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.

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);