From 5b7cdc8068e3f02ff4c6ef75bd7398af244869d4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 17 Feb 2015 01:48:08 +0000 Subject: ASoC: fsi: remove slave_id settings for DMAEngine Current fsi sets dma_slave_config :: slave_id field for DMAEngine, but it is no longer needed. Let's remove it. Signed-off-by: Kuninori Morimoto Acked-by: Mark Brown Signed-off-by: Vinod Koul --- sound/soc/sh/fsi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b87b22e88e43..dc28b03db6e5 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1371,10 +1371,9 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev shdma_chan_filter, (void *)io->dma_id, dev, is_play ? "tx" : "rx"); if (io->chan) { - struct dma_slave_config cfg; + struct dma_slave_config cfg = {}; int ret; - cfg.slave_id = io->dma_id; cfg.dst_addr = 0; /* use default addr */ cfg.src_addr = 0; /* use default addr */ cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; -- cgit v1.2.3 From 7c6cc8f2012f4146b05b8ec7238f98884100db8c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 17 Feb 2015 01:48:19 +0000 Subject: ASoC: fsi: Configure DMA slave settings Current FSI driver is assuming that dst_addr/src_addr of DMAEngine will be set by platform data. But it should be set via dmaengine_slave_config(). Special thanks to Arnd Reported-by: Arnd Bergmann Signed-off-by: Kuninori Morimoto Acked-by: Mark Brown Signed-off-by: Vinod Koul --- sound/soc/sh/fsi.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index dc28b03db6e5..bb20550b007c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -250,6 +250,7 @@ struct fsi_clk { struct fsi_priv { void __iomem *base; + phys_addr_t phys; struct fsi_master *master; struct fsi_stream playback; @@ -1374,9 +1375,15 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev struct dma_slave_config cfg = {}; int ret; - cfg.dst_addr = 0; /* use default addr */ - cfg.src_addr = 0; /* use default addr */ - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + if (is_play) { + cfg.dst_addr = fsi->phys + REG_DODT; + cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.direction = DMA_MEM_TO_DEV; + } else { + cfg.src_addr = fsi->phys + REG_DIDT; + cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.direction = DMA_DEV_TO_MEM; + } ret = dmaengine_slave_config(io->chan, &cfg); if (ret < 0) { @@ -1940,6 +1947,7 @@ static int fsi_probe(struct platform_device *pdev) /* FSI A setting */ fsi = &master->fsia; fsi->base = master->base; + fsi->phys = res->start; fsi->master = master; fsi_port_info_init(fsi, &info.port_a); fsi_handler_init(fsi, &info.port_a); @@ -1952,6 +1960,7 @@ static int fsi_probe(struct platform_device *pdev) /* FSI B setting */ fsi = &master->fsib; fsi->base = master->base + 0x40; + fsi->phys = res->start + 0x40; fsi->master = master; fsi_port_info_init(fsi, &info.port_b); fsi_handler_init(fsi, &info.port_b); -- cgit v1.2.3 From 112bdfaa525fd5993e17885861342893f15290b0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Feb 2015 15:41:02 +0000 Subject: extcon: arizona: Deobfuscate arizona_extcon_do_magic arizona_extcon_do_magic does not lend a lot of clarity to the purpose of the function, and as all the registers used are described in the datasheet there is no need to obfuscate the code. This patch renames the function to arizona_extcon_hp_clamp, as it controls clamping on the headphone output. Signed-off-by: Charles Keepax Acked-by: Lee Jones Signed-off-by: Chanwoo Choi --- drivers/extcon/extcon-arizona.c | 36 ++++++++++++++++++++---------------- include/linux/mfd/arizona/core.h | 2 +- sound/soc/codecs/arizona.c | 4 ++-- 3 files changed, 23 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 63f01c42aed4..95cf7f875bb3 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -136,18 +136,22 @@ static const char *arizona_cable[] = { static void arizona_start_hpdet_acc_id(struct arizona_extcon_info *info); -static void arizona_extcon_do_magic(struct arizona_extcon_info *info, - unsigned int magic) +static void arizona_extcon_hp_clamp(struct arizona_extcon_info *info, + bool clamp) { struct arizona *arizona = info->arizona; + unsigned int val = 0; int ret; + if (clamp) + val = ARIZONA_RMV_SHRT_HP1L; + mutex_lock(&arizona->dapm->card->dapm_mutex); - arizona->hpdet_magic = magic; + arizona->hpdet_clamp = clamp; - /* Keep the HP output stages disabled while doing the magic */ - if (magic) { + /* Keep the HP output stages disabled while doing the clamp */ + if (clamp) { ret = regmap_update_bits(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA | @@ -158,20 +162,20 @@ static void arizona_extcon_do_magic(struct arizona_extcon_info *info, ret); } - ret = regmap_update_bits(arizona->regmap, 0x225, 0x4000, - magic); + ret = regmap_update_bits(arizona->regmap, ARIZONA_HP_CTRL_1L, + ARIZONA_RMV_SHRT_HP1L, val); if (ret != 0) - dev_warn(arizona->dev, "Failed to do magic: %d\n", + dev_warn(arizona->dev, "Failed to do clamp: %d\n", ret); - ret = regmap_update_bits(arizona->regmap, 0x226, 0x4000, - magic); + ret = regmap_update_bits(arizona->regmap, ARIZONA_HP_CTRL_1R, + ARIZONA_RMV_SHRT_HP1R, val); if (ret != 0) - dev_warn(arizona->dev, "Failed to do magic: %d\n", + dev_warn(arizona->dev, "Failed to do clamp: %d\n", ret); - /* Restore the desired state while not doing the magic */ - if (!magic) { + /* Restore the desired state while not doing the clamp */ + if (!clamp) { ret = regmap_update_bits(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA | @@ -603,7 +607,7 @@ done: ARIZONA_HP_IMPEDANCE_RANGE_MASK | ARIZONA_HP_POLL, 0); - arizona_extcon_do_magic(info, 0); + arizona_extcon_hp_clamp(info, false); if (id_gpio) gpio_set_value_cansleep(id_gpio, 0); @@ -648,7 +652,7 @@ static void arizona_identify_headphone(struct arizona_extcon_info *info) if (info->mic) arizona_stop_mic(info); - arizona_extcon_do_magic(info, 0x4000); + arizona_extcon_hp_clamp(info, true); ret = regmap_update_bits(arizona->regmap, ARIZONA_ACCESSORY_DETECT_MODE_1, @@ -699,7 +703,7 @@ static void arizona_start_hpdet_acc_id(struct arizona_extcon_info *info) info->hpdet_active = true; - arizona_extcon_do_magic(info, 0x4000); + arizona_extcon_hp_clamp(info, true); ret = regmap_update_bits(arizona->regmap, ARIZONA_ACCESSORY_DETECT_MODE_1, diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 910e3aa1e965..4863548faff7 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -126,7 +126,7 @@ struct arizona { struct regmap_irq_chip_data *aod_irq_chip; struct regmap_irq_chip_data *irq_chip; - bool hpdet_magic; + bool hpdet_clamp; unsigned int hp_ena; struct mutex clk_lock; diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 29202610dd0d..fb58c7ee3780 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -840,8 +840,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, priv->arizona->hp_ena &= ~mask; priv->arizona->hp_ena |= val; - /* Force off if HPDET magic is active */ - if (priv->arizona->hpdet_magic) + /* Force off if HPDET clamp is active */ + if (priv->arizona->hpdet_clamp) val = 0; regmap_update_bits_async(arizona->regmap, ARIZONA_OUTPUT_ENABLES_1, -- cgit v1.2.3 From 96f05be37f4ece1dffba92d4ae079a486a4cf6df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Apr 2015 14:23:29 +0200 Subject: ASoC: qcom: Return an error for invalid PCM trigger command MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix a compile warning sound/soc/qcom/lpass-cpu.c: In function ‘lpass_cpu_daiops_trigger’: sound/soc/qcom/lpass-cpu.c:224:2: warning: ‘ret’ may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^ Although switch () lists the most of existing cases, it's still better to cover the rest as an error properly. Signed-off-by: Takashi Iwai Acked-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 6698d058de29..dc790abaa331 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -194,7 +194,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; + int ret = -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: -- cgit v1.2.3 From d1acba2fdebb449bad01e7cf77a9f9730dfaca6e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Apr 2015 06:00:45 +0000 Subject: ASoC: rsnd: set dmaen->chan = NULL when error case rsnd_dmaen_quit() is assuming dmaen->chan is NULL if it failed to get DMAEngine channel. but, current dmaen->chan might have error value when error case (this driver is checking it by IS_ERR_OR_NULL()) This patch makes sure dmaen->chan is NULL when error case. Otherwise, it will contact to unknown address in rsnd_dmaen_quit() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/dma.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ac3756f6af60..144308f15fb3 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -156,6 +156,7 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, (void *)id); } if (IS_ERR_OR_NULL(dmaen->chan)) { + dmaen->chan = NULL; dev_err(dev, "can't get dma channel\n"); goto rsnd_dma_channel_err; } -- cgit v1.2.3 From ac98b4c015b50b1e452f8d55b612320be7f80825 Mon Sep 17 00:00:00 2001 From: Jin Yao Date: Mon, 13 Apr 2015 14:20:54 +0800 Subject: ASoC: Intel: Remove invalid kfree of devm allocated data kbuild robot reports following warning: "sound/soc/intel/haswell/sst-haswell-ipc.c:2204:1-6: WARNING: invalid free of devm_ allocated data" As julia explains to me, the memory allocated with devm_kalloc is freed automatically on failure of a probe function. So this kfree should be removed otherwise the double free will be got in error handler path. Signed-off-by: Jin Yao Signed-off-by: Mark Brown --- sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 344a1e9bbce5..324eceb07b25 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2201,7 +2201,6 @@ dma_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); -- cgit v1.2.3 From 828fa8ce5a8d75169f16740c28c8a1b7c13dd96b Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 15 Apr 2015 13:29:05 +0200 Subject: ALSA: hda - simplify azx_has_pm_runtime Because AZX_DCAPS_PM_RUNTIME is always defined as non-zero, the initial part of the expression can be skipped. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be1b7ded8d82..0efdb094d21c 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -404,7 +404,7 @@ struct azx { ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) #define azx_has_pm_runtime(chip) \ - (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) + ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME) /* PCM setup */ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) -- cgit v1.2.3 From 3047755588e71b67c3f60409686fabf8506357e9 Mon Sep 17 00:00:00 2001 From: Scott Wood Date: Wed, 15 Apr 2015 18:16:47 -0500 Subject: ALSA: intel8x0: Check pci_iomap() success for DEVICE_ALI DEVICE_ALI previously would jump to port_inited after calling pci_iomap(), bypassing the check for bmaddr being NULL. Signed-off-by: Scott Wood Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 749069aa6997..b120925223ae 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card, chip->bmaddr = pci_iomap(pci, 3, 0); else chip->bmaddr = pci_iomap(pci, 1, 0); + + port_inited: if (!chip->bmaddr) { dev_err(card->dev, "Controller space ioremap problem\n"); snd_intel8x0_free(chip); return -EIO; } - - port_inited: chip->bdbars_count = bdbars[device_type]; /* initialize offsets */ -- cgit v1.2.3 From 7d4b5e978ad350916b5c3995490b09c4e59cec4a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Apr 2015 15:14:53 +0200 Subject: ALSA: hda - Fix regression for slave SPDIF setups The commit [a551d91473e5: ALSA: hda - Use regmap for command verb caches, too] introduced a regression due to a typo in the conversion; the IEC958 status bits of slave digital devices aren't updated correctly. This patch corrects it. Fixes: a551d91473e5 ('ALSA: hda - Use regmap for command verb caches, too') Reported-and-tested-by: Markus Trippelsdorf 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 e70a7fb393dd..873ed1bce12b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2529,7 +2529,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, if (!d) return; for (; *d; d++) - snd_hdac_regmap_update(&codec->core, nid, + snd_hdac_regmap_update(&codec->core, *d, AC_VERB_SET_DIGI_CONVERT_1, mask, val); } -- cgit v1.2.3 From 28ecc0b658e2ac882faa80e7ff1d72d144299bd0 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 15 Apr 2015 00:08:15 -0300 Subject: ASoC: fsl_ssi: Fix platform_get_irq() error handling We should check whether platform_get_irq() returns a negative number and propagate the error in this case. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e8bb8eef1d16..0d48804218b1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1357,7 +1357,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return ssi_private->irq; } -- cgit v1.2.3 From 427ced4b203dfea4f08b1298cd1f88e19039fca4 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 16 Apr 2015 20:17:46 +0800 Subject: ASoC: tfa9879: Fix return value check in tfa9879_i2c_probe() In case of error, the function devm_kzalloc() returns NULL not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun Acked-by: Peter Rosin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tfa9879.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 16f1b71edb55..aab0af681e8c 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -280,8 +280,8 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c, int i; tfa9879 = devm_kzalloc(&i2c->dev, sizeof(*tfa9879), GFP_KERNEL); - if (IS_ERR(tfa9879)) - return PTR_ERR(tfa9879); + if (!tfa9879) + return -ENOMEM; i2c_set_clientdata(i2c, tfa9879); -- cgit v1.2.3 From c479163a1b6ab424786fbcd9225b4e3c1c58eb0b Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 16 Apr 2015 20:18:02 +0800 Subject: ASoC: samsung: s3c24xx-i2s: Fix return value check in s3c24xx_iis_dev_probe() In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun Reviewed-by: Krzysztof Kozlowski Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 326d3c3804e3..5bf723689692 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -461,8 +461,8 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return -ENOENT; } s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + if (IS_ERR(s3c24xx_i2s.regs)) + return PTR_ERR(s3c24xx_i2s.regs); s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; -- cgit v1.2.3 From f4d770317997f89bb6997ee3e8dd495cb8356ae9 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 17 Apr 2015 16:19:46 +0300 Subject: ALSA: hda - potential (but unlikely) uninitialized variable This function is a bit unusual because it accepts negative values as "conn_len". It's theoretically possible for both "cache_len" and "conn_len" to be -ENOSPC and in that case we would oops trying to run memcmp() on the uninitialized "list" pointer. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ee6230767c64..baaf7ed06875 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,8 +582,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, /* Get Cache connections info */ cache_len = snd_hda_get_conn_list(codec, nid, &list); - if (cache_len != conn_len - || memcmp(list, conn, conn_len)) { + if (cache_len >= 0 && (cache_len != conn_len || + memcmp(list, conn, conn_len) != 0)) { snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); if (cache_len > 0) { snd_iprintf(buffer, " "); -- cgit v1.2.3 From d09a6b4a1412149c133a58b53f50e9c05a95e834 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 16 Apr 2015 21:46:29 +0800 Subject: ASoC: Intel: sst_byt: remove kfree for memory allocated with devm_kzalloc It's not necessary to free memory allocated with devm_kzalloc and using kfree leads to a double free. Signed-off-by: Wei Yongjun Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 1efb33b36303..a839dbfa5218 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -759,7 +759,6 @@ fw_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(byt); return err; } -- cgit v1.2.3 From bc26d4d06e337ade069f33d3f4377593b24e6e36 Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov Date: Sat, 18 Apr 2015 02:53:25 +0300 Subject: sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND) A deadlock can be initiated by userspace via ioctl(SNDCTL_SEQ_OUTOFBAND) on /dev/sequencer with TMR_ECHO midi event. In this case the control flow is: sound_ioctl() -> case SND_DEV_SEQ: case SND_DEV_SEQ2: sequencer_ioctl() -> case SNDCTL_SEQ_OUTOFBAND: spin_lock_irqsave(&lock,flags); play_event(); -> case EV_TIMING: seq_timing_event() -> case TMR_ECHO: seq_copy_to_input() -> spin_lock_irqsave(&lock,flags); It seems that spin_lock_irqsave() around play_event() is not necessary, because the only other call location in seq_startplay() makes the call without acquiring spinlock. So, the patch just removes spinlocks around play_event(). By the way, it removes unreachable code in seq_timing_event(), since (seq_mode == SEQ_2) case is handled in the beginning. Compile tested only. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov Signed-off-by: Takashi Iwai --- sound/oss/sequencer.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c0eea1dfe90f..f19da4b47c1d 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec) break; case TMR_ECHO: - if (seq_mode == SEQ_2) - seq_copy_to_input(event_rec, 8); - else - { - parm = (parm << 8 | SEQ_ECHO); - seq_copy_to_input((unsigned char *) &parm, 4); - } + parm = (parm << 8 | SEQ_ECHO); + seq_copy_to_input((unsigned char *) &parm, 4); break; default:; @@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a int mode = translate_mode(file); struct synth_info inf; struct seq_event_rec event_rec; - unsigned long flags; int __user *p = arg; orig_dev = dev = dev >> 4; @@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a case SNDCTL_SEQ_OUTOFBAND: if (copy_from_user(&event_rec, arg, sizeof(event_rec))) return -EFAULT; - spin_lock_irqsave(&lock,flags); play_event(event_rec.arr); - spin_unlock_irqrestore(&lock,flags); return 0; case SNDCTL_MIDI_INFO: -- cgit v1.2.3 From 9476d369d7b39348945c297da5f2935904229813 Mon Sep 17 00:00:00 2001 From: Gabriele Mazzotta Date: Sun, 19 Apr 2015 19:00:40 +0200 Subject: ALSA: hda - Mute headphone pin on suspend on XPS13 9333 Muting the headphone output pin right before the codec suspension prevents pop noises when headphones are plugged in (except for a barely audible click noise). This solution allows to truly save some power when headphones are plugged in unlike the previous solution (033b0a7ca9c: "ALSA: hda - Pop noises fix for XPS13 9333") Signed-off-by: Gabriele Mazzotta Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b18b9c67b262..231d0e4b9a95 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } -static unsigned int alc_power_filter_xps13(struct hda_codec *codec, - hda_nid_t nid, - unsigned int power_state) +static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int hp_pin = spec->gen.autocfg.hp_pins[0]; - /* Avoid pop noises when headphones are plugged in */ - if (spec->gen.hp_jack_present) - if (nid == codec->core.afg || nid == 0x02 || nid == 0x15) - return AC_PWRST_D0; - return snd_hda_gen_path_power_filter(codec, nid, power_state); + /* Prevent pop noises when headphones are plugged in */ + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + msleep(20); } static void alc_fixup_dell_xps13(struct hda_codec *codec, @@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, struct hda_input_mux *imux = &spec->gen.input_mux; int i; - spec->shutup = alc_no_shutup; - codec->power_filter = alc_power_filter_xps13; + spec->shutup = alc_shutup_dell_xps13; /* Make the internal mic the default input source. */ for (i = 0; i < imux->num_items; i++) { -- cgit v1.2.3 From f4c1a311d8dc55c90c39e9cf7b003254a769574d Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 20 Apr 2015 17:33:57 +0800 Subject: ALSA: hda - only sync BCLK to the display clock for Haswell & Broadwell Only Intel Haswell and Broadwell have a separate HD-A controller (PCI device 3) for display audio, which needs to get 24MHz HD-A link BCLK from the variable display core clock through vendor specific registers EM4 & EM5. Other platforms (Baytrail, Braswell and Skylake) don't have this feature. So this patch checks the PCI device ID of the controller in haswell_set_bclk() and only sync BCLK for HSW and BDW. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_i915.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 52a85d87c23c..3052a2b095f7 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -55,6 +55,12 @@ void haswell_set_bclk(struct hda_intel *hda) int cdclk_freq; unsigned int bclk_m, bclk_n; struct i915_audio_component *acomp = &hda->audio_component; + struct pci_dev *pci = hda->chip.pci; + + /* Only Haswell/Broadwell need set BCLK */ + if (pci->device != 0x0a0c && pci->device != 0x0c0c + && pci->device != 0x0d0c && pci->device != 0x160c) + return; if (!acomp->ops) return; -- cgit v1.2.3 From 40cc2392f4b144197d05eec73c1560f42fc25def Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 21 Apr 2015 13:12:23 +0800 Subject: ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail This patch addes AZX_DCAPS_I915_POWERWELL to BYT (Baytrail). Like Braswell and Skylake, the HDMI codec on Bytrail is also in the shared power well with GPU. This power well must be turned on before we reset link to probe the codec, to avoid communication failure with the codec. The side effect is that this power is always ON in S0 because the BYT HDMI codec does not support EPSS or D3ClkStop and so the controller doesn't enter D3 at runtime, and the HDMI codec and analog codec share a single physical HD-A link and so we cannot reset the HD-A link freely when we re-enable the power to use the HDMI codec. Next step is to test if an AGP reset or double AGP reset on BYT HDMI codec is okay to bring the HDMI codec back to a functional state after restoring the power. If okay, we can bind the power on/off with the HDMI codec PM without interrupting the analog audio. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e1c210515581..34040d26c94f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,6 +297,9 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BAYTRAIL \ + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_BRASWELL \ (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) @@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, -- cgit v1.2.3 From 6d1f2f605601ec701b561eca143c03e2a22d6489 Mon Sep 17 00:00:00 2001 From: Takamichi Horikawa Date: Tue, 21 Apr 2015 11:23:57 +0900 Subject: ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module Roland SC-D70 reports its device class as vendor specific class and the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output. In the quirks table the sampling rate was hard-coded to 44100 Hz and therefore not worked when the sound module was in 48000 Hz mode. In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE but as the sound module reports incorrect bSubframeSize in its descriptors, additional change is made in format.c to detect it and to override it (which uses the existing code for Edirol SD-90). Tested both when the sound module was in 44100 Hz mode and 48000 Hz mode and both audio input and output. MIDI related part of the driver is not touched. Signed-off-by: Takamichi Horikawa Signed-off-by: Takashi Iwai --- sound/usb/format.c | 5 ++++- sound/usb/quirks-table.h | 30 ++---------------------------- 2 files changed, 6 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/usb/format.c b/sound/usb/format.c index 8bcc87cf5667..789d19ec035d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format = 1 << UAC_FORMAT_TYPE_I_PCM; } if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { - if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ && + if (((chip->usb_id == USB_ID(0x0582, 0x0016)) || + /* Edirol SD-90 */ + (chip->usb_id == USB_ID(0x0582, 0x000c))) && + /* Roland SC-D70 */ sample_width == 24 && sample_bytes == 2) sample_bytes = 3; else if (sample_width > sample_bytes * 8) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 07f984d5f516..2f6d3e9a1bcd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S24_3LE, - .channels = 2, - .iface = 0, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x01, - .ep_attr = 0x01, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_min = 44100, - .rate_max = 44100, - } + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S24_3LE, - .channels = 2, - .iface = 1, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .endpoint = 0x81, - .ep_attr = 0x01, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - .rate_min = 44100, - .rate_max = 44100, - } + .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, -- cgit v1.2.3 From 7d1b6e29327428993ba568bdd8c66734070f45e0 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 21 Apr 2015 10:48:46 +0200 Subject: ALSA: hda - fix "num_steps = 0" error on ALC256 The ALC256 does not have a mixer nid at 0x0b, and there's no loopback path (the output pins are directly connected to the DACs). This commit fixes an "num_steps = 0 for NID=0xb (ctl = Beep Playback Volume)" error (and as a result, problems with amixer/alsamixer). If there's pcbeep functionality, it certainly isn't controlled by setting an amp on 0x0b, so disable beep functionality (at least for now). Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1446517 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 231d0e4b9a95..03975d03b264 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5664,6 +5664,7 @@ static int patch_alc269(struct hda_codec *codec) break; case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; + spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */ break; } @@ -5677,8 +5678,8 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->gen.no_analog && spec->gen.beep_nid) - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid) + set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; codec->patch_ops.stream_pm = snd_hda_gen_stream_pm; -- cgit v1.2.3 From 8faf141a9903477910387af062ece04ea7d730ed Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 23 Apr 2015 14:38:13 +0530 Subject: ASoC: Intel: fix the makefile for atom code The tom code should be using SND_SST_MFLD_PLATFORM and not the baytrail one. So fix it now Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index cd9aee9871a3..3853ec2ddbc7 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -4,7 +4,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SST) += common/ # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ -obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += atom/ +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ -- cgit v1.2.3 From d32b66668c702aed0e330dc5ca186afbadcdacf8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 23 Apr 2015 15:10:53 +0800 Subject: ALSA: hda/realtek - Fix Headphone Mic doesn't recording for ALC256 Switch default pcbeep path to Line in path. Signed-off-by: Kailang Yang Tested-by: Hui Wang Cc: 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 03975d03b264..4b10cde12831 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5665,6 +5665,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */ + alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; } -- cgit v1.2.3 From e8191a8e475551b277d85cd76c3f0f52fdf42e86 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 24 Apr 2015 13:39:59 +0800 Subject: ALSA: hda - fix headset mic detection problem for one more machine We have two machines with alc256 codec in the pin quirk table, so moving the common pins to ALC256_STANDARD_PINS. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1447909 Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b10cde12831..06199e4e930f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5228,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x1b, 0x411111f0}, \ {0x1e, 0x411111f0} +#define ALC256_STANDARD_PINS \ + {0x12, 0x90a60140}, \ + {0x14, 0x90170110}, \ + {0x19, 0x411111f0}, \ + {0x1a, 0x411111f0}, \ + {0x1b, 0x411111f0}, \ + {0x1d, 0x40700001}, \ + {0x1e, 0x411111f0}, \ + {0x21, 0x02211020} + #define ALC282_STANDARD_PINS \ {0x14, 0x90170110}, \ {0x18, 0x411111f0}, \ @@ -5328,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60140}, - {0x13, 0x40000000}, - {0x14, 0x90170110}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}, - {0x21, 0x02211020}), + ALC256_STANDARD_PINS, + {0x13, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- cgit v1.2.3 From 74d6ea52aeef0236242221c6eff6d892565c5a92 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 24 Apr 2015 15:19:29 +0800 Subject: ASoC: rt5677: add register patch for PLL The PLL output will be unstable in some cases. We can fix it by setting some registers. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5677.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..c6d4e8fa8bd3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -62,6 +62,9 @@ static const struct reg_default init_list[] = { {RT5677_PR_BASE + 0x1e, 0x0000}, {RT5677_PR_BASE + 0x12, 0x0eaa}, {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_PR_BASE + 0x15, 0x0490}, + {RT5677_PR_BASE + 0x38, 0x0f71}, + {RT5677_PR_BASE + 0x39, 0x0f71}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- cgit v1.2.3 From 3168c201f7ca333d12f80f8d98bbbe3a33746f8b Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Thu, 23 Apr 2015 16:35:17 -0700 Subject: ASoC: rt5645: Add ACPI match ID This patch adds the ACPI match ID for rt5645/5650 codec Signed-off-by: Fang, Yang A Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c9a4c5be083b..e16724a11c47 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -2618,6 +2619,15 @@ static const struct i2c_device_id rt5645_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5645_acpi_match[] = { + { "10EC5645", 0 }, + { "10EC5650", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5645_acpi_match); +#endif + static int rt5645_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2834,6 +2844,7 @@ static struct i2c_driver rt5645_i2c_driver = { .driver = { .name = "rt5645", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5645_acpi_match), }, .probe = rt5645_i2c_probe, .remove = rt5645_i2c_remove, -- cgit v1.2.3 From 3e1b0c4a9d563d7fc6e22dc92613cd3237bb5ce0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 10:43:22 +0200 Subject: ALSA: hda - Fix click noise at start on Dell XPS13 Dell XPS13 produces a click noise at boot up, and Gabriele spotted out that it's triggered by the initial pin control of the mic (NID 0x19). This has to be set to Hi-Z Vref while the driver initializes to Vref 80% as a normal mic. This patch fixes the generic parser code not to override the target vref if it has been already set by the driver, and adds a proper initialization of the target vref for this pin in the Realtek driver side. Reported-and-tested-by: Gabriele Mazzotta Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 3 ++- sound/pci/hda/patch_realtek.c | 16 ++++++++++++---- 2 files changed, 14 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3d2597b7037b..788f969b1a68 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3259,7 +3259,8 @@ static int create_input_ctls(struct hda_codec *codec) val = PIN_IN; if (cfg->inputs[i].type == AUTO_PIN_MIC) val |= snd_hda_get_default_vref(codec, pin); - if (pin != spec->hp_mic_pin) + if (pin != spec->hp_mic_pin && + !snd_hda_codec_get_pin_target(codec, pin)) set_pin_target(codec, pin, val, false); if (mixer) { diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06199e4e930f..e2afd53cc14c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4190,11 +4190,18 @@ static void alc_shutup_dell_xps13(struct hda_codec *codec) static void alc_fixup_dell_xps13(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - if (action == HDA_FIXUP_ACT_PROBE) { - struct alc_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->gen.input_mux; - int i; + struct alc_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->gen.input_mux; + int i; + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + /* mic pin 0x19 must be initialized with Vref Hi-Z, otherwise + * it causes a click noise at start up + */ + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); + break; + case HDA_FIXUP_ACT_PROBE: spec->shutup = alc_shutup_dell_xps13; /* Make the internal mic the default input source. */ @@ -4204,6 +4211,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, break; } } + break; } } -- cgit v1.2.3 From ee52e56e7b12834476cd0031c5986254ba1b6317 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 10:36:11 +0200 Subject: ALSA: hda - Fix mute-LED fixed mode MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The mute-LED mode control has the fixed on/off states that are supposed to remain on/off regardless of the master switch. However, this doesn't work actually because the vmaster hook is called in the vmaster code itself. This patch fixes it by calling the hook indirectly after checking the mute LED mode. Reported-and-tested-by: Pali Rohár Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 873ed1bce12b..27333e0d8ebe 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2082,6 +2082,16 @@ static struct snd_kcontrol_new vmaster_mute_mode = { .put = vmaster_mute_mode_put, }; +/* meta hook to call each driver's vmaster hook */ +static void vmaster_hook(void *private_data, int enabled) +{ + struct hda_vmaster_mute_hook *hook = private_data; + + if (hook->mute_mode != HDA_VMUTE_FOLLOW_MASTER) + enabled = hook->mute_mode; + hook->hook(hook->codec, enabled); +} + /** * snd_hda_add_vmaster_hook - Add a vmaster hook for mute-LED * @codec: the HDA codec @@ -2100,9 +2110,9 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec, if (!hook->hook || !hook->sw_kctl) return 0; - snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec); hook->codec = codec; hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; + snd_ctl_add_vmaster_hook(hook->sw_kctl, vmaster_hook, hook); if (!expose_enum_ctl) return 0; kctl = snd_ctl_new1(&vmaster_mute_mode, hook); @@ -2128,14 +2138,7 @@ void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) */ if (hook->codec->bus->shutdown) return; - switch (hook->mute_mode) { - case HDA_VMUTE_FOLLOW_MASTER: - snd_ctl_sync_vmaster_hook(hook->sw_kctl); - break; - default: - hook->hook(hook->codec, hook->mute_mode); - break; - } + snd_ctl_sync_vmaster_hook(hook->sw_kctl); } EXPORT_SYMBOL_GPL(snd_hda_sync_vmaster_hook); -- cgit v1.2.3 From 7290006d8c0900c56d8c58428134f02c35109d17 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 10:40:45 +0200 Subject: ALSA: hda - Add mute-LED mode control to Thinkpad MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds the missing flag to enable "Mute-LED Mode" mixer enum ctl for Thinkpads that have also the software mute-LED control. Reported-and-tested-by: Pali Rohár Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/thinkpad_helper.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 0a4ad5feb82e..d51703e30523 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -72,6 +72,7 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; + spec->vmaster_mute_enum = 1; removefunc = false; } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { -- cgit v1.2.3 From d02260824e2cad626fb2a9d62e27006d34b6dedc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 13:00:09 +0200 Subject: ALSA: emu10k1: Fix card shortname string buffer overflow Some models provide too long string for the shortname that has 32bytes including the terminator, and it results in a non-terminated string exposed to the user-space. This isn't too critical, though, as the string is stopped at the succeeding longname string. This patch fixes such entries by dropping "SB" prefix (it's enough to fit within 32 bytes, so far). Meanwhile, it also changes strcpy() with strlcpy() to make sure that this kind of problem won't happen in future, too. Cc: Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1.c | 6 ++++-- sound/pci/emu10k1/emu10k1_main.c | 4 ++-- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 37d0220a094c..db7a2e5e4a14 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -183,8 +183,10 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, } #endif - strcpy(card->driver, emu->card_capabilities->driver); - strcpy(card->shortname, emu->card_capabilities->name); + strlcpy(card->driver, emu->card_capabilities->driver, + sizeof(card->driver)); + strlcpy(card->shortname, emu->card_capabilities->name, + sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), "%s (rev.%d, serial:0x%x) at 0x%lx, irq %i", card->shortname, emu->revision, emu->serial, emu->port, emu->irq); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 54079f5d5673..4f8cf5e7e45f 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1446,7 +1446,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "SB Audigy 2 ZS Notebook [SB0530]", + .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1596,7 +1596,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, - .driver = "Audigy2", .name = "SB Audigy 2 Platinum EX [SB0280]", + .driver = "Audigy2", .name = "Audigy 2 Platinum EX [SB0280]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, -- cgit v1.2.3 From 07b0e5d49d227e3950cb13a3e8caf248ef2a310e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 14:50:39 +0200 Subject: ALSA: emux: Fix mutex deadlock at unloading The emux-synth driver has a possible AB/BA mutex deadlock at unloading the emu10k1 driver: snd_emux_free() -> snd_emux_detach_seq(): mutex_lock(&emu->register_mutex) -> snd_seq_delete_kernel_client() -> snd_seq_free_client(): mutex_lock(®ister_mutex) snd_seq_release() -> snd_seq_free_client(): mutex_lock(®ister_mutex) -> snd_seq_delete_all_ports() -> snd_emux_unuse(): mutex_lock(&emu->register_mutex) Basically snd_emux_detach_seq() doesn't need a protection of emu->register_mutex as it's already being unregistered. So, we can get rid of this for avoiding the deadlock. Cc: Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_seq.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 7778b8e19782..188fda0effb0 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -124,12 +124,10 @@ snd_emux_detach_seq(struct snd_emux *emu) if (emu->voices) snd_emux_terminate_all(emu); - mutex_lock(&emu->register_mutex); if (emu->client >= 0) { snd_seq_delete_kernel_client(emu->client); emu->client = -1; } - mutex_unlock(&emu->register_mutex); } -- cgit v1.2.3 From 30e5f003ff4b2be86f71733b6c9b11355d66584c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Apr 2015 16:39:19 +0200 Subject: ALSA: hda - Fix missing va_end() call in snd_hda_codec_pcm_new() Reported by coverity CID 1296024. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 27333e0d8ebe..b49feff0a319 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -873,14 +873,15 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, struct hda_pcm *pcm; va_list args; - va_start(args, fmt); pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return NULL; pcm->codec = codec; kref_init(&pcm->kref); + va_start(args, fmt); pcm->name = kvasprintf(GFP_KERNEL, fmt, args); + va_end(args); if (!pcm->name) { kfree(pcm); return NULL; -- cgit v1.2.3 From 53f9b3baa937e0cbdd75ea11b3c824462e4359b2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 21 Apr 2015 12:19:49 +0800 Subject: ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Current code uses wrong mask when setting RT5645_DMIC_2_DP_GPIO12 bit, fix it. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e16724a11c47..3153aa0fe51b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2742,7 +2742,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, case RT5645_DMIC_DATA_GPIO12: regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, - RT5645_DMIC_1_DP_MASK, RT5645_DMIC_2_DP_GPIO12); + RT5645_DMIC_2_DP_MASK, RT5645_DMIC_2_DP_GPIO12); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, RT5645_GP12_PIN_MASK, RT5645_GP12_PIN_DMIC2_SDA); -- cgit v1.2.3 From 1c94e65c668f44d2c69ae7e7fc268ab3268fba3e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2015 17:11:44 +0200 Subject: ALSA: emux: Fix mutex deadlock in OSS emulation The OSS emulation in synth-emux helper has a potential AB/BA deadlock at the simultaneous closing and opening: close -> snd_seq_release() -> sne_seq_free_client() -> snd_seq_delete_all_ports(): takes client->ports_mutex -> port_delete() -> snd_emux_unuse(): takes emux->register_mutex open -> snd_seq_oss_open() -> snd_emux_open_seq_oss(): takes emux->register_mutex -> snd_seq_event_port_attach() -> snd_seq_create_port(): takes client->ports_mutex This patch addresses the deadlock by reducing the rance taking emux->register_mutex in snd_emux_open_seq_oss(). The lock is needed for the refcount handling, so move it locally. The calls in emux_seq.c are already with the mutex, thus they are replaced with the version without mutex lock/unlock. Cc: Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_oss.c | 11 +---------- sound/synth/emux/emux_seq.c | 27 +++++++++++++++++++++------ 2 files changed, 22 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index ab37add269ae..82e350e9501c 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -118,12 +118,8 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (snd_BUG_ON(!arg || !emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); - - if (!snd_emux_inc_count(emu)) { - mutex_unlock(&emu->register_mutex); + if (!snd_emux_inc_count(emu)) return -EFAULT; - } memset(&callback, 0, sizeof(callback)); callback.owner = THIS_MODULE; @@ -135,7 +131,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (p == NULL) { snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return -ENOMEM; } @@ -148,8 +143,6 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) reset_port_mode(p, arg->seq_mode); snd_emux_reset_port(p); - - mutex_unlock(&emu->register_mutex); return 0; } @@ -195,13 +188,11 @@ snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg) if (snd_BUG_ON(!emu)) return -ENXIO; - mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); snd_soundfont_close_check(emu->sflist, SF_CLIENT_NO(p->chset.port)); snd_seq_event_port_detach(p->chset.client, p->chset.port); snd_emux_dec_count(emu); - mutex_unlock(&emu->register_mutex); return 0; } diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 188fda0effb0..a0209204ae48 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -267,8 +267,8 @@ snd_emux_event_input(struct snd_seq_event *ev, int direct, void *private_data, /* * increment usage count */ -int -snd_emux_inc_count(struct snd_emux *emu) +static int +__snd_emux_inc_count(struct snd_emux *emu) { emu->used++; if (!try_module_get(emu->ops.owner)) @@ -282,12 +282,21 @@ snd_emux_inc_count(struct snd_emux *emu) return 1; } +int snd_emux_inc_count(struct snd_emux *emu) +{ + int ret; + + mutex_lock(&emu->register_mutex); + ret = __snd_emux_inc_count(emu); + mutex_unlock(&emu->register_mutex); + return ret; +} /* * decrease usage count */ -void -snd_emux_dec_count(struct snd_emux *emu) +static void +__snd_emux_dec_count(struct snd_emux *emu) { module_put(emu->card->module); emu->used--; @@ -296,6 +305,12 @@ snd_emux_dec_count(struct snd_emux *emu) module_put(emu->ops.owner); } +void snd_emux_dec_count(struct snd_emux *emu) +{ + mutex_lock(&emu->register_mutex); + __snd_emux_dec_count(emu); + mutex_unlock(&emu->register_mutex); +} /* * Routine that is called upon a first use of a particular port @@ -315,7 +330,7 @@ snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_init_port(p); - snd_emux_inc_count(emu); + __snd_emux_inc_count(emu); mutex_unlock(&emu->register_mutex); return 0; } @@ -338,7 +353,7 @@ snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info) mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); - snd_emux_dec_count(emu); + __snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return 0; } -- cgit v1.2.3 From 7241ea558c6715501e777396b5fc312c372e11d9 Mon Sep 17 00:00:00 2001 From: Peter Zubaj Date: Tue, 28 Apr 2015 21:57:29 +0200 Subject: ALSA: emu10k1: Emu10k2 32 bit DMA mode Looks like audigy emu10k2 (probably emu10k1 - sb live too) support two modes for DMA. Second mode is useful for 64 bit os with more then 2 GB of ram (fixes problems with big soundfont loading) 1) 32MB from 2 GB address space using 8192 pages (used now as default) 2) 16MB from 4 GB address space using 4096 pages Mode is set using HCFG_EXPANDED_MEM flag in HCFG register. Also format of emu10k2 page table is then different. Signed-off-by: Peter Zubaj Tested-by: Takashi Iwai Cc: Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 14 +++++++++----- sound/pci/emu10k1/emu10k1_callback.c | 4 ++-- sound/pci/emu10k1/emu10k1_main.c | 17 ++++++++++++----- sound/pci/emu10k1/emupcm.c | 2 +- sound/pci/emu10k1/memory.c | 11 ++++++----- 5 files changed, 30 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 0de95ccb92cf..5bd134651f5e 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -41,7 +41,8 @@ #define EMUPAGESIZE 4096 #define MAXREQVOICES 8 -#define MAXPAGES 8192 +#define MAXPAGES0 4096 /* 32 bit mode */ +#define MAXPAGES1 8192 /* 31 bit mode */ #define RESERVED 0 #define NUM_MIDI 16 #define NUM_G 64 /* use all channels */ @@ -50,8 +51,7 @@ /* FIXME? - according to the OSS driver the EMU10K1 needs a 29 bit DMA mask */ #define EMU10K1_DMA_MASK 0x7fffffffUL /* 31bit */ -#define AUDIGY_DMA_MASK 0x7fffffffUL /* 31bit FIXME - 32 should work? */ - /* See ALSA bug #1276 - rlrevell */ +#define AUDIGY_DMA_MASK 0xffffffffUL /* 32bit mode */ #define TMEMSIZE 256*1024 #define TMEMSIZEREG 4 @@ -466,8 +466,11 @@ #define MAPB 0x0d /* Cache map B */ -#define MAP_PTE_MASK 0xffffe000 /* The 19 MSBs of the PTE indexed by the PTI */ -#define MAP_PTI_MASK 0x00001fff /* The 13 bit index to one of the 8192 PTE dwords */ +#define MAP_PTE_MASK0 0xfffff000 /* The 20 MSBs of the PTE indexed by the PTI */ +#define MAP_PTI_MASK0 0x00000fff /* The 12 bit index to one of the 4096 PTE dwords */ + +#define MAP_PTE_MASK1 0xffffe000 /* The 19 MSBs of the PTE indexed by the PTI */ +#define MAP_PTI_MASK1 0x00001fff /* The 13 bit index to one of the 8192 PTE dwords */ /* 0x0e, 0x0f: Not used */ @@ -1704,6 +1707,7 @@ struct snd_emu10k1 { unsigned short model; /* subsystem id */ unsigned int card_type; /* EMU10K1_CARD_* */ unsigned int ecard_ctrl; /* ecard control bits */ + unsigned int address_mode; /* address mode */ unsigned long dma_mask; /* PCI DMA mask */ unsigned int delay_pcm_irq; /* in samples */ int max_cache_pages; /* max memory size / PAGE_SIZE */ diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 874cd76c7b7f..d2c7ea3a7610 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -415,7 +415,7 @@ start_voice(struct snd_emux_voice *vp) snd_emu10k1_ptr_write(hw, Z2, ch, 0); /* invalidate maps */ - temp = (hw->silent_page.addr << 1) | MAP_PTI_MASK; + temp = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); snd_emu10k1_ptr_write(hw, MAPA, ch, temp); snd_emu10k1_ptr_write(hw, MAPB, ch, temp); #if 0 @@ -436,7 +436,7 @@ start_voice(struct snd_emux_voice *vp) snd_emu10k1_ptr_write(hw, CDF, ch, sample); /* invalidate maps */ - temp = ((unsigned int)hw->silent_page.addr << 1) | MAP_PTI_MASK; + temp = ((unsigned int)hw->silent_page.addr << hw_address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); snd_emu10k1_ptr_write(hw, MAPA, ch, temp); snd_emu10k1_ptr_write(hw, MAPB, ch, temp); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 4f8cf5e7e45f..a4548147c621 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -282,7 +282,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ - silent_page = (emu->silent_page.addr << 1) | MAP_PTI_MASK; + silent_page = (emu->silent_page.addr << emu->address_mode) | (emu->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); for (ch = 0; ch < NUM_G; ch++) { snd_emu10k1_ptr_write(emu, MAPA, ch, silent_page); snd_emu10k1_ptr_write(emu, MAPB, ch, silent_page); @@ -348,6 +348,11 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(reg | A_IOCFG_GPOUT0, emu->port + A_IOCFG); } + if (emu->address_mode == 0) { + /* use 16M in 4G */ + outl(inl(emu->port + HCFG) | HCFG_EXPANDED_MEM, emu->port + HCFG); + } + return 0; } @@ -1902,8 +1907,10 @@ int snd_emu10k1_create(struct snd_card *card, is_audigy = emu->audigy = c->emu10k2_chip; + /* set addressing mode */ + emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ - emu->dma_mask = is_audigy ? AUDIGY_DMA_MASK : EMU10K1_DMA_MASK; + emu->dma_mask = emu->address_mode ? EMU10K1_DMA_MASK : AUDIGY_DMA_MASK; if (pci_set_dma_mask(pci, emu->dma_mask) < 0 || pci_set_consistent_dma_mask(pci, emu->dma_mask) < 0) { dev_err(card->dev, @@ -1928,7 +1935,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - 32 * 1024, &emu->ptb_pages) < 0) { + (emu->address_mode ? 32 : 16) * 1024, &emu->ptb_pages) < 0) { err = -ENOMEM; goto error; } @@ -2027,8 +2034,8 @@ int snd_emu10k1_create(struct snd_card *card, /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); - silent_page = emu->silent_page.addr << 1; - for (idx = 0; idx < MAXPAGES; idx++) + silent_page = emu->silent_page.addr << emu->address_mode; + for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); /* set up voice indices */ diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 0dc07385af0e..14a305bd8a98 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -380,7 +380,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, snd_emu10k1_ptr_write(emu, Z1, voice, 0); snd_emu10k1_ptr_write(emu, Z2, voice, 0); /* invalidate maps */ - silent_page = ((unsigned int)emu->silent_page.addr << 1) | MAP_PTI_MASK; + silent_page = ((unsigned int)emu->silent_page.addr << emu->address_mode) | (emu->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); snd_emu10k1_ptr_write(emu, MAPA, voice, silent_page); snd_emu10k1_ptr_write(emu, MAPB, voice, silent_page); /* modulation envelope */ diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index c68e6dd2fa67..4f1f69be1865 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -34,10 +34,11 @@ * aligned pages in others */ #define __set_ptb_entry(emu,page,addr) \ - (((u32 *)(emu)->ptb_pages.area)[page] = cpu_to_le32(((addr) << 1) | (page))) + (((u32 *)(emu)->ptb_pages.area)[page] = cpu_to_le32(((addr) << (emu->address_mode)) | (page))) #define UNIT_PAGES (PAGE_SIZE / EMUPAGESIZE) -#define MAX_ALIGN_PAGES (MAXPAGES / UNIT_PAGES) +#define MAX_ALIGN_PAGES0 (MAXPAGES0 / UNIT_PAGES) +#define MAX_ALIGN_PAGES1 (MAXPAGES1 / UNIT_PAGES) /* get aligned page from offset address */ #define get_aligned_page(offset) ((offset) >> PAGE_SHIFT) /* get offset address from aligned page */ @@ -124,7 +125,7 @@ static int search_empty_map_area(struct snd_emu10k1 *emu, int npages, struct lis } page = blk->mapped_page + blk->pages; } - size = MAX_ALIGN_PAGES - page; + size = (emu->address_mode ? MAX_ALIGN_PAGES1 : MAX_ALIGN_PAGES0) - page; if (size >= max_size) { *nextp = pos; return page; @@ -181,7 +182,7 @@ static int unmap_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) q = get_emu10k1_memblk(p, mapped_link); end_page = q->mapped_page; } else - end_page = MAX_ALIGN_PAGES; + end_page = (emu->address_mode ? MAX_ALIGN_PAGES1 : MAX_ALIGN_PAGES0); /* remove links */ list_del(&blk->mapped_link); @@ -307,7 +308,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst if (snd_BUG_ON(!emu)) return NULL; if (snd_BUG_ON(runtime->dma_bytes <= 0 || - runtime->dma_bytes >= MAXPAGES * EMUPAGESIZE)) + runtime->dma_bytes >= (emu->address_mode ? MAXPAGES1 : MAXPAGES0) * EMUPAGESIZE)) return NULL; hdr = emu->memhdr; if (snd_BUG_ON(!hdr)) -- cgit v1.2.3 From 60a8d62b8497c23eb3d48149af7e55dac2dd83a2 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 28 Apr 2015 11:27:39 +0800 Subject: ASoC: rt5677: fixed wrong DMIC ref clock DMIC clock source is not from codec system clock directly. it is generated from the division of system clock. And it should be 256 * sample rate of AIF1. Signed-off-by: Bard Liao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c6d4e8fa8bd3..84d162d91ff6 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -904,7 +904,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int idx = rl6231_calc_dmic_clk(rt5677->sysclk); + int idx = rl6231_calc_dmic_clk(rt5677->lrck[RT5677_AIF1] << 8); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); -- cgit v1.2.3