static int set_sw_params(snd_pcm_t *pcm,
snd_pcm_sw_params_t *sw_params,
snd_spcm_xrun_type_t xrun_type)
{
int err;
err = snd_pcm_sw_params_current(pcm, sw_params);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_start_threshold(pcm, sw_params, (pcm->buffer_size / pcm->period_size) * pcm->period_size);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_avail_min(pcm, sw_params, pcm->period_size);
if (err < 0)
return err;
switch (xrun_type) {
case SND_SPCM_XRUN_STOP:
err = snd_pcm_sw_params_set_stop_threshold(pcm, sw_params, pcm->buffer_size);
break;
case SND_SPCM_XRUN_IGNORE:
err = snd_pcm_sw_params_set_stop_threshold(pcm, sw_params, pcm->boundary);
break;
default:
return -EINVAL;
}
if (err < 0)
return err;
err = snd_pcm_sw_params_set_xfer_align(pcm, sw_params, 1);
if (err < 0)
return err;
err = snd_pcm_sw_params(pcm, sw_params);
if (err < 0)
return err;
return 0;
}
static void set_params(void)
{
hwparams.format=SND_PCM_FORMAT_S16_LE;
hwparams.channels=2;
hwparams.rate=44100;
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_format(handle, params, hwparams.format);
snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
snd_pcm_hw_params_set_rate_near(handle, params, &hwparams.rate, 0);
buffer_time=0;
snd_pcm_hw_params_get_buffer_time_max(params,&buffer_time, 0);
period_time=125000;
snd_pcm_hw_params_set_period_time_near(handle, params,&period_time, 0);
buffer_time = 500000;
snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, 0);
/*monotonic = */snd_pcm_hw_params_is_monotonic(params);
/*can_pause = */snd_pcm_hw_params_can_pause(params);
printf("sizeof(params) : %d\n",sizeof(params));
snd_pcm_hw_params(handle, params);
snd_pcm_uframes_t buffer_size;
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
size_t n=chunk_size;
snd_pcm_sw_params_set_avail_min(handle, swparams, n);
snd_pcm_uframes_t start_threshold, stop_threshold;
start_threshold=22050;
snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
stop_threshold=22050;
snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
snd_pcm_format_physical_width(hwparams.format);
}
static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams, int period, int nperiods) {
int err;
/* get the current swparams */
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
printf("Unable to determine current swparams for capture: %s\n", snd_strerror(err));
return err;
}
/* start the transfer when the buffer is full */
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, period );
if (err < 0) {
printf("Unable to set start threshold mode for capture: %s\n", snd_strerror(err));
return err;
}
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, -1 );
if (err < 0) {
printf("Unable to set start threshold mode for capture: %s\n", snd_strerror(err));
return err;
}
/* allow the transfer when at least period_size samples can be processed */
err = snd_pcm_sw_params_set_avail_min(handle, swparams, 1 );
if (err < 0) {
printf("Unable to set avail min for capture: %s\n", snd_strerror(err));
return err;
}
/* align all transfers to 1 sample */
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
if (err < 0) {
printf("Unable to set transfer align for capture: %s\n", snd_strerror(err));
return err;
}
/* write the parameters to the playback device */
err = snd_pcm_sw_params(handle, swparams);
if (err < 0) {
printf("Unable to set sw params for capture: %s\n", snd_strerror(err));
return err;
}
return 0;
}
bool AudioInputALSA::PrepSwParams(void)
{
snd_pcm_sw_params_t* swparams;
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_uframes_t boundary;
if (AlsaBad(snd_pcm_sw_params_current(pcm_handle, swparams),
"failed to get swparams"))
return false;
if (AlsaBad(snd_pcm_sw_params_get_boundary(swparams, &boundary),
"failed to get boundary"))
return false;
// explicit start, not auto start
if (AlsaBad(snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams,
boundary), "failed to set start threshold"))
return false;
if (AlsaBad(snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams,
boundary), "failed to set stop threshold"))
return false;
if (AlsaBad(snd_pcm_sw_params(pcm_handle, swparams),
"failed to set software parameters"))
return false;
return true;
}
static int ao_alsa_init(dtaudio_output_t *aout, dtaudio_para_t *para)
{
memcpy(&wrapper->para, para, sizeof(dtaudio_para_t));
snd_pcm_t *alsa_handle;
snd_pcm_hw_params_t *alsa_hwparams;
snd_pcm_sw_params_t *alsa_swparams;
snd_pcm_uframes_t size;
snd_pcm_uframes_t boundary;
alsa_ctx_t *ctx = (alsa_ctx_t *)malloc(sizeof(*ctx));
if (!ctx) {
return -1;
}
wrapper->ao_priv = ctx;
int afmt = format_to_alsa(wrapper->para.bps);
uint32_t channels = wrapper->para.dst_channels;
uint32_t sr = wrapper->para.dst_samplerate;
int bytes_per_sample = snd_pcm_format_physical_width(afmt) * channels / 8;
int err = 0;
err = snd_pcm_open(&alsa_handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
free(ctx);
wrapper->ao_priv = NULL;
return -1;
}
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
err = snd_pcm_hw_params_any(alsa_handle, alsa_hwparams);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_access(alsa_handle, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_format(alsa_handle, alsa_hwparams, afmt);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_channels_near(alsa_handle, alsa_hwparams,
&channels);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_rate_near(alsa_handle, alsa_hwparams, &sr, NULL);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params(alsa_handle, alsa_hwparams);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
/******************************************************************
* Set HW Params Finish
******************************************************************/
/* buffer size means the entire size of alsa pcm buffer size*/
err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &size);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
ctx->buf_size = size * bytes_per_sample;
/*period size means count processed every interrupt*/
err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &size, NULL);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
ctx->trunk_size = size * bytes_per_sample;
dt_info(TAG, "outburst: %d, bytes_per_sample: %d, buffersize: %d\n",
ctx->trunk_size, bytes_per_sample, ctx->buf_size);
/******************************************************************
* set sw params
******************************************************************/
err = snd_pcm_sw_params_current(alsa_handle, alsa_swparams);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
#if SND_LIB_VERSION >= 0x000901
err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
#else
boundary = 0x7fffffff;
#endif
err = snd_pcm_sw_params_set_start_threshold(alsa_handle, alsa_swparams, size);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
err = snd_pcm_sw_params_set_stop_threshold(alsa_handle, alsa_swparams,
boundary);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
#if SND_LIB_VERSION >= 0x000901
err = snd_pcm_sw_params_set_silence_size(alsa_handle, alsa_swparams, boundary);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
#endif
err = snd_pcm_sw_params(alsa_handle, alsa_swparams);
if (err < 0) {
dt_error(TAG, "%s,%d: %s\n", __func__, __LINE__, snd_strerror(err));
return -1;
}
ctx->pause_support = snd_pcm_hw_params_can_pause(alsa_hwparams);
//buf size limit
ctx->buf_threshold = bytes_per_sample * sr * DEFAULT_TIME_SIZE / 1000;
dt_info(TAG, "alsa audio init ok! outburst:%d thres:%d \n", ctx->trunk_size,
ctx->buf_threshold);
ctx->handle = alsa_handle;
ctx->channels = wrapper->para.dst_channels;
ctx->bps = wrapper->para.bps;
ctx->samplerate = wrapper->para.dst_samplerate;
return 0;
}
static void set_params(void)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size;
int err;
size_t n;
unsigned int rate;
snd_pcm_uframes_t start_threshold, stop_threshold;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
error("Broken configuration for this PCM: no configurations available");
exit(EXIT_FAILURE);
}
if (mmap_flag) {
snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof());
snd_pcm_access_mask_none(mask);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
err = snd_pcm_hw_params_set_access_mask(handle, params, mask);
} else if (interleaved)
err = snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
else
err = snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_NONINTERLEAVED);
if (err < 0) {
error("Access type not available");
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
if (err < 0) {
error("Sample format non available");
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
if (err < 0) {
error("Channels count non available");
exit(EXIT_FAILURE);
}
#if 0
err = snd_pcm_hw_params_set_periods_min(handle, params, 2);
assert(err >= 0);
#endif
rate = hwparams.rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &hwparams.rate, 0);
assert(err >= 0);
if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) {
if (!quiet_mode) {
char plugex[64];
const char *pcmname = snd_pcm_name(handle);
fprintf(stderr, "Warning: rate is not accurate (requested = %iHz, got = %iHz)\n", rate, hwparams.rate);
if (! pcmname || strchr(snd_pcm_name(handle), ':'))
*plugex = 0;
else
snprintf(plugex, sizeof(plugex), "(-Dplug:%s)",
snd_pcm_name(handle));
fprintf(stderr, " please, try the plug plugin %s\n",
plugex);
}
}
rate = hwparams.rate;
if (buffer_time == 0 && buffer_frames == 0) {
err = snd_pcm_hw_params_get_buffer_time_max(params,
&buffer_time, 0);
assert(err >= 0);
if (buffer_time > 500000)
buffer_time = 500000;
}
if (period_time == 0 && period_frames == 0) {
if (buffer_time > 0)
period_time = buffer_time / 4;
else
period_frames = buffer_frames / 4;
}
if (period_time > 0)
err = snd_pcm_hw_params_set_period_time_near(handle, params,
&period_time, 0);
else
err = snd_pcm_hw_params_set_period_size_near(handle, params,
&period_frames, 0);
assert(err >= 0);
if (buffer_time > 0) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
&buffer_time, 0);
} else {
err = snd_pcm_hw_params_set_buffer_size_near(handle, params,
&buffer_frames);
}
assert(err >= 0);
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
error("Unable to install hw params:");
snd_pcm_hw_params_dump(params, log);
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
if (chunk_size == buffer_size) {
error("Can't use period equal to buffer size (%lu == %lu)",
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
if (avail_min < 0)
n = chunk_size;
else
n = (double) rate * avail_min / 1000000;
err = snd_pcm_sw_params_set_avail_min(handle, swparams, n);
if (err < 0) {
error("Setting sw params failed\n");
}
/* round up to closest transfer boundary */
n = buffer_size;
if (start_delay <= 0) {
start_threshold = n + (double) rate * start_delay / 1000000;
} else
start_threshold = (double) rate * start_delay / 1000000;
if (start_threshold < 1)
start_threshold = 1;
if (start_threshold > n)
start_threshold = n;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err >= 0);
if (stop_delay <= 0)
stop_threshold = buffer_size + (double) rate * stop_delay / 1000000;
else
stop_threshold = (double) rate * stop_delay / 1000000;
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
assert(err >= 0);
if (snd_pcm_sw_params(handle, swparams) < 0) {
error("unable to install sw params:");
snd_pcm_sw_params_dump(swparams, log);
exit(EXIT_FAILURE);
}
if (verbose)
snd_pcm_dump(handle, log);
bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
bits_per_frame = bits_per_sample * hwparams.channels;
chunk_bytes = chunk_size * bits_per_frame / 8;
audiobuf = realloc(audiobuf, chunk_bytes);
if (audiobuf == NULL) {
error("not enough memory");
exit(EXIT_FAILURE);
}
// fprintf(stderr, "real chunk_size = %i, frags = %i, total = %i\n", chunk_size, setup.buf.block.frags, setup.buf.block.frags * chunk_size);
/* stereo VU-meter isn't always available... */
if (vumeter == VUMETER_STEREO) {
if (hwparams.channels != 2 || !interleaved || verbose > 2)
vumeter = VUMETER_MONO;
}
/* show mmap buffer arragment */
if (mmap_flag && verbose) {
const snd_pcm_channel_area_t *areas;
snd_pcm_uframes_t offset;
int i;
err = snd_pcm_mmap_begin(handle, &areas, &offset, &chunk_size);
if (err < 0) {
error("snd_pcm_mmap_begin problem: %s", snd_strerror(err));
exit(EXIT_FAILURE);
}
for (i = 0; i < hwparams.channels; i++)
fprintf(stderr, "mmap_area[%i] = %p,%u,%u (%u)\n", i, areas[i].addr, areas[i].first, areas[i].step, snd_pcm_format_physical_width(hwparams.format));
/* not required, but for sure */
snd_pcm_mmap_commit(handle, offset, 0);
}
buffer_frames = buffer_size; /* for position test */
}
bool open()
{
// Open the Alsa playback device.
int err=-1,count=0;
unsigned int freakuency = frequency;
while((count < 5) && (err < 0)) {
err = snd_pcm_open
( &handle, ALSA_OUTPUT_NAME, SND_PCM_STREAM_PLAYBACK, 0 );
if(err < 0) {
count++;
qWarning()<<"QAudioOutput::open() err="<<err<<", count="<<count;
}
}
if (( err < 0)||(handle == 0)) {
qWarning( "QAudioOuput: snd_pcm_open: error %d", err );
return false;
}
snd_pcm_nonblock( handle, 0 );
// Set the desired HW parameters.
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca( &hwparams );
err = snd_pcm_hw_params_any( handle, hwparams );
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_any: err %d", err);
return false;
}
err = snd_pcm_hw_params_set_access( handle, hwparams, access );
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_access: err %d", err);
return false;
}
err = snd_pcm_hw_params_set_format( handle, hwparams,
( bitsPerSample == 16 ? SND_PCM_FORMAT_S16
: SND_PCM_FORMAT_U8 ) );
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_format: err %d", err);
return false;
}
err = snd_pcm_hw_params_set_channels
( handle, hwparams, (unsigned int)channels );
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_channels: err %d", err);
return false;
}
err = snd_pcm_hw_params_set_rate_near
( handle, hwparams, &freakuency, 0 );
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_rate_near: err %d", err);
return false;
}
#ifndef ALSA_BUFFER_SIZE
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_buffer_time_near: err %d",err);
}
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_get_buffer_size: err %d",err);
}
#else
buffer_size = ALSA_BUFFER_SIZE;
err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_buffer_time_near: err %d",err);
}
#endif
#ifndef ALSA_PERIOD_SIZE
period_time = buffer_time/4;
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_period_time_near: err %d",err);
}
#else
period_size = ALSA_PERIOD_SIZE;
err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, 0);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params_set_period_size_near: err %d",err);
}
#endif
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_hw_params: err %d", err);
return false;
}
int dir;
unsigned int vval, vval2;
snd_pcm_access_t val;
snd_pcm_format_t fval;
snd_pcm_subformat_t sval;
qLog(QAudioOutput) << "PCM handle name =" << snd_pcm_name(handle);
qLog(QAudioOutput) << "PCM state =" << snd_pcm_state_name(snd_pcm_state(handle));
snd_pcm_hw_params_get_access(hwparams,&val);
vval = (unsigned int)val;
if ( (int)vval != (int)access ) {
qWarning( "QAudioInput: access type not set, want %s got %s",
snd_pcm_access_name((snd_pcm_access_t)access),
snd_pcm_access_name((snd_pcm_access_t)vval) );
access = (snd_pcm_access_t)vval;
}
qLog(QAudioOutput) << "access type =" << snd_pcm_access_name((snd_pcm_access_t)vval);
snd_pcm_hw_params_get_format(hwparams, &fval);
vval = (unsigned int)fval;
if ( (int)vval != (int)format ) {
qWarning( "QAudioInput: format type not set, want %s got %s",
snd_pcm_format_name((snd_pcm_format_t)format),
snd_pcm_format_name((snd_pcm_format_t)vval) );
format = (snd_pcm_format_t)vval;
}
qLog(QAudioOutput) << QString("format = '%1' (%2)").arg(snd_pcm_format_name((snd_pcm_format_t)vval))
.arg(snd_pcm_format_description((snd_pcm_format_t)vval))
.toLatin1().constData();
snd_pcm_hw_params_get_subformat(hwparams,&sval);
vval = (unsigned int)sval;
qLog(QAudioOutput) << QString("subformat = '%1' (%2)").arg(snd_pcm_subformat_name((snd_pcm_subformat_t)vval))
.arg(snd_pcm_subformat_description((snd_pcm_subformat_t)vval))
.toLatin1().constData();
snd_pcm_hw_params_get_channels(hwparams, &vval);
if ( (int)vval != (int)channels ) {
qWarning( "QAudioInput: channels type not set, want %d got %d",channels,vval);
channels = vval;
}
qLog(QAudioOutput) << "channels =" << vval;
snd_pcm_hw_params_get_rate(hwparams, &vval, &dir);
if ( (int)vval != (int)frequency ) {
qWarning( "QAudioInput: frequency type not set, want %d got %d",frequency,vval);
frequency = vval;
}
qLog(QAudioOutput) << "rate =" << vval << "bps";
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
qLog(QAudioOutput) << "period time =" << period_time << "us";
snd_pcm_hw_params_get_period_size(hwparams,&period_size, &dir);
qLog(QAudioOutput) << "period size =" << (int)period_size << "frames";
qLog(QAudioOutput) << "buffer time =" << (int)buffer_time << "us";
qLog(QAudioOutput) << "buffer size =" << (int)buffer_size << "frames";
snd_pcm_hw_params_get_periods(hwparams, &vval, &dir);
periods = vval;
qLog(QAudioOutput) << "periods per buffer =" << vval << "frames";
snd_pcm_hw_params_get_rate_numden(hwparams, &vval, &vval2);
qLog(QAudioOutput) << QString("exact rate = %1/%2 bps").arg(vval).arg(vval2).toLatin1().constData();
vval = snd_pcm_hw_params_get_sbits(hwparams);
qLog(QAudioOutput) << "significant bits =" << vval;
vval = snd_pcm_hw_params_is_batch(hwparams);
qLog(QAudioOutput) << "is batch =" << vval;
vval = snd_pcm_hw_params_is_block_transfer(hwparams);
qLog(QAudioOutput) << "is block transfer =" << vval;
vval = snd_pcm_hw_params_is_double(hwparams);
qLog(QAudioOutput) << "is double =" << vval;
vval = snd_pcm_hw_params_is_half_duplex(hwparams);
qLog(QAudioOutput) << "is half duplex =" << vval;
vval = snd_pcm_hw_params_is_joint_duplex(hwparams);
qLog(QAudioOutput) << "is joint duplex =" << vval;
vval = snd_pcm_hw_params_can_overrange(hwparams);
qLog(QAudioOutput) << "can overrange =" << vval;
vval = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams);
qLog(QAudioOutput) << "can mmap =" << vval;
vval = snd_pcm_hw_params_can_pause(hwparams);
qLog(QAudioOutput) << "can pause =" << vval;
vval = snd_pcm_hw_params_can_resume(hwparams);
qLog(QAudioOutput) << "can resume =" << vval;
vval = snd_pcm_hw_params_can_sync_start(hwparams);
qLog(QAudioOutput) << "can sync start =" << vval;
// Set the desired SW parameters.
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_sw_params_current: err %d",err);
}
err = snd_pcm_sw_params_set_start_threshold(handle,swparams,buffer_size);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_sw_params_set_start_threshold: err %d",err);
}
err = snd_pcm_sw_params_set_stop_threshold(handle,swparams,buffer_size);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_sw_params_set_stop_threshold: err %d",err);
}
err = snd_pcm_sw_params_set_avail_min(handle, swparams,period_size);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_sw_params_set_avail_min: err %d",err);
}
err = snd_pcm_sw_params(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioOutput: snd_pcm_sw_params: err %d",err);
}
// Prepare for audio output.
snd_pcm_prepare( handle );
return true;
}
static void set_params(void)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size;
int err;
size_t n;
snd_pcm_uframes_t xfer_align;
unsigned int rate;
snd_pcm_uframes_t start_threshold, stop_threshold;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
error(_("Broken configuration for this PCM: no configurations available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
error(_("Access type not available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
if (err < 0) {
error(_("Sample format non available"));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
if (err < 0) {
error(_("Channels count non available"));
exit(EXIT_FAILURE);
}
#if 0
err = snd_pcm_hw_params_set_periods_min(handle, params, 2);
assert(err >= 0);
#endif
rate = hwparams.rate;
err = snd_pcm_hw_params_set_rate_near(handle, params, &hwparams.rate, 0);
assert(err >= 0);
if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) {
if (!quiet_mode) {
char plugex[64];
const char *pcmname = snd_pcm_name(handle);
fprintf(stderr, _("Warning: rate is not accurate (requested = %iHz, got = %iHz)\n"), rate, hwparams.rate);
if (! pcmname || strchr(snd_pcm_name(handle), ':'))
*plugex = 0;
else
snprintf(plugex, sizeof(plugex), "(-Dplug:%s)",
snd_pcm_name(handle));
fprintf(stderr, _(" please, try the plug plugin %s\n"),
plugex);
}
}
rate = hwparams.rate;
if (buffer_time == 0 && buffer_frames == 0) {
err = snd_pcm_hw_params_get_buffer_time_max(params,
&buffer_time, 0);
assert(err >= 0);
if (buffer_time > 500000)
buffer_time = 500000;
}
if (period_time == 0 && period_frames == 0) {
if (buffer_time > 0)
period_time = buffer_time / 4;
else
period_frames = buffer_frames / 4;
}
if (period_time > 0)
err = snd_pcm_hw_params_set_period_time_near(handle, params,
&period_time, 0);
else
err = snd_pcm_hw_params_set_period_size_near(handle, params,
&period_frames, 0);
assert(err >= 0);
if (buffer_time > 0) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
&buffer_time, 0);
} else {
err = snd_pcm_hw_params_set_buffer_size_near(handle, params,
&buffer_frames);
}
assert(err >= 0);
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
error(_("Unable to install hw params:"));
snd_pcm_hw_params_dump(params, log);
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
if (chunk_size == buffer_size) {
error(_("Can't use period equal to buffer size (%lu == %lu)"),
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
err = snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
if (err < 0) {
error(_("Unable to obtain xfer align\n"));
exit(EXIT_FAILURE);
}
if (sleep_min)
xfer_align = 1;
err = snd_pcm_sw_params_set_sleep_min(handle, swparams,
sleep_min);
assert(err >= 0);
if (avail_min < 0)
n = chunk_size;
else
n = (double) rate * avail_min / 1000000;
err = snd_pcm_sw_params_set_avail_min(handle, swparams, n);
// round up to closest transfer boundary
n = (buffer_size / xfer_align) * xfer_align;
start_threshold = n;
if (start_threshold < 1)
start_threshold = 1;
if (start_threshold > n)
start_threshold = n;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err >= 0);
stop_threshold = buffer_size;
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
assert(err >= 0);
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, xfer_align);
assert(err >= 0);
if (snd_pcm_sw_params(handle, swparams) < 0) {
error(_("unable to install sw params:"));
snd_pcm_sw_params_dump(swparams, log);
exit(EXIT_FAILURE);
}
bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
bits_per_frame = bits_per_sample * hwparams.channels;
chunk_bytes = chunk_size * bits_per_frame / 8;
audiobuf = realloc(audiobuf, chunk_bytes);
if (audiobuf == NULL) {
error(_("not enough memory"));
exit(EXIT_FAILURE);
}
// fprintf(stderr, "real chunk_size = %i, frags = %i, total = %i\n", chunk_size, setup.buf.block.frags, setup.buf.block.frags * chunk_size);
}
int ai_alsa_setup(audio_in_t *ai)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
int buffer_size;
int err;
unsigned int rate;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(ai->alsa.handle, params);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PcmBrokenConfig);
return -1;
}
err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableAccessType);
return -1;
}
err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableSampleFmt);
return -1;
}
err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
if (err < 0) {
ai->channels = snd_pcm_hw_params_get_channels(params);
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableChanCount,
ai->channels);
} else {
ai->channels = ai->req_channels;
}
err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0);
assert(err >= 0);
rate = err;
ai->samplerate = rate;
ai->alsa.buffer_time = 1000000;
ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
ai->alsa.buffer_time, 0);
assert(ai->alsa.buffer_time >= 0);
ai->alsa.period_time = ai->alsa.buffer_time / 4;
ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
ai->alsa.period_time, 0);
assert(ai->alsa.period_time >= 0);
err = snd_pcm_hw_params(ai->alsa.handle, params);
if (err < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallHWParams);
snd_pcm_hw_params_dump(params, ai->alsa.log);
return -1;
}
ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0);
buffer_size = snd_pcm_hw_params_get_buffer_size(params);
if (ai->alsa.chunk_size == buffer_size) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PeriodEqualsBufferSize, ai->alsa.chunk_size, (long)buffer_size);
return -1;
}
snd_pcm_sw_params_current(ai->alsa.handle, swparams);
err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0);
assert(err >= 0);
err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
assert(err >= 0);
err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
assert(err >= 0);
err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
assert(err >= 0);
assert(err >= 0);
if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallSWParams);
snd_pcm_sw_params_dump(swparams, ai->alsa.log);
return -1;
}
if (mp_msg_test(MSGT_TV, MSGL_V)) {
snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
}
ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
ai->samplesize = ai->alsa.bits_per_sample;
ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
return 0;
}
int main(int argc, char *argv[]) {
const char *dev;
int r, cap, count = 0;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_status_t *status;
snd_pcm_t *pcm;
unsigned rate = 44100;
unsigned periods = 2;
snd_pcm_uframes_t boundary, buffer_size = 44100/10; /* 100s */
int dir = 1;
struct timespec start, last_timestamp = { 0, 0 };
uint64_t start_us, last_us = 0;
snd_pcm_sframes_t last_avail = 0, last_delay = 0;
struct pollfd *pollfds;
int n_pollfd;
int64_t sample_count = 0;
struct sched_param sp;
r = -1;
#ifdef _POSIX_PRIORITY_SCHEDULING
sp.sched_priority = 5;
r = pthread_setschedparam(pthread_self(), SCHED_RR, &sp);
#endif
if (r)
printf("Could not get RT prio. :(\n");
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_status_alloca(&status);
r = clock_gettime(CLOCK_MONOTONIC, &start);
assert(r == 0);
start_us = timespec_us(&start);
dev = argc > 1 ? argv[1] : "front:AudioPCI";
cap = argc > 2 ? atoi(argv[2]) : 0;
if (cap == 0)
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_PLAYBACK, 0);
else
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_CAPTURE, 0);
assert(r == 0);
r = snd_pcm_hw_params_any(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
assert(r == 0);
r = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, NULL);
assert(r == 0);
r = snd_pcm_hw_params_set_channels(pcm, hwparams, 2);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_integer(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_near(pcm, hwparams, &periods, &dir);
assert(r == 0);
r = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_hw_params(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_current(pcm, hwparams);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
if (cap == 0)
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 1);
else
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_sw_params_set_period_event(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_sw_params_set_start_threshold(pcm, swparams, buffer_size);
assert(r == 0);
r = snd_pcm_sw_params_get_boundary(swparams, &boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_stop_threshold(pcm, swparams, boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_tstamp_mode(pcm, swparams, SND_PCM_TSTAMP_ENABLE);
assert(r == 0);
r = snd_pcm_sw_params(pcm, swparams);
assert(r == 0);
r = snd_pcm_prepare(pcm);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
/* assert(snd_pcm_hw_params_is_monotonic(hwparams) > 0); */
n_pollfd = snd_pcm_poll_descriptors_count(pcm);
assert(n_pollfd > 0);
pollfds = malloc(sizeof(struct pollfd) * n_pollfd);
assert(pollfds);
r = snd_pcm_poll_descriptors(pcm, pollfds, n_pollfd);
assert(r == n_pollfd);
printf("Starting. Buffer size is %u frames\n", (unsigned int) buffer_size);
if (cap) {
r = snd_pcm_start(pcm);
assert(r == 0);
}
for (;;) {
snd_pcm_sframes_t avail, delay;
struct timespec now, timestamp;
unsigned short revents;
int handled = 0;
uint64_t now_us, timestamp_us;
snd_pcm_state_t state;
unsigned long long pos;
r = poll(pollfds, n_pollfd, 0);
assert(r >= 0);
r = snd_pcm_poll_descriptors_revents(pcm, pollfds, n_pollfd, &revents);
assert(r == 0);
if (cap == 0)
assert((revents & ~POLLOUT) == 0);
else
assert((revents & ~POLLIN) == 0);
avail = snd_pcm_avail(pcm);
assert(avail >= 0);
r = snd_pcm_status(pcm, status);
assert(r == 0);
/* This assertion fails from time to time. ALSA seems to be broken */
/* assert(avail == (snd_pcm_sframes_t) snd_pcm_status_get_avail(status)); */
/* printf("%lu %lu\n", (unsigned long) avail, (unsigned long) snd_pcm_status_get_avail(status)); */
snd_pcm_status_get_htstamp(status, ×tamp);
delay = snd_pcm_status_get_delay(status);
state = snd_pcm_status_get_state(status);
r = clock_gettime(CLOCK_MONOTONIC, &now);
assert(r == 0);
assert(!revents || avail > 0);
if ((!cap && avail) || (cap && (unsigned)avail >= buffer_size)) {
snd_pcm_sframes_t sframes;
static const uint16_t psamples[2] = { 0, 0 };
uint16_t csamples[2];
if (cap == 0)
sframes = snd_pcm_writei(pcm, psamples, 1);
else
sframes = snd_pcm_readi(pcm, csamples, 1);
assert(sframes == 1);
handled = 1;
sample_count++;
}
if (!handled &&
memcmp(×tamp, &last_timestamp, sizeof(timestamp)) == 0 &&
avail == last_avail &&
delay == last_delay) {
/* This is boring */
continue;
}
now_us = timespec_us(&now);
timestamp_us = timespec_us(×tamp);
if (cap == 0)
pos = (unsigned long long) ((sample_count - handled - delay) * 1000000LU / 44100);
else
pos = (unsigned long long) ((sample_count - handled + delay) * 1000000LU / 44100);
if (count++ % 50 == 0)
printf("Elapsed\tCPU\tALSA\tPos\tSamples\tavail\tdelay\trevents\thandled\tstate\n");
printf("%llu\t%llu\t%llu\t%llu\t%llu\t%li\t%li\t%i\t%i\t%i\n",
(unsigned long long) (now_us - last_us),
(unsigned long long) (now_us - start_us),
(unsigned long long) (timestamp_us ? timestamp_us - start_us : 0),
pos,
(unsigned long long) sample_count,
(signed long) avail,
(signed long) delay,
revents,
handled,
state);
if (cap == 0)
/** When this assert is hit, most likely something bad
* happened, i.e. the avail jumped suddenly. */
assert((unsigned) avail <= buffer_size);
last_avail = avail;
last_delay = delay;
last_timestamp = timestamp;
last_us = now_us;
}
return 0;
}
int
ga_alsa_set_param(struct Xcap_alsa_param *param) {
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
size_t bits_per_sample;
unsigned int rate;
unsigned int buffer_time = 500000; // in the unit of microsecond
unsigned int period_time = 125000; // = buffer_time/4;
int monotonic = 0;
snd_pcm_uframes_t start_threshold, stop_threshold;
int err;
//
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
if((err = snd_pcm_hw_params_any(param->handle, hwparams)) < 0) {
ga_error("ALSA: set_param - no configurations available\n");
return -1;
}
if((err = snd_pcm_hw_params_set_access(param->handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
ga_error("ALSA: set_param - access type (interleaved) not available\n");
return -1;
}
if((err = snd_pcm_hw_params_set_format(param->handle, hwparams, param->format)) < 0) {
ga_error("ALSA: set_param - unsupported sample format.\n");
return -1;
}
if((err = snd_pcm_hw_params_set_channels(param->handle, hwparams, param->channels)) < 0) {
ga_error("ALSA: set_param - channles count not available\n");
return -1;
}
rate = param->samplerate;
if((err = snd_pcm_hw_params_set_rate_near(param->handle, hwparams, &rate, 0)) < 0) {
ga_error("ALSA: set_param - set rate failed.\n");
return -1;
}
if((double)param->samplerate*1.05 < rate || (double)param->samplerate*0.95 > rate) {
ga_error("ALSA: set_param/warning - inaccurate rate (req=%iHz, got=%iHz)\n", param->samplerate, rate);
}
//
period_time = buffer_time/4;
if((err = snd_pcm_hw_params_set_period_time_near(param->handle, hwparams, &period_time, 0)) < 0) {
ga_error("ALSA: set_param - set period time failed.\n");
return -1;
}
if((err = snd_pcm_hw_params_set_buffer_time_near(param->handle, hwparams, &buffer_time, 0)) < 0) {
ga_error("ALSA: set_param - set buffer time failed.\n");
return -1;
}
//
monotonic = snd_pcm_hw_params_is_monotonic(hwparams);
if((err = snd_pcm_hw_params(param->handle, hwparams)) < 0) {
ga_error("ALSA: set_param - unable to install hw params:");
snd_pcm_hw_params_dump(hwparams, sndlog);
return -1;
}
snd_pcm_hw_params_get_period_size(hwparams, ¶m->chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(hwparams, ¶m->buffer_size);
if(param->chunk_size == param->buffer_size) {
ga_error("ALSA: set_param - cannot use period equal to buffer size (%lu==%lu)\n",
param->chunk_size, param->buffer_size);
return -1;
}
//
snd_pcm_sw_params_current(param->handle, swparams);
err = snd_pcm_sw_params_set_avail_min(param->handle, swparams, param->chunk_size);
// start_delay = 1 for capture
start_threshold = (double) param->samplerate * /*start_delay=*/ 1 / 1000000;
if(start_threshold < 1) start_threshold = 1;
if(start_threshold > param->buffer_size) start_threshold = param->buffer_size;
if((err = snd_pcm_sw_params_set_start_threshold(param->handle, swparams, start_threshold)) < 0) {
ga_error("ALSA: set_param - set start threshold failed.\n");
return -1;
}
// stop_delay = 0
stop_threshold = param->buffer_size;
if((err = snd_pcm_sw_params_set_stop_threshold(param->handle, swparams, stop_threshold)) < 0) {
ga_error("ALSA: set_param - set stop threshold failed.\n");
return -1;
}
//
if(snd_pcm_sw_params(param->handle, swparams) < 0) {
ga_error("ALSA: set_param - unable to install sw params:");
snd_pcm_sw_params_dump(swparams, sndlog);
return -1;
}
bits_per_sample = snd_pcm_format_physical_width(param->format);
if(param->bits_per_sample != bits_per_sample) {
ga_error("ALSA: set_param - BPS/HW configuration mismatched %d != %d)\n",
param->bits_per_sample, bits_per_sample);
}
param->bits_per_frame = param->bits_per_sample * param->channels;
param->chunk_bytes = param->chunk_size * param->bits_per_frame / 8;
return 0;
}
static ALCboolean alsa_reset_playback(ALCdevice *device)
{
alsa_data *data = (alsa_data*)device->ExtraData;
snd_pcm_uframes_t periodSizeInFrames;
unsigned int periodLen, bufferLen;
snd_pcm_sw_params_t *sp = NULL;
snd_pcm_hw_params_t *hp = NULL;
snd_pcm_access_t access;
snd_pcm_format_t format;
unsigned int periods;
unsigned int rate;
const char *funcerr;
int allowmmap;
int err;
format = -1;
switch(device->FmtType)
{
case DevFmtByte:
format = SND_PCM_FORMAT_S8;
break;
case DevFmtUByte:
format = SND_PCM_FORMAT_U8;
break;
case DevFmtShort:
format = SND_PCM_FORMAT_S16;
break;
case DevFmtUShort:
format = SND_PCM_FORMAT_U16;
break;
case DevFmtInt:
format = SND_PCM_FORMAT_S32;
break;
case DevFmtUInt:
format = SND_PCM_FORMAT_U32;
break;
case DevFmtFloat:
format = SND_PCM_FORMAT_FLOAT;
break;
}
allowmmap = GetConfigValueBool("alsa", "mmap", 1);
periods = device->NumUpdates;
periodLen = (ALuint64)device->UpdateSize * 1000000 / device->Frequency;
bufferLen = periodLen * periods;
rate = device->Frequency;
snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
/* set interleaved access */
if(!allowmmap || snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0)
{
if(periods > 2)
{
periods--;
bufferLen = periodLen * periods;
}
CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
}
/* test and set format (implicitly sets sample bits) */
if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) < 0)
{
static const struct {
snd_pcm_format_t format;
enum DevFmtType fmttype;
} formatlist[] = {
{ SND_PCM_FORMAT_FLOAT, DevFmtFloat },
{ SND_PCM_FORMAT_S32, DevFmtInt },
{ SND_PCM_FORMAT_U32, DevFmtUInt },
{ SND_PCM_FORMAT_S16, DevFmtShort },
{ SND_PCM_FORMAT_U16, DevFmtUShort },
{ SND_PCM_FORMAT_S8, DevFmtByte },
{ SND_PCM_FORMAT_U8, DevFmtUByte },
};
size_t k;
for(k = 0;k < COUNTOF(formatlist);k++)
{
format = formatlist[k].format;
if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) >= 0)
{
device->FmtType = formatlist[k].fmttype;
break;
}
}
}
CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
/* test and set channels (implicitly sets frame bits) */
if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)) < 0)
{
static const enum DevFmtChannels channellist[] = {
DevFmtStereo,
DevFmtQuad,
DevFmtX51,
DevFmtX71,
DevFmtMono,
};
size_t k;
for(k = 0;k < COUNTOF(channellist);k++)
{
if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(channellist[k])) >= 0)
{
device->FmtChans = channellist[k];
break;
}
}
}
CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)));
/* set rate (implicitly constrains period/buffer parameters) */
if(snd_pcm_hw_params_set_rate_resample(data->pcmHandle, hp, 0) < 0)
ERR("Failed to disable ALSA resampler\n");
CHECK(snd_pcm_hw_params_set_rate_near(data->pcmHandle, hp, &rate, NULL));
/* set buffer time (implicitly constrains period/buffer parameters) */
CHECK(snd_pcm_hw_params_set_buffer_time_near(data->pcmHandle, hp, &bufferLen, NULL));
/* set period time (implicitly sets buffer size/bytes/time and period size/bytes) */
CHECK(snd_pcm_hw_params_set_period_time_near(data->pcmHandle, hp, &periodLen, NULL));
/* install and prepare hardware configuration */
CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
/* retrieve configuration info */
CHECK(snd_pcm_hw_params_get_access(hp, &access));
CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL));
CHECK(snd_pcm_hw_params_get_periods(hp, &periods, NULL));
snd_pcm_hw_params_free(hp);
hp = NULL;
snd_pcm_sw_params_malloc(&sp);
CHECK(snd_pcm_sw_params_current(data->pcmHandle, sp));
CHECK(snd_pcm_sw_params_set_avail_min(data->pcmHandle, sp, periodSizeInFrames));
CHECK(snd_pcm_sw_params_set_stop_threshold(data->pcmHandle, sp, periodSizeInFrames*periods));
CHECK(snd_pcm_sw_params(data->pcmHandle, sp));
#undef CHECK
snd_pcm_sw_params_free(sp);
sp = NULL;
/* Increase periods by one, since the temp buffer counts as an extra
* period */
if(access == SND_PCM_ACCESS_RW_INTERLEAVED)
device->NumUpdates = periods+1;
else
device->NumUpdates = periods;
device->UpdateSize = periodSizeInFrames;
device->Frequency = rate;
SetDefaultChannelOrder(device);
return ALC_TRUE;
error:
ERR("%s failed: %s\n", funcerr, snd_strerror(err));
if(hp) snd_pcm_hw_params_free(hp);
if(sp) snd_pcm_sw_params_free(sp);
return ALC_FALSE;
}
/**
* Create and initialize Alsa audio output device with given parameters.
*
* @param Parameters - optional parameters
* @throws AudioOutputException if output device cannot be opened
*/
AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, true, 1, 0) {
pcm_handle = NULL;
stream = SND_PCM_STREAM_PLAYBACK;
this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["CHANNELS"])->ValueAsInt();
this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["SAMPLERATE"])->ValueAsInt();
this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["FRAGMENTSIZE"])->ValueAsInt();
uint Fragments = ((DeviceCreationParameterInt*)Parameters["FRAGMENTS"])->ValueAsInt();
String Card = ((DeviceCreationParameterString*)Parameters["CARD"])->ValueAsString();
dmsg(2,("Checking if hw parameters supported...\n"));
if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
pcm_name = "hw:" + Card;
}
else {
fprintf(stderr, "Warning: your soundcard doesn't support chosen hardware parameters; ");
fprintf(stderr, "trying to compensate support lack with plughw...");
fflush(stdout);
pcm_name = "plughw:" + Card;
}
dmsg(2,("HW check completed.\n"));
int err;
snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
/* Open PCM. The last parameter of this function is the mode. */
/* If this is set to 0, the standard mode is used. Possible */
/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
/* If SND_PCM_NONBLOCK is used, read / write access to the */
/* PCM device will return immediately. If SND_PCM_ASYNC is */
/* specified, SIGIO will be emitted whenever a period has */
/* been completely processed by the soundcard. */
if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
}
/* Set access type. This can be either */
/* SND_PCM_ACCESS_RW_INTERLEAVED or */
/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
}
/* Set sample format */
#if WORDS_BIGENDIAN
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
#else // little endian
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
}
int dir = 0;
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
#if ALSA_MAJOR > 0
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
#else
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
}
/* Set number of periods. Periods used to be called fragments. */
if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
throw AudioOutputException(String("Error setting number of ") + ToString(Fragments) + " periods: " + snd_strerror(err));
}
/* Set buffer size (in frames). The resulting latency is given by */
/* latency = periodsize * periods / (rate * bytes_per_frame) */
if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
}
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_malloc(&swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
}
if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
}
if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
}
// allocate Alsa output buffer
pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
// create audio channels for this audio device to which the sampler engines can write to
for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(i, FragmentSize));
if (((DeviceCreationParameterBool*)Parameters["ACTIVE"])->ValueAsBool()) {
Play();
}
}
/**
* Allocate the memory-mapped buffer for direct sound, and set up the
* callback.
*/
static int DSDB_CreateMMAP(IDsDriverBufferImpl* pdbi)
{
snd_pcm_t *pcm = pdbi->pcm;
snd_pcm_format_t format;
snd_pcm_uframes_t frames, ofs, avail, psize, boundary;
unsigned int channels, bits_per_sample, bits_per_frame;
int err, mmap_mode;
const snd_pcm_channel_area_t *areas;
snd_pcm_hw_params_t *hw_params = pdbi->hw_params;
snd_pcm_sw_params_t *sw_params = pdbi->sw_params;
void *buf;
mmap_mode = snd_pcm_type(pcm);
if (mmap_mode == SND_PCM_TYPE_HW)
TRACE("mmap'd buffer is a direct hardware buffer.\n");
else if (mmap_mode == SND_PCM_TYPE_DMIX)
TRACE("mmap'd buffer is an ALSA dmix buffer\n");
else
TRACE("mmap'd buffer is an ALSA type %d buffer\n", mmap_mode);
err = snd_pcm_hw_params_get_period_size(hw_params, &psize, NULL);
err = snd_pcm_hw_params_get_format(hw_params, &format);
err = snd_pcm_hw_params_get_buffer_size(hw_params, &frames);
err = snd_pcm_hw_params_get_channels(hw_params, &channels);
bits_per_sample = snd_pcm_format_physical_width(format);
bits_per_frame = bits_per_sample * channels;
if (TRACE_ON(dsalsa))
ALSA_TraceParameters(hw_params, NULL, FALSE);
TRACE("format=%s frames=%ld channels=%d bits_per_sample=%d bits_per_frame=%d\n",
snd_pcm_format_name(format), frames, channels, bits_per_sample, bits_per_frame);
pdbi->mmap_buflen_frames = frames;
pdbi->mmap_buflen_bytes = snd_pcm_frames_to_bytes( pcm, frames );
snd_pcm_sw_params_current(pcm, sw_params);
snd_pcm_sw_params_set_start_threshold(pcm, sw_params, 0);
snd_pcm_sw_params_get_boundary(sw_params, &boundary);
snd_pcm_sw_params_set_stop_threshold(pcm, sw_params, boundary);
snd_pcm_sw_params_set_silence_threshold(pcm, sw_params, boundary);
snd_pcm_sw_params_set_silence_size(pcm, sw_params, 0);
snd_pcm_sw_params_set_avail_min(pcm, sw_params, 0);
err = snd_pcm_sw_params(pcm, sw_params);
avail = snd_pcm_avail_update(pcm);
if ((snd_pcm_sframes_t)avail < 0)
{
ERR("No buffer is available: %s.\n", snd_strerror(avail));
return DSERR_GENERIC;
}
if (!pdbi->mmap)
{
buf = pdbi->mmap_buffer = HeapAlloc(GetProcessHeap(), 0, pdbi->mmap_buflen_bytes);
if (!buf)
return DSERR_OUTOFMEMORY;
snd_pcm_format_set_silence(format, buf, pdbi->mmap_buflen_frames);
pdbi->mmap_pos = 0;
}
else
{
err = snd_pcm_mmap_begin(pcm, &areas, &ofs, &avail);
if ( err < 0 )
{
ERR("Can't map sound device for direct access: %s/%d\n", snd_strerror(err), err);
return DSERR_GENERIC;
}
snd_pcm_format_set_silence(format, areas->addr, pdbi->mmap_buflen_frames);
pdbi->mmap_pos = ofs + snd_pcm_mmap_commit(pcm, ofs, 0);
pdbi->mmap_buffer = areas->addr;
}
TRACE("created mmap buffer of %ld frames (%d bytes) at %p\n",
frames, pdbi->mmap_buflen_bytes, pdbi->mmap_buffer);
return DS_OK;
}
bool setParameters (unsigned int sampleRate, int numChannels, int bufferSize)
{
if (handle == nullptr)
return false;
JUCE_ALSA_LOG ("ALSADevice::setParameters(" << deviceID << ", "
<< (int) sampleRate << ", " << numChannels << ", " << bufferSize << ")");
snd_pcm_hw_params_t* hwParams;
snd_pcm_hw_params_alloca (&hwParams);
if (snd_pcm_hw_params_any (handle, hwParams) < 0)
{
// this is the error message that aplay returns when an error happens here,
// it is a bit more explicit that "Invalid parameter"
error = "Broken configuration for this PCM: no configurations available";
return false;
}
if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw..
isInterleaved = true;
else if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_NONINTERLEAVED) >= 0)
isInterleaved = false;
else
{
jassertfalse;
return false;
}
enum { isFloatBit = 1 << 16, isLittleEndianBit = 1 << 17, onlyUseLower24Bits = 1 << 18 };
const int formatsToTry[] = { SND_PCM_FORMAT_FLOAT_LE, 32 | isFloatBit | isLittleEndianBit,
SND_PCM_FORMAT_FLOAT_BE, 32 | isFloatBit,
SND_PCM_FORMAT_S32_LE, 32 | isLittleEndianBit,
SND_PCM_FORMAT_S32_BE, 32,
SND_PCM_FORMAT_S24_3LE, 24 | isLittleEndianBit,
SND_PCM_FORMAT_S24_3BE, 24,
SND_PCM_FORMAT_S24_LE, 32 | isLittleEndianBit | onlyUseLower24Bits,
SND_PCM_FORMAT_S16_LE, 16 | isLittleEndianBit,
SND_PCM_FORMAT_S16_BE, 16 };
bitDepth = 0;
for (int i = 0; i < numElementsInArray (formatsToTry); i += 2)
{
if (snd_pcm_hw_params_set_format (handle, hwParams, (_snd_pcm_format) formatsToTry [i]) >= 0)
{
const int type = formatsToTry [i + 1];
bitDepth = type & 255;
converter.reset (createConverter (isInput, bitDepth,
(type & isFloatBit) != 0,
(type & isLittleEndianBit) != 0,
(type & onlyUseLower24Bits) != 0,
numChannels,
isInterleaved));
break;
}
}
if (bitDepth == 0)
{
error = "device doesn't support a compatible PCM format";
JUCE_ALSA_LOG ("Error: " + error);
return false;
}
int dir = 0;
unsigned int periods = 4;
snd_pcm_uframes_t samplesPerPeriod = (snd_pcm_uframes_t) bufferSize;
if (JUCE_ALSA_FAILED (snd_pcm_hw_params_set_rate_near (handle, hwParams, &sampleRate, 0))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_channels (handle, hwParams, (unsigned int ) numChannels))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_periods_near (handle, hwParams, &periods, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_period_size_near (handle, hwParams, &samplesPerPeriod, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params (handle, hwParams)))
{
return false;
}
snd_pcm_uframes_t frames = 0;
if (JUCE_ALSA_FAILED (snd_pcm_hw_params_get_period_size (hwParams, &frames, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_get_periods (hwParams, &periods, &dir)))
latency = 0;
else
latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency..)
JUCE_ALSA_LOG ("frames: " << (int) frames << ", periods: " << (int) periods
<< ", samplesPerPeriod: " << (int) samplesPerPeriod);
snd_pcm_sw_params_t* swParams;
snd_pcm_sw_params_alloca (&swParams);
snd_pcm_uframes_t boundary;
if (JUCE_ALSA_FAILED (snd_pcm_sw_params_current (handle, swParams))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_get_boundary (swParams, &boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_silence_threshold (handle, swParams, 0))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_silence_size (handle, swParams, boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_start_threshold (handle, swParams, samplesPerPeriod))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_stop_threshold (handle, swParams, boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params (handle, swParams)))
{
return false;
}
#if JUCE_ALSA_LOGGING
// enable this to dump the config of the devices that get opened
snd_output_t* out;
snd_output_stdio_attach (&out, stderr, 0);
snd_pcm_hw_params_dump (hwParams, out);
snd_pcm_sw_params_dump (swParams, out);
#endif
numChannelsRunning = numChannels;
return true;
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
unsigned int alsa_buffer_time = 500000; /* 0.5 s */
unsigned int alsa_fragcount = 16;
int err;
int block;
strarg_t device;
snd_pcm_uframes_t chunk_size;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t boundary;
const opt_t subopts[] = {
{"block", OPT_ARG_BOOL, &block, NULL},
{"device", OPT_ARG_STR, &device, str_maxlen},
{NULL}
};
char alsa_device[ALSA_DEVICE_SIZE + 1];
// make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
channels, format);
alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
snd_lib_error_set_handler(alsa_error_handler);
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
switch (format)
{
case AF_FORMAT_S8:
alsa_format = SND_PCM_FORMAT_S8;
break;
case AF_FORMAT_U8:
alsa_format = SND_PCM_FORMAT_U8;
break;
case AF_FORMAT_U16_LE:
alsa_format = SND_PCM_FORMAT_U16_LE;
break;
case AF_FORMAT_U16_BE:
alsa_format = SND_PCM_FORMAT_U16_BE;
break;
case AF_FORMAT_AC3_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_IEC61937_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case AF_FORMAT_AC3_BE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_IEC61937_BE:
alsa_format = SND_PCM_FORMAT_S16_BE;
break;
case AF_FORMAT_U32_LE:
alsa_format = SND_PCM_FORMAT_U32_LE;
break;
case AF_FORMAT_U32_BE:
alsa_format = SND_PCM_FORMAT_U32_BE;
break;
case AF_FORMAT_S32_LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
case AF_FORMAT_S32_BE:
alsa_format = SND_PCM_FORMAT_S32_BE;
break;
case AF_FORMAT_U24_LE:
alsa_format = SND_PCM_FORMAT_U24_3LE;
break;
case AF_FORMAT_U24_BE:
alsa_format = SND_PCM_FORMAT_U24_3BE;
break;
case AF_FORMAT_S24_LE:
alsa_format = SND_PCM_FORMAT_S24_3LE;
break;
case AF_FORMAT_S24_BE:
alsa_format = SND_PCM_FORMAT_S24_3BE;
break;
case AF_FORMAT_FLOAT_LE:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AF_FORMAT_FLOAT_BE:
alsa_format = SND_PCM_FORMAT_FLOAT_BE;
break;
case AF_FORMAT_MU_LAW:
alsa_format = SND_PCM_FORMAT_MU_LAW;
break;
case AF_FORMAT_A_LAW:
alsa_format = SND_PCM_FORMAT_A_LAW;
break;
default:
alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
break;
}
//subdevice parsing
// set defaults
block = 1;
/* switch for spdif
* sets opening sequence for SPDIF
* sets also the playback and other switches 'on the fly'
* while opening the abstract alias for the spdif subdevice
* 'iec958'
*/
if (AF_FORMAT_IS_IEC61937(format)) {
device.str = "iec958";
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels);
}
else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
case 1:
case 2:
device.str = "default";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
break;
case 4:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
// hack - use the converter plugin
device.str = "plug:surround40";
else
device.str = "surround40";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
break;
case 6:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround51";
else
device.str = "surround51";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
break;
case 8:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround71";
else
device.str = "surround71";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
break;
default:
device.str = "default";
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
parse_device(alsa_device, device.str, device.len);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
if (!alsa_handler) {
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
mp_msg(MSGT_AO,MSGL_V,"alsa-init: opening device in %sblocking mode\n", block ? "" : "non-");
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
{
if (err != -EBUSY && !block) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device reopened in blocking mode\n");
}
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
snd_strerror(err));
return 0;
}
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
snd_strerror(err));
return 0;
}
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_INFO,
MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
else if (AF_FORMAT_IS_IEC61937(ao_data.format))
ao_data.format = AF_FORMAT_IEC61937_LE;
else
ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
snd_strerror(err));
return 0;
}
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler, since that allows users to choose the resampler,
even per file if desired */
#if SND_LIB_VERSION >= 0x010009
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
snd_strerror(err));
return 0;
}
bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
bytes_per_sample *= ao_data.channels;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
}
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
snd_strerror(err));
return 0;
}
// end setting hw-params
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
return 0;
}
else {
ao_data.buffersize = bufsize * bytes_per_sample;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
}
ao_data.outburst = chunk_size * bytes_per_sample;
/* setting software parameters */
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
snd_strerror(err));
return 0;
}
#else
boundary = 0x7fffffff;
#endif
/* start playing when one period has been written */
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
snd_strerror(err));
return 0;
}
/* disable underrun reporting */
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* play silence when there is an underrun */
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
return 1;
} // end init
static int sndo_pcm_initialize(sndo_pcm_t *pcm)
{
int err;
snd_pcm_uframes_t boundary;
snd_pcm_uframes_t p_period_size = ~0UL, c_period_size = ~0UL;
snd_pcm_uframes_t p_buffer_size = ~0UL, c_buffer_size = ~0UL;
if (pcm->playback) {
err = snd_pcm_hw_params(pcm->playback, pcm->p_hw_params);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_period_size(pcm->p_hw_params, &p_period_size, NULL);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_buffer_size(pcm->p_hw_params, &p_buffer_size);
if (err < 0)
return err;
}
if (pcm->capture) {
err = snd_pcm_hw_params(pcm->capture, pcm->c_hw_params);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_period_size(pcm->c_hw_params, &c_period_size, NULL);
if (err < 0)
return err;
err = snd_pcm_hw_params_get_buffer_size(pcm->c_hw_params, &c_buffer_size);
if (err < 0)
return err;
}
if (p_period_size < c_period_size)
pcm->transfer_block = p_period_size;
else
pcm->transfer_block = c_period_size;
if (p_buffer_size < c_buffer_size)
pcm->ring_size = p_buffer_size;
else
pcm->ring_size = c_buffer_size;
if (pcm->playback) {
err = snd_pcm_sw_params_get_boundary(pcm->p_sw_params, &boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_start_threshold(pcm->playback, pcm->p_sw_params, boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_stop_threshold(pcm->playback, pcm->p_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_xfer_align(pcm->playback, pcm->p_sw_params, 1);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_avail_min(pcm->playback, pcm->p_sw_params, pcm->transfer_block);
if (err < 0)
return err;
err = snd_pcm_sw_params(pcm->playback, pcm->p_sw_params);
if (err < 0)
return err;
}
if (pcm->capture) {
err = snd_pcm_sw_params_get_boundary(pcm->c_sw_params, &boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_start_threshold(pcm->capture, pcm->c_sw_params, boundary);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_stop_threshold(pcm->capture, pcm->c_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_xfer_align(pcm->capture, pcm->c_sw_params, 1);
if (err < 0)
return err;
err = snd_pcm_sw_params_set_avail_min(pcm->capture, pcm->c_sw_params, pcm->transfer_block);
if (err < 0)
return err;
err = snd_pcm_sw_params(pcm->capture, pcm->c_sw_params);
if (err < 0)
return err;
}
pcm->initialized = 1;
return 0;
}
status_t setSoftwareParams(alsa_handle_t *handle)
{
snd_pcm_sw_params_t * softwareParams;
int err;
snd_pcm_uframes_t bufferSize = 0;
snd_pcm_uframes_t periodSize = 0;
snd_pcm_uframes_t startThreshold, stopThreshold;
if (snd_pcm_sw_params_malloc(&softwareParams) < 0) {
LOG_ALWAYS_FATAL("Failed to allocate ALSA software parameters!");
return NO_INIT;
}
// Get the current software parameters
err = snd_pcm_sw_params_current(handle->handle, softwareParams);
if (err < 0) {
LOGE("Unable to get software parameters: %s", snd_strerror(err));
goto done;
}
// Configure ALSA to start the transfer when the buffer is almost full.
snd_pcm_get_params(handle->handle, &bufferSize, &periodSize);
if (handle->devices & AudioSystem::DEVICE_OUT_ALL) {
// For playback, configure ALSA to start the transfer when the
// buffer is full.
startThreshold = bufferSize - 1;
stopThreshold = bufferSize;
} else {
// For recording, configure ALSA to start the transfer on the
// first frame.
startThreshold = 1;
stopThreshold = bufferSize;
}
err = snd_pcm_sw_params_set_start_threshold(handle->handle, softwareParams,
startThreshold);
if (err < 0) {
LOGE("Unable to set start threshold to %lu frames: %s",
startThreshold, snd_strerror(err));
goto done;
}
err = snd_pcm_sw_params_set_stop_threshold(handle->handle, softwareParams,
stopThreshold);
if (err < 0) {
LOGE("Unable to set stop threshold to %lu frames: %s",
stopThreshold, snd_strerror(err));
goto done;
}
// Allow the transfer to start when at least periodSize samples can be
// processed.
err = snd_pcm_sw_params_set_avail_min(handle->handle, softwareParams,
periodSize);
if (err < 0) {
LOGE("Unable to configure available minimum to %lu: %s",
periodSize, snd_strerror(err));
goto done;
}
// Commit the software parameters back to the device.
err = snd_pcm_sw_params(handle->handle, softwareParams);
if (err < 0) LOGE("Unable to configure software parameters: %s",
snd_strerror(err));
done:
snd_pcm_sw_params_free(softwareParams);
return err;
}
static size_t
alsa_configure (void)
{
//<init:
size_t chunk_bytes, bits_per_sample, bits_per_frame = 0;
snd_pcm_uframes_t chunk_size, buffer_size = 0;
snd_pcm_hw_params_t *params;
unsigned int rate = DEFAULT_SPEED;
int err;
snd_pcm_hw_params_alloca (¶ms);
//>
//<defaults:
err = snd_pcm_hw_params_any (AHandle, params);
if (err < 0)
{
fprintf (stderr,
"Broken configuration for this PCM: no configurations available");
exit (EXIT_FAILURE);
}
//>
//<Format:
err = snd_pcm_hw_params_set_format (AHandle, params, DEFAULT_FORMAT);
if (err < 0)
{
fprintf (stderr, "Sample format non available");
exit (EXIT_FAILURE);
}
//>
//<Channels:
err = snd_pcm_hw_params_set_channels (AHandle, params, 1);
if (err < 0)
{
fprintf (stderr, "Channels count non available");
exit (EXIT_FAILURE);
}
//>
//<Rate:
err = snd_pcm_hw_params_set_rate_near (AHandle, params, &rate, 0);
assert (err >= 0);
//>
//<Access Mode:
err = snd_pcm_hw_params_set_access (AHandle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
{
fprintf (stderr, "Access type not available");
exit (EXIT_FAILURE);
}
//>
//< Set things explicitly if DEBUG
#ifdef DEBUG
//<Compute buffer_time:
unsigned int period_time = 0;
unsigned int buffer_time = 0;
snd_pcm_uframes_t period_frames = 0;
snd_pcm_uframes_t buffer_frames = 0;
// affected by defined buffer_size (e.g. via asoundrc)
if (buffer_time == 0 && buffer_frames == 0)
{
err = snd_pcm_hw_params_get_buffer_time (params, &buffer_time, 0);
assert (err >= 0);
if (buffer_time > 500000) //usecs
buffer_time = 500000;
}
//>
//<Compute period_time:
if (period_time == 0 && period_frames == 0)
{
if (buffer_time > 0)
period_time = buffer_time / 4;
else
period_frames = buffer_frames / 4;
}
if (period_time > 0)
err = snd_pcm_hw_params_set_period_time_near (AHandle, params,
&period_time, 0);
else
err = snd_pcm_hw_params_set_period_size_near (AHandle, params,
&period_frames, 0);
assert (err >= 0);
if (buffer_time > 0)
{
err = snd_pcm_hw_params_set_buffer_time_near (AHandle, params,
&buffer_time, 0);
}
else
{
err = snd_pcm_hw_params_set_buffer_size_near (AHandle, params,
&buffer_frames);
}
assert (err >= 0);
//>
#endif
//>
//<Commit hw params:
err = snd_pcm_hw_params (AHandle, params);
if (err < 0)
{
fprintf (stderr, "Unable to install hw params:");
exit (EXIT_FAILURE);
}
//>
//<finalize chunk_size and buffer_size:
snd_pcm_hw_params_get_period_size (params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size (params, &buffer_size);
if (chunk_size == buffer_size)
{
fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)",
chunk_size, buffer_size);
exit (EXIT_FAILURE);
}
//>
//< If DEBUG: SW Params Configure transfer:
#ifdef DEBUG
size_t n;
snd_pcm_uframes_t xfer_align;
snd_pcm_uframes_t start_threshold, stop_threshold;
int start_delay = 5;
int stop_delay = 0;
snd_pcm_sw_params_t *swParams;
snd_pcm_sw_params_alloca (&swParams);
snd_pcm_sw_params_current (AHandle, swParams);
err = snd_pcm_sw_params_get_xfer_align (swParams, &xfer_align);
if (err < 0)
{
fprintf (stderr, "Unable to obtain xfer align\n");
exit (EXIT_FAILURE);
}
// round up to closest transfer boundary
n = (buffer_size / xfer_align) * xfer_align;
if (start_delay <= 0)
{
start_threshold =
(snd_pcm_uframes_t) (n + (double) rate * start_delay / 1000000);
}
else
start_threshold =
(snd_pcm_uframes_t) ((double) rate * start_delay / 1000000);
if (start_threshold < 1)
start_threshold = 1;
if (start_threshold > n)
start_threshold = n;
err =
snd_pcm_sw_params_set_start_threshold (AHandle, swParams,
start_threshold);
assert (err >= 0);
if (stop_delay <= 0)
stop_threshold =
(snd_pcm_uframes_t) (buffer_size +
(double) rate * stop_delay / 1000000);
else
stop_threshold =
(snd_pcm_uframes_t) ((double) rate * stop_delay / 1000000);
err =
snd_pcm_sw_params_set_stop_threshold (AHandle, swParams, stop_threshold);
assert (err >= 0);
err = snd_pcm_sw_params_set_xfer_align (AHandle, swParams, xfer_align);
assert (err >= 0);
if (snd_pcm_sw_params (AHandle, swParams) < 0)
{
fprintf (stderr, "unable to install sw params:");
exit (EXIT_FAILURE);
}
#endif
//>
bits_per_sample = snd_pcm_format_physical_width (DEFAULT_FORMAT);
bits_per_frame = bits_per_sample * 1; //mono
chunk_bytes = chunk_size * bits_per_frame / 8;
return chunk_bytes;
}
static int
sa_alsa_open_stream(
simpleaudio *sa,
const char *backend_device,
sa_direction_t sa_stream_direction,
sa_format_t sa_format,
unsigned int rate, unsigned int channels,
char *app_name, char *stream_name )
{
snd_pcm_t *pcm;
int error;
char *be_device;
if ( ! backend_device ) {
be_device = "default";
} else {
be_device = alloca(32);
if ( strchr(backend_device, ':') )
snprintf(be_device, 32, "%s", backend_device);
else if ( strchr(backend_device, ',') )
snprintf(be_device, 32, "plughw:%s", backend_device);
else
snprintf(be_device, 32, "plughw:%s,0", backend_device);
}
error = snd_pcm_open(&pcm,
be_device,
sa_stream_direction == SA_STREAM_RECORD ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
0 /*mode*/);
if (error) {
fprintf(stderr, "E: Cannot create ALSA stream: %s\n", snd_strerror(error));
return 0;
}
snd_pcm_format_t pcm_format;
switch ( sa->format ) {
case SA_SAMPLE_FORMAT_FLOAT:
pcm_format = SND_PCM_FORMAT_FLOAT;
break;
case SA_SAMPLE_FORMAT_S16:
pcm_format = SND_PCM_FORMAT_S16;
break;
default:
assert(0);
}
/* set up ALSA hardware params */
error = snd_pcm_set_params(pcm,
pcm_format,
SND_PCM_ACCESS_RW_INTERLEAVED,
channels,
rate,
1 /* soft_resample (allow) */,
(unsigned int)-1 /* latency (allow max to avoid underruns) */);
if (error) {
fprintf(stderr, "E: %s\n", snd_strerror(error));
snd_pcm_close(pcm);
return 0;
}
#if 0
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
error = snd_pcm_sw_params_current(pcm, swparams);
if (error) {
fprintf(stderr, "E: %s\n", snd_strerror(error));
snd_pcm_close(pcm);
return NULL;
}
snd_pcm_sw_params_set_start_threshold(pcm, swparams, NFRAMES_VAL);
snd_pcm_sw_params_set_stop_threshold(pcm, swparams, NFRAMES_VAL);
snd_pcm_sw_params_set_silence_threshold(pcm, swparams, NFRAMES_VAL);
error = snd_pcm_sw_params(pcm, swparams);
if (error) {
fprintf(stderr, "E: %s\n", snd_strerror(error));
snd_pcm_close(pcm);
return NULL;
}
#endif
sa->backend_handle = pcm;
sa->backend_framesize = sa->channels * sa->samplesize;
return 1;
}
static int alsa_set_params(snd_pcm_t *pcm_handle, int rw, int bits, int stereo, int rate)
{
snd_pcm_hw_params_t *hwparams=NULL;
snd_pcm_sw_params_t *swparams=NULL;
int dir;
uint exact_uvalue;
unsigned long exact_ulvalue;
int channels;
int periods=ALSA_PERIODS;
int periodsize=ALSA_PERIOD_SIZE;
int err;
int format;
/* Allocate the snd_pcm_hw_params_t structure on the stack. */
snd_pcm_hw_params_alloca(&hwparams);
/* Init hwparams with full configuration space */
if (snd_pcm_hw_params_any(pcm_handle, hwparams) < 0) {
ms_warning("alsa_set_params: Cannot configure this PCM device.\n");
return -1;
}
if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
ms_warning("alsa_set_params: Error setting access.\n");
return -1;
}
/* Set sample format */
format=SND_PCM_FORMAT_S16;
if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, format) < 0) {
ms_warning("alsa_set_params: Error setting format.\n");
return -1;
}
/* Set number of channels */
if (stereo) channels=2;
else channels=1;
if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels) < 0) {
ms_warning("alsa_set_params: Error setting channels.\n");
return -1;
}
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
exact_uvalue=rate;
dir=0;
if ((err=snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_uvalue, &dir))<0){
ms_warning("alsa_set_params: Error setting rate to %i:%s",rate,snd_strerror(err));
return -1;
}
if (dir != 0) {
ms_warning("alsa_set_params: The rate %d Hz is not supported by your hardware.\n "
"==> Using %d Hz instead.\n", rate, exact_uvalue);
}
/* choose greater period size when rate is high */
periodsize=periodsize*(rate/8000);
/* Set buffer size (in frames). The resulting latency is given by */
/* latency = periodsize * periods / (rate * bytes_per_frame) */
/* set period size */
exact_ulvalue=periodsize;
dir=0;
if (snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &exact_ulvalue, &dir) < 0) {
ms_warning("alsa_set_params: Error setting period size.\n");
return -1;
}
if (dir != 0) {
ms_warning("alsa_set_params: The period size %d is not supported by your hardware.\n "
"==> Using %d instead.\n", periodsize, (int)exact_ulvalue);
}
periodsize=exact_ulvalue;
/* Set number of periods. Periods used to be called fragments. */
exact_uvalue=periods;
dir=0;
if (snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &exact_uvalue, &dir) < 0) {
ms_warning("alsa_set_params: Error setting periods.\n");
return -1;
}
if (dir != 0) {
ms_warning("alsa_set_params: The number of periods %d is not supported by your hardware.\n "
"==> Using %d instead.\n", periods, exact_uvalue);
}
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if ((err=snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
ms_warning("alsa_set_params: Error setting HW params:%s",snd_strerror(err));
return -1;
}
/*prepare sw params */
if (rw){
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(pcm_handle, swparams);
if ((err=snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams,periodsize*4 ))<0){
ms_warning("alsa_set_params: Error setting start threshold:%s",snd_strerror(err));
}
if ((err=snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams,periodsize*periods ))<0){
ms_warning("alsa_set_params: Error setting start threshold:%s",snd_strerror(err));
}
if ((err=snd_pcm_sw_params(pcm_handle, swparams))<0){
ms_warning("alsa_set_params: Error setting SW params:%s",snd_strerror(err));
return -1;
}
}
return 0;
}
bool QAudioOutputPrivate::open()
{
if(opened)
return true;
#ifdef DEBUG_AUDIO
QTime now(QTime::currentTime());
qDebug()<<now.second()<<"s "<<now.msec()<<"ms :open()";
#endif
timeStamp.restart();
elapsedTimeOffset = 0;
int dir;
int err=-1;
int count=0;
unsigned int freakuency=settings.frequency();
QString dev = QLatin1String(m_device.constData());
QList<QByteArray> devices = QAudioDeviceInfoInternal::availableDevices(QAudio::AudioOutput);
if(dev.compare(QLatin1String("default")) == 0) {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(devices.first().constData());
#else
dev = QLatin1String("hw:0,0");
#endif
} else {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(m_device);
#else
int idx = 0;
char *name;
QString shortName = QLatin1String(m_device.mid(m_device.indexOf('=',0)+1).constData());
while(snd_card_get_name(idx,&name) == 0) {
if(qstrncmp(shortName.toLocal8Bit().constData(),name,shortName.length()) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
#endif
}
// Step 1: try and open the device
while((count < 5) && (err < 0)) {
err=snd_pcm_open(&handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_PLAYBACK,0);
if(err < 0)
count++;
}
if (( err < 0)||(handle == 0)) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
return false;
}
snd_pcm_nonblock( handle, 0 );
// Step 2: Set the desired HW parameters.
snd_pcm_hw_params_alloca( &hwparams );
bool fatal = false;
QString errMessage;
unsigned int chunks = 8;
err = snd_pcm_hw_params_any( handle, hwparams );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_any: err = %1").arg(err);
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_resample( handle, hwparams, 1 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_rate_resample: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_access( handle, hwparams, access );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_access: err = %1").arg(err);
}
}
if ( !fatal ) {
err = setFormat();
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_format: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_channels( handle, hwparams, (unsigned int)settings.channels() );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_channels: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_near( handle, hwparams, &freakuency, 0 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_rate_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_buffer_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_period_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &chunks, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_periods_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params: err = %1").arg(err);
}
}
if( err < 0) {
qWarning()<<errMessage;
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
return false;
}
snd_pcm_hw_params_get_buffer_size(hwparams,&buffer_frames);
buffer_size = snd_pcm_frames_to_bytes(handle,buffer_frames);
snd_pcm_hw_params_get_period_size(hwparams,&period_frames, &dir);
period_size = snd_pcm_frames_to_bytes(handle,period_frames);
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
// Step 3: Set the desired SW parameters.
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(handle, swparams);
snd_pcm_sw_params_set_start_threshold(handle,swparams,period_frames);
snd_pcm_sw_params_set_stop_threshold(handle,swparams,buffer_frames);
snd_pcm_sw_params_set_avail_min(handle, swparams,period_frames);
snd_pcm_sw_params(handle, swparams);
// Step 4: Prepare audio
if(audioBuffer == 0)
audioBuffer = new char[snd_pcm_frames_to_bytes(handle,buffer_frames)];
snd_pcm_prepare( handle );
snd_pcm_start(handle);
// Step 5: Setup callback and timer fallback
snd_async_add_pcm_handler(&ahandler, handle, async_callback, this);
bytesAvailable = bytesFree();
// Step 6: Start audio processing
timer->start(period_time/1000);
clockStamp.restart();
timeStamp.restart();
elapsedTimeOffset = 0;
errorState = QAudio::NoError;
totalTimeValue = 0;
opened = true;
return true;
}
static void *_msynth_thread_main(void *arg)
{
int i;
int err;
int sample;
int processed;
struct sampleclock sc = {0, 0, 0.0f, 0.0f};
msynth_frame fb = NULL;
puts("synthread: started");
/* Begin initialization of ALSA */
/* snd_pcm_open(pcm_handle, device_name, stream_type, open_mode) */
err = snd_pcm_open(&pcm, config.device_name, SND_PCM_STREAM_PLAYBACK, 0);
ALSERT("opening audio device");
/* First ALSA hardware settings */
/* Allocate HW params */
err = snd_pcm_hw_params_malloc(&hw_p);
ALSERT("allocating hw params");
/* Get default HW parameters */
err = snd_pcm_hw_params_any(pcm, hw_p);
ALSERT("requesting hw default params");
/* Disable software resampling */
err = snd_pcm_hw_params_set_rate_resample(pcm, hw_p, config.resample);
ALSERT("disabling software resampling");
/* Configure sample rate */
if (config.srate != -1)
srate = config.srate;
else {
/* Get maximum hardware samplerate */
err = snd_pcm_hw_params_get_rate_max(hw_p, &srate, &dir);
ALSERT("requesting maximum hardware samplerate");
}
/* Set sample rate
* snd_pcm_hw_params_set_rate_near(pcn, hw_p, *rate,
* 0 (== set exact rate))
*/
err = snd_pcm_hw_params_set_rate_near(pcm, hw_p, &srate, 0);
ALSERT("setting samplerate");
printf("synthread: Detected samplerate of %u\n", srate);
/* RW interleaved access (means we will use the snd_pcm_writei function) */
err = snd_pcm_hw_params_set_access(pcm, hw_p,
SND_PCM_ACCESS_RW_INTERLEAVED);
ALSERT("setting access mode");
/* native-endian 16-bit signed sample format */
err = snd_pcm_hw_params_set_format(pcm, hw_p, SND_PCM_FORMAT_S16);
ALSERT("setting sample format");
/* Switch to 2.0ch audio */
err = snd_pcm_hw_params_set_channels(pcm, hw_p, 2);
ALSERT("switching to 2.0ch audio");
if (config.buffer_time != -1) {
/* Set configured buffer time */
err = snd_pcm_hw_params_set_buffer_time_near(pcm, hw_p,
&config.buffer_time, &dir);
ALSERT("setting buffer time");
/* Fetch resulting size */
err = snd_pcm_hw_params_get_buffer_size(hw_p, &buffer_size);
ALSERT("getting buffer size");
} else {
/* Since we want the interactive synthesizer to be very responsive
* select a minimum buffer size.
*/
err = snd_pcm_hw_params_set_buffer_size_first(pcm, hw_p, &buffer_size);
ALSERT("setting buffer size");
}
printf("synthread: Selected buffersize of %lu\n", buffer_size);
printf("synthread: Response delay is approximately %.2f ms\n",
(double)buffer_size / (double)srate * 1000.0);
if (config.period_time != -1) {
err = snd_pcm_hw_params_set_period_time_near(pcm, hw_p,
&config.period_time, &dir);
ALSERT("setting period time");
err = snd_pcm_hw_params_get_period_size(hw_p, &period_size, &dir);
ALSERT("getting period size");
} else {
/* Select a period time of half the buffer time
* since this improves processing performance
*/
err = snd_pcm_hw_params_set_period_size_last(pcm, hw_p, &period_size, &dir);
ALSERT("setting period size");
}
if (dir)
printf("synthread: Selected period size near %lu\n", period_size);
else
printf("synthread: Selected period size of %lu\n", period_size);
/* write hw parameters to device */
err = snd_pcm_hw_params(pcm, hw_p);
ALSERT("writing hw params");
/* Begin ALSA software side setup */
/* Allocate SW params */
err = snd_pcm_sw_params_malloc(&sw_p);
ALSERT("allocating sw params");
/* Request software current settings */
err = snd_pcm_sw_params_current(pcm, sw_p);
ALSERT("requesting current sw params");
/* Automatically start playing after the first period is written */
err = snd_pcm_sw_params_set_start_threshold(pcm, sw_p, period_size);
ALSERT("settings start threshold");
/* XRUN when buffer is empty */
err = snd_pcm_sw_params_set_stop_threshold(pcm, sw_p, buffer_size * 2);
ALSERT("setting stop threshold");
/* Block synthesizer when there is not at least period frames available */
err = snd_pcm_sw_params_set_avail_min(pcm, sw_p, period_size);
ALSERT("setting minimum free frames");
/* Write software parameters */
err = snd_pcm_sw_params(pcm, sw_p);
ALSERT("writing sw params");
printf("synthread: Audio system configured.\n");
/* Prepare device for playback */
err = snd_pcm_prepare(pcm);
ALSERT("preparing device");
/* Allocate a period sized framebuffer
* (yup an audio buffer is in ALSA speak indeed called a framebuffer)
*/
fb = malloc(sizeof(struct _msynth_frame) * period_size);
if (!fb) {
perror("malloc framebuffer failed");
exit(1);
}
/* Set sampleclock to 0 */
sc.samples = 0;
sc.samplerate = srate;
sc.cycle = 0.0f;
sc.seconds = 0.0f;
/* Initially the <root> variable describing the flow of sound within
* the synthesizer is configured to be a NULL signal. (silence)
*/
/* Signal successful completion of initialization */
pthread_mutex_lock(&mutex);
pthread_cond_signal(&cond);
pthread_mutex_unlock(&mutex);
/* -------------- Main loop --------------- */
while (!shutdown) {
/* Only during generation we need the synth tree to be static */
synth_lock_graphs();
for (i = 0; i < period_size; i++) {
/* Evaluate variables */
ssv_eval(sc);
sample = (int)(32767.5 * ssv_get_var_eval("left") * volume);
/* Clip samples */
if (sample > 32767) sample = 32767;
if (sample < -32768) sample = -32768;
fb[i].left = (short)sample;
sample = (int)(32767.5 * ssv_get_var_eval("right") * volume);
/* Clip samples */
if (sample > 32767) sample = 32767;
if (sample < -32768) sample = -32768;
fb[i].right = (short)sample;
sc = sc_from_samples(sc.samplerate, sc.samples + 1);
}
synth_unlock_graphs();
/* Send audio to sound card */
processed = 0;
while (processed != period_size) {
err = snd_pcm_writei(pcm, fb + processed, period_size - processed);
/* Retry on interruption by signal */
if (err == -EAGAIN)
continue;
/* Recover from XRUN/suspend */
if (err < 0) {
err = synth_recover(err);
ALSERT("sending audio");
continue;
}
/* Update processed samples */
processed += err;
}
}
puts("synthread: shutting down");
/* Dump any statistics recorded */
if (recover_resumes)
printf("synthread: Device was resumed %i times\n", recover_resumes);
if (recover_xruns)
printf("synthread: %i xrun recoveries were needed\n", recover_xruns);
printf("synthread: processed %i samples\n", sc.samples);
/* Wait for playback to complete */
err = snd_pcm_drain(pcm);
ALSERT("draining device");
/* Shutdown PCM device */
err = snd_pcm_close(pcm);
ALSERT("closing audio device");
/* Clean up parameters */
snd_pcm_hw_params_free(hw_p);
snd_pcm_sw_params_free(sw_p);
return NULL;
}
/* set up an input or output device. Return 0 on success, -1 on failure. */
static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate,
int nfrags, int frag_size)
{
int bufsizeforthis, err;
snd_pcm_hw_params_t* hw_params;
snd_pcm_sw_params_t* sw_params;
unsigned int tmp_uint;
snd_pcm_uframes_t tmp_snd_pcm_uframes;
snd_pcm_hw_params_alloca(&hw_params);
snd_pcm_sw_params_alloca(&sw_params);
if (sys_verbose)
post((out ? "configuring sound output..." :
"configuring sound input..."));
/* set hardware parameters... */
/* get the default params */
err = snd_pcm_hw_params_any(dev->a_handle, hw_params);
check_error(err, out, "snd_pcm_hw_params_any");
/* try to set interleaved access */
err = snd_pcm_hw_params_set_access(dev->a_handle,
hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
return (-1);
check_error(err, out, "snd_pcm_hw_params_set_access");
#if 0 /* enable this to print out which formats are available */
{
int i;
for (i = 0; i <= SND_PCM_FORMAT_LAST; i++)
fprintf(stderr, "%d -> %d\n",
i, snd_pcm_hw_params_test_format(dev->a_handle, hw_params, i));
}
#endif
/* Try to set 32 bit format first */
err = snd_pcm_hw_params_set_format(dev->a_handle,
hw_params, SND_PCM_FORMAT_S32);
if (err < 0)
{
err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
SND_PCM_FORMAT_S24_3LE);
if (err < 0)
{
err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
SND_PCM_FORMAT_S16);
check_error(err, out, "_pcm_hw_params_set_format");
dev->a_sampwidth = 2;
}
else dev->a_sampwidth = 3;
}
else dev->a_sampwidth = 4;
if (sys_verbose)
post("Sample width set to %d bytes", dev->a_sampwidth);
/* set the subformat */
err = snd_pcm_hw_params_set_subformat(dev->a_handle,
hw_params, SND_PCM_SUBFORMAT_STD);
check_error(err, out, "snd_pcm_hw_params_set_subformat");
/* set the number of channels */
tmp_uint = *channels;
err = snd_pcm_hw_params_set_channels_near(dev->a_handle,
hw_params, &tmp_uint);
check_error(err, out, "snd_pcm_hw_params_set_channels");
if (tmp_uint != (unsigned)*channels)
post("ALSA: set %s channels to %d", (out?"output":"input"), tmp_uint);
*channels = tmp_uint;
dev->a_channels = *channels;
/* set the sampling rate */
err = snd_pcm_hw_params_set_rate_near(dev->a_handle, hw_params,
(unsigned int *)rate, 0);
check_error(err, out, "snd_pcm_hw_params_set_rate_min");
#if 0
err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir);
post("input sample rate %d", err);
#endif
/* post("frag size %d, nfrags %d", frag_size, nfrags); */
/* set "period size" */
tmp_snd_pcm_uframes = frag_size;
err = snd_pcm_hw_params_set_period_size_near(dev->a_handle,
hw_params, &tmp_snd_pcm_uframes, 0);
check_error(err, out, "snd_pcm_hw_params_set_period_size_near");
/* set the buffer size */
tmp_snd_pcm_uframes = nfrags * frag_size;
err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle,
hw_params, &tmp_snd_pcm_uframes);
check_error(err, out, "snd_pcm_hw_params_set_buffer_size_near");
err = snd_pcm_hw_params(dev->a_handle, hw_params);
check_error(err, out, "snd_pcm_hw_params");
/* set up the buffer */
bufsizeforthis = DEFDACBLKSIZE * dev->a_sampwidth * *channels;
if (alsa_snd_buf)
{
if (alsa_snd_bufsize < bufsizeforthis)
{
if (!(alsa_snd_buf = realloc(alsa_snd_buf, bufsizeforthis)))
{
post("out of memory");
return (-1);
}
memset(alsa_snd_buf, 0, bufsizeforthis);
alsa_snd_bufsize = bufsizeforthis;
}
}
else
{
if (!(alsa_snd_buf = (void *)malloc(bufsizeforthis)))
{
post("out of memory");
return (-1);
}
memset(alsa_snd_buf, 0, bufsizeforthis);
alsa_snd_bufsize = bufsizeforthis;
}
err = snd_pcm_sw_params_current(dev->a_handle, sw_params);
if (err < 0)
{
post("Unable to determine current swparams for %s: %s\n",
(out ? "output" : "input"), snd_strerror(err));
return (-1);
}
err = snd_pcm_sw_params_set_stop_threshold(dev->a_handle, sw_params,
0x7fffffff);
if (err < 0)
{
post("Unable to set start threshold mode for %s: %s\n",
(out ? "output" : "input"), snd_strerror(err));
return (-1);
}
err = snd_pcm_sw_params_set_avail_min(dev->a_handle, sw_params, 4);
if (err < 0)
{
post("Unable to set avail min for %s: %s\n",
(out ? "output" : "input"), snd_strerror(err));
return (-1);
}
err = snd_pcm_sw_params(dev->a_handle, sw_params);
if (err < 0)
{
post("Unable to set sw params for %s: %s\n",
(out ? "output" : "input"), snd_strerror(err));
return (-1);
}
return (0);
}
static int aplaypop_open(void)
{
int err;
snd_pcm_t *handle;
if (pcm_handle)
return 0;
snd_pcm_info_t *info;
snd_pcm_info_alloca(&info);
snd_output_t *log;
err = snd_output_stdio_attach(&log, stderr, 0);
assert(err == 0);
err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0);
if (err != 0) {
fprintf(stderr, "snd_pcm_open(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_nonblock(handle, 0);
if (err != 0) {
fprintf(stderr, "snd_pcm_nonblock(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_info(handle, info);
if (err != 0) {
fprintf(stderr, "snd_pcm_info(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
// DOESN'T WORK!
err = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE,
SND_PCM_ACCESS_RW_INTERLEAVED, CHANNELS, RATE, 1, 50000);
if (err != 0) {
fprintf(stderr, "snd_pcm_set_params(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
// RIGHT WAY:
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
unsigned int channels = CHANNELS;
unsigned int rate = RATE;
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err != 0) {
fprintf(stderr, "Broken configuration for this PCM: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err != 0) {
fprintf(stderr, "snd_pcm_hw_params_set_access(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_format(handle, hwparams, format);
if (err != 0) {
fprintf(stderr, "snd_pcm_hw_params_set_format(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, channels);
if (err != 0) {
fprintf(stderr, "snd_pcm_hw_params_set_channels(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, 0);
if (err != 0) {
fprintf(stderr, "snd_pcm_hw_params_set_rate_near(): %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
/*
unsigned buffer_time = 0;
snd_pcm_uframes_t buffer_frames = 0;
if (buffer_time == 0 && buffer_frames == 0) {
err = snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time, 0);
assert(err == 0);
if (buffer_time > 500000)
buffer_time = 500000;
}
unsigned period_time = 0;
snd_pcm_uframes_t period_frames = 0;
if (period_time == 0 && period_frames == 0) {
if (buffer_time > 0)
period_time = buffer_time / 4;
else
period_frames = buffer_frames / 4;
}
if (period_time > 0)
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0);
else
err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_frames, 0);
assert(err == 0);
if (buffer_time > 0)
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0);
else
err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_frames);
assert(err == 0);
int monotonic = snd_pcm_hw_params_is_monotonic(hwparams);
int can_pause = snd_pcm_hw_params_can_pause(hwparams);
*/
err = snd_pcm_hw_params(handle, hwparams);
if (err != 0) {
fprintf(stderr, "snd_pcm_hw_params(): %s\n", snd_strerror(err));
snd_pcm_hw_params_dump(hwparams, log);
exit(EXIT_FAILURE);
}
snd_pcm_uframes_t chunk_size = 0;
snd_pcm_hw_params_get_period_size(hwparams, &chunk_size, 0);
snd_pcm_uframes_t buffer_size;
snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
if (chunk_size == buffer_size) {
fprintf(stderr, "Can't use period equal to buffer size (%lu == %lu)",
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
err = snd_pcm_sw_params_set_avail_min(handle, swparams, chunk_size);
assert(err == 0);
/* round up to closest transfer boundary */
int start_delay = 0;
snd_pcm_uframes_t start_threshold;
if (start_delay <= 0)
start_threshold = buffer_size + (double) rate * start_delay / 1000000;
else
start_threshold = (double) rate * start_delay / 1000000;
start_threshold = start_threshold < 1 ? 1 : start_threshold > buffer_size ? buffer_size : start_threshold;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err == 0);
int stop_delay = 0;
snd_pcm_uframes_t stop_threshold;
if (stop_delay <= 0)
stop_threshold = buffer_size + (double) rate * stop_delay / 1000000;
else
stop_threshold = (double) rate * stop_delay / 1000000;
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
assert(err == 0);
err = snd_pcm_sw_params(handle, swparams);
if (err != 0) {
fprintf(stderr, "snd_pcm_sw_params(): %s\n", snd_strerror(err));
snd_pcm_sw_params_dump(swparams, log);
exit(EXIT_FAILURE);
}
// END OF THE RIGHT WAY
// snd_pcm_dump(handle, log);
size_t bits_per_sample = snd_pcm_format_physical_width(format);
size_t bits_per_frame = bits_per_sample * channels;
size_t chunk_bytes = chunk_size * bits_per_frame / 8;
//audiobuf = realloc(audiobuf, chunk_bytes);
fprintf(stderr, "%s: %s, Rate %d Hz, Channels=%u\n",
snd_pcm_format_name(format), snd_pcm_format_description(format),
rate, channels);
fprintf(stderr, " bits_per_sample=%u, bits_per_frame=%u, chunk_bytes=%u\n",
bits_per_sample, bits_per_frame, chunk_bytes);
frame_bytes = bits_per_frame / 8;
pcm_handle = handle;
return 0;
}
bool QAlsaAudioInput::open()
{
#ifdef DEBUG_AUDIO
QTime now(QTime::currentTime());
qDebug()<<now.second()<<"s "<<now.msec()<<"ms :open()";
#endif
clockStamp.restart();
timeStamp.restart();
elapsedTimeOffset = 0;
int dir;
int err = 0;
int count=0;
unsigned int sampleRate=settings.sampleRate();
if (!settings.isValid()) {
qWarning("QAudioInput: open error, invalid format.");
} else if (settings.sampleRate() <= 0) {
qWarning("QAudioInput: open error, invalid sample rate (%d).",
settings.sampleRate());
} else {
err = -1;
}
if (err == 0) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit errorChanged(errorState);
return false;
}
QString dev = QString(QLatin1String(m_device.constData()));
QList<QByteArray> devices = QAlsaAudioDeviceInfo::availableDevices(QAudio::AudioInput);
if(dev.compare(QLatin1String("default")) == 0) {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
if (devices.size() > 0)
dev = QLatin1String(devices.first());
else
return false;
#else
dev = QLatin1String("hw:0,0");
#endif
} else {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(m_device);
#else
int idx = 0;
char *name;
QString shortName = QLatin1String(m_device.mid(m_device.indexOf('=',0)+1).constData());
while(snd_card_get_name(idx,&name) == 0) {
if(qstrncmp(shortName.toLocal8Bit().constData(),name,shortName.length()) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
#endif
}
// Step 1: try and open the device
while((count < 5) && (err < 0)) {
err=snd_pcm_open(&handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_CAPTURE,0);
if(err < 0)
count++;
}
if (( err < 0)||(handle == 0)) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit stateChanged(deviceState);
return false;
}
snd_pcm_nonblock( handle, 0 );
// Step 2: Set the desired HW parameters.
snd_pcm_hw_params_alloca( &hwparams );
bool fatal = false;
QString errMessage;
unsigned int chunks = 8;
err = snd_pcm_hw_params_any( handle, hwparams );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_any: err = %1").arg(err);
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_resample( handle, hwparams, 1 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_rate_resample: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_access( handle, hwparams, access );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_access: err = %1").arg(err);
}
}
if ( !fatal ) {
err = setFormat();
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_format: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_channels( handle, hwparams, (unsigned int)settings.channelCount() );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_channels: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_near( handle, hwparams, &sampleRate, 0 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_rate_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_buffer_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_period_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &chunks, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_periods_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params: err = %1").arg(err);
}
}
if( err < 0) {
qWarning()<<errMessage;
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit stateChanged(deviceState);
return false;
}
snd_pcm_hw_params_get_buffer_size(hwparams,&buffer_frames);
buffer_size = snd_pcm_frames_to_bytes(handle,buffer_frames);
snd_pcm_hw_params_get_period_size(hwparams,&period_frames, &dir);
period_size = snd_pcm_frames_to_bytes(handle,period_frames);
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
// Step 3: Set the desired SW parameters.
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(handle, swparams);
snd_pcm_sw_params_set_start_threshold(handle,swparams,period_frames);
snd_pcm_sw_params_set_stop_threshold(handle,swparams,buffer_frames);
snd_pcm_sw_params_set_avail_min(handle, swparams,period_frames);
snd_pcm_sw_params(handle, swparams);
// Step 4: Prepare audio
ringBuffer.resize(buffer_size);
snd_pcm_prepare( handle );
snd_pcm_start(handle);
// Step 5: Setup timer
bytesAvailable = checkBytesReady();
if(pullMode)
connect(audioSource,SIGNAL(readyRead()),this,SLOT(userFeed()));
// Step 6: Start audio processing
chunks = buffer_size/period_size;
timer->start(period_time*chunks/2000);
errorState = QAudio::NoError;
totalTimeValue = 0;
return true;
}
static int initialize_device(struct audio_info_struct *ai)
{
snd_pcm_hw_params_t *hw;
int i;
snd_pcm_format_t format;
unsigned int rate;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
snd_pcm_sw_params_t *sw;
snd_pcm_uframes_t boundary;
snd_pcm_hw_params_alloca(&hw);
if (snd_pcm_hw_params_any(ai->handle, hw) < 0) {
fprintf(stderr, "initialize_device(): no configuration available\n");
return -1;
}
if (snd_pcm_hw_params_set_access(ai->handle, hw, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
fprintf(stderr, "initialize_device(): device does not support interleaved access\n");
return -1;
}
format = SND_PCM_FORMAT_UNKNOWN;
for (i = 0; i < NUM_FORMATS; ++i) {
if (ai->format == format_map[i].mpg123) {
format = format_map[i].alsa;
break;
}
}
if (format == SND_PCM_FORMAT_UNKNOWN) {
fprintf(stderr, "initialize_device(): invalid sample format %d\n", ai->format);
errno = EINVAL;
return -1;
}
if (snd_pcm_hw_params_set_format(ai->handle, hw, format) < 0) {
fprintf(stderr, "initialize_device(): cannot set format %s\n", snd_pcm_format_name(format));
return -1;
}
if (snd_pcm_hw_params_set_channels(ai->handle, hw, ai->channels) < 0) {
fprintf(stderr, "initialize_device(): cannot set %d channels\n", ai->channels);
return -1;
}
rate = ai->rate;
if (snd_pcm_hw_params_set_rate_near(ai->handle, hw, &rate, NULL) < 0) {
fprintf(stderr, "initialize_device(): cannot set rate %u\n", rate);
return -1;
}
if (!rates_match(ai->rate, rate)) {
fprintf(stderr, "initialize_device(): rate %ld not available, using %u\n", ai->rate, rate);
/* return -1; */
}
buffer_size = rate * BUFFER_LENGTH;
if (snd_pcm_hw_params_set_buffer_size_near(ai->handle, hw, &buffer_size) < 0) {
fprintf(stderr, "initialize_device(): cannot set buffer size\n");
return -1;
}
period_size = buffer_size / 4;
if (snd_pcm_hw_params_set_period_size_near(ai->handle, hw, &period_size, NULL) < 0) {
fprintf(stderr, "initialize_device(): cannot set period size\n");
return -1;
}
if (snd_pcm_hw_params(ai->handle, hw) < 0) {
fprintf(stderr, "initialize_device(): cannot set hw params\n");
return -1;
}
snd_pcm_sw_params_alloca(&sw);
if (snd_pcm_sw_params_current(ai->handle, sw) < 0) {
fprintf(stderr, "initialize_device(): cannot get sw params\n");
return -1;
}
/* start playing after the first write */
if (snd_pcm_sw_params_set_start_threshold(ai->handle, sw, 1) < 0) {
fprintf(stderr, "initialize_device(): cannot set start threshold\n");
return -1;
}
if (snd_pcm_sw_params_get_boundary(sw, &boundary) < 0) {
fprintf(stderr, "initialize_device(): cannot get boundary\n");
return -1;
}
/* never stop on underruns */
if (snd_pcm_sw_params_set_stop_threshold(ai->handle, sw, boundary) < 0) {
fprintf(stderr, "initialize_device(): cannot set stop threshold\n");
return -1;
}
/* wake up on every interrupt */
if (snd_pcm_sw_params_set_avail_min(ai->handle, sw, 1) < 0) {
fprintf(stderr, "initialize_device(): cannot set min avail\n");
return -1;
}
#if 0
/* always write as many frames as possible */
if (snd_pcm_sw_params_set_xfer_align(ai->handle, sw, 1) < 0) {
fprintf(stderr, "initialize_device(): cannot set transfer alignment\n");
return -1;
}
#endif
/* play silence when there is an underrun */
if (snd_pcm_sw_params_set_silence_size(ai->handle, sw, boundary) < 0) {
fprintf(stderr, "initialize_device(): cannot set silence size\n");
return -1;
}
if (snd_pcm_sw_params(ai->handle, sw) < 0) {
fprintf(stderr, "initialize_device(): cannot set sw params\n");
return -1;
}
return 0;
}
static snd_pcm_uframes_t set_params(snd_pcm_t *handle, snd_pcm_stream_t stream)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t chunk_size = 0;
snd_pcm_uframes_t start_threshold;
unsigned period_time = 0;
unsigned buffer_time = 0;
size_t chunk_bytes = 0;
int err;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(handle, params);
if (err < 0) {
printf("Broken configuration for this PCM: no configurations available");
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
printf("Access type not available");
exit(EXIT_FAILURE);
}
printf("format = %s, channels = %d, rate = %d\n",
snd_pcm_format_name(hwparams.format),hwparams.channels,hwparams.rate);
//err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
printf("Sample format non available");
exit(EXIT_FAILURE);
}
//err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
err = snd_pcm_hw_params_set_channels(handle, params, 2);
if (err < 0) {
printf("Channels count non available");
exit(EXIT_FAILURE);
}
#if 0
//add by yjc 2012/08/21
err = set_audio_clk_freq(hwparams.rate);
if (err < 0){
printf("set_audio_clk_freq fail..........\n");
exit(EXIT_FAILURE);
}
#endif
err = snd_pcm_hw_params_set_rate(handle, params, hwparams.rate, 0);
if (err < 0) {
printf("Rate non available");
exit(EXIT_FAILURE);
}
err = snd_pcm_hw_params_get_buffer_time_max(params,
&buffer_time, 0);
assert(err >= 0);
if (buffer_time > 500000)
buffer_time = 500000;
period_time = buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(handle, params,
&period_time, 0);
assert(err >= 0);
err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
&buffer_time, 0);
assert(err >= 0);
err = snd_pcm_hw_params(handle, params);
if (err < 0) {
printf("Unable to install hw params:");
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
if (chunk_size == buffer_size) {
printf("Can't use period equal to buffer size (%lu == %lu)",
chunk_size, buffer_size);
exit(EXIT_FAILURE);
}
snd_pcm_sw_params_current(handle, swparams);
err = snd_pcm_sw_params_set_avail_min(handle, swparams, chunk_size);
if(stream == SND_PCM_STREAM_PLAYBACK)
start_threshold = (buffer_size / chunk_size) * chunk_size;
else
start_threshold = 1;
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
assert(err >= 0);
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, buffer_size);
assert(err >= 0);
if (snd_pcm_sw_params(handle, swparams) < 0) {
printf("unable to install sw params:");
exit(EXIT_FAILURE);
}
//bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
//bits_per_frame = bits_per_sample * hwparams.channels;
bits_per_frame = bits_per_sample * 2;
chunk_bytes = chunk_size * bits_per_frame / 8;
printf("chunk_size = %d,chunk_bytes = %d,buffer_size = %d\n\n",
(int)chunk_size,chunk_bytes,(int)buffer_size);
return chunk_size;
}
int ai_alsa_setup(audio_in_t *ai)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size, period_size;
int err;
int dir;
unsigned int rate;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(ai->alsa.handle, params);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Broken configuration for this PCM: no configurations available.\n");
return -1;
}
err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Access type not available.\n");
return -1;
}
err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Sample format not available.\n");
return -1;
}
err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
if (err < 0) {
snd_pcm_hw_params_get_channels(params, &ai->channels);
mp_tmsg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n",
ai->channels);
} else {
ai->channels = ai->req_channels;
}
dir = 0;
rate = ai->req_samplerate;
err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, &rate, &dir);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Cannot set samplerate.\n");
}
ai->samplerate = rate;
dir = 0;
ai->alsa.buffer_time = 1000000;
err = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
&ai->alsa.buffer_time, &dir);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Cannot set buffer time.\n");
}
dir = 0;
ai->alsa.period_time = ai->alsa.buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
&ai->alsa.period_time, &dir);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Cannot set period time.\n");
}
err = snd_pcm_hw_params(ai->alsa.handle, params);
if (err < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Unable to install hardware parameters: %s", snd_strerror(err));
snd_pcm_hw_params_dump(params, ai->alsa.log);
return -1;
}
dir = -1;
snd_pcm_hw_params_get_period_size(params, &period_size, &dir);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
ai->alsa.chunk_size = period_size;
if (period_size == buffer_size) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size);
return -1;
}
snd_pcm_sw_params_current(ai->alsa.handle, swparams);
err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
mp_tmsg(MSGT_TV, MSGL_ERR, "Unable to install software parameters:\n");
snd_pcm_sw_params_dump(swparams, ai->alsa.log);
return -1;
}
if (mp_msg_test(MSGT_TV, MSGL_V)) {
snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
}
ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
ai->samplesize = ai->alsa.bits_per_sample;
ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
return 0;
}
static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams, int playback)
{
#ifndef ALSAAPI9
int err;
snd_pcm_uframes_t ps,ops;
snd_pcm_uframes_t bs,obs;
/* get the current swparams */
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
check_error(err,"Unable to determine current swparams for playback");
return err;
}
/* AUTOSTART: start the transfer on each write/commit ??? */
snd_pcm_sw_params_get_start_threshold(swparams, &obs);
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
if (err < 0) {
check_error(err,"Unable to set start threshold mode");
return err;
}
snd_pcm_sw_params_get_start_threshold(swparams, &bs);
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("sw_params: got start_thresh_hold= %d (was %d)",(int) bs,(int)obs);
#endif
/* AUTOSTOP: never stop the machine */
snd_pcm_sw_params_get_stop_threshold(swparams, &obs);
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, (snd_pcm_uframes_t)-1);
if (err < 0) {
check_error(err,"Unable to set stop threshold mode");
return err;
}
snd_pcm_sw_params_get_stop_threshold(swparams, &bs);
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("sw_params: set stop_thresh_hold= %d (was %d)", (int) bs,(int)obs);
#endif
/* AUTOSILENCE: silence if overrun.... */
snd_pcm_sw_params_get_silence_threshold (swparams, &ops);
if ((err = snd_pcm_sw_params_set_silence_threshold (handle, swparams, alsamm_period_size)) < 0) {
check_error (err,"cannot set silence threshold for");
return -1;
}
snd_pcm_sw_params_get_silence_threshold (swparams, &ps);
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("sw_params: set silence_threshold = %d (was %d)", (int) ps,(int)ops);
#endif
snd_pcm_sw_params_get_silence_size (swparams, &ops);
if ((err = snd_pcm_sw_params_set_silence_size(handle, swparams, alsamm_period_size)) < 0) {
check_error (err,"cannot set silence size for");
return -1;
}
snd_pcm_sw_params_get_silence_size (swparams, &ps);
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("sw_params: set silence_size = %d (was %d)", (int) ps,(int)ops);
#endif
/* AVAIL: allow the transfer when at least period_size samples can be processed */
snd_pcm_sw_params_get_avail_min(swparams, &ops);
err = snd_pcm_sw_params_set_avail_min(handle, swparams, alsamm_transfersize/2);
if (err < 0) {
check_error(err,"Unable to set avail min for");
return err;
}
snd_pcm_sw_params_get_avail_min(swparams, &ps);
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("sw_params: set avail_min= %d (was %d)", (int) ps, (int) ops);
#endif
/* write the parameters to the playback device */
err = snd_pcm_sw_params(handle, swparams);
if (err < 0) {
check_error(err,"Unable to set sw params");
return err;
}
#ifdef ALSAMM_DEBUG
if(sys_verbose)
post("set sw finished");
#endif
#else
post("alsa: need version 1.0 or above for mmap operation");
#endif /* ALSAAPI9 */
return 0;
}
