Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/ak4624',...

Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/ak4624', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next

sound/soc/atmel/atmel_ssc_dai.c

@@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		rcmr =	  SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
 			| SSC_BF(RCMR_STTDLY, 1)
 			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
 			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
 			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
 
@@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
 			| SSC_BF(RCMR_STTDLY, START_DELAY)
 			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
-			| SSC_BF(RCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
 			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
 			| SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
 					   SSC_CKS_PIN : SSC_CKS_CLOCK);

sound/soc/au1x/dbdma2.c

@@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	stype = substream->stream;
 	pcd = to_dmadata(substream);
 
-	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
-	    "runtime->min_align %d\n",
+	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu "
+	    "runtime->min_align %lu\n",
 		(unsigned long)runtime->dma_area,
 		(unsigned long)runtime->dma_addr, runtime->dma_bytes,
 		runtime->min_align);

sound/soc/codecs/ak4642.c

@@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev)
 
 	of_property_read_string(np, "clock-output-names", &clk_name);
 
-	clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
-				      (parent_clk_name) ? 0 : CLK_IS_ROOT,
-				      rate);
+	clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate);
 	if (!IS_ERR(clk))
 		of_clk_add_provider(np, of_clk_src_simple_get, clk);
 

sound/soc/codecs/arizona.c

@@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 
 	switch (arizona->type) {
 	case WM8997:
+	case CS47L24:
+	case WM1831:
 		break;
 	default:
 		ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
@@ -1134,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
 		   int event)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
-	unsigned int mask = 0x3 << w->shift;
 	unsigned int val;
 
 	switch (event) {
@@ -1148,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w,
 		return 0;
 	}
 
-	snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val);
+	snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val);
 
 	return 0;
 }
@@ -2047,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 			init_ratio, Fref, refdiv);
 
 	while (div <= ARIZONA_FLL_MAX_REFDIV) {
-		for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+		/* start from init_ratio because this may already give a
+		 * fractional N.K
+		 */
+		for (ratio = init_ratio; ratio > 0; ratio--) {
+			if (target % (ratio * Fref)) {
+				cfg->refdiv = refdiv;
+				cfg->fratio = ratio - 1;
+				arizona_fll_dbg(fll,
+					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
+					Fref, refdiv, div, ratio);
+				return ratio;
+			}
+		}
+
+		for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO;
 		     ratio++) {
 			if ((ARIZONA_FLL_VCO_CORNER / 2) /
 			    (fll->vco_mult * ratio) < Fref) {
@@ -2073,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
 			}
 		}
 
-		for (ratio = init_ratio - 1; ratio > 0; ratio--) {
-			if (target % (ratio * Fref)) {
-				cfg->refdiv = refdiv;
-				cfg->fratio = ratio - 1;
-				arizona_fll_dbg(fll,
-					"pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n",
-					Fref, refdiv, div, ratio);
-				return ratio;
-			}
-		}
-
 		div *= 2;
 		Fref /= 2;
 		refdiv++;

sound/soc/codecs/cs47l24.c

@@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = {
 	{ "IN2L PGA", NULL, "IN2L" },
 	{ "IN2R PGA", NULL, "IN2R" },
 
+	{ "Audio Trace DSP", NULL, "DSP2" },
+	{ "Audio Trace DSP", NULL, "SYSCLK" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 
@@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = {
 			.formats = CS47L24_FORMATS,
 		},
 	},
+	{
+		.name = "cs47l24-cpu-trace",
+		.capture = {
+			.stream_name = "Audio Trace CPU",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+		.compress_new = snd_soc_new_compress,
+	},
+	{
+		.name = "cs47l24-dsp-trace",
+		.capture = {
+			.stream_name = "Audio Trace DSP",
+			.channels_min = 1,
+			.channels_max = 6,
+			.rates = CS47L24_RATES,
+			.formats = CS47L24_FORMATS,
+		},
+	},
 };
 
 static int cs47l24_open(struct snd_compr_stream *stream)
@@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream)
 
 	if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) {
 		n_adsp = 2;
+	} else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) {
+		n_adsp = 1;
 	} else {
 		dev_err(arizona->dev,
 			"No suitable compressed stream for DAI '%s'\n",
@@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data)
 {
 	struct cs47l24_priv *priv = data;
 	struct arizona *arizona = priv->core.arizona;
-	int ret;
+	int serviced = 0;
+	int i, ret;
 
-	ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]);
-	if (ret == -ENODEV) {
+	for (i = 1; i <= 2; ++i) {
+		ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]);
+		if (ret != -ENODEV)
+			serviced++;
+	}
+
+	if (!serviced) {
 		dev_err(arizona->dev, "Spurious compressed data IRQ\n");
 		return IRQ_NONE;
 	}
@@ -1160,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = {
 static struct snd_soc_platform_driver cs47l24_compr_platform = {
 	.compr_ops = &cs47l24_compr_ops,
 };
+
 static int cs47l24_probe(struct platform_device *pdev)
 {
 	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
@@ -1228,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "Failed to register platform: %d\n", ret);
 		return ret;
 	}
+
 	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24,
 				      cs47l24_dai, ARRAY_SIZE(cs47l24_dai));
-
 	if (ret < 0) {
 		dev_err(&pdev->dev, "Failed to register codec: %d\n", ret);
 		snd_soc_unregister_platform(&pdev->dev);
@@ -1241,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev)
 
 static int cs47l24_remove(struct platform_device *pdev)
 {
+	struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	wm_adsp2_remove(&cs47l24->core.adsp[1]);
+	wm_adsp2_remove(&cs47l24->core.adsp[2]);
+
 	return 0;
 }
 

sound/soc/codecs/wm5102.c

@@ -2098,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev)
 
 static int wm5102_remove(struct platform_device *pdev)
 {
+	struct wm5102_priv *wm5102 = platform_get_drvdata(pdev);
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	wm_adsp2_remove(&wm5102->core.adsp[0]);
+
 	return 0;
 }
 

sound/soc/codecs/wm5110.c

@@ -2437,10 +2437,16 @@ static int wm5110_probe(struct platform_device *pdev)
 
 static int wm5110_remove(struct platform_device *pdev)
 {
+	struct wm5110_priv *wm5110 = platform_get_drvdata(pdev);
+	int i;
+
 	snd_soc_unregister_platform(&pdev->dev);
 	snd_soc_unregister_codec(&pdev->dev);
 	pm_runtime_disable(&pdev->dev);
 
+	for (i = 0; i < WM5110_NUM_ADSP; i++)
+		wm_adsp2_remove(&wm5110->core.adsp[i]);
+
 	return 0;
 }
 

sound/soc/codecs/wm_adsp.h

@@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[];
 
 int wm_adsp1_init(struct wm_adsp *dsp);
 int wm_adsp2_init(struct wm_adsp *dsp);
+void wm_adsp2_remove(struct wm_adsp *dsp);
 int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec);
 int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec);
 int wm_adsp1_event(struct snd_soc_dapm_widget *w,