Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "A relative calm release at this time with a flat diffstat.  The only
  significant change in the ALSA core side is the support for more than
  32 card instances, configurable via kconfig.  Other than that, in both
  ASoC and other parts, mostly some improvements and fixes on the driver
  side.

   - hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
   - hda: Haswell HDMI audio fixes, runtime PM improvements
   - hda: Intel BayTrail support, ALC5505 DSP support
   - es1968: MediaForte M56VAP support
   - usb-audio: Improved support for Yamaha/Roland devices
   - usb-audio: M2Tech hiFace, Audio Advantage Micro II support
   - hdspm: wordclock fixes
   - ASoC: Pending fixes for WM8962
   - ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
   - ASoC: Generalisation of the Bluetooth and HDMI stub drivers
   - ASoC: SSM2518 and RT5640 codec drivers.
   - ASoC: Tegra CPUs with RT5640 machine driver
   - ASoC: AC'97 refactoring bug fixes
   - ASoC: ADAU1701 driver fixes
   - Clean up of *_set_drvdata() in a wide range of drivers"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
  ALSA: vmaster: Fix the regression of missing vmaster hook call
  ALSA: hda - Add Dell SSID to support Headset Mic recording
  ASoC: adau1701: remove control_data assignment
  ASoC: adau1701: more direct regmap usage
  ASoC: ac97: fixup multi-platform AC'97 module build failure
  ASoC: pxa2xx: fixup multi-platform AC'97 build failures
  ASoC: tegra20-ac97: Remove unused variable
  ASoC: tegra20-ac97: Remove duplicate error message
  ALSA: usb-audio: Add Audio Advantage Micro II
  ASoC: tas5086: fix Mid-Z implementation
  ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
  ALSA: Replace the magic number 44 with const
  ALSA: hda - Fix the max length of control name in generic parser
  ALSA: hda - Guess what, it's two more Dell headset mic quirks
  ALSA: hda - Yet another Dell headset mic quirk
  ALSA: hda - Add support for ALC5505 DSP power-save mode
  ASoC: mfld: Remove unused variable
  ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
  ALSA: usb-audio: claim autodetected PCM interfaces all at once
  ALSA: usb-audio: remove superfluous Roland quirks
  ...

5 files changed, 3 insertions, 70 deletions

diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index b41fffc056fb..b16abbbf7764 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -49,24 +49,8 @@ static const struct snd_pcm_hardware snd_mxs_hardware = {
 	.fifo_size		= 32,
 };
 
-static bool filter(struct dma_chan *chan, void *param)
-{
-	struct mxs_pcm_dma_params *dma_params = param;
-
-	if (!mxs_dma_is_apbx(chan))
-		return false;
-
-	if (chan->chan_id != dma_params->chan_num)
-		return false;
-
-	chan->private = &dma_params->dma_data;
-
-	return true;
-}
-
 static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = {
 	.pcm_hardware = &snd_mxs_hardware,
-	.compat_filter_fn = filter,
 	.prealloc_buffer_size = 64 * 1024,
 };
 
@@ -74,8 +58,6 @@ int mxs_pcm_platform_register(struct device *dev)
 {
 	return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config,
 		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
-		SND_DMAENGINE_PCM_FLAG_NO_DT |
-		SND_DMAENGINE_PCM_FLAG_COMPAT |
 		SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX);
 }
 EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
index 3aa918f9ed3e..bc685b67cac7 100644
--- a/sound/soc/mxs/mxs-pcm.h
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -19,13 +19,6 @@
 #ifndef _MXS_PCM_H
 #define _MXS_PCM_H
 
-#include <linux/fsl/mxs-dma.h>
-
-struct mxs_pcm_dma_params {
-	struct mxs_dma_data dma_data;
-	int chan_num;
-};
-
 int mxs_pcm_platform_register(struct device *dev);
 void mxs_pcm_platform_unregister(struct device *dev);
 
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index d31dc52fa862..49d870034bc3 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -26,8 +26,6 @@
 #include <linux/clk.h>
 #include <linux/delay.h>
 #include <linux/time.h>
-#include <linux/fsl/mxs-dma.h>
-#include <linux/pinctrl/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -605,8 +603,6 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
 	struct mxs_saif *saif = dev_get_drvdata(dai->dev);
 
 	snd_soc_dai_set_drvdata(dai, saif);
-	dai->playback_dma_data = &saif->dma_param;
-	dai->capture_dma_data = &saif->dma_param;
 
 	return 0;
 }
@@ -665,9 +661,8 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
 static int mxs_saif_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
-	struct resource *iores, *dmares;
+	struct resource *iores;
 	struct mxs_saif *saif;
-	struct pinctrl *pinctrl;
 	int ret = 0;
 	struct device_node *master;
 
@@ -707,12 +702,6 @@ static int mxs_saif_probe(struct platform_device *pdev)
 
 	mxs_saif[saif->id] = saif;
 
-	pinctrl = devm_pinctrl_get_select_default(&pdev->dev);
-	if (IS_ERR(pinctrl)) {
-		ret = PTR_ERR(pinctrl);
-		return ret;
-	}
-
 	saif->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(saif->clk)) {
 		ret = PTR_ERR(saif->clk);
@@ -727,22 +716,6 @@ static int mxs_saif_probe(struct platform_device *pdev)
 	if (IS_ERR(saif->base))
 		return PTR_ERR(saif->base);
 
-	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!dmares) {
-		/*
-		 * TODO: This is a temporary solution and should be changed
-		 * to use generic DMA binding later when the helplers get in.
-		 */
-		ret = of_property_read_u32(np, "fsl,saif-dma-channel",
-					   &saif->dma_param.chan_num);
-		if (ret) {
-			dev_err(&pdev->dev, "failed to get dma channel\n");
-			return ret;
-		}
-	} else {
-		saif->dma_param.chan_num = dmares->start;
-	}
-
 	saif->irq = platform_get_irq(pdev, 0);
 	if (saif->irq < 0) {
 		ret = saif->irq;
@@ -759,14 +732,6 @@ static int mxs_saif_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	saif->dma_param.dma_data.chan_irq = platform_get_irq(pdev, 1);
-	if (saif->dma_param.dma_data.chan_irq < 0) {
-		ret = saif->dma_param.dma_data.chan_irq;
-		dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
-			ret);
-		return ret;
-	}
-
 	platform_set_drvdata(pdev, saif);
 
 	ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component,
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
index 3cb342e5bc90..53eaa4bf0e27 100644
--- a/sound/soc/mxs/mxs-saif.h
+++ b/sound/soc/mxs/mxs-saif.h
@@ -117,7 +117,6 @@ struct mxs_saif {
 	unsigned int mclk_in_use;
 	void __iomem *base;
 	int irq;
-	struct mxs_pcm_dma_params dma_param;
 	unsigned int id;
 	unsigned int master_id;
 	unsigned int cur_rate;
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index b1d9b5ebeeeb..1b134d72f120 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -90,17 +90,11 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.name		= "HiFi Tx",
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
-		.codec_name	= "sgtl5000.0-000a",
-		.cpu_dai_name	= "mxs-saif.0",
-		.platform_name	= "mxs-saif.0",
 		.ops		= &mxs_sgtl5000_hifi_ops,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
-		.codec_name	= "sgtl5000.0-000a",
-		.cpu_dai_name	= "mxs-saif.1",
-		.platform_name	= "mxs-saif.1",
 		.ops		= &mxs_sgtl5000_hifi_ops,
 	},
 };
@@ -116,7 +110,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *saif_np[2], *codec_np;
-	int i, ret = 0;
+	int i;
 
 	if (!np)
 		return 1; /* no device tree */
@@ -142,7 +136,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev)
 	of_node_put(saif_np[0]);
 	of_node_put(saif_np[1]);
 
-	return ret;
+	return 0;
 }
 
 static int mxs_sgtl5000_probe(struct platform_device *pdev)