Merge tag 'asoc-v3.19-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.19

There's a lot more fixes here than I'd like since I've been lax in
sending things throughout the release cycle but there's only two in
generic code and they've had quite some time to cook in -next so
hopefully on balance are OK.

The two generic fixes are a fix for crashes on capture DAIs in the
compress code and a fix for error handling on probe failures which would
have been harmless in the past but now oopses with the new dynamic probe
code.

16 files changed, 93 insertions, 35 deletions

diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7752860f7230..4c23381727a1 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev)
 	if (ret)
 		goto err_clk_disable;
 
+	return 0;
+
 err_clk_disable:
 	clk_disable_unprepare(i2s->clk);
 	return ret;
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index e5f2fb884bf3..30c673cdc12e 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
 static const char * const pcm512x_dsp_program_texts[] = {
 	"FIR interpolation with de-emphasis",
 	"Low latency IIR with de-emphasis",
-	"Fixed process flow",
 	"High attenuation with de-emphasis",
+	"Fixed process flow",
 	"Ringing-less low latency FIR",
 };
 
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 2cd4fe463102..1d1c7f8a9af2 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream,
 		RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
 	dev_dbg(codec->dev, "format val = 0x%x\n", val);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
-	else
-		snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+	snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+	snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index c0fbe1881439..918ada9738b0 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2083,10 +2083,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_PRE_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2);
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0);
 		break;
+
 	default:
 		return 0;
 	}
@@ -2101,10 +2105,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
 
 	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
+	case SND_SOC_DAPM_PRE_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2);
+		break;
+
+	case SND_SOC_DAPM_POST_PMU:
 		regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0);
 		break;
+
 	default:
 		return 0;
 	}
@@ -2212,9 +2220,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,
 
 static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
-		0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU),
+		0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
+		SND_SOC_DAPM_POST_PMU),
 	SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT,
-		0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU),
+		0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU |
+		SND_SOC_DAPM_POST_PMU),
 
 	/* Input Side */
 	/* micbias */
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
index 1d1205702d23..9f2dced046de 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
 	struct ts3a227e *ts3a227e;
 	struct device *dev = &i2c->dev;
 	int ret;
+	unsigned int acc_reg;
 
 	ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
 	if (ts3a227e == NULL)
@@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
 			   INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
 			   ADC_COMPLETE_INT_DISABLE);
 
+	/* Read jack status because chip might not trigger interrupt at boot. */
+	regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+	ts3a227e_new_jack_state(ts3a227e, acc_reg);
+	ts3a227e_jack_report(ts3a227e);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4d2d2b1380d5..75b87c5c0f04 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 	{ "Right Capture PGA", NULL, "Right Capture Mux" },
 	{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },
 
-	{ "AIFOUTL", "Left",  "ADCL" },
-	{ "AIFOUTL", "Right", "ADCR" },
-	{ "AIFOUTR", "Left",  "ADCL" },
-	{ "AIFOUTR", "Right", "ADCR" },
+	{ "AIFOUTL Mux", "Left", "ADCL" },
+	{ "AIFOUTL Mux", "Right", "ADCR" },
+	{ "AIFOUTR Mux", "Left", "ADCL" },
+	{ "AIFOUTR Mux", "Right", "ADCR" },
+
+	{ "AIFOUTL", NULL, "AIFOUTL Mux" },
+	{ "AIFOUTR", NULL, "AIFOUTR Mux" },
 
 	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCL", NULL, "Left Capture PGA" },
@@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
 };
 
 static const struct snd_soc_dapm_route dac_intercon[] = {
-	{ "DACL", "Right", "AIFINR" },
-	{ "DACL", "Left",  "AIFINL" },
+	{ "DACL Mux", "Left", "AIFINL" },
+	{ "DACL Mux", "Right", "AIFINR" },
+
+	{ "DACR Mux", "Left", "AIFINL" },
+	{ "DACR Mux", "Right", "AIFINR" },
+
+	{ "DACL", NULL, "DACL Mux" },
 	{ "DACL", NULL, "CLK_DSP" },
 
-	{ "DACR", "Right", "AIFINR" },
-	{ "DACR", "Left",  "AIFINL" },
+	{ "DACR", NULL, "DACR Mux" },
 	{ "DACR", NULL, "CLK_DSP" },
 
 	{ "Charge pump", NULL, "SYSCLK" },
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 031a1ae71d94..a96eb497a379 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -556,7 +556,7 @@ static struct {
 	{ 22050, 2 },
 	{ 24000, 2 },
 	{ 16000, 3 },
-	{ 11250, 4 },
+	{ 11025, 4 },
 	{ 12000, 4 },
 	{  8000, 5 },
 };
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 91a550f4a10d..5e793bbb6b02 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -302,7 +302,7 @@
 #define ESAI_xCCR_xFP_MASK	(((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
 #define ESAI_xCCR_xFP(v)	((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
 #define ESAI_xCCR_xDC_SHIFT     9
-#define ESAI_xCCR_xDC_WIDTH	4
+#define ESAI_xCCR_xDC_WIDTH	5
 #define ESAI_xCCR_xDC_MASK	(((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
 #define ESAI_xCCR_xDC(v)	((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
 #define ESAI_xCCR_xPSR_SHIFT	8
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a65f17d57ffb..059496ed9ad7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1362,9 +1362,9 @@ 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", np->full_name);
-		return -ENXIO;
+		return ssi_private->irq;
 	}
 
 	/* Are the RX and the TX clocks locked? */
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 4caacb05a623..cd146d4fa805 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	if (ret)
 		goto clk_fail;
 	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
 	data->card.dai_link = &data->dai;
 	data->card.dapm_widgets = imx_wm8962_dapm_widgets;
 	data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fb9240fdc9b7..7fe3009b1c43 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node,
 }
 
 /* Decrease the reference count of the device nodes */
-static int asoc_simple_card_unref(struct platform_device *pdev)
+static int asoc_simple_card_unref(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
 
@@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
 		return ret;
 
 err:
-	asoc_simple_card_unref(pdev);
+	asoc_simple_card_unref(&priv->snd_card);
 	return ret;
 }
 
@@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 		snd_soc_jack_free_gpios(&simple_card_mic_jack, 1,
 					&simple_card_mic_jack_gpio);
 
-	return asoc_simple_card_unref(pdev);
+	return asoc_simple_card_unref(card);
 }
 
 static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
index ef2e8b5766a1..b3f9489794a6 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -706,6 +706,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 	struct list_head *block_list)
 {
 	struct sst_mem_block *block, *tmp;
+	struct sst_block_allocator ba_tmp = *ba;
 	u32 end = ba->offset + ba->size, block_end;
 	int err;
 
@@ -730,9 +731,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 		if (ba->offset >= block->offset && ba->offset < block_end) {
 
 			/* align ba to block boundary */
-			ba->size -= block_end - ba->offset;
-			ba->offset = block_end;
-			err = block_alloc_contiguous(dsp, ba, block_list);
+			ba_tmp.size -= block_end - ba->offset;
+			ba_tmp.offset = block_end;
+			err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
 			if (err < 0)
 				return -ENOMEM;
 
@@ -767,10 +768,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
 			list_move(&block->list, &dsp->used_block_list);
 			list_add(&block->module_list, block_list);
 			/* align ba to block boundary */
-			ba->size -= block_end - ba->offset;
-			ba->offset = block_end;
+			ba_tmp.size -= block_end - ba->offset;
+			ba_tmp.offset = block_end;
 
-			err = block_alloc_contiguous(dsp, ba, block_list);
+			err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
 			if (err < 0)
 				return -ENOMEM;
 
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 3f8c48231364..5bf14040c24a 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1228,6 +1228,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	struct sst_dsp *sst = hsw->dsp;
 	unsigned long flags;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n");
+		return 0;
+	}
+
 	/* dont free DSP streams that are not commited */
 	if (!stream->commited)
 		goto out;
@@ -1415,6 +1420,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 	u32 header;
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n");
+		return 0;
+	}
+
+	if (stream->commited) {
+		dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream alloc", stream->host_id);
 
 	header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
@@ -1519,6 +1534,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 {
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream pause", stream->reply.stream_hw_id);
 
 	ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
@@ -1535,6 +1555,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
 {
 	int ret;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n");
+		return 0;
+	}
+
 	trace_ipc_request("stream resume", stream->reply.stream_hw_id);
 
 	ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
@@ -1550,6 +1575,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
 {
 	int ret, tries = 10;
 
+	if (!stream) {
+		dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n");
+		return 0;
+	}
+
 	/* dont reset streams that are not commited */
 	if (!stream->commited)
 		return 0;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8b79cafab1e2..c7eb9dd67f60 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_CBM_CFS:
 		/* McBSP slave. FS clock as output */
 		regs->srgr2	|= FSGM;
-		regs->pcr0	|= FSXM;
+		regs->pcr0	|= FSXM | FSRM;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		/* McBSP slave */
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 13d8507333b8..dcc26eda0539 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -335,6 +335,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
 			    SNDRV_PCM_FMTBIT_S24_LE),
 	},
 	.ops = &rockchip_i2s_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static const struct snd_soc_component_driver rockchip_i2s_component = {
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 590a82f01d0b..025c38fbe3c0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 			rtd->dai_link->stream_name);
 
 		ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
-				1, 0, &be_pcm);
+				rtd->dai_link->dpcm_playback,
+				rtd->dai_link->dpcm_capture, &be_pcm);
 		if (ret < 0) {
 			dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n",
 				rtd->dai_link->name);
@@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 
 		rtd->pcm = be_pcm;
 		rtd->fe_compr = 1;
-		be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
-		be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+		if (rtd->dai_link->dpcm_playback)
+			be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+		else if (rtd->dai_link->dpcm_capture)
+			be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
 		memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
 	} else
 		memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));