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authorMark Brown <broonie@opensource.wolfsonmicro.com>2012-12-10 00:22:07 +0900
committerMark Brown <broonie@opensource.wolfsonmicro.com>2012-12-10 00:22:07 +0900
commit1bd202e4c7745459aca6616cd127b2d2bbd29901 (patch)
tree94ab16eaf7ee76c4d495780df8d72c8e765a61cd /sound
parent57769541b4bb696bf69c3350ca09187e04ebe7d4 (diff)
parent1b3bc060fb008ddd75fe60c876c24784a517c10c (diff)
Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/davinci/davinci-evm.c5
-rw-r--r--sound/soc/davinci/davinci-mcasp.c152
-rw-r--r--sound/soc/davinci/davinci-mcasp.h15
-rw-r--r--sound/soc/davinci/davinci-pcm.c53
-rw-r--r--sound/soc/davinci/davinci-pcm.h2
5 files changed, 141 insertions, 86 deletions
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 6fac5af1329..d55e6477bff 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -71,6 +71,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
if (ret < 0)
return ret;
+ /* set the CPU system clock */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
return 0;
}
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 714e51e5be5..55e2bf652be 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -199,6 +199,7 @@
#define ACLKXE BIT(5)
#define TX_ASYNC BIT(6)
#define ACLKXPOL BIT(7)
+#define ACLKXDIV_MASK 0x1f
/*
* DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
@@ -207,6 +208,7 @@
#define ACLKRE BIT(5)
#define RX_ASYNC BIT(6)
#define ACLKRPOL BIT(7)
+#define ACLKRDIV_MASK 0x1f
/*
* DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
@@ -215,6 +217,7 @@
#define AHCLKXDIV(val) (val)
#define AHCLKXPOL BIT(14)
#define AHCLKXE BIT(15)
+#define AHCLKXDIV_MASK 0xfff
/*
* DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
@@ -223,6 +226,7 @@
#define AHCLKRDIV(val) (val)
#define AHCLKRPOL BIT(14)
#define AHCLKRE BIT(15)
+#define AHCLKRDIV_MASK 0xfff
/*
* DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits
@@ -473,6 +477,23 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
void __iomem *base = dev->base;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_B:
+ case SND_SOC_DAIFMT_AC97:
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+ break;
+ default:
+ /* configure a full-word SYNC pulse (LRCLK) */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+
+ /* make 1st data bit occur one ACLK cycle after the frame sync */
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, FSXDLY(1));
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, FSRDLY(1));
+ break;
+ }
+
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
/* codec is clock and frame slave */
@@ -482,8 +503,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
- mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG,
- ACLKX | AHCLKX | AFSX);
+ mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX);
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
@@ -554,59 +574,75 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
return 0;
}
-static int davinci_config_channel_size(struct davinci_audio_dev *dev,
- int channel_size)
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
{
- u32 fmt = 0;
- u32 mask, rotate;
-
- switch (channel_size) {
- case DAVINCI_AUDIO_WORD_8:
- fmt = 0x03;
- rotate = 6;
- mask = 0x000000ff;
- break;
+ struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
- case DAVINCI_AUDIO_WORD_12:
- fmt = 0x05;
- rotate = 5;
- mask = 0x00000fff;
+ switch (div_id) {
+ case 0: /* MCLK divider */
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG,
+ AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG,
+ AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
break;
- case DAVINCI_AUDIO_WORD_16:
- fmt = 0x07;
- rotate = 4;
- mask = 0x0000ffff;
+ case 1: /* BCLK divider */
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG,
+ ACLKXDIV(div - 1), ACLKXDIV_MASK);
+ mcasp_mod_bits(dev->base + DAVINCI_MCASP_ACLKRCTL_REG,
+ ACLKRDIV(div - 1), ACLKRDIV_MASK);
break;
- case DAVINCI_AUDIO_WORD_20:
- fmt = 0x09;
- rotate = 3;
- mask = 0x000fffff;
+ case 2: /* BCLK/LRCLK ratio */
+ dev->bclk_lrclk_ratio = div;
break;
- case DAVINCI_AUDIO_WORD_24:
- fmt = 0x0B;
- rotate = 2;
- mask = 0x00ffffff;
- break;
+ default:
+ return -EINVAL;
+ }
- case DAVINCI_AUDIO_WORD_28:
- fmt = 0x0D;
- rotate = 1;
- mask = 0x0fffffff;
- break;
+ return 0;
+}
- case DAVINCI_AUDIO_WORD_32:
- fmt = 0x0F;
- rotate = 0;
- mask = 0xffffffff;
- break;
+static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
- default:
- return -EINVAL;
+ if (dir == SND_SOC_CLOCK_OUT) {
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+ mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
+ } else {
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+ mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AHCLKX);
}
+ return 0;
+}
+
+static int davinci_config_channel_size(struct davinci_audio_dev *dev,
+ int word_length)
+{
+ u32 fmt;
+ u32 rotate = (32 - word_length) / 4;
+ u32 mask = (1ULL << word_length) - 1;
+
+ /*
+ * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
+ * callback, take it into account here. That allows us to for example
+ * send 32 bits per channel to the codec, while only 16 of them carry
+ * audio payload.
+ * The clock ratio is given for a full period of data (both left and
+ * right channels), so it has to be divided by 2.
+ */
+ if (dev->bclk_lrclk_ratio)
+ word_length = dev->bclk_lrclk_ratio / 2;
+
+ /* mapping of the XSSZ bit-field as described in the datasheet */
+ fmt = (word_length >> 1) - 1;
+
mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
RXSSZ(fmt), RXSSZ(0x0F));
mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
@@ -709,8 +745,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* bit stream is MSB first with no delay */
/* DSP_B mode */
- mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG,
- AHCLKXE);
mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask);
mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD);
@@ -720,14 +754,10 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
else
printk(KERN_ERR "playback tdm slot %d not supported\n",
dev->tdm_slots);
-
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
} else {
/* bit stream is MSB first with no delay */
/* DSP_B mode */
mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD);
- mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG,
- AHCLKRE);
mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask);
if ((dev->tdm_slots >= 2) && (dev->tdm_slots <= 32))
@@ -736,8 +766,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
else
printk(KERN_ERR "capture tdm slot %d not supported\n",
dev->tdm_slots);
-
- mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
}
}
@@ -800,19 +828,27 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_U8:
case SNDRV_PCM_FORMAT_S8:
dma_params->data_type = 1;
- word_length = DAVINCI_AUDIO_WORD_8;
+ word_length = 8;
break;
case SNDRV_PCM_FORMAT_U16_LE:
case SNDRV_PCM_FORMAT_S16_LE:
dma_params->data_type = 2;
- word_length = DAVINCI_AUDIO_WORD_16;
+ word_length = 16;
+ break;
+
+ case SNDRV_PCM_FORMAT_U24_3LE:
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ dma_params->data_type = 3;
+ word_length = 24;
break;
+ case SNDRV_PCM_FORMAT_U24_LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
- word_length = DAVINCI_AUDIO_WORD_32;
+ word_length = 32;
break;
default:
@@ -880,13 +916,18 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
.trigger = davinci_mcasp_trigger,
.hw_params = davinci_mcasp_hw_params,
.set_fmt = davinci_mcasp_set_dai_fmt,
-
+ .set_clkdiv = davinci_mcasp_set_clkdiv,
+ .set_sysclk = davinci_mcasp_set_sysclk,
};
#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
SNDRV_PCM_FMTBIT_U8 | \
SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_U16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_U24_LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | \
+ SNDRV_PCM_FMTBIT_U24_3LE | \
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_U32_LE)
@@ -1089,7 +1130,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dev->tdm_slots = pdata->tdm_slots;
dev->num_serializer = pdata->num_serializer;
dev->serial_dir = pdata->serial_dir;
- dev->codec_fmt = pdata->codec_fmt;
dev->version = pdata->version;
dev->txnumevt = pdata->txnumevt;
dev->rxnumevt = pdata->rxnumevt;
@@ -1098,6 +1138,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
+ dma_data->sram_pool = pdata->sram_pool;
dma_data->sram_size = pdata->sram_size_playback;
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
mem->start);
@@ -1115,6 +1156,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
+ dma_data->sram_pool = pdata->sram_pool;
dma_data->sram_size = pdata->sram_size_capture;
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
mem->start);
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 0de9ed6ce03..0edd3b5a37f 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -23,26 +23,14 @@
#include "davinci-pcm.h"
-#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000
+#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000
#define DAVINCI_MCASP_I2S_DAI 0
#define DAVINCI_MCASP_DIT_DAI 1
-enum {
- DAVINCI_AUDIO_WORD_8 = 0,
- DAVINCI_AUDIO_WORD_12,
- DAVINCI_AUDIO_WORD_16,
- DAVINCI_AUDIO_WORD_20,
- DAVINCI_AUDIO_WORD_24,
- DAVINCI_AUDIO_WORD_32,
- DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */
-};
-
struct davinci_audio_dev {
struct davinci_pcm_dma_params dma_params[2];
void __iomem *base;
- int sample_rate;
struct device *dev;
- unsigned int codec_fmt;
/* McASP specific data */
int tdm_slots;
@@ -50,6 +38,7 @@ struct davinci_audio_dev {
u8 num_serializer;
u8 *serial_dir;
u8 version;
+ u8 bclk_lrclk_ratio;
/* McASP FIFO related */
u8 txnumevt;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 93ea3bf567e..afab81f844a 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -16,6 +16,7 @@
#include <linux/slab.h>
#include <linux/dma-mapping.h>
#include <linux/kernel.h>
+#include <linux/genalloc.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -23,7 +24,6 @@
#include <sound/soc.h>
#include <asm/dma.h>
-#include <mach/sram.h>
#include "davinci-pcm.h"
@@ -67,13 +67,9 @@ static struct snd_pcm_hardware pcm_hardware_playback = {
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME|
SNDRV_PCM_INFO_BATCH),
.formats = DAVINCI_PCM_FMTBITS,
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
+ .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT,
.rate_min = 8000,
- .rate_max = 96000,
+ .rate_max = 192000,
.channels_min = 2,
.channels_max = 384,
.buffer_bytes_max = 128 * 1024,
@@ -90,13 +86,9 @@ static struct snd_pcm_hardware pcm_hardware_capture = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_BATCH),
.formats = DAVINCI_PCM_FMTBITS,
- .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_KNOT),
+ .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT,
.rate_min = 8000,
- .rate_max = 96000,
+ .rate_max = 192000,
.channels_min = 2,
.channels_max = 384,
.buffer_bytes_max = 128 * 1024,
@@ -259,7 +251,9 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
}
}
-static int allocate_sram(struct snd_pcm_substream *substream, unsigned size,
+#ifdef CONFIG_GENERIC_ALLOCATOR
+static int allocate_sram(struct snd_pcm_substream *substream,
+ struct gen_pool *sram_pool, unsigned size,
struct snd_pcm_hardware *ppcm)
{
struct snd_dma_buffer *buf = &substream->dma_buffer;
@@ -271,9 +265,10 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size,
return 0;
ppcm->period_bytes_max = size;
- iram_virt = sram_alloc(size, &iram_phys);
+ iram_virt = (void *)gen_pool_alloc(sram_pool, size);
if (!iram_virt)
goto exit1;
+ iram_phys = gen_pool_virt_to_phys(sram_pool, (unsigned)iram_virt);
iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL);
if (!iram_dma)
goto exit2;
@@ -285,11 +280,33 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size,
return 0;
exit2:
if (iram_virt)
- sram_free(iram_virt, size);
+ gen_pool_free(sram_pool, (unsigned)iram_virt, size);
exit1:
return -ENOMEM;
}
+static void davinci_free_sram(struct snd_pcm_substream *substream,
+ struct snd_dma_buffer *iram_dma)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct gen_pool *sram_pool = prtd->params->sram_pool;
+
+ gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes);
+}
+#else
+static int allocate_sram(struct snd_pcm_substream *substream,
+ struct gen_pool *sram_pool, unsigned size,
+ struct snd_pcm_hardware *ppcm)
+{
+ return 0;
+}
+
+static void davinci_free_sram(struct snd_pcm_substream *substream,
+ struct snd_dma_buffer *iram_dma)
+{
+}
+#endif
+
/*
* Only used with ping/pong.
* This is called after runtime->dma_addr, period_bytes and data_type are valid
@@ -676,7 +693,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
&pcm_hardware_playback : &pcm_hardware_capture;
- allocate_sram(substream, params->sram_size, ppcm);
+ allocate_sram(substream, params->sram_pool, params->sram_size, ppcm);
snd_soc_set_runtime_hwparams(substream, ppcm);
/* ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
@@ -819,7 +836,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
buf->area = NULL;
iram_dma = buf->private_data;
if (iram_dma) {
- sram_free(iram_dma->area, iram_dma->bytes);
+ davinci_free_sram(substream, iram_dma);
kfree(iram_dma);
}
}
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index fc4d01cdd8c..b6ef7039dd0 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -12,6 +12,7 @@
#ifndef _DAVINCI_PCM_H
#define _DAVINCI_PCM_H
+#include <linux/genalloc.h>
#include <linux/platform_data/davinci_asp.h>
#include <mach/edma.h>
@@ -20,6 +21,7 @@ struct davinci_pcm_dma_params {
unsigned short acnt;
dma_addr_t dma_addr; /* device physical address for DMA */
unsigned sram_size;
+ struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */
enum dma_event_q asp_chan_q; /* event queue number for ASP channel */
enum dma_event_q ram_chan_q; /* event queue number for RAM channel */
unsigned char data_type; /* xfer data type */