---

yaml
---
r: 170155
b: refs/heads/master
c: bb26276
h: refs/heads/master
i:
  170153: 5355b44
  170151: f2cdf27
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: aa983d9d63c38f596fb87754205da9b7a8d2f6fd
+refs/heads/master: bb26276744a80d066681836f4d49c70010b129d6

trunk/include/sound/soc-dai.h

@@ -30,7 +30,6 @@ struct snd_pcm_substream;
 #define SND_SOC_DAIFMT_DSP_A		3 /* L data MSB after FRM LRC */
 #define SND_SOC_DAIFMT_DSP_B		4 /* L data MSB during FRM LRC */
 #define SND_SOC_DAIFMT_AC97		5 /* AC97 */
-#define SND_SOC_DAIFMT_PDM		6 /* Pulse density modulation */
 
 /* left and right justified also known as MSB and LSB respectively */
 #define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J

trunk/sound/soc/codecs/wm8711.c

@@ -19,6 +19,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -632,9 +633,9 @@ static int __init wm8711_modinit(void)
 	}
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	ret = spi_register_driver(&wm8731_spi_driver);
+	ret = spi_register_driver(&wm8711_spi_driver);
 	if (ret != 0) {
-		printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n",
+		printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n",
 		       ret);
 	}
 #endif
@@ -648,7 +649,7 @@ static void __exit wm8711_exit(void)
 	i2c_del_driver(&wm8711_i2c_driver);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8731_spi_driver);
+	spi_unregister_driver(&wm8711_spi_driver);
 #endif
 }
 module_exit(wm8711_exit);

trunk/sound/soc/codecs/wm8974.c

@@ -281,38 +281,36 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec)
 }
 
 struct pll_ {
-	unsigned int pre_div:1;
+	unsigned int pre_div:4; /* prescale - 1 */
 	unsigned int n:4;
 	unsigned int k;
 };
 
+static struct pll_ pll_div;
+
 /* The size in bits of the pll divide multiplied by 10
  * to allow rounding later */
 #define FIXED_PLL_SIZE ((1 << 24) * 10)
 
-static void pll_factors(struct pll_ *pll_div,
-			unsigned int target, unsigned int source)
+static void pll_factors(unsigned int target, unsigned int source)
 {
 	unsigned long long Kpart;
 	unsigned int K, Ndiv, Nmod;
 
-	/* There is a fixed divide by 4 in the output path */
-	target *= 4;
-
 	Ndiv = target / source;
 	if (Ndiv < 6) {
-		source /= 2;
-		pll_div->pre_div = 1;
+		source >>= 1;
+		pll_div.pre_div = 1;
 		Ndiv = target / source;
 	} else
-		pll_div->pre_div = 0;
+		pll_div.pre_div = 0;
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
 			"WM8974 N value %u outwith recommended range!\n",
 			Ndiv);
 
-	pll_div->n = Ndiv;
+	pll_div.n = Ndiv;
 	Nmod = target % source;
 	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
 
@@ -327,14 +325,13 @@ static void pll_factors(struct pll_ *pll_div,
 	/* Move down to proper range now rounding is done */
 	K /= 10;
 
-	pll_div->k = K;
+	pll_div.k = K;
 }
 
 static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	struct pll_ pll_div;
 	u16 reg;
 
 	if (freq_in == 0 || freq_out == 0) {
@@ -348,7 +345,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		return 0;
 	}
 
-	pll_factors(&pll_div, freq_out, freq_in);
+	pll_factors(freq_out*4, freq_in);
 
 	snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n);
 	snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18);

trunk/sound/soc/codecs/wm8993.c

@@ -1572,15 +1572,33 @@ static int wm8993_i2c_probe(struct i2c_client *i2c,
 	/* Use automatic clock configuration */
 	snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0);
 
-	wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff,
-				      wm8993->pdata.lineout2_diff,
-				      wm8993->pdata.lineout1fb,
-				      wm8993->pdata.lineout2fb,
-				      wm8993->pdata.jd_scthr,
-				      wm8993->pdata.jd_thr,
-				      wm8993->pdata.micbias1_lvl,
-				      wm8993->pdata.micbias2_lvl);
-
+	if (!wm8993->pdata.lineout1_diff)
+		snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
+				    WM8993_LINEOUT1_MODE,
+				    WM8993_LINEOUT1_MODE);
+	if (!wm8993->pdata.lineout2_diff)
+		snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
+				    WM8993_LINEOUT2_MODE,
+				    WM8993_LINEOUT2_MODE);
+
+	if (wm8993->pdata.lineout1fb)
+		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+				    WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
+
+	if (wm8993->pdata.lineout2fb)
+		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
+
+	/* Apply the microphone bias/detection configuration - the
+	 * platform data is directly applicable to the register. */
+	snd_soc_update_bits(codec, WM8993_MICBIAS,
+			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
+			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
+			    wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT |
+			    wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT |
+			    wm8993->pdata.micbias1_lvl |
+			    wm8993->pdata.micbias1_lvl << 1);
+
 	ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	if (ret != 0)
 		goto err;

trunk/sound/soc/codecs/wm_hubs.c

@@ -738,41 +738,6 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes);
 
-int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
-				  int lineout1_diff, int lineout2_diff,
-				  int lineout1fb, int lineout2fb,
-				  int jd_scthr, int jd_thr, int micbias1_lvl,
-				  int micbias2_lvl)
-{
-	if (!lineout1_diff)
-		snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
-				    WM8993_LINEOUT1_MODE,
-				    WM8993_LINEOUT1_MODE);
-	if (!lineout2_diff)
-		snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
-				    WM8993_LINEOUT2_MODE,
-				    WM8993_LINEOUT2_MODE);
-
-	if (lineout1fb)
-		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
-				    WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
-
-	if (lineout2fb)
-		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
-				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
-
-	snd_soc_update_bits(codec, WM8993_MICBIAS,
-			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
-			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
-			    jd_scthr << WM8993_JD_SCTHR_SHIFT |
-			    jd_thr << WM8993_JD_THR_SHIFT |
-			    micbias1_lvl |
-			    micbias2_lvl << WM8993_MICB2_LVL_SHIFT);
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata);
-
 MODULE_DESCRIPTION("Shared support for Wolfson hubs products");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");

trunk/sound/soc/codecs/wm_hubs.h

@@ -20,10 +20,5 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
 extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int);
-extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *,
-					 int lineout1_diff, int lineout2_diff,
-					 int lineout1fb, int lineout2fb,
-					 int jd_scthr, int jd_thr,
-					 int micbias1_lvl, int micbias2_lvl);
 
 #endif

trunk/sound/soc/davinci/davinci-i2s.c

@@ -397,8 +397,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	dma_params->acnt  = dma_params->data_type;
-	dma_params->fifo_level = 0;
-
 	rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1);
 	xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1);
 

trunk/sound/soc/davinci/davinci-mcasp.c

@@ -714,13 +714,16 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	struct davinci_pcm_dma_params *dma_params =
 					&dev->dma_params[substream->stream];
 	int word_length;
-	u8 fifo_level;
+	u8 numevt;
 
 	davinci_hw_common_param(dev, substream->stream);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		fifo_level = dev->txnumevt;
+		numevt = dev->txnumevt;
 	else
-		fifo_level = dev->rxnumevt;
+		numevt = dev->rxnumevt;
+
+	if (!numevt)
+		numevt = 1;
 
 	if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
 		davinci_hw_dit_param(dev);
@@ -748,12 +751,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (dev->version == MCASP_VERSION_2 && !fifo_level)
-		dma_params->acnt = 4;
-	else
+	if (dev->version == MCASP_VERSION_2) {
+		dma_params->data_type *= numevt;
+		dma_params->acnt = 4 * numevt;
+	} else
 		dma_params->acnt = dma_params->data_type;
 
-	dma_params->fifo_level = fifo_level;
 	davinci_config_channel_size(dev, word_length);
 
 	return 0;

trunk/sound/soc/davinci/davinci-pcm.c

@@ -66,53 +66,38 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	dma_addr_t dma_pos;
 	dma_addr_t src, dst;
 	unsigned short src_bidx, dst_bidx;
-	unsigned short src_cidx, dst_cidx;
 	unsigned int data_type;
 	unsigned short acnt;
 	unsigned int count;
-	unsigned int fifo_level;
 
 	period_size = snd_pcm_lib_period_bytes(substream);
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
-	fifo_level = prtd->params->fifo_level;
 
 	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
 		"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
-	if (fifo_level)
-		count /= fifo_level;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		src = dma_pos;
 		dst = prtd->params->dma_addr;
 		src_bidx = data_type;
 		dst_bidx = 0;
-		src_cidx = data_type * fifo_level;
-		dst_cidx = 0;
 	} else {
 		src = prtd->params->dma_addr;
 		dst = dma_pos;
 		src_bidx = 0;
 		dst_bidx = data_type;
-		src_cidx = 0;
-		dst_cidx = data_type * fifo_level;
 	}
 
 	acnt = prtd->params->acnt;
 	edma_set_src(lch, src, INCR, W8BIT);
 	edma_set_dest(lch, dst, INCR, W8BIT);
-
-	edma_set_src_index(lch, src_bidx, src_cidx);
-	edma_set_dest_index(lch, dst_bidx, dst_cidx);
-
-	if (!fifo_level)
-		edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
-	else
-		edma_set_transfer_params(lch, acnt, fifo_level, count,
-							fifo_level, ABSYNC);
+	edma_set_src_index(lch, src_bidx, 0);
+	edma_set_dest_index(lch, dst_bidx, 0);
+	edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
 
 	prtd->period++;
 	if (unlikely(prtd->period >= runtime->periods))

trunk/sound/soc/davinci/davinci-pcm.h

@@ -23,7 +23,6 @@ struct davinci_pcm_dma_params {
 	enum dma_event_q eventq_no;	/* event queue number */
 	unsigned char data_type;	/* xfer data type */
 	unsigned char convert_mono_stereo;
-	unsigned int fifo_level;
 };