static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
int i;
struct snd_msnd *chip = snd_pcm_substream_chip(substream);
void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF;
chip->capture_sample_size = snd_pcm_format_width(params_format(params));
chip->capture_channels = params_channels(params);
chip->capture_sample_rate = params_rate(params);
for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) {
writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize);
writew(chip->capture_channels, pDAQ + DAQDS_wChannels);
writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate);
}
return 0;
}
static int jz_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct device *dev = dai->dev;
struct jz_pcm *jz_pcm = dev_get_drvdata(dai->dev);
int fmt_width = snd_pcm_format_width(params_format(params));
enum dma_slave_buswidth buswidth;
int trigger;
PCM_DEBUG_MSG("enter %s, substream = %s\n",
__func__,
(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? "playback" : "capture");
if (!((1 << params_format(params)) & JZ_PCM_FORMATS) ||
params_channels(params) != 1 ||
params_rate(params) != 8000) {
dev_err(dai->dev, "hw params not inval channel %d params %x\n",
params_channels(params), params_format(params));
return -EINVAL;
}
/* format 8 bit or 16 bit*/
if (fmt_width == 8)
buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE;
else
buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
jz_pcm->tx_dma_data.buswidth = buswidth;
jz_pcm->tx_dma_data.max_burst = (PCM_TFIFO_DEPTH * buswidth)/2;
trigger = PCM_TFIFO_DEPTH - (jz_pcm->tx_dma_data.max_burst/(int)buswidth);
__pcm_set_oss_sample_size(dev, fmt_width == 8 ? 0 : 1);
__pcm_set_transmit_trigger(dev, trigger);
snd_soc_dai_set_dma_data(dai, substream, (void *)&jz_pcm->tx_dma_data);
} else {
jz_pcm->rx_dma_data.buswidth = buswidth;
jz_pcm->rx_dma_data.max_burst = (PCM_RFIFO_DEPTH * buswidth)/2;
trigger = jz_pcm->rx_dma_data.max_burst/(int)buswidth;
__pcm_set_iss_sample_size(dev, fmt_width == 8 ? 0 : 1);
__pcm_set_receive_trigger(dev, trigger);
snd_soc_dai_set_dma_data(dai, substream, (void *)&jz_pcm->rx_dma_data);
}
return 0;
}
static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_dummy_pcm *dpcm = runtime->private_data;
unsigned int bps;
bps = runtime->rate * runtime->channels;
bps *= snd_pcm_format_width(runtime->format);
bps /= 8;
if (bps <= 0)
return -EINVAL;
dpcm->pcm_bps = bps;
dpcm->pcm_jiffie = bps / HZ;
dpcm->pcm_size = snd_pcm_lib_buffer_bytes(substream);
dpcm->pcm_count = snd_pcm_lib_period_bytes(substream);
dpcm->pcm_irq_pos = 0;
dpcm->pcm_buf_pos = 0;
return 0;
}
static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
u32 width = snd_pcm_format_width(params_format(params));
u32 channels = params_channels(params);
u32 bclk, mask, val, ret;
bclk = params_rate(params) * esai_priv->slot_width * 2;
ret = fsl_esai_set_bclk(dai, tx, bclk);
if (ret)
return ret;
/* Use Normal mode to support monaural audio */
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ?
ESAI_xCR_xMOD_NETWORK : 0);
regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx),
ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR);
mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
(tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
(tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0);
val = ESAI_xCR_xSWS(esai_priv->slot_width, width) | (tx ? ESAI_xCR_PADC : 0);
regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val);
/* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */
regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC,
ESAI_PRRC_PDC_MASK, ESAI_PRRC_PDC(ESAI_GPIO));
regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC,
ESAI_PCRC_PC_MASK, ESAI_PCRC_PC(ESAI_GPIO));
return 0;
}
/**
* fsl_ssi_prepare: prepare the SSI.
*
* Most of the SSI registers have been programmed in the startup function,
* but the word length must be programmed here. Unfortunately, programming
* the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
* cause a problem with supporting simultaneous playback and capture. If
* the SSI is already playing a stream, then that stream may be temporarily
* stopped when you start capture.
*
* Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
* clock master.
*/
static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
if (substream == ssi_private->first_stream) {
u32 wl;
/* The SSI should always be disabled at this points (SSIEN=0) */
wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
/* In synchronous mode, the SSI uses STCCR for capture */
clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
}
return 0;
}
static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_dummy_pcm *dpcm = runtime->private_data;
int bps;
bps = snd_pcm_format_width(runtime->format) * runtime->rate * runtime->channels / 8;
if (bps <= 0)
return -EINVAL;
dpcm->pcm_bps = bps;
dpcm->pcm_hz = HZ;
dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream);
dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream);
dpcm->pcm_irq_pos = 0;
dpcm->pcm_buf_pos = 0;
snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes));
return 0;
}
/**
* fsl_ssi_hw_params - program the sample size
*
* Most of the SSI registers have been programmed in the startup function,
* but the word length must be programmed here. Unfortunately, programming
* the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
* cause a problem with supporting simultaneous playback and capture. If
* the SSI is already playing a stream, then that stream may be temporarily
* stopped when you start capture.
*
* Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
* clock master.
*/
static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
{
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
unsigned int sample_size =
snd_pcm_format_width(params_format(hw_params));
u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
int enabled = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
unsigned int channels = params_channels(hw_params);
/*
* If we're in synchronous mode, and the SSI is already enabled,
* then STCCR is already set properly.
*/
if (enabled && ssi_private->cpu_dai_drv.symmetric_rates)
return 0;
/*
* FIXME: The documentation says that SxCCR[WL] should not be
* modified while the SSI is enabled. The only time this can
* happen is if we're trying to do simultaneous playback and
* capture in asynchronous mode. Unfortunately, I have been enable
* to get that to work at all on the P1022DS. Therefore, we don't
* bother to disable/enable the SSI when setting SxCCR[WL], because
* the SSI will stop anyway. Maybe one day, this will get fixed.
*/
/* In synchronous mode, the SSI uses STCCR for capture */
if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
ssi_private->cpu_dai_drv.symmetric_rates)
write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
else
write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK,
channels == 1 ? 0 : ssi_private->i2s_mode);
return 0;
}
static int jz_dmic_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
struct device *dev = dai->dev;
int channels = params_channels(params);
int rate = params_rate(params);
struct jz_dmic *jz_dmic = dev_get_drvdata(dai->dev);
int fmt_width = snd_pcm_format_width(params_format(params));
enum dma_slave_buswidth buswidth;
int trigger;
DMIC_DEBUG_MSG("enter %s, substream = %s\n",__func__,"capture");
if (!((1 << params_format(params)) & JZ_DMIC_FORMATS)
||channels < 1||channels > 4|| rate > 48000||rate < 8000
||fmt_width != 16) {
dev_err(dai->dev, "hw params not inval channel %d params %x rate %d fmt_width %d\n",
channels, params_format(params),rate,fmt_width);
return -EINVAL;
}
if (jz_dmic->unpack_enable)
buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
else
buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
jz_dmic->rx_dma_data.buswidth = buswidth;
jz_dmic->rx_dma_data.max_burst = (DMIC_FIFO_DEPTH * buswidth)/2;
trigger = jz_dmic->rx_dma_data.max_burst/(int)buswidth;
__dmic_set_request(dev, trigger);
__dmic_set_chnum(dev, channels - 1);
dmic_set_rate(dev,rate);
snd_soc_dai_set_dma_data(dai, substream, (void *)&jz_dmic->rx_dma_data);
} else {
dev_err(dai->dev, "DMIC is a capture device\n");
}
return 0;
}
static void set_wm8785_params(struct oxygen *chip,
struct snd_pcm_hw_params *params)
{
unsigned int value;
wm8785_write(chip, WM8785_R7, 0);
value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST;
if (params_rate(params) <= 48000)
value |= WM8785_OSR_SINGLE;
else if (params_rate(params) <= 96000)
value |= WM8785_OSR_DOUBLE;
else
value |= WM8785_OSR_QUAD;
wm8785_write(chip, WM8785_R0, value);
if (snd_pcm_format_width(params_format(params)) <= 16)
value = WM8785_WL_16;
else
value = WM8785_WL_24;
wm8785_write(chip, WM8785_R1, value);
}
int
snd_send(FILE * fp, size_t n)
{
snd_pcm_format_t format;
unsigned int nchannels;
snd_pcm_uframes_t period;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_current(pcm, params);
snd_pcm_hw_params_get_format(params, &format);
snd_pcm_hw_params_get_channels(params, &nchannels);
snd_pcm_hw_params_get_period_size(params, &period, 0);
int framesize =
snd_pcm_format_width(format) / 8 * nchannels;
unsigned char buf[period * framesize * 128];
size_t l;
while (n > sizeof(buf)) {
if ((l = fread(buf, 1, sizeof(buf), fp))) {
switch (snd_pcm_writei(pcm, buf, l / framesize)) {
case -EBADF:
return -1;
case -EPIPE:
#ifndef NDEBUG
snd_pcm_recover(pcm, -EPIPE, 0);
#else
snd_pcm_prepare(pcm);
#endif
}
} else
goto EOS;
n -= l;
}
if ((l = fread(buf, 1, n, fp)))
snd_pcm_writei(pcm, buf, l / framesize);
if (l < n)
EOS:if (ftell(fp) > 0)
eputs("Unexpected end of stream");
return snd_pcm_drain(pcm);
}
/**
* fsl_ssi_prepare: prepare the SSI.
*
* Most of the SSI registers have been programmed in the startup function,
* but the word length must be programmed here. Unfortunately, programming
* the SxCCR.WL bits requires the SSI to be temporarily disabled. This can
* cause a problem with supporting simultaneous playback and capture. If
* the SSI is already playing a stream, then that stream may be temporarily
* stopped when you start capture.
*
* Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
* clock master.
*/
static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
u32 wl;
wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
else
clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
return 0;
}
static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
int i;
struct snd_msnd *chip = snd_pcm_substream_chip(substream);
void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF;
chip->play_sample_size = snd_pcm_format_width(params_format(params));
chip->play_channels = params_channels(params);
chip->play_sample_rate = params_rate(params);
for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) {
writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize);
writew(chip->play_channels, pDAQ + DAQDS_wChannels);
writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate);
}
/* dont do this here:
* snd_msnd_calibrate_adc(chip->play_sample_rate);
*/
return 0;
}
static int loopback_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct loopback_pcm *dpcm = runtime->private_data;
struct loopback_cable *cable = dpcm->cable;
int bps, salign;
salign = (snd_pcm_format_width(runtime->format) *
runtime->channels) / 8;
bps = salign * runtime->rate;
if (bps <= 0 || salign <= 0)
return -EINVAL;
dpcm->buf_pos = 0;
dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size);
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
dpcm->silent_size = dpcm->pcm_buffer_size;
snd_pcm_format_set_silence(runtime->format, runtime->dma_area,
runtime->buffer_size * runtime->channels);
}
dpcm->irq_pos = 0;
dpcm->period_update_pending = 0;
dpcm->pcm_bps = bps;
dpcm->pcm_salign = salign;
dpcm->pcm_period_size = frames_to_bytes(runtime, runtime->period_size);
mutex_lock(&dpcm->loopback->cable_lock);
if (!(cable->valid & ~(1 << substream->stream)) ||
(get_setup(dpcm)->notify &&
substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
params_change(substream);
cable->valid |= 1 << substream->stream;
mutex_unlock(&dpcm->loopback->cable_lock);
return 0;
}
/* hw_params callback */
static int snd_bcm2835_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct bcm2835_alsa_stream *alsa_stream = runtime->private_data;
int err;
audio_info(" .. IN\n");
err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
if (err < 0) {
audio_error
(" pcm_lib_malloc failed to allocated pages for buffers\n");
return err;
}
alsa_stream->channels = params_channels(params);
alsa_stream->params_rate = params_rate(params);
alsa_stream->pcm_format_width = snd_pcm_format_width(params_format(params));
audio_info(" .. OUT\n");
return err;
}
static void generate_sine(uint8_t *frames, int channel, int count, double *_phase) {
double phase = *_phase;
double max_phase = 1.0 / freq;
double step = 1.0 / (double)rate;
double res;
float fres;
int chn;
int32_t ires;
int8_t *samp8 = (int8_t*) frames;
int16_t *samp16 = (int16_t*) frames;
int32_t *samp32 = (int32_t*) frames;
float *samp_f = (float*) frames;
int sample_size_bits = snd_pcm_format_width(format);
while (count-- > 0) {
//res = sin((phase * 2 * M_PI) / max_phase - M_PI) * 32767;
//res = sin((phase * 2 * M_PI) / max_phase - M_PI) * 32767;
//res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */
//if (res > 0) res = 10000;
//if (res < 0) res = -10000;
/* printf("%e\n",res); */
//ires = res;
//ires = ((16 - (count & 0xf)) <<24);
//ires = 0;
for(chn=0;chn<channels;chn++) {
if (sample_size_bits == 8) {
if (chn==channel) {
res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * 0x03fffffff; /* Don't use MAX volume */
ires = res;
*samp8++ = ires >> 24;
//*samp8++ = 0x12;
} else {
*samp8++ = 0;
}
} else if (sample_size_bits == 16) {
// returns 1 if successful
// enc: 0 for PCM, 1 for ULAW, 2 for ALAW (see DirectAudio.h)
int getFormatFromAlsaFormat(snd_pcm_format_t alsaFormat,
int* sampleSizeInBytes, int* significantBits,
int* isSigned, int* isBigEndian, int* enc) {
*sampleSizeInBytes = (snd_pcm_format_physical_width(alsaFormat) + 7) / 8;
*significantBits = snd_pcm_format_width(alsaFormat);
// defaults
*enc = 0; // PCM
*isSigned = (snd_pcm_format_signed(alsaFormat) > 0);
*isBigEndian = (snd_pcm_format_big_endian(alsaFormat) > 0);
// non-PCM formats
if (alsaFormat == SND_PCM_FORMAT_MU_LAW) { // Mu-Law
*sampleSizeInBytes = 8; *enc = 1; *significantBits = *sampleSizeInBytes;
}
else if (alsaFormat == SND_PCM_FORMAT_A_LAW) { // A-Law
*sampleSizeInBytes = 8; *enc = 2; *significantBits = *sampleSizeInBytes;
}
else if (snd_pcm_format_linear(alsaFormat) < 1) {
return 0;
}
return (*sampleSizeInBytes > 0);
}
bool VoiceStreamer::initDevice() {
int err;
if ((err = snd_output_stdio_attach(&output, stdout, 0)) < 0) {
printf("Output failed: %s\n", snd_strerror(err));
return false;
}
if ((err = snd_pcm_open(&phandle, "default", SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK)) < 0) {
printf("Playback open error: %s\n", snd_strerror(err));
return false;
}
if ((err = snd_pcm_open(&chandle, "default" , SND_PCM_STREAM_CAPTURE,
SND_PCM_NONBLOCK)) < 0) {
printf("Record open error: %s\n", snd_strerror(err));
return false;
}
latency = latency_min - 4;
buffer = new char[(latency_max*snd_pcm_format_width(format)/8)*2];
// if (setparams_p(phandle, &latency) < 0)
// return false;
if (setparams_c(chandle, &latency) < 0)
return false;
if ((err = snd_pcm_start(chandle)) < 0) {
printf("Go error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
frames_in = frames_out = 0;
in_max = 0;
return true;
}
static int arizona_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
unsigned int val;
int base = dai->driver->base;
const int *rates;
int i, ret, bclk_target;
int bclk, lrclk, wl, frame, sr_val;
//printk("Arizona: hw params start\n");
if (params_rate(params) % 8000)
rates = &arizona_44k1_bclk_rates[0];
else
rates = &arizona_48k_bclk_rates[0];
/* Force BCLK to stereo for I2S */
bclk_target = snd_soc_params_to_bclk(params);
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
if (val & ARIZONA_AIF1_FMT_MASK && params_channels(params) == 1) {
arizona_aif_err(dai, "Forcing stereo mode\n");
bclk_target *= 2;
}
for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
if (rates[i] >= bclk_target &&
rates[i] % params_rate(params) == 0) {
bclk = i;
break;
}
}
if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) {
arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
params_rate(params));
return -EINVAL;
}
for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
if (arizona_sr_vals[i] == params_rate(params))
break;
if (i == ARRAY_SIZE(arizona_sr_vals)) {
arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
params_rate(params));
return -EINVAL;
}
sr_val = i;
lrclk = rates[bclk] / params_rate(params);
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
wl = snd_pcm_format_width(params_format(params));
frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
/*
* We will need to be more flexible than this in future,
* currently we use a single sample rate for SYSCLK.
*/
switch (dai_priv->clk) {
case ARIZONA_CLK_SYSCLK:
/* SR2 is forced to 8kHz */
if (params_rate(params) != 8000) {
snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
ARIZONA_AIF1_RATE_MASK, 0);
} else {
snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_2,
ARIZONA_SAMPLE_RATE_2_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
ARIZONA_AIF1_RATE_MASK, 1);
}
break;
case ARIZONA_CLK_ASYNCCLK:
snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
ARIZONA_AIF1_RATE_MASK, 8);
break;
default:
arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
return -EINVAL;
}
snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
ARIZONA_AIF1TX_BCPF_MASK, lrclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
ARIZONA_AIF1RX_BCPF_MASK, lrclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
ARIZONA_AIF1TX_WL_MASK |
ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
ARIZONA_AIF1RX_WL_MASK |
ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
// HLL will add this in
//ret=wm2000_check_poll();
//printk("Arizona: hw params start 1\n");
return 0; //ret;
}
/* this may get called several times by oss emulation */
static int sst_byt_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct sst_byt_priv_data *pdata = snd_soc_component_get_drvdata(component);
struct sst_byt_pcm_data *pcm_data = &pdata->pcm[substream->stream];
struct sst_byt *byt = pdata->byt;
u32 rate, bits;
u8 channels;
int ret, playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
dev_dbg(rtd->dev, "PCM: hw_params, pcm_data %p\n", pcm_data);
ret = sst_byt_stream_type(byt, pcm_data->stream,
1, 1, !playback);
if (ret < 0) {
dev_err(rtd->dev, "failed to set stream format %d\n", ret);
return ret;
}
rate = params_rate(params);
ret = sst_byt_stream_set_rate(byt, pcm_data->stream, rate);
if (ret < 0) {
dev_err(rtd->dev, "could not set rate %d\n", rate);
return ret;
}
bits = snd_pcm_format_width(params_format(params));
ret = sst_byt_stream_set_bits(byt, pcm_data->stream, bits);
if (ret < 0) {
dev_err(rtd->dev, "could not set formats %d\n",
params_rate(params));
return ret;
}
channels = (u8)(params_channels(params) & 0xF);
ret = sst_byt_stream_set_channels(byt, pcm_data->stream, channels);
if (ret < 0) {
dev_err(rtd->dev, "could not set channels %d\n",
params_rate(params));
return ret;
}
snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
ret = sst_byt_stream_buffer(byt, pcm_data->stream,
substream->dma_buffer.addr,
params_buffer_bytes(params));
if (ret < 0) {
dev_err(rtd->dev, "PCM: failed to set DMA buffer %d\n", ret);
return ret;
}
ret = sst_byt_stream_commit(byt, pcm_data->stream);
if (ret < 0) {
dev_err(rtd->dev, "PCM: failed stream commit %d\n", ret);
return ret;
}
return 0;
}
static int
QSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
int status = 0;
int format = 0;
SDL_AudioFormat test_format = 0;
int found = 0;
snd_pcm_channel_setup_t csetup;
snd_pcm_channel_params_t cparams;
/* Initialize all variables that we clean on shutdown */
this->hidden =
(struct SDL_PrivateAudioData *) SDL_calloc(1,
(sizeof
(struct
SDL_PrivateAudioData)));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData));
/* Initialize channel transfer parameters to default */
QSA_InitAudioParams(&cparams);
/* Initialize channel direction: capture or playback */
this->hidden->iscapture = iscapture;
/* Find deviceid and cardid by device name for playback */
if ((!this->hidden->iscapture) && (devname != NULL)) {
uint32_t device;
int32_t status;
/* Search in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_playback_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_playback_device[device].deviceno;
this->hidden->cardno = qsa_playback_device[device].cardno;
break;
}
device++;
if (device >= qsa_playback_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such playback device");
}
} while (1);
}
/* Find deviceid and cardid by device name for capture */
if ((this->hidden->iscapture) && (devname != NULL)) {
/* Search in the capture devices */
uint32_t device;
int32_t status;
/* Searching in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_capture_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_capture_device[device].deviceno;
this->hidden->cardno = qsa_capture_device[device].cardno;
break;
}
device++;
if (device >= qsa_capture_devices) {
QSA_CloseDevice(this);
return SDL_SetError("No such capture device");
}
} while (1);
}
/* Check if SDL requested default audio device */
if (devname == NULL) {
/* Open system default audio device */
if (!this->hidden->iscapture) {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
} else {
/* Open requested audio device */
if (!this->hidden->iscapture) {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
}
/* Check if requested device is opened */
if (status < 0) {
this->hidden->audio_handle = NULL;
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_open", status);
}
if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) {
/* Disable QSA MMAP plugin for buggy audio drivers */
status =
snd_pcm_plugin_set_disable(this->hidden->audio_handle,
PLUGIN_DISABLE_MMAP);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_set_disable", status);
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
found = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format); !found;) {
/* if match found set format to equivalent QSA format */
switch (test_format) {
case AUDIO_U8:
{
format = SND_PCM_SFMT_U8;
found = 1;
}
break;
case AUDIO_S8:
{
format = SND_PCM_SFMT_S8;
found = 1;
}
break;
case AUDIO_S16LSB:
{
format = SND_PCM_SFMT_S16_LE;
found = 1;
}
break;
case AUDIO_S16MSB:
{
format = SND_PCM_SFMT_S16_BE;
found = 1;
}
break;
case AUDIO_U16LSB:
{
format = SND_PCM_SFMT_U16_LE;
found = 1;
}
break;
case AUDIO_U16MSB:
{
format = SND_PCM_SFMT_U16_BE;
found = 1;
}
break;
case AUDIO_S32LSB:
{
format = SND_PCM_SFMT_S32_LE;
found = 1;
}
break;
case AUDIO_S32MSB:
{
format = SND_PCM_SFMT_S32_BE;
found = 1;
}
break;
case AUDIO_F32LSB:
{
format = SND_PCM_SFMT_FLOAT_LE;
found = 1;
}
break;
case AUDIO_F32MSB:
{
format = SND_PCM_SFMT_FLOAT_BE;
found = 1;
}
break;
default:
{
break;
}
}
if (!found) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono/stereo/4ch/6ch/8ch audio */
cparams.format.voices = this->spec.channels;
/* Set rate */
cparams.format.rate = this->spec.freq;
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_channel_params", status);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0, sizeof(csetup));
if (!this->hidden->iscapture) {
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
csetup.channel = SND_PCM_CHANNEL_CAPTURE;
}
/* Setup an audio channel */
if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Unable to setup channel");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->pcm_len = this->spec.size;
if (this->hidden->pcm_len == 0) {
this->hidden->pcm_len =
csetup.buf.block.frag_size * this->spec.channels *
(snd_pcm_format_width(format) / 8);
}
/*
* Allocate memory to the audio buffer and initialize with silence
* (Note that buffer size must be a multiple of fragment size, so find
* closest multiple)
*/
this->hidden->pcm_buf =
(Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len);
if (this->hidden->pcm_buf == NULL) {
QSA_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->pcm_buf, this->spec.silence,
this->hidden->pcm_len);
/* get the file descriptor */
if (!this->hidden->iscapture) {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (this->hidden->audio_fd < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_file_descriptor", status);
}
/* Prepare an audio channel */
if (!this->hidden->iscapture) {
/* Prepare audio playback */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Prepare audio capture */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_prepare", status);
}
/* We're really ready to rock and roll. :-) */
return 0;
}
static int wm8776_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8776_priv *wm8776 = snd_soc_codec_get_drvdata(codec);
int iface_reg, iface;
int ratio_shift, master;
int i;
switch (dai->driver->id) {
case WM8776_DAI_DAC:
iface_reg = WM8776_DACIFCTRL;
master = 0x80;
ratio_shift = 4;
break;
case WM8776_DAI_ADC:
iface_reg = WM8776_ADCIFCTRL;
master = 0x100;
ratio_shift = 0;
break;
default:
return -EINVAL;
}
/* Set word length */
switch (snd_pcm_format_width(params_format(params))) {
case 16:
iface = 0;
break;
case 20:
iface = 0x10;
break;
case 24:
iface = 0x20;
break;
case 32:
iface = 0x30;
break;
default:
dev_err(codec->dev, "Unsupported sample size: %i\n",
snd_pcm_format_width(params_format(params)));
return -EINVAL;
}
/* Only need to set MCLK/LRCLK ratio if we're master */
if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) {
for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) {
if (wm8776->sysclk[dai->driver->id] / params_rate(params)
== mclk_ratios[i])
break;
}
if (i == ARRAY_SIZE(mclk_ratios)) {
dev_err(codec->dev,
"Unable to configure MCLK ratio %d/%d\n",
wm8776->sysclk[dai->driver->id], params_rate(params));
return -EINVAL;
}
dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]);
snd_soc_update_bits(codec, WM8776_MSTRCTRL,
0x7 << ratio_shift, i << ratio_shift);
} else {
dev_dbg(codec->dev, "DAI in slave mode\n");
}
snd_soc_update_bits(codec, iface_reg, 0x30, iface);
return 0;
}
static int es705_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
unsigned int base = dai->driver->base;
unsigned int msg, resp;
int ret, val;
u32 cmd_block[3];
return 0;
if (escore_priv.flag.is_fw_ready == 0) {
pr_warn("%s(port-%c): es705 firmware is not ready, abort\n",
__func__, PORT_NAME(base));
return 0;
}
/* word length */
//if (es705_ports[PORT_ID(base)].wl != params_format(params))
{
printk("++%s(port-%c)++: format=%d\n", __func__, PORT_NAME(base),
params_format(params));
val = snd_pcm_format_width(params_format(params)) - 1;
cmd_block[0] = ES705_PORT_PARAM_ID + base + ES705_PORT_WORDLENGHT;
cmd_block[1] = ES705_PORT_SET_PARAM + val;
cmd_block[2] = 0xffffffff;
ret = escore_write_block(&escore_priv, cmd_block);
if (ret < 0) {
dev_err(dai->dev, "escore_cmd: send [%08x %08x] failed\n",
cmd_block[0], cmd_block[2]);
goto out;
}
es705_ports[PORT_ID(base)].wl = params_format(params);
}
/* sample rate */
//if (es705_ports[PORT_ID(base)].rate != params_rate(params))
{
printk("++%s(port-%c)++: rate=%d\n", __func__, PORT_NAME(base),
params_rate(params));
msg = ES705_PORT_GET_PARAM + base + ES705_PORT_CLOCK;
ret = escore_cmd(&escore_priv, msg, &resp);
if (ret < 0) {
dev_err(dai->dev, "escore_cmd: send %08x failed\n", msg);
goto out;
}
val = (resp & 0x100) + (params_rate(params) / 1000);
cmd_block[0] = ES705_PORT_PARAM_ID + base + ES705_PORT_CLOCK;
cmd_block[1] = ES705_PORT_SET_PARAM + val;
cmd_block[2] = 0xffffffff;
ret = escore_write_block(&escore_priv, cmd_block);
if (ret < 0) {
dev_err(dai->dev, "escore_cmd: send [%08x %08x] failed\n",
cmd_block[0], cmd_block[2]);
goto out;
}
es705_ports[PORT_ID(base)].rate = params_rate(params);
}
out:
return ret;
}
static int uni_player_prepare_pcm(struct uniperif *player,
struct snd_pcm_runtime *runtime)
{
int output_frame_size, slot_width, clk_div;
/* Force slot width to 32 in I2S mode (HW constraint) */
if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
SND_SOC_DAIFMT_I2S)
slot_width = 32;
else
slot_width = snd_pcm_format_width(runtime->format);
output_frame_size = slot_width * runtime->channels;
clk_div = player->mclk / runtime->rate;
/*
* For 32 bits subframe clk_div must be a multiple of 128,
* for 16 bits must be a multiple of 64
*/
if ((slot_width == 32) && (clk_div % 128)) {
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
if ((slot_width == 16) && (clk_div % 64)) {
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
/*
* Number of bits per subframe (which is one channel sample)
* on output - Transfer 16 or 32 bits from FIFO
*/
switch (slot_width) {
case 32:
SET_UNIPERIF_I2S_FMT_NBIT_32(player);
SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player);
break;
case 16:
SET_UNIPERIF_I2S_FMT_NBIT_16(player);
SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player);
break;
default:
dev_err(player->dev, "subframe format not supported\n");
return -EINVAL;
}
/* Configure data memory format */
switch (runtime->format) {
case SNDRV_PCM_FORMAT_S16_LE:
/* One data word contains two samples */
SET_UNIPERIF_CONFIG_MEM_FMT_16_16(player);
break;
case SNDRV_PCM_FORMAT_S32_LE:
/*
* Actually "16 bits/0 bits" means "32/28/24/20/18/16 bits
* on the left than zeros (if less than 32 bytes)"... ;-)
*/
SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player);
break;
default:
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
/* Set rounding to off */
SET_UNIPERIF_CTRL_ROUNDING_OFF(player);
/* Set clock divisor */
SET_UNIPERIF_CTRL_DIVIDER(player, clk_div / (2 * output_frame_size));
/* Number of channelsmust be even*/
if ((runtime->channels % 2) || (runtime->channels < 2) ||
(runtime->channels > 10)) {
dev_err(player->dev, "%s: invalid nb of channels\n", __func__);
return -EINVAL;
}
SET_UNIPERIF_I2S_FMT_NUM_CH(player, runtime->channels / 2);
/* Set 1-bit audio format to disabled */
SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player);
SET_UNIPERIF_I2S_FMT_ORDER_MSB(player);
/* No iec958 formatting as outputting to DAC */
SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player);
return 0;
}
static int max98373_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component);
unsigned int sampling_rate = 0;
unsigned int chan_sz = 0;
/* pcm mode configuration */
switch (snd_pcm_format_width(params_format(params))) {
case 16:
chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_16;
break;
case 24:
chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_24;
break;
case 32:
chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_32;
break;
default:
dev_err(component->dev, "format unsupported %d\n",
params_format(params));
goto err;
}
max98373->ch_size = snd_pcm_format_width(params_format(params));
regmap_update_bits(max98373->regmap,
MAX98373_R2024_PCM_DATA_FMT_CFG,
MAX98373_PCM_MODE_CFG_CHANSZ_MASK, chan_sz);
dev_dbg(component->dev, "format supported %d",
params_format(params));
/* sampling rate configuration */
switch (params_rate(params)) {
case 8000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_8000;
break;
case 11025:
sampling_rate = MAX98373_PCM_SR_SET1_SR_11025;
break;
case 12000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_12000;
break;
case 16000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_16000;
break;
case 22050:
sampling_rate = MAX98373_PCM_SR_SET1_SR_22050;
break;
case 24000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_24000;
break;
case 32000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_32000;
break;
case 44100:
sampling_rate = MAX98373_PCM_SR_SET1_SR_44100;
break;
case 48000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_48000;
break;
default:
dev_err(component->dev, "rate %d not supported\n",
params_rate(params));
goto err;
}
/* set DAI_SR to correct LRCLK frequency */
regmap_update_bits(max98373->regmap,
MAX98373_R2027_PCM_SR_SETUP_1,
MAX98373_PCM_SR_SET1_SR_MASK,
sampling_rate);
regmap_update_bits(max98373->regmap,
MAX98373_R2028_PCM_SR_SETUP_2,
MAX98373_PCM_SR_SET2_SR_MASK,
sampling_rate << MAX98373_PCM_SR_SET2_SR_SHIFT);
/* set sampling rate of IV */
if (max98373->interleave_mode &&
sampling_rate > MAX98373_PCM_SR_SET1_SR_16000)
regmap_update_bits(max98373->regmap,
MAX98373_R2028_PCM_SR_SETUP_2,
MAX98373_PCM_SR_SET2_IVADC_SR_MASK,
sampling_rate - 3);
else
regmap_update_bits(max98373->regmap,
MAX98373_R2028_PCM_SR_SETUP_2,
MAX98373_PCM_SR_SET2_IVADC_SR_MASK,
sampling_rate);
return max98373_set_clock(component, params);
err:
return -EINVAL;
}
static int pcm512x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
int alen;
int gpio;
int clock_output;
int master_mode;
int ret;
dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n",
params_rate(params),
params_channels(params));
switch (snd_pcm_format_width(params_format(params))) {
case 16:
alen = PCM512x_ALEN_16;
break;
case 20:
alen = PCM512x_ALEN_20;
break;
case 24:
alen = PCM512x_ALEN_24;
break;
case 32:
alen = PCM512x_ALEN_32;
break;
default:
dev_err(codec->dev, "Bad frame size: %d\n",
snd_pcm_format_width(params_format(params)));
return -EINVAL;
}
switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
ret = regmap_update_bits(pcm512x->regmap,
PCM512x_BCLK_LRCLK_CFG,
PCM512x_BCKP
| PCM512x_BCKO | PCM512x_LRKO,
0);
if (ret != 0) {
dev_err(codec->dev,
"Failed to enable slave mode: %d\n", ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
PCM512x_DCAS, 0);
if (ret != 0) {
dev_err(codec->dev,
"Failed to enable clock divider autoset: %d\n",
ret);
return ret;
}
return 0;
case SND_SOC_DAIFMT_CBM_CFM:
clock_output = PCM512x_BCKO | PCM512x_LRKO;
master_mode = PCM512x_RLRK | PCM512x_RBCK;
break;
case SND_SOC_DAIFMT_CBM_CFS:
clock_output = PCM512x_BCKO;
master_mode = PCM512x_RBCK;
break;
default:
return -EINVAL;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_I2S_1,
PCM512x_ALEN, alen);
if (ret != 0) {
dev_err(codec->dev, "Failed to set frame size: %d\n", ret);
return ret;
}
if (pcm512x->pll_out) {
ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_A, 0x11);
if (ret != 0) {
dev_err(codec->dev, "Failed to set FLEX_A: %d\n", ret);
return ret;
}
ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_B, 0xff);
if (ret != 0) {
dev_err(codec->dev, "Failed to set FLEX_B: %d\n", ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
PCM512x_IDFS | PCM512x_IDBK
| PCM512x_IDSK | PCM512x_IDCH
| PCM512x_IDCM | PCM512x_DCAS
| PCM512x_IPLK,
PCM512x_IDFS | PCM512x_IDBK
| PCM512x_IDSK | PCM512x_IDCH
| PCM512x_DCAS);
if (ret != 0) {
dev_err(codec->dev,
"Failed to ignore auto-clock failures: %d\n",
ret);
return ret;
}
} else {
ret = regmap_update_bits(pcm512x->regmap, PCM512x_ERROR_DETECT,
PCM512x_IDFS | PCM512x_IDBK
| PCM512x_IDSK | PCM512x_IDCH
| PCM512x_IDCM | PCM512x_DCAS
| PCM512x_IPLK,
PCM512x_IDFS | PCM512x_IDBK
| PCM512x_IDSK | PCM512x_IDCH
| PCM512x_DCAS | PCM512x_IPLK);
if (ret != 0) {
dev_err(codec->dev,
"Failed to ignore auto-clock failures: %d\n",
ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN,
PCM512x_PLLE, 0);
if (ret != 0) {
dev_err(codec->dev, "Failed to disable pll: %d\n", ret);
return ret;
}
}
ret = pcm512x_set_dividers(dai, params);
if (ret != 0)
return ret;
if (pcm512x->pll_out) {
ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_REF,
PCM512x_SREF, PCM512x_SREF_GPIO);
if (ret != 0) {
dev_err(codec->dev,
"Failed to set gpio as pllref: %d\n", ret);
return ret;
}
gpio = PCM512x_GREF_GPIO1 + pcm512x->pll_in - 1;
ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_PLLIN,
PCM512x_GREF, gpio);
if (ret != 0) {
dev_err(codec->dev,
"Failed to set gpio %d as pllin: %d\n",
pcm512x->pll_in, ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_PLL_EN,
PCM512x_PLLE, PCM512x_PLLE);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable pll: %d\n", ret);
return ret;
}
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_BCLK_LRCLK_CFG,
PCM512x_BCKP | PCM512x_BCKO | PCM512x_LRKO,
clock_output);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable clock output: %d\n", ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_MASTER_MODE,
PCM512x_RLRK | PCM512x_RBCK,
master_mode);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable master mode: %d\n", ret);
return ret;
}
if (pcm512x->pll_out) {
gpio = PCM512x_G1OE << (pcm512x->pll_out - 1);
ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN,
gpio, gpio);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable gpio %d: %d\n",
pcm512x->pll_out, ret);
return ret;
}
gpio = PCM512x_GPIO_OUTPUT_1 + pcm512x->pll_out - 1;
ret = regmap_update_bits(pcm512x->regmap, gpio,
PCM512x_GxSL, PCM512x_GxSL_PLLCK);
if (ret != 0) {
dev_err(codec->dev, "Failed to output pll on %d: %d\n",
ret, pcm512x->pll_out);
return ret;
}
gpio = PCM512x_G1OE << (4 - 1);
ret = regmap_update_bits(pcm512x->regmap, PCM512x_GPIO_EN,
gpio, gpio);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable gpio %d: %d\n",
4, ret);
return ret;
}
gpio = PCM512x_GPIO_OUTPUT_1 + 4 - 1;
ret = regmap_update_bits(pcm512x->regmap, gpio,
PCM512x_GxSL, PCM512x_GxSL_PLLLK);
if (ret != 0) {
dev_err(codec->dev,
"Failed to output pll lock on %d: %d\n",
ret, 4);
return ret;
}
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE,
PCM512x_RQSY, PCM512x_RQSY_HALT);
if (ret != 0) {
dev_err(codec->dev, "Failed to halt clocks: %d\n", ret);
return ret;
}
ret = regmap_update_bits(pcm512x->regmap, PCM512x_SYNCHRONIZE,
PCM512x_RQSY, PCM512x_RQSY_RESUME);
if (ret != 0) {
dev_err(codec->dev, "Failed to resume clocks: %d\n", ret);
return ret;
}
return 0;
}
static ALCboolean qsa_reset_playback(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
int32_t format=-1;
switch(device->FmtType)
{
case DevFmtByte:
format=SND_PCM_SFMT_S8;
break;
case DevFmtUByte:
format=SND_PCM_SFMT_U8;
break;
case DevFmtShort:
format=SND_PCM_SFMT_S16_LE;
break;
case DevFmtUShort:
format=SND_PCM_SFMT_U16_LE;
break;
case DevFmtInt:
format=SND_PCM_SFMT_S32_LE;
break;
case DevFmtUInt:
format=SND_PCM_SFMT_U32_LE;
break;
case DevFmtFloat:
format=SND_PCM_SFMT_FLOAT_LE;
break;
}
/* we actually don't want to block on writes */
snd_pcm_nonblock_mode(data->pcmHandle, 1);
/* Disable mmap to control data transfer to the audio device */
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_BUFFER_PARTIAL_BLOCKS);
// configure a sound channel
memset(&data->cparams, 0, sizeof(data->cparams));
data->cparams.channel=SND_PCM_CHANNEL_PLAYBACK;
data->cparams.mode=SND_PCM_MODE_BLOCK;
data->cparams.start_mode=SND_PCM_START_FULL;
data->cparams.stop_mode=SND_PCM_STOP_STOP;
data->cparams.buf.block.frag_size=device->UpdateSize*
ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType);
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
data->cparams.format.interleave=1;
data->cparams.format.rate=device->Frequency;
data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans);
data->cparams.format.format=format;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
int original_rate=data->cparams.format.rate;
int original_voices=data->cparams.format.voices;
int original_format=data->cparams.format.format;
int it;
int jt;
for (it=0; it<1; it++)
{
/* Check for second pass */
if (it==1)
{
original_rate=ratelist[0].rate;
original_voices=channellist[0].channels;
original_format=formatlist[0].format;
}
do {
/* At first downgrade sample format */
jt=0;
do {
if (formatlist[jt].format==data->cparams.format.format)
{
data->cparams.format.format=formatlist[jt+1].format;
break;
}
if (formatlist[jt].format==0)
{
data->cparams.format.format=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.format==0)
{
data->cparams.format.format=original_format;
/* At secod downgrade sample rate */
jt=0;
do {
if (ratelist[jt].rate==data->cparams.format.rate)
{
data->cparams.format.rate=ratelist[jt+1].rate;
break;
}
if (ratelist[jt].rate==0)
{
data->cparams.format.rate=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.rate==0)
{
data->cparams.format.rate=original_rate;
data->cparams.format.format=original_format;
/* At third downgrade channels number */
jt=0;
do {
if(channellist[jt].channels==data->cparams.format.voices)
{
data->cparams.format.voices=channellist[jt+1].channels;
break;
}
if (channellist[jt].channels==0)
{
data->cparams.format.voices=0;
break;
}
jt++;
} while(1);
}
if (data->cparams.format.voices==0)
{
break;
}
}
data->cparams.buf.block.frag_size=device->UpdateSize*
data->cparams.format.voices*
snd_pcm_format_width(data->cparams.format.format)/8;
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
continue;
}
else
{
break;
}
} while(1);
if (data->cparams.format.voices!=0)
{
break;
}
}
if (data->cparams.format.voices==0)
{
return ALC_FALSE;
}
}
if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0)
{
return ALC_FALSE;
}
memset(&data->csetup, 0, sizeof(data->csetup));
data->csetup.channel=SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_setup(data->pcmHandle, &data->csetup)<0)
{
return ALC_FALSE;
}
/* now fill back to the our AL device */
device->Frequency=data->cparams.format.rate;
switch (data->cparams.format.voices)
{
case 1:
device->FmtChans=DevFmtMono;
break;
case 2:
device->FmtChans=DevFmtStereo;
break;
case 4:
device->FmtChans=DevFmtQuad;
break;
case 6:
device->FmtChans=DevFmtX51;
break;
case 7:
device->FmtChans=DevFmtX61;
break;
case 8:
device->FmtChans=DevFmtX71;
break;
default:
device->FmtChans=DevFmtMono;
break;
}
switch (data->cparams.format.format)
{
case SND_PCM_SFMT_S8:
device->FmtType=DevFmtByte;
break;
case SND_PCM_SFMT_U8:
device->FmtType=DevFmtUByte;
break;
case SND_PCM_SFMT_S16_LE:
device->FmtType=DevFmtShort;
break;
case SND_PCM_SFMT_U16_LE:
device->FmtType=DevFmtUShort;
break;
case SND_PCM_SFMT_S32_LE:
device->FmtType=DevFmtInt;
break;
case SND_PCM_SFMT_U32_LE:
device->FmtType=DevFmtUInt;
break;
case SND_PCM_SFMT_FLOAT_LE:
device->FmtType=DevFmtFloat;
break;
default:
device->FmtType=DevFmtShort;
break;
}
SetDefaultChannelOrder(device);
device->UpdateSize=data->csetup.buf.block.frag_size/
(ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType));
device->NumUpdates=data->csetup.buf.block.frags;
data->size=data->csetup.buf.block.frag_size;
data->buffer=malloc(data->size);
if (!data->buffer)
{
return ALC_FALSE;
}
return ALC_TRUE;
}
static int NTO_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int rval;
int format;
Uint16 test_format;
int twidth;
int found;
#ifdef DEBUG_AUDIO
fprintf(stderr, "NTO_OpenAudio\n");
#endif
audio_handle = NULL;
this->enabled = 0;
if ( pcm_buf != NULL ) {
free((Uint8 *)pcm_buf);
pcm_buf = NULL;
}
/* initialize channel transfer parameters to default */
init_pcm_cparams(&cparams);
/* Open the audio device */
rval = snd_pcm_open_preferred(&audio_handle, &card_no, &device_no, OPEN_FLAGS);
if ( rval < 0 ) {
SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
return(-1);
}
/* set to nonblocking mode */
if ((rval = snd_pcm_nonblock_mode(audio_handle, 1))<0) //I assume 1 means on
{
SDL_SetError("snd_pcm_nonblock_mode failed: %s\n", snd_strerror(rval));
return(-1);
}
/* enable count status parameter */
if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
{
SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
return(-1);
}
/* Try for a closest match on audio format */
format = 0;
found = 0; // can't use format as SND_PCM_SFMT_U8 = 0 in nto
for ( test_format = SDL_FirstAudioFormat(spec->format); !found ; )
{
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
#endif
/* if match found set format to equivalent ALSA format */
switch ( test_format ) {
case AUDIO_U8:
format = SND_PCM_SFMT_U8;
cparams.buf.block.frag_size = spec->samples * spec->channels;
found = 1;
break;
case AUDIO_S8:
format = SND_PCM_SFMT_S8;
cparams.buf.block.frag_size = spec->samples * spec->channels;
found = 1;
break;
case AUDIO_S16LSB:
format = SND_PCM_SFMT_S16_LE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_S16MSB:
format = SND_PCM_SFMT_S16_BE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_U16LSB:
format = SND_PCM_SFMT_U16_LE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
case AUDIO_U16MSB:
format = SND_PCM_SFMT_U16_BE;
cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
found = 1;
break;
default:
break;
}
if ( ! found ) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if ( test_format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono or stereo audio (currently only two channels supported) */
cparams.format.voices = spec->channels;
#ifdef DEBUG_AUDIO
fprintf(stderr,"intializing channels %d\n", cparams.format.voices);
#endif
/* Set rate */
cparams.format.rate = spec->freq ;
/* Setup the transfer parameters according to cparams */
rval = snd_pcm_plugin_params(audio_handle, &cparams);
if (rval < 0) {
SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
return(-1);
}
/* Make sure channel is setup right one last time */
memset( &csetup, 0, sizeof( csetup ) );
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
{
SDL_SetError("Unable to setup playback channel\n" );
return(-1);
}
else
{
#ifdef DEBUG_AUDIO
fprintf(stderr,"requested format: %d\n",cparams.format.format);
fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
fprintf(stderr,"real format: %d\n", csetup.format.format );
fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
#endif // DEBUG_AUDIO
}
/* Allocate memory to the audio buffer and initialize with silence
(Note that buffer size must be a multiple of fragment size, so find closest multiple)
*/
twidth = snd_pcm_format_width(format);
if (twidth < 0) {
printf("snd_pcm_format_width failed\n");
twidth = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr,"format is %d bits wide\n",twidth);
#endif
pcm_len = spec->size ;
#ifdef DEBUG_AUDIO
fprintf(stderr,"pcm_len set to %d\n", pcm_len);
#endif
if (pcm_len == 0)
{
pcm_len = csetup.buf.block.frag_size;
}
pcm_buf = (Uint8*)malloc(pcm_len);
if (pcm_buf == NULL) {
SDL_SetError("pcm_buf malloc failed\n");
return(-1);
}
memset(pcm_buf,spec->silence,pcm_len);
#ifdef DEBUG_AUDIO
fprintf(stderr,"pcm_buf malloced and silenced.\n");
#endif
/* get the file descriptor */
if( (audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
{
fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
}
/* Trigger audio playback */
rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
if (rval < 0) {
SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
return(-1);
}
this->enabled = 1;
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct gus_pcm_private *pcmp = runtime->private_data;
struct snd_gus_card * gus = pcmp->gus;
unsigned long flags;
unsigned char voice_ctrl, ramp_ctrl;
unsigned short rate;
unsigned int curr, begin, end;
unsigned short vol;
unsigned char pan;
unsigned int voice;
spin_lock_irqsave(&pcmp->lock, flags);
if (pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE) {
spin_unlock_irqrestore(&pcmp->lock, flags);
return;
}
pcmp->flags |= SNDRV_GF1_PCM_PFLG_ACTIVE;
pcmp->final_volume = 0;
spin_unlock_irqrestore(&pcmp->lock, flags);
rate = snd_gf1_translate_freq(gus, runtime->rate << 4);
/* enable WAVE IRQ */
voice_ctrl = snd_pcm_format_width(runtime->format) == 16 ? 0x24 : 0x20;
/* enable RAMP IRQ + rollover */
ramp_ctrl = 0x24;
if (pcmp->blocks == 1) {
voice_ctrl |= 0x08; /* loop enable */
ramp_ctrl &= ~0x04; /* disable rollover */
}
for (voice = 0; voice < pcmp->voices; voice++) {
begin = pcmp->memory + voice * (pcmp->dma_size / runtime->channels);
curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels;
end = curr + (pcmp->block_size / runtime->channels);
end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1;
/*
snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, "
"ctrl=0x%x, ramp=0x%x, rate=0x%x\n",
curr, begin, end, voice_ctrl, ramp_ctrl, rate);
*/
pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8;
vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
spin_lock_irqsave(&gus->reg_lock, flags);
snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
snd_gf1_write8(gus, SNDRV_GF1_VB_PAN, pan);
snd_gf1_write16(gus, SNDRV_GF1_VW_FREQUENCY, rate);
snd_gf1_write_addr(gus, SNDRV_GF1_VA_START, begin << 4, voice_ctrl & 4);
snd_gf1_write_addr(gus, SNDRV_GF1_VA_END, end << 4, voice_ctrl & 4);
snd_gf1_write_addr(gus, SNDRV_GF1_VA_CURRENT, curr << 4, voice_ctrl & 4);
snd_gf1_write16(gus, SNDRV_GF1_VW_VOLUME, SNDRV_GF1_MIN_VOLUME << 4);
snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_RATE, 0x2f);
snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_START, SNDRV_GF1_MIN_OFFSET);
snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_END, vol >> 8);
snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
if (!gus->gf1.enh_mode) {
snd_gf1_delay(gus);
snd_gf1_write8(gus, SNDRV_GF1_VB_VOLUME_CONTROL, ramp_ctrl);
}
spin_unlock_irqrestore(&gus->reg_lock, flags);
}
spin_lock_irqsave(&gus->reg_lock, flags);
for (voice = 0; voice < pcmp->voices; voice++) {
snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
if (gus->gf1.enh_mode)
snd_gf1_write8(gus, SNDRV_GF1_VB_MODE, 0x00); /* deactivate voice */
snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
voice_ctrl &= ~0x20;
}
voice_ctrl |= 0x20;
if (!gus->gf1.enh_mode) {
snd_gf1_delay(gus);
for (voice = 0; voice < pcmp->voices; voice++) {
snd_gf1_select_voice(gus, pcmp->pvoices[voice]->number);
snd_gf1_write8(gus, SNDRV_GF1_VB_ADDRESS_CONTROL, voice_ctrl);
voice_ctrl &= ~0x20; /* disable IRQ for next voice */
}
}
spin_unlock_irqrestore(&gus->reg_lock, flags);
}
static int arizona_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona *arizona = priv->arizona;
int base = dai->driver->base;
const int *rates;
int i, ret, val;
int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
int bclk, lrclk, wl, frame, bclk_target;
if (params_rate(params) % 8000)
rates = &arizona_44k1_bclk_rates[0];
else
rates = &arizona_48k_bclk_rates[0];
bclk_target = snd_soc_params_to_bclk(params);
if (chan_limit && chan_limit < params_channels(params)) {
arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit);
bclk_target /= params_channels(params);
bclk_target *= chan_limit;
}
/* Force stereo for I2S mode */
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
arizona_aif_dbg(dai, "Forcing stereo mode\n");
bclk_target *= 2;
}
for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
if (rates[i] >= bclk_target &&
rates[i] % params_rate(params) == 0) {
bclk = i;
break;
}
}
if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) {
arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
params_rate(params));
return -EINVAL;
}
lrclk = rates[bclk] / params_rate(params);
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
wl = snd_pcm_format_width(params_format(params));
frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
ret = arizona_hw_params_rate(substream, params, dai);
if (ret != 0)
return ret;
snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
ARIZONA_AIF1TX_BCPF_MASK, lrclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
ARIZONA_AIF1RX_BCPF_MASK, lrclk);
snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
ARIZONA_AIF1TX_WL_MASK |
ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
ARIZONA_AIF1RX_WL_MASK |
ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
return 0;
}
/*
* set sampling rate according to the substream
*/
static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
{
struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
struct snd_ps3_avsetting_info avs;
int ret;
avs = card->avs;
pr_debug("%s: called freq=%d width=%d\n", __func__,
substream->runtime->rate,
snd_pcm_format_width(substream->runtime->format));
pr_debug("%s: before freq=%d width=%d\n", __func__,
card->avs.avs_audio_rate, card->avs.avs_audio_width);
/* sample rate */
switch (substream->runtime->rate) {
case 44100:
avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K;
break;
case 48000:
avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
break;
case 88200:
avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K;
break;
case 96000:
avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K;
break;
default:
pr_info("%s: invalid rate %d\n", __func__,
substream->runtime->rate);
return 1;
}
/* width */
switch (snd_pcm_format_width(substream->runtime->format)) {
case 16:
avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
break;
case 24:
avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24;
break;
default:
pr_info("%s: invalid width %d\n", __func__,
snd_pcm_format_width(substream->runtime->format));
return 1;
}
memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8);
if (memcmp(&card->avs, &avs, sizeof(avs))) {
pr_debug("%s: after freq=%d width=%d\n", __func__,
card->avs.avs_audio_rate, card->avs.avs_audio_width);
card->avs = avs;
snd_ps3_change_avsetting(card);
ret = 0;
} else
ret = 1;
/* check CS non-audio bit and mute accordingly */
if (avs.avs_cs_info[0] & 0x02)
ps3av_audio_mute_analog(1); /* mute if non-audio */
else
ps3av_audio_mute_analog(0);
return ret;
}
