = 0; @@ -571,3 +571,4 @@ module_exit(alsa_tea575x_module_exit) EXPORT_SYMBOL(snd_tea575x_init); EXPORT_SYMBOL(snd_tea575x_exit); +EXPORT_SYMBOL(snd_tea575x_set_freq); -- cgit v1.2.3 From b35cc8225845112a616e3a2266d2fde5ab13d3ab Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 5 Sep 2012 15:32:18 +0300 Subject: ALSA: compress_core: integer overflow in snd_compr_allocate_buffer() These are 32 bit values that come from the user, we need to check for integer overflows or we could end up allocating a smaller buffer than expected. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index eb60cb8dbb8a..68fe02c7400a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -407,6 +407,10 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL; -- cgit v1.2.3 From 62cbde1868b16e7cf1ed115cdfb9cbe82e230f0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 14 Sep 2012 11:58:54 +0200 Subject: ALSA: hda - Add mic-mute LED control for HP laptop Some of new HP laptops have a LED for microphone (or recording) mute, and it's controlled by GPIO pin 3. Bind this with the capture switch to turn it on/off properly by the mixer change. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 84 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 74 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index eef9c6cad25d..bb6c50e7b5da 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -103,6 +103,7 @@ enum { STAC_HP_ZEPHYR, STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, + STAC_92HD83XXX_HP_MIC_LED, STAC_92HD83XXX_MODELS }; @@ -215,6 +216,9 @@ struct sigmatel_spec { unsigned int vref_mute_led_nid; /* pin NID for mute-LED vref control */ unsigned int vref_led; + unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */ + bool mic_mute_led_on; /* current mic mute state */ + /* stream */ unsigned int stream_delay; @@ -1679,6 +1683,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_HP_ZEPHYR] = "hp-zephyr", [STAC_92HD83XXX_HP_LED] = "hp-led", [STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led", + [STAC_92HD83XXX_HP_MIC_LED] = "hp-mic-led", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1703,6 +1708,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, + "HP Folio", STAC_92HD83XXX_HP_MIC_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, @@ -2791,18 +2798,27 @@ stac_control_new(struct sigmatel_spec *spec, return knew; } -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) +static struct snd_kcontrol_new * +add_control_temp(struct sigmatel_spec *spec, + const struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) { struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, HDA_SUBDEV_AMP_FLAG); if (!knew) - return -ENOMEM; + return NULL; knew->index = idx; knew->private_value = val; - return 0; + return knew; +} + +static int stac92xx_add_control_temp(struct sigmatel_spec *spec, + const struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) +{ + return add_control_temp(spec, ktemp, idx, name, val) ? 0 : -ENOMEM; } static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec, @@ -3245,18 +3261,56 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, return 0; } +static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, + unsigned int dir_mask, unsigned int data); + +/* hook for controlling mic-mute LED GPIO */ +static int stac92xx_capture_sw_put_led(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + int err; + bool mute; + + err = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + if (err <= 0) + return err; + mute = !(ucontrol->value.integer.value[0] && + ucontrol->value.integer.value[1]); + if (spec->mic_mute_led_on != mute) { + spec->mic_mute_led_on = mute; + if (mute) + spec->gpio_data |= spec->mic_mute_led_gpio; + else + spec->gpio_data &= ~spec->mic_mute_led_gpio; + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); + } + return err; +} + static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol, unsigned long sw, int idx) { + struct sigmatel_spec *spec = codec->spec; + struct snd_kcontrol_new *knew; int err; + err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, "Capture Volume", vol); if (err < 0) return err; - err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_MUTE, idx, - "Capture Switch", sw); - if (err < 0) - return err; + + knew = add_control_temp(spec, + &stac92xx_control_templates[STAC_CTL_WIDGET_MUTE], + idx, "Capture Switch", sw); + if (!knew) + return -ENOMEM; + /* add a LED hook for some HP laptops */ + if (spec->mic_mute_led_gpio) + knew->put = stac92xx_capture_sw_put_led; + return 0; } @@ -5579,6 +5633,9 @@ again: case STAC_92HD83XXX_HP_INV_LED: default_polarity = 1; break; + case STAC_92HD83XXX_HP_MIC_LED: + spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ + break; } if (find_mute_led_cfg(codec, default_polarity)) @@ -5597,6 +5654,13 @@ again: } } + if (spec->mic_mute_led_gpio) { + spec->gpio_mask |= spec->mic_mute_led_gpio; + spec->gpio_dir |= spec->mic_mute_led_gpio; + spec->mic_mute_led_on = true; + spec->gpio_data |= spec->mic_mute_led_gpio; + } + err = stac92xx_parse_auto_config(codec); if (!err) { if (spec->board_config < 0) { -- cgit v1.2.3 From 4dc040a0b34890d2adc0d63da6e9bfb4eb791b19 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 17 Sep 2012 11:51:25 +0530 Subject: ALSA: compress - move the buffer check Commit ALSA: compress_core: integer overflow in snd_compr_allocate_buffer() added a new error check for input params. this add new routine for input checks and moves buffer overflow check to this new routine. This allows the error value to be propogated to user space Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 68fe02c7400a..bd7f28e89254 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -407,10 +407,6 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, unsigned int buffer_size; void *buffer; - if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) - return -EINVAL; - buffer_size = params->buffer.fragment_size * params->buffer.fragments; if (stream->ops->copy) { buffer = NULL; @@ -429,6 +425,16 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_input(struct snd_compr_params *params) +{ + /* first let's check the buffer parameter's */ + if (params->buffer.fragment_size == 0 || + params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + return -EINVAL; + + return 0; +} + static int snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) { @@ -447,11 +453,17 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) retval = -EFAULT; goto out; } + + retval = snd_compress_check_input(params); + if (retval) + goto out; + retval = snd_compr_allocate_buffer(stream, params); if (retval) { retval = -ENOMEM; goto out; } + retval = stream->ops->set_params(stream, params); if (retval) goto out; -- cgit v1.2.3 From fb4a9779f1267539292268d7363e17180eb4ee5c Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 17 Sep 2012 11:51:26 +0530 Subject: ALSA: Compress - add codec parameter checks Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/compress_params.h | 1 + sound/core/compress_offload.c | 10 ++++++++++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h index da4a456de032..602dc6c45d1a 100644 --- a/include/sound/compress_params.h +++ b/include/sound/compress_params.h @@ -72,6 +72,7 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 /* * Profile and modes are listed with bit masks. This allows for a diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index bd7f28e89254..c40ae573346d 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -432,6 +432,16 @@ static int snd_compress_check_input(struct snd_compr_params *params) params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) return -EINVAL; + /* now codec parameters */ + if (params->codec.id == 0 || params->codec.id > SND_AUDIOCODEC_MAX) + return -EINVAL; + + if (params->codec.ch_in == 0 || params->codec.ch_out == 0) + return -EINVAL; + + if (!(params->codec.sample_rate & SNDRV_PCM_RATE_8000_192000)) + return -EINVAL; + return 0; } -- cgit v1.2.3 From d279fae8a41690ec1b20c07be8c6f42f8af27a17 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 17 Sep 2012 13:10:23 +0800 Subject: ALSA: hda - Add another pci id for Haswell board A new PCI id 0x0d0c for Haswell HDA Controller. [root@SKBM04SDP ~]# lspci |grep Audio 00:03.0 Audio device: Intel Corporation Device 0d0c (rev 02) 00:1b.0 Audio device: Intel Corporation Lynx Point HD Audio Controller Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 195d84726187..207d0a20f317 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3431,6 +3431,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + { PCI_DEVICE(0x8086, 0x0d0c), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From d41789b2660e5b18b8401bf83ebcd502916c2cb5 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 17 Sep 2012 13:34:31 +0800 Subject: ASoC: mx27vis: retrieve gpio numbers from platform_data Rather than including mach/iomux-mx27.h to define gpio numbers and set up the pins, the patch moves all these into machine code and has the gpio numbers passed to driver via platform_data. As the result, we can remove the mach/iomux-mx27.h inclusion from driver. Signed-off-by: Shawn Guo Acked-by: Javier Martin Signed-off-by: Mark Brown --- arch/arm/mach-imx/mach-imx27_visstrim_m10.c | 42 ++++++++++++++++++++++++++++- include/linux/platform_data/asoc-mx27vis.h | 11 ++++++++ sound/soc/fsl/mx27vis-aic32x4.c | 42 ++++++++++++++--------------- 3 files changed, 72 insertions(+), 23 deletions(-) create mode 100644 include/linux/platform_data/asoc-mx27vis.h (limited to 'sound') diff --git a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c index f264ddddd47c..562722959666 100644 --- a/arch/arm/mach-imx/mach-imx27_visstrim_m10.c +++ b/arch/arm/mach-imx/mach-imx27_visstrim_m10.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include @@ -58,6 +59,11 @@ #define EXPBOARD_BIT1 (GPIO_PORTD + 27) #define EXPBOARD_BIT0 (GPIO_PORTD + 28) +#define AMP_GAIN_0 (GPIO_PORTF + 9) +#define AMP_GAIN_1 (GPIO_PORTF + 8) +#define AMP_MUTE_SDL (GPIO_PORTE + 5) +#define AMP_MUTE_SDR (GPIO_PORTF + 7) + static const int visstrim_m10_pins[] __initconst = { /* UART1 (console) */ PE12_PF_UART1_TXD, @@ -139,6 +145,11 @@ static const int visstrim_m10_pins[] __initconst = { EXPBOARD_BIT2 | GPIO_GPIO | GPIO_IN | GPIO_PUEN, EXPBOARD_BIT1 | GPIO_GPIO | GPIO_IN | GPIO_PUEN, EXPBOARD_BIT0 | GPIO_GPIO | GPIO_IN | GPIO_PUEN, + /* Audio AMP control */ + AMP_GAIN_0 | GPIO_GPIO | GPIO_OUT, + AMP_GAIN_1 | GPIO_GPIO | GPIO_OUT, + AMP_MUTE_SDL | GPIO_GPIO | GPIO_OUT, + AMP_MUTE_SDR | GPIO_GPIO | GPIO_OUT, }; static struct gpio visstrim_m10_version_gpios[] = { @@ -166,6 +177,26 @@ static const struct gpio visstrim_m10_gpios[] __initconst = { .flags = GPIOF_DIR_OUT | GPIOF_INIT_LOW, .label = "usbotg_cs", }, + { + .gpio = AMP_GAIN_0, + .flags = GPIOF_DIR_OUT, + .label = "amp-gain-0", + }, + { + .gpio = AMP_GAIN_1, + .flags = GPIOF_DIR_OUT, + .label = "amp-gain-1", + }, + { + .gpio = AMP_MUTE_SDL, + .flags = GPIOF_DIR_OUT, + .label = "amp-mute-sdl", + }, + { + .gpio = AMP_MUTE_SDR, + .flags = GPIOF_DIR_OUT, + .label = "amp-mute-sdr", + }, }; /* Camera */ @@ -405,6 +436,14 @@ static const struct imx_ssi_platform_data visstrim_m10_ssi_pdata __initconst = { .flags = IMX_SSI_DMA | IMX_SSI_SYN, }; +/* Audio */ +static const struct snd_mx27vis_platform_data snd_mx27vis_pdata __initconst = { + .amp_gain0_gpio = AMP_GAIN_0, + .amp_gain1_gpio = AMP_GAIN_1, + .amp_mutel_gpio = AMP_MUTE_SDL, + .amp_muter_gpio = AMP_MUTE_SDR, +}; + static void __init visstrim_m10_revision(void) { int exp_version = 0; @@ -463,7 +502,8 @@ static void __init visstrim_m10_board_init(void) imx27_add_fec(NULL); imx_add_gpio_keys(&visstrim_gpio_keys_platform_data); platform_add_devices(platform_devices, ARRAY_SIZE(platform_devices)); - imx_add_platform_device("mx27vis", 0, NULL, 0, NULL, 0); + imx_add_platform_device("mx27vis", 0, NULL, 0, &snd_mx27vis_pdata, + sizeof(snd_mx27vis_pdata)); platform_device_register_resndata(NULL, "soc-camera-pdrv", 0, NULL, 0, &iclink_tvp5150, sizeof(iclink_tvp5150)); gpio_led_register_device(0, &visstrim_m10_led_data); diff --git a/include/linux/platform_data/asoc-mx27vis.h b/include/linux/platform_data/asoc-mx27vis.h new file mode 100644 index 000000000000..409adcd04d04 --- /dev/null +++ b/include/linux/platform_data/asoc-mx27vis.h @@ -0,0 +1,11 @@ +#ifndef __PLATFORM_DATA_ASOC_MX27VIS_H +#define __PLATFORM_DATA_ASOC_MX27VIS_H + +struct snd_mx27vis_platform_data { + int amp_gain0_gpio; + int amp_gain1_gpio; + int amp_mutel_gpio; + int amp_muter_gpio; +}; + +#endif /* __PLATFORM_DATA_ASOC_MX27VIS_H */ diff --git a/sound/soc/fsl/mx27vis-aic32x4.c b/sound/soc/fsl/mx27vis-aic32x4.c index f6d04ad4bb39..2b76877b1789 100644 --- a/sound/soc/fsl/mx27vis-aic32x4.c +++ b/sound/soc/fsl/mx27vis-aic32x4.c @@ -26,13 +26,13 @@ #include #include #include +#include #include #include #include #include #include #include -#include #include "../codecs/tlv320aic32x4.h" #include "imx-ssi.h" @@ -41,20 +41,12 @@ #define MX27VIS_AMP_GAIN 0 #define MX27VIS_AMP_MUTE 1 -#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) -#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) -#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) -#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) - static int mx27vis_amp_gain; static int mx27vis_amp_mute; - -static const int mx27vis_amp_pins[] = { - MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, - MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, -}; +static int mx27vis_amp_gain0_gpio; +static int mx27vis_amp_gain1_gpio; +static int mx27vis_amp_mutel_gpio; +static int mx27vis_amp_muter_gpio; static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -109,13 +101,13 @@ static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, switch (reg) { case MX27VIS_AMP_GAIN: - gpio_set_value(MX27VIS_PIN_G0, value & 1); - gpio_set_value(MX27VIS_PIN_G1, value >> 1); + gpio_set_value(mx27vis_amp_gain0_gpio, value & 1); + gpio_set_value(mx27vis_amp_gain1_gpio, value >> 1); mx27vis_amp_gain = value; break; case MX27VIS_AMP_MUTE: - gpio_set_value(MX27VIS_PIN_SDL, value & 1); - gpio_set_value(MX27VIS_PIN_SDR, value >> 1); + gpio_set_value(mx27vis_amp_mutel_gpio, value & 1); + gpio_set_value(mx27vis_amp_muter_gpio, value >> 1); mx27vis_amp_mute = value; break; } @@ -190,8 +182,19 @@ static struct snd_soc_card mx27vis_aic32x4 = { static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { + struct snd_mx27vis_platform_data *pdata = pdev->dev.platform_data; int ret; + if (!pdata) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + + mx27vis_amp_gain0_gpio = pdata->amp_gain0_gpio; + mx27vis_amp_gain1_gpio = pdata->amp_gain1_gpio; + mx27vis_amp_mutel_gpio = pdata->amp_mutel_gpio; + mx27vis_amp_muter_gpio = pdata->amp_muter_gpio; + mx27vis_aic32x4.dev = &pdev->dev; ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { @@ -213,11 +216,6 @@ static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); - ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, - ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) - printk(KERN_ERR "ASoC: unable to setup gpios\n"); - return ret; } -- cgit v1.2.3 From 4b527b6516ab1f0af8aaedd02dbf71ee2c1180f4 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 18 Sep 2012 14:26:59 +0200 Subject: ALSA: hda - limit internal mic boost for Asus X202E When the input gain for the internal mic is set to its maximum level, the background noise becomes so high - and any relevant signal clipped - that the setting becomes unusable. It is better to limit the amplification. BugLink: https://bugs.launchpad.net/bugs/1052460 Signed-off-by: David Henningsson Cc: [v3.5+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5f36a179eb5..74fb6fd6768a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3666,6 +3666,32 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) update_power_state(codec, 0x21, AC_PWRST_D3); } +/* + * pin fix-up + */ +enum { + VIA_FIXUP_INTMIC_BOOST, +}; + +static void via_fixup_intmic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + override_mic_boost(codec, 0x30, 0, 2, 40); +} + +static const struct hda_fixup via_fixups[] = { + [VIA_FIXUP_INTMIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = via_fixup_intmic_boost, + }, +}; + +static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), + {} +}; + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3682,6 +3708,9 @@ static int patch_vt2002P(struct hda_codec *codec) override_mic_boost(codec, 0x29, 0, 3, 40); add_secret_dac_path(codec); + snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); if (err < 0) { -- cgit v1.2.3 From 13c57e5b868b4f023f6436d8c6a079eaffd7f3a8 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 14 Sep 2012 16:14:34 -0500 Subject: ASoC: fsl: use snd_soc_register_card to register the card Use snd_soc_register_card() instead of platform_device_alloc("soc-audio") to register the sound card from the machine drivers. The use of platform_device_alloc is deprecated. Although several other drivers still use platform_device_alloc(), the Freescale drivers were not using it to pass driver data. Instead of fixing the driver data usage, it's better to replace the deprecated code. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 32 +++++++------------------------- sound/soc/fsl/p1022_ds.c | 31 ++++++------------------------- 2 files changed, 13 insertions(+), 50 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 60bcba1bc30e..9ff9318c52b9 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -192,7 +192,6 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; int ret = -ENODEV; const char *sprop; @@ -341,34 +340,22 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) machine_data->card.probe = mpc8610_hpcd_machine_probe; machine_data->card.remove = mpc8610_hpcd_machine_remove; machine_data->card.name = pdev->name; /* The platform driver name */ + machine_data->card.owner = THIS_MODULE; + machine_data->card.dev = &pdev->dev; machine_data->card.num_links = 2; machine_data->card.dai_link = machine_data->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &machine_data->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&machine_data->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); - goto error_sound; + dev_err(&pdev->dev, "could not register card\n"); + goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; -error_sound: - platform_device_put(sound_device); error: kfree(machine_data); error_alloc: @@ -383,17 +370,12 @@ error_alloc: */ static int __devexit mpc8610_hpcd_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(machine_data); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 50adf4032bcc..144d49603637 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -202,7 +202,6 @@ static int p1022_ds_probe(struct platform_device *pdev) container_of(dev, struct platform_device, dev); struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; - struct platform_device *sound_device = NULL; struct machine_data *mdata; int ret = -ENODEV; const char *sprop; @@ -349,36 +348,23 @@ static int p1022_ds_probe(struct platform_device *pdev) mdata->card.probe = p1022_ds_machine_probe; mdata->card.remove = p1022_ds_machine_remove; mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.owner = THIS_MODULE; + mdata->card.dev = &pdev->dev; mdata->card.num_links = 2; mdata->card.dai_link = mdata->dai; - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&pdev->dev, "platform device alloc failed\n"); - ret = -ENOMEM; - goto error; - } - - /* Associate the card data with the sound device */ - platform_set_drvdata(sound_device, &mdata->card); - /* Register with ASoC */ - ret = platform_device_add(sound_device); + ret = snd_soc_register_card(&mdata->card); if (ret) { - dev_err(&pdev->dev, "platform device add failed\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } - dev_set_drvdata(&pdev->dev, sound_device); of_node_put(codec_np); return 0; error: - if (sound_device) - platform_device_put(sound_device); - kfree(mdata); error_put: of_node_put(codec_np); @@ -392,17 +378,12 @@ error_put: */ static int __devexit p1022_ds_remove(struct platform_device *pdev) { - struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); - struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); struct machine_data *mdata = container_of(card, struct machine_data, card); - platform_device_unregister(sound_device); - + snd_soc_unregister_card(card); kfree(mdata); - sound_device->dev.platform_data = NULL; - - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- cgit v1.2.3 From d55438beb2329493cb54df5175d83be65a8d5100 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 14 Sep 2012 16:14:36 -0500 Subject: ASoC: fsl: remove unnecessary call to dma_unmap_single Remove a call to dma_unmap_single() from the PowerPC ASoC DMA driver. The buffer is allocated and not actually mapped, so the unmap call doesn't make sense. It was probably left over from some early version of the driver. This bug was unnoticed for so long because the DMA mapping functions normally don't do anything on PowerPC. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 96bb92dd174c..6feb26500580 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -823,12 +823,6 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) if (dma_private->irq) free_irq(dma_private->irq, dma_private); - if (dma_private->ld_buf_phys) { - dma_unmap_single(dev, dma_private->ld_buf_phys, - sizeof(dma_private->link), - DMA_TO_DEVICE); - } - /* Deallocate the fsl_dma_private structure */ dma_free_coherent(dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); -- cgit v1.2.3 From 4c2474c007867c102c96482f3bacb1fdf209958c Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 14 Sep 2012 16:14:37 -0500 Subject: ASoC: wm8960: add support for big-endian audio samples PowerPC ASoC drivers frequently use the _BE variants of the SNDRV_PCM_FORMAT macros, so we need to look for those as well. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7cb0d07ca8a6..066250e3f7f4 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -557,18 +557,25 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; + snd_pcm_format_t format = params_format(params); int i; /* bit size */ - switch (params_format(params)) { + switch (format) { case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: break; case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: iface |= 0x0004; break; case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: iface |= 0x0008; break; + default: + dev_err(codec->dev, "unsupported format %i\n", format); + return -EINVAL; } /* Update filters for the new rate */ -- cgit v1.2.3 From 86767b7d5b3cdbd105e7d7066d671b52aa208188 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Sep 2012 13:57:27 +0200 Subject: ASoC: Avoid recalculating the bitmask for SOC_ENUM controls For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +++- sound/soc/codecs/twl4030.c | 8 +++----- sound/soc/soc-core.c | 16 ++++++---------- sound/soc/soc-dapm.c | 22 ++++++++-------------- 4 files changed, 20 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 313b7660562c..91244a096c19 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -160,7 +161,8 @@ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .max = xmax, .texts = xtexts } + .max = xmax, .texts = xtexts, \ + .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0} #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) #define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 391fcfc7b63b..2548f5c56885 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -999,7 +999,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; - unsigned short mask, bitmask; + unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -1007,18 +1007,16 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits(codec, e->reg, mask, val); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e5b0713e6f3c..9a6daf997319 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2413,16 +2413,14 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] - = (val >> e->shift_l) & (bitmask - 1); + = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2443,19 +2441,17 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; - unsigned int mask, bitmask; + unsigned int mask; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } return snd_soc_update_bits_locked(codec, e->reg, mask, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f7999e949acb..a18d115bc507 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -355,12 +355,10 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mux: { struct soc_enum *e = (struct soc_enum *) w->kcontrol_news[i].private_value; - int val, item, bitmask; + int val, item; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = soc_widget_read(w, e->reg); - item = (val >> e->shift_l) & (bitmask - 1); + item = (val >> e->shift_l) & e->mask; p->connect = 0; for (i = 0; i < e->max; i++) { @@ -2677,15 +2675,13 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, bitmask; + unsigned int val; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; val = snd_soc_read(widget->codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); + ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; if (e->shift_l != e->shift_r) ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & (bitmask - 1); + (val >> e->shift_r) & e->mask; return 0; } @@ -2709,22 +2705,20 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; - unsigned int mask, bitmask; + unsigned int mask; struct snd_soc_dapm_update update; int wi; - for (bitmask = 1; bitmask < e->max; bitmask <<= 1) - ; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; - mask = (bitmask - 1) << e->shift_l; + mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { if (ucontrol->value.enumerated.item[1] > e->max - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= (bitmask - 1) << e->shift_r; + mask |= e->mask << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); -- cgit v1.2.3 From d37777a13b4c4b32e274013846d739e5bbbf6f8d Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 18 Sep 2012 17:20:05 +0800 Subject: ASoC: imx-pcm-dma: check kzalloc return value in function snd_imx_open It fixed smatch warning: sound/soc/fsl/imx-pcm-dma.c:112 snd_imx_open() error: potential null dereference 'dma_data'. (kzalloc returns null) Signed-off-by: Richard Zhao Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-dma.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 48f9d886f020..a23505a9ae64 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -109,6 +109,9 @@ static int snd_imx_open(struct snd_pcm_substream *substream) dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + if (!dma_data) + return -ENOMEM; + dma_data->peripheral_type = dma_params->shared_peripheral ? IMX_DMATYPE_SSI_SP : IMX_DMATYPE_SSI; dma_data->priority = DMA_PRIO_HIGH; -- cgit v1.2.3 From f31e08e160d646952a5142e3f92f3fcdc645a09b Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 18 Sep 2012 17:20:06 +0800 Subject: ASoC: imx-pcm-dma: open function failed when snd_dmaengine_pcm_open fail snd_imx_open should return error code returned by snd_dmaengine_pcm_open. Signed-off-by: Richard Zhao Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index a23505a9ae64..9a15bc4bd570 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -120,7 +120,7 @@ static int snd_imx_open(struct snd_pcm_substream *substream) ret = snd_dmaengine_pcm_open(substream, filter, dma_data); if (ret) { kfree(dma_data); - return 0; + return ret; } snd_dmaengine_pcm_set_data(substream, dma_data); -- cgit v1.2.3 From f3a50c95e275c2e553e2a6dcc646eef3daf98a3e Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 18 Sep 2012 16:38:30 +0800 Subject: ASoC: imx-audmux: remove null check of audmux_base in audmux_read_file When audmux_read_file is called, it means the driver is already initialised successfully, so we don't need to check audmux_base. It also fix smatch warning: sound/soc/fsl/imx-audmux.c:78 audmux_read_file() warn: possible memory leak of 'buf' Signed-off-by: Richard Zhao Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index e7c800ebbd75..524ce6210cee 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -74,9 +74,6 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (!audmux_base) - return -ENOSYS; - if (audmux_clk) clk_prepare_enable(audmux_clk); -- cgit v1.2.3 From f515b67381de0a4a28d639e0f1bb587a3a49f0d2 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 13 Sep 2012 17:43:11 -0400 Subject: ASoC: fsl: mpc5200 combine psc_dma platform data The mpc5200_psc_ac97 and mpc5200_psc_i2s modules rely on shared platform data with mpc5200_dma. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 24 ++++-------------------- sound/soc/fsl/mpc5200_dma.h | 3 +++ sound/soc/fsl/mpc5200_psc_ac97.c | 5 +++++ sound/soc/fsl/mpc5200_psc_i2s.c | 5 +++++ 4 files changed, 17 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9a3f7c5ab687..9997c039bb24 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -370,7 +370,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct platform_device *op) +int mpc5200_audio_dma_create(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,8 +487,9 @@ out_unmap: iounmap(regs); return ret; } +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); -static int mpc5200_hpcd_remove(struct platform_device *op) +int mpc5200_audio_dma_destroy(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -510,24 +511,7 @@ static int mpc5200_hpcd_remove(struct platform_device *op) return 0; } - -static struct of_device_id mpc5200_hpcd_match[] = { - { .compatible = "fsl,mpc5200-pcm", }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); - -static struct platform_driver mpc5200_hpcd_of_driver = { - .probe = mpc5200_hpcd_probe, - .remove = mpc5200_hpcd_remove, - .driver = { - .owner = THIS_MODULE, - .name = "mpc5200-pcm-audio", - .of_match_table = mpc5200_hpcd_match, - } -}; - -module_platform_driver(mpc5200_hpcd_of_driver); +EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy); MODULE_AUTHOR("Grant Likely "); MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver"); diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index a3c0cd5382fb..dff253fde29a 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -81,4 +81,7 @@ to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) return &psc_dma->playback; } +int mpc5200_audio_dma_create(struct platform_device *op); +int mpc5200_audio_dma_destroy(struct platform_device *op); + #endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */ diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index ffa00a2eb770..9a094535fb26 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -278,6 +278,10 @@ static int __devinit psc_ac97_of_probe(struct platform_device *op) struct snd_ac97 ac97; struct mpc52xx_psc __iomem *regs; + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + rc = snd_soc_register_dais(&op->dev, psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); if (rc != 0) { dev_err(&op->dev, "Failed to register DAI\n"); @@ -303,6 +307,7 @@ static int __devinit psc_ac97_of_probe(struct platform_device *op) static int __devexit psc_ac97_of_remove(struct platform_device *op) { + mpc5200_audio_dma_destroy(op); snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_ac97_dai)); return 0; } diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 7b530327553a..c0b7a23ebbfd 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -156,6 +156,10 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op) struct psc_dma *psc_dma; struct mpc52xx_psc __iomem *regs; + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai)); if (rc != 0) { pr_err("Failed to register DAI\n"); @@ -200,6 +204,7 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op) static int __devexit psc_i2s_of_remove(struct platform_device *op) { + mpc5200_audio_dma_destroy(op); snd_soc_unregister_dais(&op->dev, ARRAY_SIZE(psc_i2s_dai)); return 0; } -- cgit v1.2.3 From a4f7b70dc73d611680a485150f2b11bcf23a2d01 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 13 Sep 2012 17:43:12 -0400 Subject: ASoC: fsl: mpc5200 add missing information to snd_soc_dai_driver Add missing dai_driver information to avoid these runtime errors [ 16.433788] asoc: error - multiple DAI f0002c00.i2s registered with no name [ 16.453551] Failed to register DAI [ 16.461222] mpc5200-psc-i2s: probe of f0002c00.i2s failed with error -22 [ 16.475242] asoc: error - multiple DAI f0002000.ac97 registered with no name [ 16.488087] mpc5200-psc-ac97 f0002000.ac97: Failed to register DAI [ 16.502222] mpc5200-psc-ac97: probe of f0002000.ac97 failed with error -22 Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 5 +++++ sound/soc/fsl/mpc5200_psc_i2s.c | 3 +++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9a094535fb26..a313c0ae36db 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -237,15 +237,18 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = { static struct snd_soc_dai_driver psc_ac97_dai[] = { { + .name = "mpc5200-psc-ac97.0", .ac97_control = 1, .probe = psc_ac97_probe, .playback = { + .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 6, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { + .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, @@ -254,8 +257,10 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { .ops = &psc_ac97_analog_ops, }, { + .name = "mpc5200-psc-ac97.1", .ac97_control = 1, .playback = { + .stream_name = "AC97 SPDIF", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index c0b7a23ebbfd..ba1f0a66358f 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -130,13 +130,16 @@ static const struct snd_soc_dai_ops psc_i2s_dai_ops = { }; static struct snd_soc_dai_driver psc_i2s_dai[] = {{ + .name = "mpc5200-psc-i2s.0", .playback = { + .stream_name = "I2S Playback", .channels_min = 2, .channels_max = 2, .rates = PSC_I2S_RATES, .formats = PSC_I2S_FORMATS, }, .capture = { + .stream_name = "I2S Capture", .channels_min = 2, .channels_max = 2, .rates = PSC_I2S_RATES, -- cgit v1.2.3 From 22bab8ceda6ea2885c6315d07c37b4ef69aefb0c Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 13 Sep 2012 17:43:13 -0400 Subject: ASoC: fsl: cleanup headers in pcm030-audio-fabric Remove unreferenced header files. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index b3af55dcde9d..1353e8fecd7b 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -12,22 +12,13 @@ #include #include -#include #include -#include #include #include -#include -#include -#include -#include -#include #include #include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" -#include "../codecs/wm9712.h" #define DRV_NAME "pcm030-audio-fabric" -- cgit v1.2.3 From 2cbde7abfdd8c3e2c1293b7096477e8bcf10b755 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Sep 2012 20:15:22 +0800 Subject: ASoC: wm8776: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 75 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 60 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 879c356a9045..c32249ddb2e0 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -37,18 +38,46 @@ enum wm8776_chip_type { /* codec private data */ struct wm8776_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk[2]; }; -static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { - 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ - 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ - 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ - 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ - 0xa6, 0x01, 0x01 +static const struct reg_default wm8776_reg_defaults[] = { + { 0, 0x79 }, + { 1, 0x79 }, + { 2, 0x79 }, + { 3, 0xff }, + { 4, 0xff }, + { 5, 0xff }, + { 6, 0x00 }, + { 7, 0x90 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x22 }, + { 11, 0x22 }, + { 12, 0x22 }, + { 13, 0x08 }, + { 14, 0xcf }, + { 15, 0xcf }, + { 16, 0x7b }, + { 17, 0x00 }, + { 18, 0x32 }, + { 19, 0x00 }, + { 20, 0xa6 }, + { 21, 0x01 }, + { 22, 0x01 }, }; +static bool wm8776_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8776_RESET: + return true; + default: + return false; + } +} + static int wm8776_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8776_RESET, 0); @@ -306,6 +335,8 @@ static int wm8776_set_sysclk(struct snd_soc_dai *dai, static int wm8776_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -313,7 +344,7 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8776->regmap); /* Disable the global powerdown; DAPM does the rest */ snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); @@ -396,10 +427,9 @@ static int wm8776_resume(struct snd_soc_codec *codec) static int wm8776_probe(struct snd_soc_codec *codec) { - struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8776->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -434,9 +464,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = { .suspend = wm8776_suspend, .resume = wm8776_resume, .set_bias_level = wm8776_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8776_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8776_reg, .controls = wm8776_snd_controls, .num_controls = ARRAY_SIZE(wm8776_snd_controls), @@ -452,6 +479,18 @@ static const struct of_device_id wm8776_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8776_of_match); +static const struct regmap_config wm8776_regmap = { + .reg_bits = 7, + .val_bits = 9, + .max_register = WM8776_RESET, + + .reg_defaults = wm8776_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8776_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8776_volatile, +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8776_spi_probe(struct spi_device *spi) { @@ -463,7 +502,10 @@ static int __devinit wm8776_spi_probe(struct spi_device *spi) if (wm8776 == NULL) return -ENOMEM; - wm8776->control_type = SND_SOC_SPI; + wm8776->regmap = devm_regmap_init_spi(spi, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + spi_set_drvdata(spi, wm8776); ret = snd_soc_register_codec(&spi->dev, @@ -501,8 +543,11 @@ static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, if (wm8776 == NULL) return -ENOMEM; + wm8776->regmap = devm_regmap_init_i2c(i2c, &wm8776_regmap); + if (IS_ERR(wm8776->regmap)) + return PTR_ERR(wm8776->regmap); + i2c_set_clientdata(i2c, wm8776); - wm8776->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8776, wm8776_dai, ARRAY_SIZE(wm8776_dai)); -- cgit v1.2.3 From 3706163140939bccd58fba739a9820f1d5eebeaf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Sep 2012 11:46:58 +0800 Subject: ASoC: wm8960: Support shared LRCLK If the LRCLK is shared and the WM8960 is clock master then we should enable the LRCM bit to tell the device that it should drive LRCLK when either ADC or DAC is enabled rather than separately driving the two LRCLKs. Signed-off-by: Mark Brown --- include/sound/wm8960.h | 2 ++ sound/soc/codecs/wm8960.c | 11 +++++++++++ 2 files changed, 13 insertions(+) (limited to 'sound') diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index 74e9a95529c5..b5a1ab9ebb4a 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -19,6 +19,8 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ int dres; /* Discharge resistance for headphone outputs */ + + bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; #endif diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 066250e3f7f4..782faa0a3b49 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1036,6 +1036,7 @@ static const struct regmap_config wm8960_regmap = { static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1048,6 +1049,16 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata && pdata->shared_lrclk) { + ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, + 0x4, 0x4); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", + ret); + return ret; + } + } + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From 35ecf7cd96a79d92c1b8433c950a827a2a723db9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Sep 2012 12:53:59 +0800 Subject: ASoC: wm8961: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 426 ++++++++++++++++++---------------------------- 1 file changed, 161 insertions(+), 265 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 719fb69a17c8..4ea64d6e68e2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -31,277 +32,158 @@ #define WM8961_MAX_REGISTER 0xFC -static u16 wm8961_reg_defaults[] = { - 0x009F, /* R0 - Left Input volume */ - 0x009F, /* R1 - Right Input volume */ - 0x0000, /* R2 - LOUT1 volume */ - 0x0000, /* R3 - ROUT1 volume */ - 0x0020, /* R4 - Clocking1 */ - 0x0008, /* R5 - ADC & DAC Control 1 */ - 0x0000, /* R6 - ADC & DAC Control 2 */ - 0x000A, /* R7 - Audio Interface 0 */ - 0x01F4, /* R8 - Clocking2 */ - 0x0000, /* R9 - Audio Interface 1 */ - 0x00FF, /* R10 - Left DAC volume */ - 0x00FF, /* R11 - Right DAC volume */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0040, /* R14 - Audio Interface 2 */ - 0x0000, /* R15 - Software Reset */ - 0x0000, /* R16 */ - 0x007B, /* R17 - ALC1 */ - 0x0000, /* R18 - ALC2 */ - 0x0032, /* R19 - ALC3 */ - 0x0000, /* R20 - Noise Gate */ - 0x00C0, /* R21 - Left ADC volume */ - 0x00C0, /* R22 - Right ADC volume */ - 0x0120, /* R23 - Additional control(1) */ - 0x0000, /* R24 - Additional control(2) */ - 0x0000, /* R25 - Pwr Mgmt (1) */ - 0x0000, /* R26 - Pwr Mgmt (2) */ - 0x0000, /* R27 - Additional Control (3) */ - 0x0000, /* R28 - Anti-pop */ - 0x0000, /* R29 */ - 0x005F, /* R30 - Clocking 3 */ - 0x0000, /* R31 */ - 0x0000, /* R32 - ADCL signal path */ - 0x0000, /* R33 - ADCR signal path */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 */ - 0x0000, /* R39 */ - 0x0000, /* R40 - LOUT2 volume */ - 0x0000, /* R41 - ROUT2 volume */ - 0x0000, /* R42 */ - 0x0000, /* R43 */ - 0x0000, /* R44 */ - 0x0000, /* R45 */ - 0x0000, /* R46 */ - 0x0000, /* R47 - Pwr Mgmt (3) */ - 0x0023, /* R48 - Additional Control (4) */ - 0x0000, /* R49 - Class D Control 1 */ - 0x0000, /* R50 */ - 0x0003, /* R51 - Class D Control 2 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0106, /* R56 - Clocking 4 */ - 0x0000, /* R57 - DSP Sidetone 0 */ - 0x0000, /* R58 - DSP Sidetone 1 */ - 0x0000, /* R59 */ - 0x0000, /* R60 - DC Servo 0 */ - 0x0000, /* R61 - DC Servo 1 */ - 0x0000, /* R62 */ - 0x015E, /* R63 - DC Servo 3 */ - 0x0010, /* R64 */ - 0x0010, /* R65 - DC Servo 5 */ - 0x0000, /* R66 */ - 0x0001, /* R67 */ - 0x0003, /* R68 - Analogue PGA Bias */ - 0x0000, /* R69 - Analogue HP 0 */ - 0x0060, /* R70 */ - 0x01FB, /* R71 - Analogue HP 2 */ - 0x0000, /* R72 - Charge Pump 1 */ - 0x0065, /* R73 */ - 0x005F, /* R74 */ - 0x0059, /* R75 */ - 0x006B, /* R76 */ - 0x0038, /* R77 */ - 0x000C, /* R78 */ - 0x000A, /* R79 */ - 0x006B, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 - Charge Pump B */ - 0x0087, /* R83 */ - 0x0000, /* R84 */ - 0x005C, /* R85 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - Write Sequencer 1 */ - 0x0000, /* R88 - Write Sequencer 2 */ - 0x0000, /* R89 - Write Sequencer 3 */ - 0x0000, /* R90 - Write Sequencer 4 */ - 0x0000, /* R91 - Write Sequencer 5 */ - 0x0000, /* R92 - Write Sequencer 6 */ - 0x0000, /* R93 - Write Sequencer 7 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0000, /* R104 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 */ - 0x0000, /* R109 */ - 0x0000, /* R110 */ - 0x0000, /* R111 */ - 0x0000, /* R112 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 */ - 0x0000, /* R117 */ - 0x0000, /* R118 */ - 0x0000, /* R119 */ - 0x0000, /* R120 */ - 0x0000, /* R121 */ - 0x0000, /* R122 */ - 0x0000, /* R123 */ - 0x0000, /* R124 */ - 0x0000, /* R125 */ - 0x0000, /* R126 */ - 0x0000, /* R127 */ - 0x0000, /* R128 */ - 0x0000, /* R129 */ - 0x0000, /* R130 */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 */ - 0x0000, /* R135 */ - 0x0000, /* R136 */ - 0x0000, /* R137 */ - 0x0000, /* R138 */ - 0x0000, /* R139 */ - 0x0000, /* R140 */ - 0x0000, /* R141 */ - 0x0000, /* R142 */ - 0x0000, /* R143 */ - 0x0000, /* R144 */ - 0x0000, /* R145 */ - 0x0000, /* R146 */ - 0x0000, /* R147 */ - 0x0000, /* R148 */ - 0x0000, /* R149 */ - 0x0000, /* R150 */ - 0x0000, /* R151 */ - 0x0000, /* R152 */ - 0x0000, /* R153 */ - 0x0000, /* R154 */ - 0x0000, /* R155 */ - 0x0000, /* R156 */ - 0x0000, /* R157 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0030, /* R196 */ - 0x0006, /* R197 */ - 0x0000, /* R198 */ - 0x0060, /* R199 */ - 0x0000, /* R200 */ - 0x003F, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 */ - 0x0001, /* R205 */ - 0x0000, /* R206 */ - 0x0181, /* R207 */ - 0x0005, /* R208 */ - 0x0008, /* R209 */ - 0x0008, /* R210 */ - 0x0000, /* R211 */ - 0x013B, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0070, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0003, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0001, /* R226 */ - 0x0008, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0004, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0080, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0052, /* R245 */ - 0x0110, /* R246 */ - 0x0040, /* R247 */ - 0x0000, /* R248 */ - 0x0030, /* R249 */ - 0x0000, /* R250 */ - 0x0000, /* R251 */ - 0x0001, /* R252 - General test 1 */ +static const struct reg_default wm8961_reg_defaults[] = { + { 0, 0x009F }, /* R0 - Left Input volume */ + { 1, 0x009F }, /* R1 - Right Input volume */ + { 2, 0x0000 }, /* R2 - LOUT1 volume */ + { 3, 0x0000 }, /* R3 - ROUT1 volume */ + { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0008 }, /* R5 - ADC & DAC Control 1 */ + { 6, 0x0000 }, /* R6 - ADC & DAC Control 2 */ + { 7, 0x000A }, /* R7 - Audio Interface 0 */ + { 8, 0x01F4 }, /* R8 - Clocking2 */ + { 9, 0x0000 }, /* R9 - Audio Interface 1 */ + { 10, 0x00FF }, /* R10 - Left DAC volume */ + { 11, 0x00FF }, /* R11 - Right DAC volume */ + + { 14, 0x0040 }, /* R14 - Audio Interface 2 */ + + { 17, 0x007B }, /* R17 - ALC1 */ + { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x0032 }, /* R19 - ALC3 */ + { 20, 0x0000 }, /* R20 - Noise Gate */ + { 21, 0x00C0 }, /* R21 - Left ADC volume */ + { 22, 0x00C0 }, /* R22 - Right ADC volume */ + { 23, 0x0120 }, /* R23 - Additional control(1) */ + { 24, 0x0000 }, /* R24 - Additional control(2) */ + { 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */ + { 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */ + { 27, 0x0000 }, /* R27 - Additional Control (3) */ + { 28, 0x0000 }, /* R28 - Anti-pop */ + + { 30, 0x005F }, /* R30 - Clocking 3 */ + + { 32, 0x0000 }, /* R32 - ADCL signal path */ + { 33, 0x0000 }, /* R33 - ADCR signal path */ + + { 40, 0x0000 }, /* R40 - LOUT2 volume */ + { 41, 0x0000 }, /* R41 - ROUT2 volume */ + + { 47, 0x0000 }, /* R47 - Pwr Mgmt (3) */ + { 48, 0x0023 }, /* R48 - Additional Control (4) */ + { 49, 0x0000 }, /* R49 - Class D Control 1 */ + + { 51, 0x0003 }, /* R51 - Class D Control 2 */ + + { 56, 0x0106 }, /* R56 - Clocking 4 */ + { 57, 0x0000 }, /* R57 - DSP Sidetone 0 */ + { 58, 0x0000 }, /* R58 - DSP Sidetone 1 */ + + { 60, 0x0000 }, /* R60 - DC Servo 0 */ + { 61, 0x0000 }, /* R61 - DC Servo 1 */ + + { 63, 0x015E }, /* R63 - DC Servo 3 */ + + { 65, 0x0010 }, /* R65 - DC Servo 5 */ + + { 68, 0x0003 }, /* R68 - Analogue PGA Bias */ + { 69, 0x0000 }, /* R69 - Analogue HP 0 */ + + { 71, 0x01FB }, /* R71 - Analogue HP 2 */ + { 72, 0x0000 }, /* R72 - Charge Pump 1 */ + + { 82, 0x0000 }, /* R82 - Charge Pump B */ + + { 87, 0x0000 }, /* R87 - Write Sequencer 1 */ + { 88, 0x0000 }, /* R88 - Write Sequencer 2 */ + { 89, 0x0000 }, /* R89 - Write Sequencer 3 */ + { 90, 0x0000 }, /* R90 - Write Sequencer 4 */ + { 91, 0x0000 }, /* R91 - Write Sequencer 5 */ + { 92, 0x0000 }, /* R92 - Write Sequencer 6 */ + { 93, 0x0000 }, /* R93 - Write Sequencer 7 */ + + { 252, 0x0001 }, /* R252 - General test 1 */ }; struct wm8961_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk; }; -static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8961_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: case WM8961_WRITE_SEQUENCER_7: case WM8961_DC_SERVO_1: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8961_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8961_LEFT_INPUT_VOLUME: + case WM8961_RIGHT_INPUT_VOLUME: + case WM8961_LOUT1_VOLUME: + case WM8961_ROUT1_VOLUME: + case WM8961_CLOCKING1: + case WM8961_ADC_DAC_CONTROL_1: + case WM8961_ADC_DAC_CONTROL_2: + case WM8961_AUDIO_INTERFACE_0: + case WM8961_CLOCKING2: + case WM8961_AUDIO_INTERFACE_1: + case WM8961_LEFT_DAC_VOLUME: + case WM8961_RIGHT_DAC_VOLUME: + case WM8961_AUDIO_INTERFACE_2: + case WM8961_SOFTWARE_RESET: + case WM8961_ALC1: + case WM8961_ALC2: + case WM8961_ALC3: + case WM8961_NOISE_GATE: + case WM8961_LEFT_ADC_VOLUME: + case WM8961_RIGHT_ADC_VOLUME: + case WM8961_ADDITIONAL_CONTROL_1: + case WM8961_ADDITIONAL_CONTROL_2: + case WM8961_PWR_MGMT_1: + case WM8961_PWR_MGMT_2: + case WM8961_ADDITIONAL_CONTROL_3: + case WM8961_ANTI_POP: + case WM8961_CLOCKING_3: + case WM8961_ADCL_SIGNAL_PATH: + case WM8961_ADCR_SIGNAL_PATH: + case WM8961_LOUT2_VOLUME: + case WM8961_ROUT2_VOLUME: + case WM8961_PWR_MGMT_3: + case WM8961_ADDITIONAL_CONTROL_4: + case WM8961_CLASS_D_CONTROL_1: + case WM8961_CLASS_D_CONTROL_2: + case WM8961_CLOCKING_4: + case WM8961_DSP_SIDETONE_0: + case WM8961_DSP_SIDETONE_1: + case WM8961_DC_SERVO_0: + case WM8961_DC_SERVO_1: + case WM8961_DC_SERVO_3: + case WM8961_DC_SERVO_5: + case WM8961_ANALOGUE_PGA_BIAS: + case WM8961_ANALOGUE_HP_0: + case WM8961_ANALOGUE_HP_2: + case WM8961_CHARGE_PUMP_1: + case WM8961_CHARGE_PUMP_B: + case WM8961_WRITE_SEQUENCER_1: + case WM8961_WRITE_SEQUENCER_2: + case WM8961_WRITE_SEQUENCER_3: + case WM8961_WRITE_SEQUENCER_4: + case WM8961_WRITE_SEQUENCER_5: + case WM8961_WRITE_SEQUENCER_6: + case WM8961_WRITE_SEQUENCER_7: + case WM8961_GENERAL_TEST_1: + return true; + default: + return false; } } @@ -958,11 +840,12 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { + struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -975,9 +858,9 @@ static int wm8961_probe(struct snd_soc_codec *codec) } /* This isn't volatile - readback doesn't correspond to write */ - codec->cache_bypass = 1; + regcache_cache_bypass(wm8961->regmap, true); reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); - codec->cache_bypass = 0; + regcache_cache_bypass(wm8961->regmap, false); dev_info(codec->dev, "WM8961 family %d revision %c\n", (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) @@ -1066,10 +949,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8961 = { .suspend = wm8961_suspend, .resume = wm8961_resume, .set_bias_level = wm8961_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8961_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8961_reg_defaults, - .volatile_register = wm8961_volatile_register, +}; + +static const struct regmap_config wm8961_regmap = { + .reg_bits = 8, + .val_bits = 16, + .max_register = WM8961_MAX_REGISTER, + + .reg_defaults = wm8961_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8961_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = wm8961_volatile, + .readable_reg = wm8961_readable, }; static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, @@ -1083,6 +975,10 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, if (wm8961 == NULL) return -ENOMEM; + wm8961->regmap = devm_regmap_init_i2c(i2c, &wm8961_regmap); + if (IS_ERR(wm8961->regmap)) + return PTR_ERR(wm8961->regmap); + i2c_set_clientdata(i2c, wm8961); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From b306e84f9a15e465812d9b66f8d6ecadae806f4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Sep 2012 13:31:38 +0800 Subject: ASoC: wm8961: Move device identification and reset to I2C probe This is more idiomatic as it means we verify that the device is there prior to trying to do the card probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 60 ++++++++++++++++++++++++++--------------------- 1 file changed, 33 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4ea64d6e68e2..f387670d0d75 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -187,11 +187,6 @@ static bool wm8961_readable(struct device *dev, unsigned int reg) } } -static int wm8961_reset(struct snd_soc_codec *codec) -{ - return snd_soc_write(codec, WM8961_SOFTWARE_RESET, 0); -} - /* * The headphone output supports special anti-pop sequences giving * silent power up and power down. @@ -840,7 +835,6 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { - struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; u16 reg; @@ -851,27 +845,6 @@ static int wm8961_probe(struct snd_soc_codec *codec) return ret; } - reg = snd_soc_read(codec, WM8961_SOFTWARE_RESET); - if (reg != 0x1801) { - dev_err(codec->dev, "Device is not a WM8961: ID=0x%x\n", reg); - return -EINVAL; - } - - /* This isn't volatile - readback doesn't correspond to write */ - regcache_cache_bypass(wm8961->regmap, true); - reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); - regcache_cache_bypass(wm8961->regmap, false); - dev_info(codec->dev, "WM8961 family %d revision %c\n", - (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, - ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) - + 'A'); - - ret = wm8961_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - /* Enable class W */ reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B); reg |= WM8961_CP_DYN_PWR_MASK; @@ -968,6 +941,7 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8961_priv *wm8961; + unsigned int val; int ret; wm8961 = devm_kzalloc(&i2c->dev, sizeof(struct wm8961_priv), @@ -979,6 +953,38 @@ static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8961->regmap)) return PTR_ERR(wm8961->regmap); + ret = regmap_read(wm8961->regmap, WM8961_SOFTWARE_RESET, &val); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + return ret; + } + + if (val != 0x1801) { + dev_err(&i2c->dev, "Device is not a WM8961: ID=0x%x\n", val); + return -EINVAL; + } + + /* This isn't volatile - readback doesn't correspond to write */ + regcache_cache_bypass(wm8961->regmap, true); + ret = regmap_read(wm8961->regmap, WM8961_RIGHT_INPUT_VOLUME, &val); + regcache_cache_bypass(wm8961->regmap, false); + + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip revision: %d\n", ret); + return ret; + } + + dev_info(&i2c->dev, "WM8961 family %d revision %c\n", + (val & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, + ((val & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) + + 'A'); + + ret = regmap_write(wm8961->regmap, WM8961_SOFTWARE_RESET, 0x1801); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8961); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From 715a170563843a1f55ae4c8484bc4732d69d2288 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 18 Sep 2012 09:49:46 -0700 Subject: ALSA: usb-audio: set period_bytes in substream. Set the peiod_bytes member of snd_usb_substream. It was no longer being set, but will be needed to resume properly in a future commit. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f782ce19bf5a..786f7a05e9a6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -486,6 +486,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, } if (changed) { + subs->period_bytes = params_period_bytes(hw_params); + mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ stop_endpoints(subs, 0, 0, 0); -- cgit v1.2.3 From 35ec7aa29833de350f51922736aefe22ebf76c4d Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 18 Sep 2012 09:49:47 -0700 Subject: ALSA: usb-audio: Don't require hw_params in endpoint. Change the interface to configure an endpoint so that it doesn't require a hw_params struct. This will allow it to be called from prepare instead of hw_params, configuring it after system resume. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 31 ++++++++++++++++++------------- sound/usb/endpoint.h | 5 ++++- sound/usb/pcm.c | 16 +++++++++++++--- 3 files changed, 35 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a83a18dbac25..152bfd48a311 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -567,20 +567,19 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) * configure a data endpoint */ static int data_ep_set_params(struct snd_usb_endpoint *ep, - struct snd_pcm_hw_params *hw_params, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) { unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms; - int period_bytes = params_period_bytes(hw_params); - int format = params_format(hw_params); int is_playback = usb_pipeout(ep->pipe); - int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) * - params_channels(hw_params); + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; ep->datainterval = fmt->datainterval; ep->stride = frame_bits >> 3; - ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; + ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; /* calculate max. frequency */ if (ep->maxpacksize) { @@ -693,7 +692,6 @@ out_of_memory: * configure a sync endpoint */ static int sync_ep_set_params(struct snd_usb_endpoint *ep, - struct snd_pcm_hw_params *hw_params, struct audioformat *fmt) { int i; @@ -736,7 +734,10 @@ out_of_memory: * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * * @ep: the snd_usb_endpoint to configure - * @hw_params: the hardware parameters + * @pcm_format: the audio fomat. + * @channels: the number of audio channels. + * @period_bytes: the number of bytes in one alsa period. + * @rate: the frame rate. * @fmt: the USB audio format information * @sync_ep: the sync endpoint to use, if any * @@ -745,7 +746,10 @@ out_of_memory: * An endpoint that is already running can not be reconfigured. */ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, - struct snd_pcm_hw_params *hw_params, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep) { @@ -765,9 +769,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) - ep->freqn = get_usb_full_speed_rate(params_rate(hw_params)); + ep->freqn = get_usb_full_speed_rate(rate); else - ep->freqn = get_usb_high_speed_rate(params_rate(hw_params)); + ep->freqn = get_usb_high_speed_rate(rate); /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -777,10 +781,11 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, switch (ep->type) { case SND_USB_ENDPOINT_TYPE_DATA: - err = data_ep_set_params(ep, hw_params, fmt, sync_ep); + err = data_ep_set_params(ep, pcm_format, channels, + period_bytes, fmt, sync_ep); break; case SND_USB_ENDPOINT_TYPE_SYNC: - err = sync_ep_set_params(ep, hw_params, fmt); + err = sync_ep_set_params(ep, fmt); break; default: err = -EINVAL; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index cbbbdf226d66..6376ccf10fd4 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -9,7 +9,10 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, int ep_num, int direction, int type); int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, - struct snd_pcm_hw_params *hw_params, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int rate, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 786f7a05e9a6..62ab4fd5880a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -491,14 +491,24 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, mutex_lock(&subs->stream->chip->shutdown_mutex); /* format changed */ stop_endpoints(subs, 0, 0, 0); - ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt, + ret = snd_usb_endpoint_set_params(subs->data_endpoint, + format, + channels, + subs->period_bytes, + rate, + fmt, subs->sync_endpoint); if (ret < 0) goto unlock; if (subs->sync_endpoint) - ret = snd_usb_endpoint_set_params(subs->sync_endpoint, - hw_params, fmt, NULL); + ret = snd_usb_endpoint_set_params(subs->data_endpoint, + format, + channels, + subs->period_bytes, + rate, + fmt, + NULL); unlock: mutex_unlock(&subs->stream->chip->shutdown_mutex); } -- cgit v1.2.3 From 61a709504b079110cd5b12ea9a4590ffea687a5c Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 18 Sep 2012 09:49:48 -0700 Subject: ALSA: usb-audio: Move configuration to prepare. Move interface and endpoint configuration from hw_params to prepare callback. During system suspend/resume when the USB device power isn't cycled the interface and endpoint configuration need to be set before audio playback can continue. Resume involves another call to prepare but not to hw_params, moving it here allows a playing stream to continue after resume. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 + sound/usb/pcm.c | 134 ++++++++++++++++++++++++++++++------------------------- 2 files changed, 76 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index 23b6f23bd36a..6cc883c3567d 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -107,6 +107,8 @@ struct snd_usb_substream { int interface; /* current interface */ int endpoint; /* assigned endpoint */ struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ + snd_pcm_format_t pcm_format; /* current audio format (for hw_params callback) */ + unsigned int channels; /* current number of channels (for hw_params callback) */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ unsigned int altset_idx; /* USB data format: index of alternate setting */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 62ab4fd5880a..ae783d40f55a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -82,8 +82,7 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream /* * find a matching audio format */ -static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned int format, - unsigned int rate, unsigned int channels) +static struct audioformat *find_format(struct snd_usb_substream *subs) { struct list_head *p; struct audioformat *found = NULL; @@ -92,16 +91,17 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!(fp->formats & (1uLL << format))) + if (!(fp->formats & (1uLL << subs->pcm_format))) continue; - if (fp->channels != channels) + if (fp->channels != subs->channels) continue; - if (rate < fp->rate_min || rate > fp->rate_max) + if (subs->cur_rate < fp->rate_min || + subs->cur_rate > fp->rate_max) continue; if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) { unsigned int i; for (i = 0; i < fp->nr_rates; i++) - if (fp->rate_table[i] == rate) + if (fp->rate_table[i] == subs->cur_rate) break; if (i >= fp->nr_rates) continue; @@ -435,6 +435,42 @@ add_sync_ep: return 0; } +/* + * configure endpoint params + * + * called during initial setup and upon resume + */ +static int configure_endpoint(struct snd_usb_substream *subs) +{ + int ret; + + mutex_lock(&subs->stream->chip->shutdown_mutex); + /* format changed */ + stop_endpoints(subs, 0, 0, 0); + ret = snd_usb_endpoint_set_params(subs->data_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + subs->sync_endpoint); + if (ret < 0) + goto unlock; + + if (subs->sync_endpoint) + ret = snd_usb_endpoint_set_params(subs->data_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + +unlock: + mutex_unlock(&subs->stream->chip->shutdown_mutex); + return ret; +} + /* * hw_params callback * @@ -450,75 +486,32 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, { struct snd_usb_substream *subs = substream->runtime->private_data; struct audioformat *fmt; - unsigned int channels, rate, format; - int ret, changed; + int ret; ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; - format = params_format(hw_params); - rate = params_rate(hw_params); - channels = params_channels(hw_params); - fmt = find_format(subs, format, rate, channels); + subs->pcm_format = params_format(hw_params); + subs->period_bytes = params_period_bytes(hw_params); + subs->channels = params_channels(hw_params); + subs->cur_rate = params_rate(hw_params); + + fmt = find_format(subs); if (!fmt) { snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", - format, rate, channels); + subs->pcm_format, subs->cur_rate, subs->channels); return -EINVAL; } - changed = subs->cur_audiofmt != fmt || - subs->period_bytes != params_period_bytes(hw_params) || - subs->cur_rate != rate; if ((ret = set_format(subs, fmt)) < 0) return ret; - if (subs->cur_rate != rate) { - struct usb_host_interface *alts; - struct usb_interface *iface; - iface = usb_ifnum_to_if(subs->dev, fmt->iface); - alts = &iface->altsetting[fmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, fmt->iface, alts, fmt, rate); - if (ret < 0) - return ret; - subs->cur_rate = rate; - } - - if (changed) { - subs->period_bytes = params_period_bytes(hw_params); + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; - mutex_lock(&subs->stream->chip->shutdown_mutex); - /* format changed */ - stop_endpoints(subs, 0, 0, 0); - ret = snd_usb_endpoint_set_params(subs->data_endpoint, - format, - channels, - subs->period_bytes, - rate, - fmt, - subs->sync_endpoint); - if (ret < 0) - goto unlock; - - if (subs->sync_endpoint) - ret = snd_usb_endpoint_set_params(subs->data_endpoint, - format, - channels, - subs->period_bytes, - rate, - fmt, - NULL); -unlock: - mutex_unlock(&subs->stream->chip->shutdown_mutex); - } - - if (ret == 0) { - subs->interface = fmt->iface; - subs->altset_idx = fmt->altset_idx; - } - - return ret; + return 0; } /* @@ -549,6 +542,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = runtime->private_data; + struct usb_host_interface *alts; + struct usb_interface *iface; + int ret; if (! subs->cur_audiofmt) { snd_printk(KERN_ERR "usbaudio: no format is specified!\n"); @@ -558,6 +554,24 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (snd_BUG_ON(!subs->data_endpoint)) return -EIO; + ret = set_format(subs, subs->cur_audiofmt); + if (ret < 0) + return ret; + + iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); + alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; + ret = snd_usb_init_sample_rate(subs->stream->chip, + subs->cur_audiofmt->iface, + alts, + subs->cur_audiofmt, + subs->cur_rate); + if (ret < 0) + return ret; + + ret = configure_endpoint(subs); + if (ret < 0) + return ret; + /* some unit conversions in runtime */ subs->data_endpoint->maxframesize = bytes_to_frames(runtime, subs->data_endpoint->maxpacksize); -- cgit v1.2.3 From 384dc085c32285e6548511bf80c5d5a5b246ed24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Sep 2012 14:49:31 +0200 Subject: ALSA: usb-audio: Avoid unnecessary EP setups in prepare The recent fix for USB suspend breakage moved the code to set up EP from hw_params to prepare, but it means also the EP setup might be called multiple times unnecessarily because the prepare callback can be called multiple times without starting the stream (e.g. OSS emulation). This patch adds a new flag to struct snd_usb_substream indicating whether the setup of EP is required, and do it only when necessary, i.e. right after hw_params or suspend. Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 ++ sound/usb/card.h | 1 + sound/usb/pcm.c | 10 +++++++--- 3 files changed, 10 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 4a469f0cb6d4..561bb74fd364 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -646,6 +646,8 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) list_for_each(p, &chip->pcm_list) { as = list_entry(p, struct snd_usb_stream, list); snd_pcm_suspend_all(as->pcm); + as->substream[0].need_setup_ep = + as->substream[1].need_setup_ep = true; } } } else { diff --git a/sound/usb/card.h b/sound/usb/card.h index 6cc883c3567d..afa4f9e9b27a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -125,6 +125,7 @@ struct snd_usb_substream { struct snd_usb_endpoint *data_endpoint; struct snd_usb_endpoint *sync_endpoint; unsigned long flags; + bool need_setup_ep; /* (re)configure EP at prepare? */ u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ae783d40f55a..55e19e1b80ec 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -510,6 +510,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; + subs->need_setup_ep = true; return 0; } @@ -568,9 +569,12 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (ret < 0) return ret; - ret = configure_endpoint(subs); - if (ret < 0) - return ret; + if (subs->need_setup_ep) { + ret = configure_endpoint(subs); + if (ret < 0) + return ret; + subs->need_setup_ep = false; + } /* some unit conversions in runtime */ subs->data_endpoint->maxframesize = -- cgit v1.2.3 From c40bd914a80e067ad87aa5c587e590873ebb97e2 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 19 Sep 2012 12:19:47 +0200 Subject: ALSA: hda - avoid non-standard "Docking" name in mixers The standard name (and what PulseAudio picks up) is "Dock Mic", not "Docking Mic" or "Docking-Station". Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 14 +++++++------- sound/pci/hda/patch_conexant.c | 4 ++-- 2 files changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a82cce4b551..cdd43eadbc67 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1810,7 +1810,7 @@ static const struct hda_input_mux ad1981_hp_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, - { "Docking-Station", 0x1 }, + { "Dock Mic", 0x1 }, { "Mix", 0x2 }, }, }; @@ -1834,8 +1834,8 @@ static const struct snd_kcontrol_new ad1981_hp_mixers[] = { */ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), /* FIXME: does this laptop have analog CD connection? */ @@ -3565,7 +3565,7 @@ static const struct hda_amp_list ad1884_loopbacks[] = { static const char * const ad1884_slave_vols[] = { "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", + "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", NULL }; @@ -3626,7 +3626,7 @@ static const struct hda_input_mux ad1984_thinkpad_capture_source = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, { "Mix", 0x3 }, - { "Docking-Station", 0x4 }, + { "Dock Mic", 0x4 }, }, }; @@ -3655,8 +3655,8 @@ static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7d70210d7623..b871b01332de 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1710,8 +1710,8 @@ static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Docking Mic Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Docking Mic Switch", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Volume", 0x15, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Switch", 0x15, 0x00, HDA_INPUT), {} }; -- cgit v1.2.3 From c10514394ef9e8de93a4ad8c8904d71dcd82c122 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 20 Sep 2012 10:20:41 +0200 Subject: ALSA: usb - disable broken hw volume for Tenx TP6911 While going through Ubuntu bugs, I discovered this patch being posted and a confirmation that the patch works as expected. Finding out how the hw volume really works would be preferrable to just disabling the broken one, but this would be better than nothing. Credit: sndfnsdfin (qawsnews) BugLink: https://bugs.launchpad.net/bugs/559939 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f40ba823163..fe56c9da38e9 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1267,6 +1267,13 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* disable non-functional volume control */ master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME); break; + case USB_ID(0x1130, 0xf211): + snd_printk(KERN_INFO + "usbmixer: volume control quirk for Tenx TP6911 Audio Headset\n"); + /* disable non-functional volume control */ + channels = 0; + break; + } if (channels > 0) first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); -- cgit v1.2.3 From 739572a545b8ab6faf9e266c3ed83ea202073699 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 20 Sep 2012 15:41:21 +0200 Subject: ALSA: hda - use both input paths on Conexant auto parser On the Thinkpad W520 - and probably several other machines with Conexant 506x chips - the Dock Mic and Mic are connected to the same two selector nodes. This patch will make Dock Mic take one selector node and Mic take the other, when possible. Without the patch, both paths would take the first selector, leading to the normal Mic's volume being controlled by "Dock Mic Boost". (On other machines, this could instead fixup similar problems between Mic and Line In, for example.) BugLink: https://bugs.launchpad.net/bugs/1037642 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b871b01332de..c03d3b8e42b2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3540,8 +3540,9 @@ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, hda_nid_t pin, hda_nid_t *srcp, bool do_select, int depth) { + struct conexant_spec *spec = codec->spec; hda_nid_t conn[HDA_MAX_NUM_INPUTS]; - int i, nums; + int startidx, i, nums; switch (get_wcaps_type(get_wcaps(codec, mux))) { case AC_WID_AUD_IN: @@ -3565,14 +3566,25 @@ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, depth++; if (depth == 2) return -1; + + /* Try to rotate around connections to avoid one boost controlling + another input path as well */ + startidx = 0; + for (i = 0; i < spec->private_imux.num_items; i++) + if (spec->imux_info[i].pin == pin) { + startidx = i; + break; + } + for (i = 0; i < nums; i++) { - int ret = __select_input_connection(codec, conn[i], pin, srcp, + int j = (i + startidx) % nums; + int ret = __select_input_connection(codec, conn[j], pin, srcp, do_select, depth); if (ret >= 0) { if (do_select) snd_hda_codec_write(codec, mux, 0, - AC_VERB_SET_CONNECT_SEL, i); - return i; + AC_VERB_SET_CONNECT_SEL, j); + return j; } } return -1; -- cgit v1.2.3 From 0534951ba493a97eee646f62101cf88fac2308c6 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 20 Sep 2012 13:57:27 -0500 Subject: ASoC: wm8960: remove 'dres' field from platform data structure The 'dres' field (discharge resistance for headphone outputs) is no longer used in the driver, so remove it. It was used in the original version of the driver when entering standby from off, but we stopped using it when we switched from having a single startup sequence to having separate cap and capless sequences. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- include/sound/wm8960.h | 2 -- sound/soc/codecs/wm8960.c | 5 ----- 2 files changed, 7 deletions(-) (limited to 'sound') diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index b5a1ab9ebb4a..e8ce8ee7d62d 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -18,8 +18,6 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ - int dres; /* Discharge resistance for headphone outputs */ - bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 782faa0a3b49..f0f6f6601785 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -962,11 +962,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); } else { - if (pdata->dres > WM8960_DRES_MAX) { - dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); - pdata->dres = 0; - } - if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; } -- cgit v1.2.3 From 148728f1f7ee7c878a7f57ca6132891552967689 Mon Sep 17 00:00:00 2001 From: Oleksij Rempel Date: Fri, 21 Sep 2012 17:44:58 +0200 Subject: ALSA: hda - Add external mic quirk for Asus Zenbook UX31A Signed-off-by: Oleksij Rempel Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1907ddad3840..3d440fc7c200 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6189,6 +6189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.2.3 From cf55e904516947597d75fd3844acc24891a95772 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Fri, 21 Sep 2012 20:45:19 -0300 Subject: ALSA: hda/via - don't report presence on HPs with no presence support If headphone jack can't detect plug presence, and we have the jack in the jack table, snd_hda_jack_detect will return the plug as always present (as it'll be considered as a phantom jack). The problem is that when this happens, line out pins will always be disabled, resulting in no sound if there are no headphones connected. This was reported as a no sound problem after suspend on http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests on first initialization before the phantom jack is added, but on resume we reexecute the initialization code, and via_hp_automute starts reporting HP always present with the jack now on the table. BugLink: https://bugs.launchpad.net/bugs/1052499 Signed-off-by: Herton Ronaldo Krzesinski Cc: [v3.6+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 74fb6fd6768a..5a45a912aedc 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1672,7 +1672,8 @@ static void via_hp_automute(struct hda_codec *codec) struct via_spec *spec = codec->spec; if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] && - (spec->codec_type != VT1708 || spec->vt1708_jack_detect)) + (spec->codec_type != VT1708 || spec->vt1708_jack_detect) && + is_jack_detectable(codec, spec->autocfg.hp_pins[0])) present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (spec->smart51_enabled) -- cgit v1.2.3 From 172d3b209622785ce7c4f4104319df06d9814b62 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 21 Sep 2012 18:39:05 -0500 Subject: ALSA: hda - force use of SSYNC bits SSYNC bits are typically used to start multiple streams synchronously. It makes sense to use them for a single stream for a more predictable startup sequence. The transfers only start once the DMA and FIFOs are ready. This results in a better correlation between timestamps and number of samples played. Credits to Kar Leong Wang for suggesting this improvement. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 36 ++++++++++++++++-------------------- 1 file changed, 16 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 207d0a20f317..f4070a4f39fd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1986,14 +1986,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_lock(&chip->reg_lock); - if (nsync > 1) { - /* first, set SYNC bits of corresponding streams */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) | sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); - } + + /* first, set SYNC bits of corresponding streams */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) | sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) | sbits); + snd_pcm_group_for_each_entry(s, substream) { if (s->pcm->card != substream->pcm->card) continue; @@ -2011,8 +2011,6 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } spin_unlock(&chip->reg_lock); if (start) { - if (nsync == 1) - return 0; /* wait until all FIFOs get ready */ for (timeout = 5000; timeout; timeout--) { nwait = 0; @@ -2045,16 +2043,14 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) cpu_relax(); } } - if (nsync > 1) { - spin_lock(&chip->reg_lock); - /* reset SYNC bits */ - if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) - azx_writel(chip, OLD_SSYNC, - azx_readl(chip, OLD_SSYNC) & ~sbits); - else - azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); - spin_unlock(&chip->reg_lock); - } + spin_lock(&chip->reg_lock); + /* reset SYNC bits */ + if (chip->driver_caps & AZX_DCAPS_OLD_SSYNC) + azx_writel(chip, OLD_SSYNC, + azx_readl(chip, OLD_SSYNC) & ~sbits); + else + azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); + spin_unlock(&chip->reg_lock); return 0; } -- cgit v1.2.3 From 90accc58a6946e7245993da6079f88d8c29cb731 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 21 Sep 2012 18:39:06 -0500 Subject: ALSA: hda - use LPIB for delay estimation DMA Position in Buffer (DPIB) should be used for ring buffer management, while LPIB register provides information on the number of samples transfered on the link. The difference between the two pieces of information corresponds to hardware/DMA buffering. This patch reports this difference in runtime->delay, and removes the use of the COMBO mode on recent Intel hardware. Credits to Takashi Iwai for an initial patch. [rebased to for-next branch and replaced snd_printk() with snd_printdd() by tiwai] Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 36 +++++++++++++++++++++++++++++------- 1 file changed, 29 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f4070a4f39fd..1c622e5e397c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -555,6 +555,7 @@ enum { #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ #define AZX_DCAPS_POSFIX_COMBO (1 << 24) /* Use COMBO as default */ +#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -2143,6 +2144,27 @@ static unsigned int azx_get_position(struct azx *chip, if (pos >= azx_dev->bufsize) pos = 0; + + /* calculate runtime delay from LPIB */ + if (azx_dev->substream->runtime && + chip->position_fix[stream] == POS_FIX_POSBUF && + (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) { + unsigned int lpib_pos = azx_sd_readl(azx_dev, SD_LPIB); + int delay; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = pos - lpib_pos; + else + delay = lpib_pos - pos; + if (delay < 0) + delay += azx_dev->bufsize; + if (delay >= azx_dev->period_bytes) { + snd_printdd("delay %d > period_bytes %d\n", + delay, azx_dev->period_bytes); + delay = 0; /* something is wrong */ + } + azx_dev->substream->runtime->delay = + bytes_to_frames(azx_dev->substream->runtime, delay); + } return pos; } @@ -3402,7 +3424,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | @@ -3410,26 +3432,26 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c21), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, { PCI_DEVICE(0x8086, 0x0d0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From 99df18b31db389ec6abc8ad5415c14e7bb752b58 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 21 Sep 2012 18:39:07 -0500 Subject: ALSA: hda - add PCI identifier for Intel 5 Series/3400 Tested with LPIB delay without any issues. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c622e5e397c..2a41161fa057 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3452,6 +3452,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x0d0c), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + /* 5 Series/3400 */ + { PCI_DEVICE(0x8086, 0x3b56), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From 7f51e7d30e36ac987d7a0e480d890477c5f8b04f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 20 Sep 2012 16:32:02 +0300 Subject: ASoC: twl4030: Convert to use DAI DAPM widgets Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl4030-codec twl4030-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 48 +++++++++++++++++++++++++++------------------- 1 file changed, 28 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2548f5c56885..962341df7ddc 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1237,16 +1237,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("Virtual Voice OUT"), /* DACs */ - SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Voice", NULL, SND_SOC_NOPM, 0, 0), /* Analog bypasses */ SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, @@ -1375,14 +1370,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Introducing four virtual ADC, since TWL4030 have four channel for capture */ - SND_SOC_DAPM_ADC("ADC Virtual Left1", "Left Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right1", "Right Front Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Left2", "Left Rear Capture", - SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("ADC Virtual Right2", "Right Rear Capture", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Left2", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC Virtual Right2", NULL, SND_SOC_NOPM, 0, 0), /* Analog/Digital mic path selection. TX1 Left/Right: either analog Left/Right or Digimic0 @@ -1426,6 +1417,23 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"DAC Right1", NULL, "HiFi Playback"}, + {"DAC Left1", NULL, "HiFi Playback"}, + {"DAC Right2", NULL, "HiFi Playback"}, + {"DAC Left2", NULL, "HiFi Playback"}, + {"DAC Voice", NULL, "Voice Playback"}, + + /* ADC -> Stream mapping */ + {"HiFi Capture", NULL, "ADC Virtual Left1"}, + {"HiFi Capture", NULL, "ADC Virtual Right1"}, + {"HiFi Capture", NULL, "ADC Virtual Left2"}, + {"HiFi Capture", NULL, "ADC Virtual Right2"}, + {"Voice Capture", NULL, "ADC Virtual Left1"}, + {"Voice Capture", NULL, "ADC Virtual Right1"}, + {"Voice Capture", NULL, "ADC Virtual Left2"}, + {"Voice Capture", NULL, "ADC Virtual Right2"}, + {"Digital L1 Playback Mixer", NULL, "DAC Left1"}, {"Digital R1 Playback Mixer", NULL, "DAC Right1"}, {"Digital L2 Playback Mixer", NULL, "DAC Left2"}, @@ -2170,7 +2178,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .formats = TWL4030_FORMATS, .sig_bits = 24,}, .capture = { - .stream_name = "Capture", + .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, @@ -2187,7 +2195,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .capture = { - .stream_name = "Capture", + .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, -- cgit v1.2.3 From 805238b1b76c8f4f17a92f50c12664a8e6f3564f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 20 Sep 2012 16:32:15 +0300 Subject: ASoC: twl6040: Convert to use DAI DAPM widgets Use DAPM mapping for stream events and give unique names for the streams. This change also fixes the following warning: twl6040-codec twl6040-codec: Failed to create Capture debugfs file Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 43 +++++++++++++++++++++++++++++-------------- 1 file changed, 29 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index c084c549942e..e8f97af75928 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -727,10 +727,8 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_MICRCTL, 1, 0, NULL, 0), /* ADCs */ - SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", - TWL6040_REG_MICLCTL, 2, 0), - SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", - TWL6040_REG_MICRCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Left", NULL, TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, TWL6040_REG_MICRCTL, 2, 0), /* Microphone bias */ SND_SOC_DAPM_SUPPLY("Headset Mic Bias", @@ -743,15 +741,12 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_DMICBCTL, 4, 0, NULL, 0), /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", - TWL6040_REG_HFLCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", - TWL6040_REG_HFRCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", NULL, TWL6040_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", NULL, TWL6040_REG_HFRCTL, 0, 0), /* Virtual DAC for vibra path (DL4 channel) */ - SND_SOC_DAPM_DAC("VIBRA DAC", "Vibra Playback", - SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("VIBRA DAC", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("Handsfree Left Playback", SND_SOC_NOPM, 0, 0, &hfl_mux_controls), @@ -810,6 +805,26 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + /* Stream -> DAC mapping */ + {"HSDAC Left", NULL, "Legacy Playback"}, + {"HSDAC Left", NULL, "Headset Playback"}, + {"HSDAC Right", NULL, "Legacy Playback"}, + {"HSDAC Right", NULL, "Headset Playback"}, + + {"HFDAC Left", NULL, "Legacy Playback"}, + {"HFDAC Left", NULL, "Handsfree Playback"}, + {"HFDAC Right", NULL, "Legacy Playback"}, + {"HFDAC Right", NULL, "Handsfree Playback"}, + + {"VIBRA DAC", NULL, "Legacy Playback"}, + {"VIBRA DAC", NULL, "Vibra Playback"}, + + /* ADC -> Stream mapping */ + {"ADC Left", NULL, "Legacy Capture"}, + {"ADC Left", NULL, "Capture"}, + {"ADC Right", NULL, "Legacy Capture"}, + {"ADC Right", NULL, "Capture"}, + /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, @@ -1028,14 +1043,14 @@ static struct snd_soc_dai_driver twl6040_dai[] = { { .name = "twl6040-legacy", .playback = { - .stream_name = "Playback", + .stream_name = "Legacy Playback", .channels_min = 1, .channels_max = 5, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .capture = { - .stream_name = "Capture", + .stream_name = "Legacy Capture", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, -- cgit v1.2.3 From f99ddef0d8e02884b302701fb7acb6fe51a36749 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 20 Sep 2012 10:36:43 -0400 Subject: ASoC: fsl: add PPC_MPC52xx dependency to SND_POWERPC_SOC mpc52xx socs do not define FSL_SOC but need SND_POWERPC_SOC defined to build ASoC drivers. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index d70133086ac3..4563b28bd625 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -6,7 +6,7 @@ config SND_SOC_FSL_UTILS menuconfig SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" - depends on FSL_SOC + depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. -- cgit v1.2.3 From 084011615c1c885232e8d1fb5cc3f85c9d880d14 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 20 Sep 2012 10:36:44 -0400 Subject: ASoC: fsl: pcm030-audio-fabric use snd_soc_register_card Convert pcm030-audio-fabric to use the new snd_soc_register_card api instead of the older method of registering a separate platform device. Create the dai_link to the mpc5200_psc_ac97 platform using the device tree. Convert the pcm030-audio-fabric driver to a platform-driver and add a remove function. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 65 +++++++++++++++++++++++++++---------- 1 file changed, 47 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 1353e8fecd7b..893e24034816 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -28,7 +28,6 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", - .platform_name = "mpc5200-pcm-audio", .codec_name = "wm9712-codec", }, { @@ -36,44 +35,74 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", - .platform_name = "mpc5200-pcm-audio", .codec_name = "wm9712-codec", }, }; -static struct snd_soc_card card = { +static struct snd_soc_card pcm030_card = { .name = "pcm030", .owner = THIS_MODULE, .dai_link = pcm030_fabric_dai, .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static __init int pcm030_fabric_init(void) +static int __init pcm030_fabric_probe(struct platform_device *op) { - struct platform_device *pdev; - int rc; + struct device_node *np = op->dev.of_node; + struct device_node *platform_np; + struct snd_soc_card *card = &pcm030_card; + int ret; + int i; if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; - pdev = platform_device_alloc("soc-audio", 1); - if (!pdev) { - pr_err("pcm030_fabric_init: platform_device_alloc() failed\n"); + card->dev = &op->dev; + platform_set_drvdata(op, card); + + platform_np = of_parse_phandle(np, "asoc-platform", 0); + if (!platform_np) { + dev_err(&op->dev, "ac97 not registered\n"); return -ENODEV; } - platform_set_drvdata(pdev, &card); + for (i = 0; i < card->num_links; i++) + card->dai_link[i].platform_of_node = platform_np; - rc = platform_device_add(pdev); - if (rc) { - pr_err("pcm030_fabric_init: platform_device_add() failed\n"); - platform_device_put(pdev); - return -ENODEV; - } - return 0; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + + return ret; } -module_init(pcm030_fabric_init); +static int __devexit pcm030_fabric_remove(struct platform_device *op) +{ + struct snd_soc_card *card = platform_get_drvdata(op); + int ret; + + ret = snd_soc_unregister_card(card); + + return ret; +} + +static struct of_device_id pcm030_audio_match[] = { + { .compatible = "phytec,pcm030-audio-fabric", }, + {} +}; +MODULE_DEVICE_TABLE(of, pcm030_audio_match); + +static struct platform_driver pcm030_fabric_driver = { + .probe = pcm030_fabric_probe, + .remove = __devexit_p(pcm030_fabric_remove), + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = pcm030_audio_match, + }, +}; + +module_platform_driver(pcm030_fabric_driver); MODULE_AUTHOR("Jon Smirl "); -- cgit v1.2.3 From c912fa913446b07147f6cbc1a8fa2fb20d2f7c36 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Thu, 20 Sep 2012 10:36:45 -0400 Subject: ASoC: fsl: register the wm9712-codec The mpc5200-psc-ac97 driver does not enumerate attached ac97 devices, so register the device here. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 893e24034816..4b63ec8eb372 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -22,6 +22,11 @@ #define DRV_NAME "pcm030-audio-fabric" +struct pcm030_audio_data { + struct snd_soc_card *card; + struct platform_device *codec_device; +}; + static struct snd_soc_dai_link pcm030_fabric_dai[] = { { .name = "AC97", @@ -51,14 +56,22 @@ static int __init pcm030_fabric_probe(struct platform_device *op) struct device_node *np = op->dev.of_node; struct device_node *platform_np; struct snd_soc_card *card = &pcm030_card; + struct pcm030_audio_data *pdata; int ret; int i; if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; + pdata = devm_kzalloc(&op->dev, sizeof(struct pcm030_audio_data), + GFP_KERNEL); + if (!pdata) + return -ENOMEM; + card->dev = &op->dev; - platform_set_drvdata(op, card); + platform_set_drvdata(op, pdata); + + pdata->card = card; platform_np = of_parse_phandle(np, "asoc-platform", 0); if (!platform_np) { @@ -69,6 +82,18 @@ static int __init pcm030_fabric_probe(struct platform_device *op) for (i = 0; i < card->num_links; i++) card->dai_link[i].platform_of_node = platform_np; + ret = request_module("snd-soc-wm9712"); + if (ret) + dev_err(&op->dev, "request_module returned: %d\n", ret); + + pdata->codec_device = platform_device_alloc("wm9712-codec", -1); + if (!pdata->codec_device) + dev_err(&op->dev, "platform_device_alloc() failed\n"); + + ret = platform_device_add(pdata->codec_device); + if (ret) + dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + ret = snd_soc_register_card(card); if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -78,10 +103,11 @@ static int __init pcm030_fabric_probe(struct platform_device *op) static int __devexit pcm030_fabric_remove(struct platform_device *op) { - struct snd_soc_card *card = platform_get_drvdata(op); + struct pcm030_audio_data *pdata = platform_get_drvdata(op); int ret; - ret = snd_soc_unregister_card(card); + ret = snd_soc_unregister_card(pdata->card); + platform_device_unregister(pdata->codec_device); return ret; } -- cgit v1.2.3 From dffb360e64968b95a0a0221ca52234a28cc049c9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:49 +0300 Subject: ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode When McBSP is configured in threshold mode we can use sDMA packet mode in all cases. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 47 ++++++++++++++++----------------------------- 1 file changed, 17 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2e91a86b6114..fe3debcc2d0b 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -81,9 +81,6 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) */ if (dma_data->packet_size) words = dma_data->packet_size; - else if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) - words = snd_pcm_lib_period_bytes(substream) / - (mcbsp->wlen / 8); else words = 1; @@ -251,6 +248,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, dma_data->set_threshold = omap_mcbsp_set_threshold; if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; + int divider = 0; period_words = params_period_bytes(params) / (wlen / 8); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -258,34 +256,23 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, else max_thrsh = mcbsp->max_rx_thres; /* - * If the period contains less or equal number of words, - * we are using the original threshold mode setup: - * McBSP threshold = sDMA frame size = period_size - * Otherwise we switch to sDMA packet mode: - * McBSP threshold = sDMA packet size - * sDMA frame size = period size + * Use sDMA packet mode if McBSP is in threshold mode: + * If period words less than the FIFO size the packet + * size is set to the number of period words, otherwise + * Look for the biggest threshold value which divides + * the period size evenly. */ - if (period_words > max_thrsh) { - int divider = 0; - - /* - * Look for the biggest threshold value, which - * divides the period size evenly. - */ - divider = period_words / max_thrsh; - if (period_words % max_thrsh) - divider++; - while (period_words % divider && - divider < period_words) - divider++; - if (divider == period_words) - return -EINVAL; - - pkt_size = period_words / divider; - sync_mode = OMAP_DMA_SYNC_PACKET; - } else { - sync_mode = OMAP_DMA_SYNC_FRAME; - } + divider = period_words / max_thrsh; + if (period_words % max_thrsh) + divider++; + while (period_words % divider && + divider < period_words) + divider++; + if (divider == period_words) + return -EINVAL; + + pkt_size = period_words / divider; + sync_mode = OMAP_DMA_SYNC_PACKET; } else if (channels > 1) { /* Use packet mode for non mono streams */ pkt_size = channels; -- cgit v1.2.3 From e512589c17572e7498693a3e05eefa44e622f62b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:50 +0300 Subject: ASoC: omap-pcm: Select sDMA synchronization based on packet_size Since we only have element or packet synchronization we can use the dma_data->packet_size to select the desired mode: if packet_size is 0 we use ELEMENT mode if packet_size is not 0 we use PACKET mode for sDMA synchronization. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f0feb06615f8..02eeb2e7ceda 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -165,7 +165,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) memset(&dma_params, 0, sizeof(dma_params)); dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; - dma_params.sync_mode = dma_data->sync_mode; + + if (dma_data->packet_size) + dma_params.sync_mode = OMAP_DMA_SYNC_PACKET; + else + dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dma_params.src_amode = OMAP_DMA_AMODE_POST_INC; dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT; -- cgit v1.2.3 From 061fb36db7c0187aa90b95f1ba56f6192f42b984 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:51 +0300 Subject: ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct The omap-pcm platform driver no longer needs this parameter to select between ELEMENT and PACKET mode. The selection is based on the configured packet_size. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 1 - sound/soc/omap/omap-hdmi.c | 1 - sound/soc/omap/omap-mcbsp.c | 5 +---- sound/soc/omap/omap-mcpdm.c | 2 -- sound/soc/omap/omap-pcm.h | 1 - 5 files changed, 1 insertion(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 75f5dca0e8d2..60b7b8cd1c7d 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -64,7 +64,6 @@ struct omap_dmic { static struct omap_pcm_dma_data omap_dmic_dai_dma_params = { .name = "DMIC capture", .data_type = OMAP_DMA_DATA_TYPE_S32, - .sync_mode = OMAP_DMA_SYNC_PACKET, }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index a08245d9203c..b19464697ac8 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -290,7 +290,6 @@ static __devinit int omap_hdmi_probe(struct platform_device *pdev) hdmi_data->dma_params.dma_req = hdmi_rsrc->start; hdmi_data->dma_params.name = "HDMI playback"; - hdmi_data->dma_params.sync_mode = OMAP_DMA_SYNC_PACKET; /* * TODO: We assume that there is only one DSS HDMI device. Future diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index fe3debcc2d0b..5b3baccd74c3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -225,7 +225,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; struct omap_pcm_dma_data *dma_data; - int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; + int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; @@ -272,15 +272,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; pkt_size = period_words / divider; - sync_mode = OMAP_DMA_SYNC_PACKET; } else if (channels > 1) { /* Use packet mode for non mono streams */ pkt_size = channels; - sync_mode = OMAP_DMA_SYNC_PACKET; } } - dma_data->sync_mode = sync_mode; dma_data->packet_size = pkt_size; snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index f7babb374a37..baf92da42aec 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -73,14 +73,12 @@ static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { .name = "Audio playback", .dma_req = OMAP44XX_DMA_MCPDM_DL, .data_type = OMAP_DMA_DATA_TYPE_S32, - .sync_mode = OMAP_DMA_SYNC_PACKET, .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA, }, { .name = "Audio capture", .dma_req = OMAP44XX_DMA_MCPDM_UP, .data_type = OMAP_DMA_DATA_TYPE_S32, - .sync_mode = OMAP_DMA_SYNC_PACKET, .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA, }, }; diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index b92248cbd47a..1bf47e4b60c2 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -33,7 +33,6 @@ struct omap_pcm_dma_data { unsigned long port_addr; /* transmit/receive register */ void (*set_threshold)(struct snd_pcm_substream *substream); int data_type; /* data type 8,16,32 */ - int sync_mode; /* DMA sync mode */ int packet_size; /* packet size only in PACKET mode */ }; -- cgit v1.2.3 From 03945e99a8ac4f596dad9a679816e5b4bb77e5d2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:52 +0300 Subject: ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties Based on the format of the stream the omap-pcm can decide alone what data type should be used with by the sDMA. Keep the possibility for OMAP dai drivers to tell omap-pcm if they want to use different data type. This is needed for the omap-hdmi for example which needs 32bit data type even if the stream format is S16_LE. The check if (dma_data->data_type) is safe at the moment since omap-pcm does not support 8bit samples (OMAP_DMA_DATA_TYPE_S8 == 0x00). The next step is to redefine the meaning of dma_data->data_type to unblock this limitation. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 02eeb2e7ceda..4c13a5f4eebb 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -149,6 +149,24 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } +static int omap_pcm_get_dma_type(int num_bits) +{ + int data_type; + + switch (num_bits) { + case 16: + data_type = OMAP_DMA_DATA_TYPE_S16; + break; + case 32: + data_type = OMAP_DMA_DATA_TYPE_S32; + break; + default: + data_type = -EINVAL; + break; + } + return data_type; +} + static int omap_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -163,7 +181,16 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - dma_params.data_type = dma_data->data_type; + + if (dma_data->data_type) + dma_params.data_type = dma_data->data_type; + else + dma_params.data_type = omap_pcm_get_dma_type( + snd_pcm_format_physical_width(runtime->format)); + + if (dma_params.data_type < 0) + return dma_params.data_type; + dma_params.trigger = dma_data->dma_req; if (dma_data->packet_size) @@ -195,7 +222,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * still can get an interrupt at each period bounary */ bytes = snd_pcm_lib_period_bytes(substream); - dma_params.elem_count = bytes >> dma_data->data_type; + dma_params.elem_count = bytes >> dma_params.data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); -- cgit v1.2.3 From 5a40c57af55b17dabe25854c9245171515bb5d23 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:54 +0300 Subject: ASoC: omap-mcpdm: Use platform_get_resource_* to get resources Get the needed resources in a correct way and avoid using defines for them. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index baf92da42aec..f90d5de605c3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -71,15 +71,11 @@ struct omap_mcpdm { static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { { .name = "Audio playback", - .dma_req = OMAP44XX_DMA_MCPDM_DL, .data_type = OMAP_DMA_DATA_TYPE_S32, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_DN_DATA, }, { .name = "Audio capture", - .dma_req = OMAP44XX_DMA_MCPDM_UP, .data_type = OMAP_DMA_DATA_TYPE_S32, - .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_REG_UP_DATA, }, }; @@ -452,10 +448,33 @@ static __devinit int asoc_mcpdm_probe(struct platform_device *pdev) mutex_init(&mcpdm->mutex); + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (res == NULL) + return -ENOMEM; + + omap_mcpdm_dai_dma_params[0].port_addr = res->start + MCPDM_REG_DN_DATA; + omap_mcpdm_dai_dma_params[1].port_addr = res->start + MCPDM_REG_UP_DATA; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) return -ENOMEM; + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); + if (!res) + return -ENODEV; + + omap_mcpdm_dai_dma_params[0].dma_req = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link"); + if (!res) + return -ENODEV; + + omap_mcpdm_dai_dma_params[1].dma_req = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (res == NULL) + return -ENOMEM; + if (!devm_request_mem_region(&pdev->dev, res->start, resource_size(res), "McPDM")) return -EBUSY; -- cgit v1.2.3 From 04564e3258304df607d4536de42603b4c8e21e1a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:55 +0300 Subject: ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type omap-pcm can figure out the correct dma_type based on the stream's format. In this way we can get rid of the plat/dma.h include from these drivers. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 2 -- sound/soc/omap/omap-mcbsp.c | 3 --- sound/soc/omap/omap-mcpdm.c | 3 --- 3 files changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 60b7b8cd1c7d..df0ff247f496 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -33,7 +33,6 @@ #include #include #include -#include #include #include @@ -63,7 +62,6 @@ struct omap_dmic { */ static struct omap_pcm_dma_data omap_dmic_dai_dma_params = { .name = "DMIC capture", - .data_type = OMAP_DMA_DATA_TYPE_S32, }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 5b3baccd74c3..a23064644e58 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -34,7 +34,6 @@ #include #include -#include #include #include "mcbsp.h" #include "omap-mcbsp.h" @@ -234,11 +233,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - dma_data->data_type = OMAP_DMA_DATA_TYPE_S16; wlen = 16; break; case SNDRV_PCM_FORMAT_S32_LE: - dma_data->data_type = OMAP_DMA_DATA_TYPE_S32; wlen = 32; break; default: diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index f90d5de605c3..84743d47e687 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,6 @@ #include #include -#include #include #include "omap-mcpdm.h" #include "omap-pcm.h" @@ -71,11 +70,9 @@ struct omap_mcpdm { static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { { .name = "Audio playback", - .data_type = OMAP_DMA_DATA_TYPE_S32, }, { .name = "Audio capture", - .data_type = OMAP_DMA_DATA_TYPE_S32, }, }; -- cgit v1.2.3 From f05cc9dac99ac6403d057d2cccb3c754714d2f32 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Sep 2012 15:05:56 +0300 Subject: ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type Instead of the OMAP DMA data type definition the data_type will be used to specify the number of bits the DMA word should be configured or 0 in case when based on the stream's format the omap-pcm can decide the needed DMA word size. This feature is needed for the omap-hdmi where the sDMA need to be configured for 32bit word type regardless of the audio format used. Signed-off-by: Peter Ujfalusi Tested-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/omap-hdmi.c | 3 +-- sound/soc/omap/omap-pcm.c | 3 ++- sound/soc/omap/omap-pcm.h | 3 ++- 3 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index b19464697ac8..095176738fd6 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -34,7 +34,6 @@ #include #include