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author | Takashi Iwai <tiwai@suse.de> | 2014-12-08 17:04:02 +0300 |
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committer | Takashi Iwai <tiwai@suse.de> | 2014-12-08 17:04:02 +0300 |
commit | e5edba464c11d9d42550f9e3ff97f25196ba50b2 (patch) | |
tree | 269a640ff93c68db724080f73b0e267e024af082 /sound/soc/fsl | |
parent | 77de61c3975da6f2200935c341e84018ece6ce36 (diff) | |
parent | 1810afd3e1ded09c53d4e966dddce3c7d484521f (diff) | |
download | linux-e5edba464c11d9d42550f9e3ff97f25196ba50b2.tar.xz |
Merge tag 'asoc-v3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.19
Lots and lots of changes this time around, the usual set of driver
updates and a huge bulk of cleanups from Lars-Peter. Probably the most
interesting thing for most users is the Intel driver updates which will
(with some more machine integration work) enable support for newer x86
laptops.
- Conversion of AC'97 drivers to use regmap, bringing us closer to the
removal of the ASoC level I/O code.
- Clean up a lot of old drivers that were open coding things that have
subsequently been implemented in the core.
- Some DAPM performance improvements.
- Removal of the now seldom used CODEC mutex.
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers.
- Support for Samsung boards using rt5631 as the CODEC.
- Removal of the obsolete AFEB9260 machine driver.
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks.
Diffstat (limited to 'sound/soc/fsl')
-rw-r--r-- | sound/soc/fsl/eukrea-tlv320.c | 5 | ||||
-rw-r--r-- | sound/soc/fsl/fsl-asoc-card.c | 19 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_esai.c | 12 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 17 | ||||
-rw-r--r-- | sound/soc/fsl/imx-sgtl5000.c | 6 | ||||
-rw-r--r-- | sound/soc/fsl/imx-spdif.c | 3 | ||||
-rw-r--r-- | sound/soc/fsl/imx-ssi.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-wm8962.c | 6 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_dma.c | 3 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_ac97.c | 6 |
10 files changed, 39 insertions, 40 deletions
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index eb093d5b85c4..b175b0145a42 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -105,7 +105,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) int ret; int int_port = 0, ext_port; struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; + struct device_node *ssi_np = NULL, *codec_np = NULL; eukrea_tlv320.dev = &pdev->dev; if (np) { @@ -217,8 +217,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - if (np) - of_node_put(ssi_np); + of_node_put(ssi_np); return ret; } diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 007c772f3cef..3f6959c8e2f7 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -51,6 +51,7 @@ struct codec_priv { * @sysclk_freq[2]: SYSCLK rates for set_sysclk() * @sysclk_dir[2]: SYSCLK directions for set_sysclk() * @sysclk_id[2]: SYSCLK ids for set_sysclk() + * @slot_width: Slot width of each frame * * Note: [1] for tx and [0] for rx */ @@ -58,6 +59,7 @@ struct cpu_priv { unsigned long sysclk_freq[2]; u32 sysclk_dir[2]; u32 sysclk_id[2]; + u32 slot_width; }; /** @@ -125,7 +127,12 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); - if (priv->card.set_bias_level) + /* + * If codec-dai is DAI Master and all configurations are already in the + * set_bias_level(), bypass the remaining settings in hw_params(). + * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. + */ + if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) return 0; /* Specific configurations of DAIs starts from here */ @@ -137,6 +144,15 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return ret; } + if (cpu_priv->slot_width) { + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, + cpu_priv->slot_width); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + } + return 0; } @@ -448,6 +464,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a645e296199e..ca319d59f843 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -513,10 +513,15 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, u32 width = snd_pcm_format_width(params_format(params)); u32 channels = params_channels(params); u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 slot_width = width; u32 bclk, mask, val; int ret; - bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots; + /* Override slot_width if being specifially set */ + if (esai_priv->slot_width) + slot_width = esai_priv->slot_width; + + bclk = params_rate(params) * slot_width * esai_priv->slots; ret = fsl_esai_set_bclk(dai, tx, bclk); if (ret) @@ -538,7 +543,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0); + val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); @@ -780,9 +785,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - /* Set a default slot size */ - esai_priv->slot_width = 32; - /* Set a default slot number */ esai_priv->slots = 2; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e6955170dc42..b6b0d25f6ace 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,8 +67,6 @@ /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * - * This driver currently only supports the SSI running in I2S slave mode. - * * The SSI has a limitation in that the samples must be in the same byte * order as the host CPU. This is because when multiple bytes are written * to the STX register, the bytes and bits must be written in the same @@ -1099,7 +1097,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { }; static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -1363,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1389,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1412,7 +1410,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_asoc_register; + goto error_irq; /* * If codec-handle property is missing from SSI node, we assume @@ -1460,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1480,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 1cb22dd034eb..1dab963a59f7 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -175,10 +175,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e1dc40143600..0c9068ebe1e7 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -74,8 +74,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) platform_set_drvdata(pdev, data); end: - if (spdif_np) - of_node_put(spdif_np); + of_node_put(spdif_np); return ret; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index ab2fdd76b693..60b0a5b1f1f1 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -382,7 +382,7 @@ static struct snd_soc_dai_driver imx_ssi_dai = { static struct snd_soc_dai_driver imx_ac97_dai = { .probe = imx_ssi_dai_probe, - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17ce6ba4..48179ffe1543 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -281,10 +281,8 @@ static int imx_wm8962_probe(struct platform_device *pdev) clk_fail: clk_disable_unprepare(data->codec_clk); fail: - if (ssi_np) - of_node_put(ssi_np); - if (codec_np) - of_node_put(codec_np); + of_node_put(ssi_np); + of_node_put(codec_np); return ret; } diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f2b5d756b1f3..0b82e209b6e3 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -327,9 +327,6 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) goto capture_alloc_err; } - if (rtd->codec->ac97) - rtd->codec->ac97->private_data = psc_dma; - return 0; capture_alloc_err: diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 24eafa2cfbf4..c6ed6ba965a9 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -237,7 +237,7 @@ 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, + .bus_control = true, .probe = psc_ac97_probe, .playback = { .stream_name = "AC97 Playback", @@ -257,7 +257,7 @@ static struct snd_soc_dai_driver psc_ac97_dai[] = { }, { .name = "mpc5200-psc-ac97.1", - .ac97_control = 1, + .bus_control = true, .playback = { .stream_name = "AC97 SPDIF", .channels_min = 1, @@ -282,7 +282,6 @@ static const struct snd_soc_component_driver psc_ac97_component = { static int psc_ac97_of_probe(struct platform_device *op) { int rc; - struct snd_ac97 ac97; struct mpc52xx_psc __iomem *regs; rc = mpc5200_audio_dma_create(op); @@ -304,7 +303,6 @@ static int psc_ac97_of_probe(struct platform_device *op) psc_dma = dev_get_drvdata(&op->dev); regs = psc_dma->psc_regs; - ac97.private_data = psc_dma; psc_dma->imr = 0; out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); |