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-rw-r--r--include/sound/soc-dai.h1
-rw-r--r--sound/soc/codecs/wm8974.c23
-rw-r--r--sound/soc/codecs/wm8993.c36
-rw-r--r--sound/soc/codecs/wm_hubs.c35
-rw-r--r--sound/soc/codecs/wm_hubs.h5
-rw-r--r--sound/soc/davinci/davinci-i2s.c2
-rw-r--r--sound/soc/davinci/davinci-mcasp.c17
-rw-r--r--sound/soc/davinci/davinci-pcm.c21
-rw-r--r--sound/soc/davinci/davinci-pcm.h1
9 files changed, 91 insertions, 50 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index e0c7fa7b1060..ca24e7f7a3f5 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -30,6 +30,7 @@ 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
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 93d66e30f109..eff29331235b 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -281,36 +281,38 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec)
}
struct pll_ {
- unsigned int pre_div:4; /* prescale - 1 */
+ unsigned int pre_div: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(unsigned int target, unsigned int source)
+static void pll_factors(struct pll_ *pll_div,
+ 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 >>= 1;
- pll_div.pre_div = 1;
+ source /= 2;
+ 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;
@@ -325,13 +327,14 @@ static void pll_factors(unsigned int target, unsigned int source)
/* 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) {
@@ -345,7 +348,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
return 0;
}
- pll_factors(freq_out*4, freq_in);
+ pll_factors(&pll_div, freq_out, 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);
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 6b32a2852603..dac397712147 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1572,33 +1572,15 @@ 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);
- 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);
-
+ 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);
+
ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret != 0)
goto err;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e542027eea89..810a563d0ebf 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -738,6 +738,41 @@ 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 <[email protected]>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index ec09cb6a2939..36d3fba1de8b 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -20,5 +20,10 @@ 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
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 4ae707048021..2ab809359c08 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -397,6 +397,8 @@ 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);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 5d1f98a4c978..50ad0519a8fa 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -714,16 +714,13 @@ 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 numevt;
+ u8 fifo_level;
davinci_hw_common_param(dev, substream->stream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- numevt = dev->txnumevt;
+ fifo_level = dev->txnumevt;
else
- numevt = dev->rxnumevt;
-
- if (!numevt)
- numevt = 1;
+ fifo_level = dev->rxnumevt;
if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
davinci_hw_dit_param(dev);
@@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- if (dev->version == MCASP_VERSION_2) {
- dma_params->data_type *= numevt;
- dma_params->acnt = 4 * numevt;
- } else
+ if (dev->version == MCASP_VERSION_2 && !fifo_level)
+ dma_params->acnt = 4;
+ else
dma_params->acnt = dma_params->data_type;
+ dma_params->fifo_level = fifo_level;
davinci_config_channel_size(dev, word_length);
return 0;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 359e99ec7244..1152d8ba8970 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -66,38 +66,53 @@ 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, 0);
- edma_set_dest_index(lch, dst_bidx, 0);
- edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC);
+
+ 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);
prtd->period++;
if (unlikely(prtd->period >= runtime->periods))
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 8746606efc89..c8b0d2baf05a 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -23,6 +23,7 @@ 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;
};
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