Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It was a busy development cycle at this time, as you can see a wide
  range of changes in diffstat.  There are no big changes but many
  refactoring and improvements.  Here we go some highlights:

  ALSA core:
   - Procfs codes were cleaned up to use seq_file
   - Procfs can be opt out via Kconfig (only for EXPERT)
   - Two types of jack API were unified finally; now both kctl and input
     jack devs are handled via a single function call.

  HD-audio:
   - Continued code restructuring for the future ASoC driver; now HDA
     controller driver is split to a core helper module.
   - Preliminary codes for Skylake audio support in HDA core.
   - Proper i915 gfx power well management for SKL & co
   - Enabled runtime PM as default for Intel HDMI/DP codecs
   - Newer Tegra chip supports
   - More quirks for Dell headsets, Alienware (with CA0132), etc.
   - A couple of DRM ELD helper API functions

  ASoC:
   - Support for loading ASoC topology maps from firmware, intended to
     be used to allow self-describing DSP firmware images to be built
     which can map controls added by the DSP to userspace without the
     kernel needing to know about individual DSP firmwares
   - Lots of refactoring to avoid direct access to snd_soc_codec where
     it's not needed supporting future refactoring
   - Big refactoring, cleanup and enhancement for the Wolfson ADSP
     driver
   - Cleanup series for TI TAS2552 and R-CAR drivers
   - Fixes and improvements on RT56xx codecs
   - Support for TI TAS571x power amplifiers
   - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
   - Support for x86 systems with RT5650 and Qualcomm Storm
   - Support for Mediatek AFE (Audio Front End) unit
   - Other various small fixes to ASoC codec drivers

  Firewire:
   - Enhanced to allow non-blocking streams to use timestamp
     synchronization
   - Improve support for DM1500 and BeBoBv3

  Misc:
   - Cleanup of old pci API functions over all PCI sound drivers
   - Fix long-standing regression of the old powermac i2c setup"

* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
  ALSA: pcm: Fix pcm_class sysfs output
  ALSA: hda-beep: Update authors dead email address
  ASoC: wm_adsp: Move DSP Rate controls into the codec
  ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
  ALSA: hda: provide default bus io ops extended hdac
  ALSA: hda: add hda link cleanup routine
  ALSA: hda: add hdac_ext stream creation and cleanup routines
  ASoC: rsrc-card: remove unused ret
  ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
  ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
  ASoC: mediatek: Add machine driver for MAX98090 codec
  ASoC: mediatek: Add AFE platform driver
  ASoC: rsnd: remove io from rsnd_mod
  ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
  ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
  ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
  ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
  ...

8 files changed, 152 insertions, 32 deletions

diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 93d7e56c6066..ccadefceeff2 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 		return ret;
 	}
 
-	dma->assigned = 1;
+	dma->assigned = true;
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 	snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
@@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream)
 		substream->runtime->private_data = NULL;
 	}
 
-	dma->assigned = 0;
+	dma->assigned = false;
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ec79c3d5e65e..5c73bea7b11e 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1,7 +1,7 @@
 /*
  * Freescale ALSA SoC Digital Audio Interface (SAI) driver.
  *
- * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ * Copyright 2012-2015 Freescale Semiconductor, Inc.
  *
  * This program is free software, you can redistribute it and/or modify it
  * under the terms of the GNU General Public License as published by the
@@ -27,6 +27,17 @@
 #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\
 		       FSL_SAI_CSR_FEIE)
 
+static u32 fsl_sai_rates[] = {
+	8000, 11025, 12000, 16000, 22050,
+	24000, 32000, 44100, 48000, 64000,
+	88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = {
+	.count = ARRAY_SIZE(fsl_sai_rates),
+	.list = fsl_sai_rates,
+};
+
 static irqreturn_t fsl_sai_isr(int irq, void *devid)
 {
 	struct fsl_sai *sai = (struct fsl_sai *)devid;
@@ -251,12 +262,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
+		sai->is_slave_mode = true;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+		sai->is_slave_mode = true;
 		break;
 	default:
 		return -EINVAL;
@@ -288,6 +301,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 	return ret;
 }
 
+static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai);
+	unsigned long clk_rate;
+	u32 savediv = 0, ratio, savesub = freq;
+	u32 id;
+	int ret = 0;
+
+	/* Don't apply to slave mode */
+	if (sai->is_slave_mode)
+		return 0;
+
+	for (id = 0; id < FSL_SAI_MCLK_MAX; id++) {
+		clk_rate = clk_get_rate(sai->mclk_clk[id]);
+		if (!clk_rate)
+			continue;
+
+		ratio = clk_rate / freq;
+
+		ret = clk_rate - ratio * freq;
+
+		/*
+		 * Drop the source that can not be
+		 * divided into the required rate.
+		 */
+		if (ret != 0 && clk_rate / ret < 1000)
+			continue;
+
+		dev_dbg(dai->dev,
+			"ratio %d for freq %dHz based on clock %ldHz\n",
+			ratio, freq, clk_rate);
+
+		if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512)
+			ratio /= 2;
+		else
+			continue;
+
+		if (ret < savesub) {
+			savediv = ratio;
+			sai->mclk_id[tx] = id;
+			savesub = ret;
+		}
+
+		if (ret == 0)
+			break;
+	}
+
+	if (savediv == 0) {
+		dev_err(dai->dev, "failed to derive required %cx rate: %d\n",
+				tx ? 'T' : 'R', freq);
+		return -EINVAL;
+	}
+
+	if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) {
+		regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+				   FSL_SAI_CR2_MSEL_MASK,
+				   FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
+		regmap_update_bits(sai->regmap, FSL_SAI_RCR2,
+				   FSL_SAI_CR2_DIV_MASK, savediv - 1);
+	} else {
+		regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+				   FSL_SAI_CR2_MSEL_MASK,
+				   FSL_SAI_CR2_MSEL(sai->mclk_id[tx]));
+		regmap_update_bits(sai->regmap, FSL_SAI_TCR2,
+				   FSL_SAI_CR2_DIV_MASK, savediv - 1);
+	}
+
+	dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n",
+			sai->mclk_id[tx], savediv, savesub);
+
+	return 0;
+}
+
 static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params,
 		struct snd_soc_dai *cpu_dai)
@@ -297,6 +383,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	unsigned int channels = params_channels(params);
 	u32 word_width = snd_pcm_format_width(params_format(params));
 	u32 val_cr4 = 0, val_cr5 = 0;
+	int ret;
+
+	if (!sai->is_slave_mode) {
+		ret = fsl_sai_set_bclk(cpu_dai, tx,
+			2 * word_width * params_rate(params));
+		if (ret)
+			return ret;
+
+		/* Do not enable the clock if it is already enabled */
+		if (!(sai->mclk_streams & BIT(substream->stream))) {
+			ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]);
+			if (ret)
+				return ret;
+
+			sai->mclk_streams |= BIT(substream->stream);
+		}
+
+	}
 
 	if (!sai->is_dsp_mode)
 		val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
@@ -322,6 +426,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int fsl_sai_hw_free(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+	if (!sai->is_slave_mode &&
+			sai->mclk_streams & BIT(substream->stream)) {
+		clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]);
+		sai->mclk_streams &= ~BIT(substream->stream);
+	}
+
+	return 0;
+}
+
+
 static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
 		struct snd_soc_dai *cpu_dai)
 {
@@ -410,7 +530,10 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE,
 			   FSL_SAI_CR3_TRCE);
 
-	return 0;
+	ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints);
+
+	return ret;
 }
 
 static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
@@ -428,6 +551,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt,
 	.hw_params	= fsl_sai_hw_params,
+	.hw_free	= fsl_sai_hw_free,
 	.trigger	= fsl_sai_trigger,
 	.startup	= fsl_sai_startup,
 	.shutdown	= fsl_sai_shutdown,
@@ -463,14 +587,18 @@ static struct snd_soc_dai_driver fsl_sai_dai = {
 		.stream_name = "CPU-Playback",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_96000,
+		.rate_min = 8000,
+		.rate_max = 192000,
+		.rates = SNDRV_PCM_RATE_KNOT,
 		.formats = FSL_SAI_FORMATS,
 	},
 	.capture = {
 		.stream_name = "CPU-Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_96000,
+		.rate_min = 8000,
+		.rate_max = 192000,
+		.rates = SNDRV_PCM_RATE_KNOT,
 		.formats = FSL_SAI_FORMATS,
 	},
 	.ops = &fsl_sai_pcm_dai_ops,
@@ -600,8 +728,9 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		sai->bus_clk = NULL;
 	}
 
-	for (i = 0; i < FSL_SAI_MCLK_MAX; i++) {
-		sprintf(tmp, "mclk%d", i + 1);
+	sai->mclk_clk[0] = sai->bus_clk;
+	for (i = 1; i < FSL_SAI_MCLK_MAX; i++) {
+		sprintf(tmp, "mclk%d", i);
 		sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp);
 		if (IS_ERR(sai->mclk_clk[i])) {
 			dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n",
@@ -664,8 +793,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (sai->sai_on_imx)
 		return imx_pcm_dma_init(pdev);
 	else
-		return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL,
-				SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+		return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
 }
 
 static const struct of_device_id fsl_sai_ids[] = {
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 34667209b607..066280953c85 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -72,13 +72,15 @@
 
 /* SAI Transmit and Recieve Configuration 2 Register */
 #define FSL_SAI_CR2_SYNC	BIT(30)
-#define FSL_SAI_CR2_MSEL_MASK	(0xff << 26)
+#define FSL_SAI_CR2_MSEL_MASK	(0x3 << 26)
 #define FSL_SAI_CR2_MSEL_BUS	0
 #define FSL_SAI_CR2_MSEL_MCLK1	BIT(26)
 #define FSL_SAI_CR2_MSEL_MCLK2	BIT(27)
 #define FSL_SAI_CR2_MSEL_MCLK3	(BIT(26) | BIT(27))
+#define FSL_SAI_CR2_MSEL(ID)	((ID) << 26)
 #define FSL_SAI_CR2_BCP		BIT(25)
 #define FSL_SAI_CR2_BCD_MSTR	BIT(24)
+#define FSL_SAI_CR2_DIV_MASK	0xff
 
 /* SAI Transmit and Recieve Configuration 3 Register */
 #define FSL_SAI_CR3_TRCE	BIT(16)
@@ -120,7 +122,7 @@
 #define FSL_SAI_CLK_MAST2	2
 #define FSL_SAI_CLK_MAST3	3
 
-#define FSL_SAI_MCLK_MAX	3
+#define FSL_SAI_MCLK_MAX	4
 
 /* SAI data transfer numbers per DMA request */
 #define FSL_SAI_MAXBURST_TX 6
@@ -132,11 +134,14 @@ struct fsl_sai {
 	struct clk *bus_clk;
 	struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
 
+	bool is_slave_mode;
 	bool is_lsb_first;
 	bool is_dsp_mode;
 	bool sai_on_imx;
 	bool synchronous[2];
 
+	unsigned int mclk_id[2];
+	unsigned int mclk_streams;
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 91eb3aef7f02..8e932219cb3a 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	if (clk != STC_TXCLK_SPDIF_ROOT)
 		goto clk_set_bypass;
 
-	/*
-	 * The S/PDIF block needs a clock of 64 * fs * txclk_df.
-	 * So request 64 * fs * (txclk_df + 1) to get rounded.
-	 */
-	ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1));
+	/* The S/PDIF block needs a clock of 64 * fs * txclk_df */
+	ret = clk_set_rate(spdif_priv->txclk[rate],
+			   64 * sample_rate * txclk_df);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to set tx clock rate\n");
 		return ret;
@@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv,
 
 	for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) {
 		for (txclk_df = 1; txclk_df <= 128; txclk_df++) {
-			rate_ideal = rate[index] * (txclk_df + 1) * 64;
+			rate_ideal = rate[index] * txclk_df * 64;
 			if (round)
 				rate_actual = clk_round_rate(clk, rate_ideal);
 			else
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0d48804218b1..c7647e066cfd 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	void __iomem *iomem;
 	char name[64];
 
-	/* SSIs that are not connected on the board should have a
-	 *      status = "disabled"
-	 * property in their device tree nodes.
-	 */
-	if (!of_device_is_available(np))
-		return -ENODEV;
-
 	of_id = of_match_device(fsl_ssi_ids, &pdev->dev);
 	if (!of_id || !of_id->data)
 		return -EINVAL;
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index d9050d946ae7..fc57da341d61 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -184,7 +184,7 @@ static enum imx_audmux_type {
 	IMX31_AUDMUX,
 } audmux_type;
 
-static struct platform_device_id imx_audmux_ids[] = {
+static const struct platform_device_id imx_audmux_ids[] = {
 	{
 		.name = "imx21-audmux",
 		.driver_data = IMX21_AUDMUX,
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 9e6493d4e7ff..bb0459018b45 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -45,11 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
-	if (ret)
-		return ret;
-
-	return 0;
+	return snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16);
 }
 
 static struct snd_soc_ops imx_mc13783_hifi_ops = {
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index cd146d4fa805..b38b98cae855 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -190,7 +190,7 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "audmux internal port setup failed\n");
 		return ret;
 	}
-	imx_audmux_v2_configure_port(ext_port,
+	ret = imx_audmux_v2_configure_port(ext_port,
 			IMX_AUDMUX_V2_PTCR_SYN,
 			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
 	if (ret) {