void S9xAlsaSoundDriver::samples_available()
{
snd_pcm_sframes_t frames_written, frames;
int bytes;
frames = snd_pcm_avail(pcm);
if (frames < 0)
{
frames = snd_pcm_recover(pcm, frames, 1);
return;
}
if (Settings.DynamicRateControl)
{
S9xUpdateDynamicRate(snd_pcm_frames_to_bytes(pcm, frames),
output_buffer_size);
}
int snes_frames_available = S9xGetSampleCount() >> 1;
if (Settings.DynamicRateControl && !Settings.SoundSync)
{
// Using rate control, we should always keep the emulator's sound buffers empty to
// maintain an accurate measurement.
if (frames < snes_frames_available)
{
S9xClearSamples();
return;
}
}
if (Settings.SoundSync && !Settings.TurboMode && !Settings.Mute)
{
snd_pcm_nonblock(pcm, 0);
frames = snes_frames_available;
}
else
{
snd_pcm_nonblock(pcm, 1);
frames = MIN(frames, snes_frames_available);
}
bytes = snd_pcm_frames_to_bytes(pcm, frames);
if (bytes <= 0)
return;
if (sound_buffer_size < bytes || sound_buffer == NULL)
{
sound_buffer = (uint8 *)realloc(sound_buffer, bytes);
sound_buffer_size = bytes;
}
S9xMixSamples(sound_buffer, frames * 2);
frames_written = 0;
while (frames_written < frames)
{
int result;
result = snd_pcm_writei(pcm,
sound_buffer +
snd_pcm_frames_to_bytes(pcm, frames_written),
frames - frames_written);
if (result < 0)
{
result = snd_pcm_recover(pcm, result, 1);
if (result < 0)
{
break;
}
}
else
{
frames_written += result;
}
}
}
bool CAESinkALSA::Initialize(AEAudioFormat &format, std::string &device)
{
m_initDevice = device;
m_initFormat = format;
/* if we are raw, correct the data format */
if (AE_IS_RAW(format.m_dataFormat))
{
m_channelLayout = GetChannelLayout(format);
format.m_dataFormat = AE_FMT_S16NE;
m_passthrough = true;
}
else
{
m_channelLayout = GetChannelLayout(format);
m_passthrough = false;
}
#if defined(HAS_AMLPLAYER) || defined(HAS_LIBAMCODEC)
if (aml_present())
{
aml_set_audio_passthrough(m_passthrough);
device = "default";
}
#endif
if (m_channelLayout.Count() == 0)
{
CLog::Log(LOGERROR, "CAESinkALSA::Initialize - Unable to open the requested channel layout");
return false;
}
format.m_channelLayout = m_channelLayout;
AEDeviceType devType = AEDeviceTypeFromName(device);
std::string AESParams;
/* digital interfaces should have AESx set, though in practice most
* receivers don't care */
if (m_passthrough || devType == AE_DEVTYPE_HDMI || devType == AE_DEVTYPE_IEC958)
GetAESParams(format, AESParams);
CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Attempting to open device \"%s\"", device.c_str());
/* get the sound config */
snd_config_t *config;
snd_config_copy(&config, snd_config);
if (!OpenPCMDevice(device, AESParams, m_channelLayout.Count(), &m_pcm, config))
{
CLog::Log(LOGERROR, "CAESinkALSA::Initialize - failed to initialize device \"%s\"", device.c_str());
snd_config_delete(config);
return false;
}
/* get the actual device name that was used */
device = snd_pcm_name(m_pcm);
m_device = device;
CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Opened device \"%s\"", device.c_str());
/* free the sound config */
snd_config_delete(config);
if (!InitializeHW(format) || !InitializeSW(format))
return false;
snd_pcm_nonblock(m_pcm, 1);
snd_pcm_prepare (m_pcm);
m_format = format;
m_formatSampleRateMul = 1.0 / (double)m_format.m_sampleRate;
return true;
}
bool QAudioInputPrivate::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 freakuency=settings.frequency();
if (!settings.isValid()) {
qWarning("QAudioOutput: open error, invalid format.");
} else if (settings.sampleRate() <= 0) {
qWarning("QAudioOutput: 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 = QAudioDeviceInfoInternal::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.channels() );
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, &freakuency, 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;
}
/*
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 snd_pcm_t *
alsa_open (int channels, unsigned samplerate, int realtime)
{ const char * device = "default" ;
snd_pcm_t *alsa_dev = NULL ;
snd_pcm_hw_params_t *hw_params ;
snd_pcm_uframes_t buffer_size ;
snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
snd_pcm_sw_params_t *sw_params ;
int err ;
if (realtime)
{ alsa_period_size = 256 ;
alsa_buffer_frames = 3 * alsa_period_size ;
}
else
{ alsa_period_size = 1024 ;
alsa_buffer_frames = 4 * alsa_period_size ;
} ;
if ((err = snd_pcm_open (&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{ fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_nonblock (alsa_dev, 0) ;
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{ fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
{ fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{ fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
{ fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_rate_near (alsa_dev, hw_params, &samplerate, 0)) < 0)
{ fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_channels (alsa_dev, hw_params, channels)) < 0)
{ fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_dev, hw_params, &alsa_buffer_frames)) < 0)
{ fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_period_size_near (alsa_dev, hw_params, &alsa_period_size, 0)) < 0)
{ fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
{ fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
/* extra check: if we have only one period, this code won't work */
snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
if (alsa_period_size == buffer_size)
{ fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
goto catch_error ;
} ;
snd_pcm_hw_params_free (hw_params) ;
if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params_current (alsa_dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
/* note: set start threshold to delay start until the ring buffer is full */
snd_pcm_sw_params_current (alsa_dev, sw_params) ;
if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, buffer_size)) < 0)
{ fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params (alsa_dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_sw_params_free (sw_params) ;
snd_pcm_reset (alsa_dev) ;
catch_error :
if (err < 0 && alsa_dev != NULL)
{ snd_pcm_close (alsa_dev) ;
return NULL ;
} ;
return alsa_dev ;
} /* alsa_open */
void SetupSound(void)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_status_t *status;
int pspeed;
int pchannels;
int format;
int buffer_time;
int period_time;
int err;
if(iDisStereo) pchannels=1;
else pchannels=2;
pspeed=48000;
format=SND_PCM_FORMAT_S16_LE;
buffer_time=500000;
period_time=buffer_time/4;
if((err=snd_pcm_open(&handle, "default",
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))<0)
{
printf("Audio open error: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_nonblock(handle, 0))<0)
{
printf("Can't set blocking moded: %s\n", snd_strerror(err));
return;
}
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
if((err=snd_pcm_hw_params_any(handle, hwparams))<0)
{
printf("Broken configuration for this PCM: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED))<0)
{
printf("Access type not available: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_format(handle, hwparams, format))<0)
{
printf("Sample format not available: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_channels(handle, hwparams, pchannels))<0)
{
printf("Channels count not available: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_rate_near(handle, hwparams, &pspeed, 0))<0)
{
printf("Rate not available: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0))<0)
{
printf("Buffer time error: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0))<0)
{
printf("Period time error: %s\n", snd_strerror(err));
return;
}
if((err=snd_pcm_hw_params(handle, hwparams))<0)
{
printf("Unable to install hw params: %s\n", snd_strerror(err));
return;
}
snd_pcm_status_alloca(&status);
if((err=snd_pcm_status(handle, status))<0)
{
printf("Unable to get status: %s\n", snd_strerror(err));
return;
}
buffer_size=snd_pcm_status_get_avail(status);
}
bool QAlsaAudioDeviceInfo::testSettings(const QAudioFormat& format) const
{
// Set nearest to closest settings that do work.
// See if what is in settings will work (return value).
int err = -1;
snd_pcm_t* pcmHandle;
snd_pcm_hw_params_t *params;
QString dev;
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = device;
if (dev.compare(QLatin1String("default")) == 0) {
QList<QByteArray> devices = availableDevices(QAudio::AudioOutput);
if (!devices.isEmpty())
dev = QLatin1String(devices.first().constData());
}
#else
if (dev.compare(QLatin1String("default")) == 0) {
dev = QLatin1String("hw:0,0");
} else {
int idx = 0;
char *name;
QString shortName = device.mid(device.indexOf(QLatin1String("="),0)+1);
while(snd_card_get_name(idx,&name) == 0) {
if(shortName.compare(QLatin1String(name)) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
}
#endif
snd_pcm_stream_t stream = mode == QAudio::AudioOutput
? SND_PCM_STREAM_PLAYBACK : SND_PCM_STREAM_CAPTURE;
if (snd_pcm_open(&pcmHandle, dev.toLocal8Bit().constData(), stream, 0) < 0)
return false;
snd_pcm_nonblock(pcmHandle, 0);
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(pcmHandle, params);
// set the values!
snd_pcm_hw_params_set_channels(pcmHandle, params, format.channelCount());
snd_pcm_hw_params_set_rate(pcmHandle, params, format.sampleRate(), 0);
snd_pcm_format_t pcmFormat = SND_PCM_FORMAT_UNKNOWN;
switch (format.sampleSize()) {
case 8:
if (format.sampleType() == QAudioFormat::SignedInt)
pcmFormat = SND_PCM_FORMAT_S8;
else if (format.sampleType() == QAudioFormat::UnSignedInt)
pcmFormat = SND_PCM_FORMAT_U8;
break;
case 16:
if (format.sampleType() == QAudioFormat::SignedInt) {
pcmFormat = format.byteOrder() == QAudioFormat::LittleEndian
? SND_PCM_FORMAT_S16_LE : SND_PCM_FORMAT_S16_BE;
} else if (format.sampleType() == QAudioFormat::UnSignedInt) {
pcmFormat = format.byteOrder() == QAudioFormat::LittleEndian
? SND_PCM_FORMAT_U16_LE : SND_PCM_FORMAT_U16_BE;
}
break;
case 32:
if (format.sampleType() == QAudioFormat::SignedInt) {
pcmFormat = format.byteOrder() == QAudioFormat::LittleEndian
? SND_PCM_FORMAT_S32_LE : SND_PCM_FORMAT_S32_BE;
} else if (format.sampleType() == QAudioFormat::UnSignedInt) {
pcmFormat = format.byteOrder() == QAudioFormat::LittleEndian
? SND_PCM_FORMAT_U32_LE : SND_PCM_FORMAT_U32_BE;
} else if (format.sampleType() == QAudioFormat::Float) {
pcmFormat = format.byteOrder() == QAudioFormat::LittleEndian
? SND_PCM_FORMAT_FLOAT_LE : SND_PCM_FORMAT_FLOAT_BE;
}
}
if (pcmFormat != SND_PCM_FORMAT_UNKNOWN)
err = snd_pcm_hw_params_set_format(pcmHandle, params, pcmFormat);
// For now, just accept only audio/pcm codec
if (!format.codec().startsWith(QLatin1String("audio/pcm")))
err = -1;
if (err >= 0 && format.channelCount() != -1) {
err = snd_pcm_hw_params_test_channels(pcmHandle, params, format.channelCount());
if (err >= 0)
err = snd_pcm_hw_params_set_channels(pcmHandle, params, format.channelCount());
}
if (err >= 0 && format.sampleRate() != -1) {
err = snd_pcm_hw_params_test_rate(pcmHandle, params, format.sampleRate(), 0);
if (err >= 0)
err = snd_pcm_hw_params_set_rate(pcmHandle, params, format.sampleRate(), 0);
}
if (err >= 0 && pcmFormat != SND_PCM_FORMAT_UNKNOWN)
err = snd_pcm_hw_params_set_format(pcmHandle, params, pcmFormat);
if (err >= 0)
err = snd_pcm_hw_params(pcmHandle, params);
snd_pcm_close(pcmHandle);
return (err == 0);
}
void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource,
int encoding, float sampleRate, int sampleSizeInBits,
int frameSize, int channels,
int isSigned, int isBigEndian, int bufferSizeInBytes) {
snd_pcm_format_mask_t* formatMask;
snd_pcm_format_t format;
int dir;
int ret = 0;
AlsaPcmInfo* info = NULL;
/* snd_pcm_uframes_t is 64 bit on 64-bit systems */
snd_pcm_uframes_t alsaPeriodSize = 0;
snd_pcm_uframes_t alsaBufferSizeInFrames = 0;
TRACE0("> DAUDIO_Open\n");
#ifdef USE_TRACE
// for using ALSA debug dump methods
if (ALSA_OUTPUT == NULL) {
snd_output_stdio_attach(&ALSA_OUTPUT, stdout, 0);
}
#endif
info = (AlsaPcmInfo*) malloc(sizeof(AlsaPcmInfo));
if (!info) {
ERROR0("Out of memory\n");
return NULL;
}
memset(info, 0, sizeof(AlsaPcmInfo));
ret = openPCMfromDeviceID(deviceID, &(info->handle), isSource, FALSE /* do open device*/);
if (ret == 0) {
// set to blocking mode
snd_pcm_nonblock(info->handle, 0);
ret = snd_pcm_hw_params_malloc(&(info->hwParams));
if (ret != 0) {
ERROR1(" snd_pcm_hw_params_malloc returned error %d\n", ret);
} else {
ret = -1;
if (getAlsaFormatFromFormat(&format, frameSize / channels, sampleSizeInBits,
isSigned, isBigEndian, encoding)) {
if (setHWParams(info,
sampleRate,
channels,
bufferSizeInBytes / frameSize,
format)) {
info->frameSize = frameSize;
#ifdef ALSA_PCM_NEW_HW_PARAMS_API
ret = snd_pcm_hw_params_get_period_size(info->hwParams, &alsaPeriodSize, &dir);
info->periodSize = (int) alsaPeriodSize;
if (ret < 0) {
ERROR1("ERROR: snd_pcm_hw_params_get_period: %s\n", snd_strerror(ret));
}
snd_pcm_hw_params_get_periods(info->hwParams, &(info->periods), &dir);
snd_pcm_hw_params_get_buffer_size(info->hwParams, &alsaBufferSizeInFrames);
info->bufferSizeInBytes = (int) alsaBufferSizeInFrames * frameSize;
#else
info->periodSize = snd_pcm_hw_params_get_period_size(info->hwParams, &dir);
info->periods = snd_pcm_hw_params_get_periods(info->hwParams, &dir);
info->bufferSizeInBytes = snd_pcm_hw_params_get_buffer_size(info->hwParams) * frameSize;
ret = 0;
#endif
TRACE3(" DAUDIO_Open: period size = %d frames, periods = %d. Buffer size: %d bytes.\n",
(int) info->periodSize, info->periods, info->bufferSizeInBytes);
}
}
}
if (ret == 0) {
// set software parameters
ret = snd_pcm_sw_params_malloc(&(info->swParams));
if (ret != 0) {
ERROR1("snd_pcm_hw_params_malloc returned error %d\n", ret);
} else {
if (!setSWParams(info)) {
ret = -1;
}
}
}
if (ret == 0) {
// prepare device
ret = snd_pcm_prepare(info->handle);
if (ret < 0) {
ERROR1("ERROR: snd_pcm_prepare: %s\n", snd_strerror(ret));
}
}
#ifdef GET_POSITION_METHOD2
if (ret == 0) {
ret = snd_pcm_status_malloc(&(info->positionStatus));
if (ret != 0) {
ERROR1("ERROR in snd_pcm_status_malloc: %s\n", snd_strerror(ret));
}
}
#endif
}
if (ret != 0) {
DAUDIO_Close((void*) info, isSource);
info = NULL;
} else {
// set to non-blocking mode
snd_pcm_nonblock(info->handle, 1);
TRACE1("< DAUDIO_Open: Opened device successfully. Handle=%p\n",
(void*) info->handle);
}
return (void*) info;
}
/*****************************************************************
* \brief init audio device
* \return STREAM_OK if success, STREAM_ERROR otherwise
*/
static int init_audio(radio_priv_t *priv)
{
int is_oss=1;
int seconds=2;
char* tmp;
if (priv->audio_inited) return 1;
/* do_capture==0 mplayer was not started with capture keyword, so disabling capture*/
if(!priv->do_capture)
return STREAM_OK;
if (!priv->radio_param->adevice){
priv->do_capture=0;
return STREAM_OK;
}
priv->do_capture=1;
mp_msg(MSGT_RADIO,MSGL_V,MSGTR_RADIO_CaptureStarting);
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
while ((tmp = strrchr(priv->radio_param->adevice, '='))){
tmp[0] = ':';
//adevice option looks like ALSA device name. Switching to ALSA
is_oss=0;
}
while ((tmp = strrchr(priv->radio_param->adevice, '.')))
tmp[0] = ',';
#endif
if(audio_in_init(&priv->audio_in, is_oss?AUDIO_IN_OSS:AUDIO_IN_ALSA)<0){
mp_msg(MSGT_RADIO, MSGL_ERR, MSGTR_RADIO_AudioInInitFailed);
}
audio_in_set_device(&priv->audio_in, priv->radio_param->adevice);
audio_in_set_channels(&priv->audio_in, priv->radio_param->achannels);
audio_in_set_samplerate(&priv->audio_in, priv->radio_param->arate);
if (audio_in_setup(&priv->audio_in) < 0) {
mp_msg(MSGT_RADIO, MSGL_ERR, MSGTR_RADIO_AudioInSetupFailed, strerror(errno));
return STREAM_ERROR;
}
#ifdef USE_OSS_AUDIO
if(is_oss)
ioctl(priv->audio_in.oss.audio_fd, SNDCTL_DSP_NONBLOCK, 0);
#endif
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
if(!is_oss)
snd_pcm_nonblock(priv->audio_in.alsa.handle,1);
#endif
priv->audio_buffer_size = seconds*priv->audio_in.samplerate*priv->audio_in.channels*
priv->audio_in.bytes_per_sample+priv->audio_in.blocksize;
if (priv->audio_buffer_size < 256*priv->audio_in.blocksize)
priv->audio_buffer_size = 256*priv->audio_in.blocksize;
mp_msg(MSGT_RADIO, MSGL_V, MSGTR_RADIO_AudioBuffer,
priv->audio_buffer_size,priv->audio_in.blocksize);
/* start capture */
priv->audio_ringbuffer = calloc(1, priv->audio_buffer_size);
if (!priv->audio_ringbuffer) {
mp_msg(MSGT_RADIO, MSGL_ERR, MSGTR_RADIO_AllocateBufferFailed,priv->audio_in.blocksize, priv->audio_buffer_size, strerror(errno));
return STREAM_ERROR;
}
priv->audio_head = 0;
priv->audio_tail = 0;
priv->audio_cnt = 0;
priv->audio_drop = 0;
priv->audio_inited = 1;
return STREAM_OK;
}
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int status;
snd_pcm_hw_params_t *params;
snd_pcm_format_t format;
snd_pcm_uframes_t frames;
Uint16 test_format;
/* Open the audio device */
status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if ( status < 0 ) {
SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));
return(-1);
}
/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(¶ms);
status = snd_pcm_hw_params_any(pcm_handle, params);
if ( status < 0 ) {
SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
/* SDL only uses interleaved sample output */
status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if ( status < 0 ) {
SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
/* Try for a closest match on audio format */
status = -1;
for ( test_format = SDL_FirstAudioFormat(spec->format);
test_format && (status < 0); ) {
switch ( test_format ) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
default:
format = 0;
break;
}
if ( format != 0 ) {
status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
}
if ( status < 0 ) {
test_format = SDL_NextAudioFormat();
}
}
if ( status < 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
ALSA_CloseAudio(this);
return(-1);
}
spec->format = test_format;
/* Set the number of channels */
status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
if ( status < 0 ) {
status = snd_pcm_hw_params_get_channels(params);
if ( (status <= 0) || (status > 2) ) {
SDL_SetError("Couldn't set audio channels");
ALSA_CloseAudio(this);
return(-1);
}
spec->channels = status;
}
/* Set the audio rate */
status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
if ( status < 0 ) {
SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
spec->freq = status;
/* Set the buffer size, in samples */
frames = spec->samples;
frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
spec->samples = frames;
snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);
/* "set" the hardware with the desired parameters */
status = snd_pcm_hw_params(pcm_handle, params);
if ( status < 0 ) {
SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
ALSA_CloseAudio(this);
return(-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
ALSA_CloseAudio(this);
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* Switch to blocking mode for playback */
snd_pcm_nonblock(pcm_handle, 0);
/* We're ready to rock and roll. :-) */
return(0);
}
int run(char *filename)
{
capture_stop = 0;
char *pcm_name = "default";
int tmp, err;
snd_pcm_info_t *info;
snd_pcm_info_alloca(&info);
err = snd_output_stdio_attach(&log, stderr, 0);
assert(err >= 0);
file_type = FORMAT_DEFAULT;
stream = SND_PCM_STREAM_CAPTURE;
file_type = FORMAT_WAVE;
command = "arecord";
start_delay = 1;
chunk_size = -1;
rhwparams.format = DEFAULT_FORMAT;
rhwparams.rate = DEFAULT_SPEED;
rhwparams.channels = 1;
file_type = FORMAT_WAVE;
// cdr:
// rhwparams.format = SND_PCM_FORMAT_S16_BE;
rhwparams.format = file_type == FORMAT_AU ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_S16_LE;
rhwparams.rate = 44100;
rhwparams.channels = 2;
err = snd_pcm_open(&handle, pcm_name, stream, open_mode);
if (err < 0) {
error(_("audio open error: %s"), snd_strerror(err));
return 1;
}
if ((err = snd_pcm_info(handle, info)) < 0) {
error(_("info error: %s"), snd_strerror(err));
return 1;
}
if (nonblock) {
err = snd_pcm_nonblock(handle, 1);
if (err < 0) {
error(_("nonblock setting error: %s"), snd_strerror(err));
return 1;
}
}
chunk_size = 1024;
hwparams = rhwparams;
audiobuf = (u_char *)malloc(1024);
if (audiobuf == NULL) {
error(_("not enough memory"));
return 1;
}
writei_func = snd_pcm_writei;
readi_func = snd_pcm_readi;
writen_func = snd_pcm_writen;
readn_func = snd_pcm_readn;
//signal(SIGINT, signal_handler);
//signal(SIGTERM, signal_handler);
//signal(SIGABRT, signal_handler);
capture(filename);
if (fmt_rec_table[file_type].end) {
fmt_rec_table[file_type].end(fd);
fd = -1;
}
stream = -1;
if (fd > 1) {
close(fd);
fd = -1;
}
if (handle) {
snd_pcm_close(handle);
handle = NULL;
}
//snd_pcm_close(handle);
//free(audiobuf);
//snd_output_close(log);
//snd_config_update_free_global();
return EXIT_SUCCESS;
}
/*****************************************************************************
* OpenAudioDev: open and set up the audio device and probe for capabilities
*****************************************************************************/
static int OpenAudioDevAlsa( demux_t *p_demux, const char *psz_device )
{
demux_sys_t *p_sys = p_demux->p_sys;
p_sys->p_alsa_pcm = NULL;
snd_pcm_hw_params_t *p_hw_params = NULL;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t chunk_size;
/* ALSA */
int i_err;
if( ( i_err = snd_pcm_open( &p_sys->p_alsa_pcm, psz_device,
SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK ) ) < 0)
{
msg_Err( p_demux, "Cannot open ALSA audio device %s (%s)",
psz_device, snd_strerror( i_err ) );
goto adev_fail;
}
if( ( i_err = snd_pcm_nonblock( p_sys->p_alsa_pcm, 1 ) ) < 0)
{
msg_Err( p_demux, "Cannot set ALSA nonblock (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Begin setting hardware parameters */
if( ( i_err = snd_pcm_hw_params_malloc( &p_hw_params ) ) < 0 )
{
msg_Err( p_demux,
"ALSA: cannot allocate hardware parameter structure (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
if( ( i_err = snd_pcm_hw_params_any( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
{
msg_Err( p_demux,
"ALSA: cannot initialize hardware parameter structure (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Set Interleaved access */
if( ( i_err = snd_pcm_hw_params_set_access( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot set access type (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Set 16 bit little endian */
if( ( i_err = snd_pcm_hw_params_set_format( p_sys->p_alsa_pcm, p_hw_params, SND_PCM_FORMAT_S16_LE ) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot set sample format (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Set sample rate */
i_err = snd_pcm_hw_params_set_rate_near( p_sys->p_alsa_pcm, p_hw_params, &p_sys->i_sample_rate, NULL );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set sample rate (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Set channels */
unsigned int channels = p_sys->b_stereo ? 2 : 1;
if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
{
channels = ( channels==1 ) ? 2 : 1;
msg_Warn( p_demux, "ALSA: cannot set channel count (%s). "
"Trying with channels=%d",
snd_strerror( i_err ),
channels );
if( ( i_err = snd_pcm_hw_params_set_channels( p_sys->p_alsa_pcm, p_hw_params, channels ) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot set channel count (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
p_sys->b_stereo = ( channels == 2 );
}
/* Set metrics for buffer calculations later */
unsigned int buffer_time;
if( ( i_err = snd_pcm_hw_params_get_buffer_time_max(p_hw_params, &buffer_time, 0) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot get buffer time max (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
if( buffer_time > 500000 ) buffer_time = 500000;
/* Set period time */
unsigned int period_time = buffer_time / 4;
i_err = snd_pcm_hw_params_set_period_time_near( p_sys->p_alsa_pcm, p_hw_params, &period_time, 0 );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set period time (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Set buffer time */
i_err = snd_pcm_hw_params_set_buffer_time_near( p_sys->p_alsa_pcm, p_hw_params, &buffer_time, 0 );
if( i_err < 0 )
{
msg_Err( p_demux, "ALSA: cannot set buffer time (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Apply new hardware parameters */
if( ( i_err = snd_pcm_hw_params( p_sys->p_alsa_pcm, p_hw_params ) ) < 0 )
{
msg_Err( p_demux, "ALSA: cannot set hw parameters (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
/* Get various buffer metrics */
snd_pcm_hw_params_get_period_size( p_hw_params, &chunk_size, 0 );
snd_pcm_hw_params_get_buffer_size( p_hw_params, &buffer_size );
if( chunk_size == buffer_size )
{
msg_Err( p_demux,
"ALSA: period cannot equal buffer size (%lu == %lu)",
chunk_size, buffer_size);
goto adev_fail;
}
int bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
int bits_per_frame = bits_per_sample * channels;
p_sys->i_alsa_chunk_size = chunk_size;
p_sys->i_alsa_frame_size = bits_per_frame / 8;
p_sys->i_max_frame_size = chunk_size * bits_per_frame / 8;
snd_pcm_hw_params_free( p_hw_params );
p_hw_params = NULL;
/* Prep device */
if( ( i_err = snd_pcm_prepare( p_sys->p_alsa_pcm ) ) < 0 )
{
msg_Err( p_demux,
"ALSA: cannot prepare audio interface for use (%s)",
snd_strerror( i_err ) );
goto adev_fail;
}
snd_pcm_start( p_sys->p_alsa_pcm );
return VLC_SUCCESS;
adev_fail:
if( p_hw_params ) snd_pcm_hw_params_free( p_hw_params );
if( p_sys->p_alsa_pcm ) snd_pcm_close( p_sys->p_alsa_pcm );
p_sys->p_alsa_pcm = NULL;
return VLC_EGENERIC;
}
/*****************************************************************************
* Open: create a handle and open an alsa device
*****************************************************************************
* This function opens an alsa device, through the alsa API.
*
* Note: the only heap-allocated string is psz_device. All the other pointers
* are references to psz_device or to stack-allocated data.
*****************************************************************************/
static int Open( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t *)p_this;
struct aout_sys_t * p_sys;
vlc_value_t val;
char psz_default_iec_device[128]; /* Buffer used to store the default
S/PDIF device */
char * psz_device, * psz_iec_device; /* device names for PCM and S/PDIF
output */
int i_vlc_pcm_format; /* Audio format for VLC's data */
int i_snd_pcm_format; /* Audio format for ALSA's data */
snd_pcm_uframes_t i_buffer_size = 0;
snd_pcm_uframes_t i_period_size = 0;
int i_channels = 0;
snd_pcm_hw_params_t *p_hw;
snd_pcm_sw_params_t *p_sw;
int i_snd_rc = -1;
unsigned int i_old_rate;
bool b_retry = true;
/* Allocate structures */
p_aout->output.p_sys = p_sys = malloc( sizeof( aout_sys_t ) );
if( p_sys == NULL )
return VLC_ENOMEM;
p_sys->b_playing = false;
p_sys->start_date = 0;
vlc_cond_init( &p_sys->wait );
vlc_mutex_init( &p_sys->lock );
/* Get device name */
if( (psz_device = config_GetPsz( p_aout, "alsa-audio-device" )) == NULL )
{
msg_Err( p_aout, "no audio device given (maybe \"default\" ?)" );
dialog_Fatal( p_aout, _("No Audio Device"), "%s",
_("No audio device name was given. You might want to " \
"enter \"default\".") );
free( p_sys );
return VLC_EGENERIC;
}
/* Choose the IEC device for S/PDIF output:
if the device is overriden by the user then it will be the one
otherwise we compute the default device based on the output format. */
if( AOUT_FMT_NON_LINEAR( &p_aout->output.output )
&& !strcmp( psz_device, DEFAULT_ALSA_DEVICE ) )
{
snprintf( psz_default_iec_device, sizeof(psz_default_iec_device),
"iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
IEC958_AES0_CON_EMPHASIS_NONE | IEC958_AES0_NONAUDIO,
IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
0,
( p_aout->output.output.i_rate == 48000 ?
IEC958_AES3_CON_FS_48000 :
( p_aout->output.output.i_rate == 44100 ?
IEC958_AES3_CON_FS_44100 : IEC958_AES3_CON_FS_32000 ) ) );
psz_iec_device = psz_default_iec_device;
}
else if( AOUT_FMT_NON_LINEAR( &p_aout->output.output ) )
{
psz_iec_device = psz_device;
}
else
{
psz_iec_device = NULL;
}
/* Choose the linear PCM format (read the comment above about FPU
and float32) */
if( vlc_CPU() & CPU_CAPABILITY_FPU )
{
i_vlc_pcm_format = VLC_CODEC_FL32;
i_snd_pcm_format = SND_PCM_FORMAT_FLOAT;
}
else
{
i_vlc_pcm_format = VLC_CODEC_S16N;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
}
/* If the variable doesn't exist then it's the first time we're called
and we have to probe the available audio formats and channels */
if ( var_Type( p_aout, "audio-device" ) == 0 )
{
Probe( p_aout, psz_device, psz_iec_device, &i_snd_pcm_format );
}
if ( var_Get( p_aout, "audio-device", &val ) < 0 )
{
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
p_aout->output.output.i_format = i_vlc_pcm_format;
if ( val.i_int == AOUT_VAR_5_1 )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT
| AOUT_CHAN_LFE;
free( psz_device );
psz_device = strdup( "surround51" );
}
else if ( val.i_int == AOUT_VAR_2F2R )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT
| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT;
free( psz_device );
psz_device = strdup( "surround40" );
}
else if ( val.i_int == AOUT_VAR_STEREO )
{
p_aout->output.output.i_physical_channels
= AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT;
}
else if ( val.i_int == AOUT_VAR_MONO )
{
p_aout->output.output.i_physical_channels = AOUT_CHAN_CENTER;
}
else if( val.i_int != AOUT_VAR_SPDIF )
{
/* This should not happen ! */
msg_Err( p_aout, "internal: can't find audio-device (%i)", val.i_int );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
#ifdef ALSA_DEBUG
snd_output_stdio_attach( &p_sys->p_snd_stderr, stderr, 0 );
#endif
/* Open the device */
if ( val.i_int == AOUT_VAR_SPDIF )
{
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_iec_device,
SND_PCM_STREAM_PLAYBACK, 0 ) ) < 0 )
{
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_iec_device, snd_strerror( i_snd_rc ) );
dialog_Fatal( p_aout, _("Audio output failed"),
_("VLC could not open the ALSA device \"%s\" (%s)."),
psz_iec_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
i_buffer_size = ALSA_SPDIF_BUFFER_SIZE;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
i_channels = 2;
i_vlc_pcm_format = VLC_CODEC_SPDIFL;
p_aout->output.i_nb_samples = i_period_size = ALSA_SPDIF_PERIOD_SIZE;
p_aout->output.output.i_bytes_per_frame = AOUT_SPDIF_SIZE;
p_aout->output.output.i_frame_length = A52_FRAME_NB;
aout_VolumeNoneInit( p_aout );
}
else
{
int i;
msg_Dbg( p_aout, "opening ALSA device `%s'", psz_device );
/* Since it seems snd_pcm_close hasn't really released the device at
the time it returns, probe if the device is available in loop for 1s.
We cannot use blocking mode since the we would wait indefinitely when
switching from a dmx device to surround51. */
for( i = 10; i >= 0; i-- )
{
if ( ( i_snd_rc = snd_pcm_open( &p_sys->p_snd_pcm, psz_device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) == -EBUSY )
{
if( i ) msleep( 100000 /* 100ms */ );
else
{
msg_Err( p_aout, "audio device: %s is already in use",
psz_device );
dialog_Fatal( p_aout, _("Audio output failed"),
_("The audio device \"%s\" is already in use."),
psz_device );
}
continue;
}
break;
}
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "cannot open ALSA device `%s' (%s)",
psz_device, snd_strerror( i_snd_rc ) );
dialog_Fatal( p_aout, _("Audio output failed"),
_("VLC could not open the ALSA device \"%s\" (%s)."),
psz_device, snd_strerror( i_snd_rc ) );
free( p_sys );
free( psz_device );
return VLC_EGENERIC;
}
/* We want blocking mode */
snd_pcm_nonblock( p_sys->p_snd_pcm, 0 );
i_buffer_size = ALSA_DEFAULT_BUFFER_SIZE;
i_channels = aout_FormatNbChannels( &p_aout->output.output );
p_aout->output.i_nb_samples = i_period_size = ALSA_DEFAULT_PERIOD_SIZE;
aout_VolumeSoftInit( p_aout );
}
/* Free psz_device so that all the remaining data is stack-allocated */
free( psz_device );
p_aout->output.pf_play = Play;
snd_pcm_hw_params_alloca(&p_hw);
snd_pcm_sw_params_alloca(&p_sw);
/* Due to some bugs in alsa with some drivers, we need to retry in s16l
if snd_pcm_hw_params fails in fl32 */
while ( b_retry )
{
b_retry = false;
/* Get Initial hardware parameters */
if ( ( i_snd_rc = snd_pcm_hw_params_any( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
msg_Err( p_aout, "unable to retrieve initial hardware parameters (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set format. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm, p_hw,
i_snd_pcm_format ) ) < 0 )
{
if( i_snd_pcm_format != SND_PCM_FORMAT_S16 )
{
i_snd_pcm_format = SND_PCM_FORMAT_S16;
i_snd_rc = snd_pcm_hw_params_set_format( p_sys->p_snd_pcm,
p_hw, i_snd_pcm_format );
}
if ( i_snd_rc < 0 )
{
msg_Err( p_aout, "unable to set stream sample size and "
"word order (%s)", snd_strerror( i_snd_rc ) );
goto error;
}
}
if( i_vlc_pcm_format != VLC_CODEC_SPDIFL )
switch( i_snd_pcm_format )
{
case SND_PCM_FORMAT_FLOAT:
i_vlc_pcm_format = VLC_CODEC_FL32;
break;
case SND_PCM_FORMAT_S16:
i_vlc_pcm_format = VLC_CODEC_S16N;
break;
}
p_aout->output.output.i_format = i_vlc_pcm_format;
if ( ( i_snd_rc = snd_pcm_hw_params_set_access( p_sys->p_snd_pcm, p_hw,
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
msg_Err( p_aout, "unable to set interleaved stream format (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set channels. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_channels( p_sys->p_snd_pcm, p_hw,
i_channels ) ) < 0 )
{
msg_Err( p_aout, "unable to set number of output channels (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Set rate. */
i_old_rate = p_aout->output.output.i_rate;
i_snd_rc = snd_pcm_hw_params_set_rate_near( p_sys->p_snd_pcm, p_hw,
&p_aout->output.output.i_rate,
NULL );
if( i_snd_rc < 0 || p_aout->output.output.i_rate != i_old_rate )
{
msg_Warn( p_aout, "The rate %d Hz is not supported by your " \
"hardware. Using %d Hz instead.\n", i_old_rate, \
p_aout->output.output.i_rate );
}
/* Set period size. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_period_size_near( p_sys->p_snd_pcm,
p_hw, &i_period_size, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to set period size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
p_aout->output.i_nb_samples = i_period_size;
/* Set buffer size. */
if ( ( i_snd_rc = snd_pcm_hw_params_set_buffer_size_near( p_sys->p_snd_pcm,
p_hw, &i_buffer_size ) ) < 0 )
{
msg_Err( p_aout, "unable to set buffer size (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Commit hardware parameters. */
if ( ( i_snd_rc = snd_pcm_hw_params( p_sys->p_snd_pcm, p_hw ) ) < 0 )
{
if ( b_retry == false &&
i_snd_pcm_format == SND_PCM_FORMAT_FLOAT)
{
b_retry = true;
i_snd_pcm_format = SND_PCM_FORMAT_S16;
p_aout->output.output.i_format = VLC_CODEC_S16N;
msg_Warn( p_aout, "unable to commit hardware configuration "
"with fl32 samples. Retrying with s16l (%s)", snd_strerror( i_snd_rc ) );
}
else
{
msg_Err( p_aout, "unable to commit hardware configuration (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
}
}
if( ( i_snd_rc = snd_pcm_hw_params_get_period_time( p_hw,
&p_sys->i_period_time, NULL ) ) < 0 )
{
msg_Err( p_aout, "unable to get period time (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Get Initial software parameters */
snd_pcm_sw_params_current( p_sys->p_snd_pcm, p_sw );
i_snd_rc = snd_pcm_sw_params_set_sleep_min( p_sys->p_snd_pcm, p_sw, 0 );
i_snd_rc = snd_pcm_sw_params_set_avail_min( p_sys->p_snd_pcm, p_sw,
p_aout->output.i_nb_samples );
/* start playing when one period has been written */
i_snd_rc = snd_pcm_sw_params_set_start_threshold( p_sys->p_snd_pcm, p_sw,
ALSA_DEFAULT_PERIOD_SIZE);
if( i_snd_rc < 0 )
{
msg_Err( p_aout, "unable to set start threshold (%s)",
snd_strerror( i_snd_rc ) );
goto error;
}
/* Commit software parameters. */
if ( snd_pcm_sw_params( p_sys->p_snd_pcm, p_sw ) < 0 )
{
msg_Err( p_aout, "unable to set software configuration" );
goto error;
}
#ifdef ALSA_DEBUG
snd_output_printf( p_sys->p_snd_stderr, "\nALSA hardware setup:\n\n" );
snd_pcm_dump_hw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr );
snd_output_printf( p_sys->p_snd_stderr, "\nALSA software setup:\n\n" );
snd_pcm_dump_sw_setup( p_sys->p_snd_pcm, p_sys->p_snd_stderr );
snd_output_printf( p_sys->p_snd_stderr, "\n" );
#endif
/* Create ALSA thread and wait for its readiness. */
if( vlc_thread_create( p_aout, "aout", ALSAThread,
VLC_THREAD_PRIORITY_OUTPUT ) )
{
msg_Err( p_aout, "cannot create ALSA thread (%m)" );
goto error;
}
return 0;
error:
snd_pcm_close( p_sys->p_snd_pcm );
#ifdef ALSA_DEBUG
snd_output_close( p_sys->p_snd_stderr );
#endif
free( p_sys );
return VLC_EGENERIC;
}
/** Plays the audio contained packets queue that is filled by the
* chatting thread with received packets.
*/
void VoiceStreamer::playbackAudio() {
int data_left, r;
packet audioPacket;
uint8_t *data_ptr;
bool done = false;
int pLatency = latency_min - 4;
size_t pframes_in, pframes_out, pin_max;
// Init playback device
if (setparams_p(phandle, &pLatency) < 0) {
fprintf(stderr, "Playback Audio Thread: exiting due to setparams_p error\n");
return;
}
pframes_in = pframes_out = 0;
pin_max = 0;
while (true) {
if (!popAudioPacket(audioPacket))
break;
data_ptr = audioPacket.data;
data_left = (int) ntohs(audioPacket.dlength);
while (pframes_in < loop_limit) {
if (data_left > (pLatency<<2))
r = pLatency;
else
r = data_left>>2;
if (r > 0) {
pframes_in += r;
if ((int) pin_max < r)
pin_max = r;
if (writebuf(phandle, (char *) data_ptr, r, &pframes_out) < 0) {
break;
}
} else if (r == 0) { // done reading input file
done = true;
break;
} else {
break;
}
if (data_left > (pLatency<<2)) {
data_ptr += (pLatency<<2);
data_left -= (pLatency<<2);
} else {
data_ptr += data_left;
data_left = 0;
}
}
if (!done) {
done = false;
snd_pcm_nonblock(phandle, 0);
snd_pcm_drain(phandle);
snd_pcm_hw_free(phandle);
pframes_in = pframes_out = pin_max = 0;
if (setparams_p(phandle, &pLatency) < 0) {
fprintf(stderr, "setparams_p < 0!\n");
break;
}
}
}
startPlayback = false;
fprintf(stderr, "Playback Audio Thread exitting!\n");
}
bool CAESinkALSA::Initialize(AEAudioFormat &format, std::string &device)
{
CAEChannelInfo channelLayout = GetChannelLayout(format, 2, 8);
m_initDevice = device;
m_initFormat = format;
ALSAConfig inconfig, outconfig;
inconfig.format = format.m_dataFormat;
inconfig.sampleRate = format.m_sampleRate;
inconfig.channels = channelLayout.Count();
/* if we are raw, correct the data format */
if (AE_IS_RAW(format.m_dataFormat))
{
inconfig.format = AE_FMT_S16NE;
m_passthrough = true;
}
else
{
m_passthrough = false;
}
#if defined(HAS_LIBAMCODEC)
if (aml_present())
{
aml_set_audio_passthrough(m_passthrough);
device = "default";
}
#endif
if (inconfig.channels == 0)
{
CLog::Log(LOGERROR, "CAESinkALSA::Initialize - Unable to open the requested channel layout");
return false;
}
AEDeviceType devType = AEDeviceTypeFromName(device);
std::string AESParams;
/* digital interfaces should have AESx set, though in practice most
* receivers don't care */
if (m_passthrough || devType == AE_DEVTYPE_HDMI || devType == AE_DEVTYPE_IEC958)
GetAESParams(format, AESParams);
CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Attempting to open device \"%s\"", device.c_str());
/* get the sound config */
snd_config_t *config;
snd_config_copy(&config, snd_config);
if (!OpenPCMDevice(device, AESParams, inconfig.channels, &m_pcm, config))
{
CLog::Log(LOGERROR, "CAESinkALSA::Initialize - failed to initialize device \"%s\"", device.c_str());
snd_config_delete(config);
return false;
}
/* get the actual device name that was used */
device = snd_pcm_name(m_pcm);
m_device = device;
CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Opened device \"%s\"", device.c_str());
/* free the sound config */
snd_config_delete(config);
if (!InitializeHW(inconfig, outconfig) || !InitializeSW(outconfig))
return false;
// we want it blocking
snd_pcm_nonblock(m_pcm, 0);
snd_pcm_prepare (m_pcm);
if (m_passthrough && inconfig.channels != outconfig.channels)
{
CLog::Log(LOGINFO, "CAESinkALSA::Initialize - could not open required number of channels");
return false;
}
// adjust format to the configuration we got
format.m_channelLayout = GetChannelLayout(format, outconfig.channels, outconfig.channels);
format.m_sampleRate = outconfig.sampleRate;
format.m_frames = outconfig.periodSize;
format.m_frameSize = outconfig.frameSize;
format.m_frameSamples = outconfig.periodSize * outconfig.channels;
format.m_dataFormat = outconfig.format;
m_format = format;
m_formatSampleRateMul = 1.0 / (double)m_format.m_sampleRate;
return true;
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SRC (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
#ifdef SND_CHMAP_API_VERSION
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) {
snd_pcm_chmap_t *chmap = snd_pcm_get_chmap (alsa->handle);
if (chmap && chmap->channels == alsa->channels) {
GstAudioChannelPosition pos[8];
if (alsa_chmap_to_channel_positions (chmap, pos))
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(alsa)->ringbuffer, pos);
}
free (chmap);
}
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
/* return 0 on success */
int alsa_open_audio(int naudioindev, int *audioindev, int nchindev,
int *chindev, int naudiooutdev, int *audiooutdev, int nchoutdev,
int *choutdev, int rate, int blocksize)
{
int err, inchans = 0, outchans = 0, subunitdir;
char devname[512];
snd_output_t* out;
int frag_size = (blocksize ? blocksize : ALSA_DEFFRAGSIZE);
int nfrags, i, iodev, dev2;
int wantinchans, wantoutchans, device;
nfrags = sys_schedadvance * (float)rate / (1e6 * frag_size);
/* save our belief as to ALSA's buffer size for later */
alsa_buf_samps = nfrags * frag_size;
alsa_nindev = alsa_noutdev = 0;
alsa_jittermax = ALSA_DEFJITTERMAX;
if (sys_verbose)
post("audio buffer set to %d", (int)(0.001 * sys_schedadvance));
for (iodev = 0; iodev < naudioindev; iodev++)
{
alsa_numbertoname(audioindev[iodev], devname, 512);
err = snd_pcm_open(&alsa_indev[alsa_nindev].a_handle, devname,
SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
check_error(err, 0, "snd_pcm_open");
if (err < 0)
continue;
alsa_indev[alsa_nindev].a_devno = audioindev[iodev];
snd_pcm_nonblock(alsa_indev[alsa_nindev].a_handle, 1);
if (sys_verbose)
post("opened input device name %s", devname);
alsa_nindev++;
}
for (iodev = 0; iodev < naudiooutdev; iodev++)
{
alsa_numbertoname(audiooutdev[iodev], devname, 512);
err = snd_pcm_open(&alsa_outdev[alsa_noutdev].a_handle, devname,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
check_error(err, 1, "snd_pcm_open");
if (err < 0)
continue;
alsa_outdev[alsa_noutdev].a_devno = audiooutdev[iodev];
snd_pcm_nonblock(alsa_outdev[alsa_noutdev].a_handle, 1);
alsa_noutdev++;
}
if (!alsa_nindev && !alsa_noutdev)
goto blewit;
/* If all the open devices support mmap_noninterleaved, let's call
Wini's code in s_audio_alsamm.c */
alsa_usemmap = 1;
for (iodev = 0; iodev < alsa_nindev; iodev++)
if (!alsaio_canmmap(&alsa_indev[iodev]))
alsa_usemmap = 0;
for (iodev = 0; iodev < alsa_noutdev; iodev++)
if (!alsaio_canmmap(&alsa_outdev[iodev]))
alsa_usemmap = 0;
if (alsa_usemmap)
{
post("using mmap audio interface");
if (alsamm_open_audio(rate, blocksize))
goto blewit;
else return (0);
}
for (iodev = 0; iodev < alsa_nindev; iodev++)
{
int channels = chindev[iodev];
if (alsaio_setup(&alsa_indev[iodev], 0, &channels, &rate,
nfrags, frag_size) < 0)
goto blewit;
inchans += channels;
}
for (iodev = 0; iodev < alsa_noutdev; iodev++)
{
int channels = choutdev[iodev];
if (alsaio_setup(&alsa_outdev[iodev], 1, &channels, &rate,
nfrags, frag_size) < 0)
goto blewit;
outchans += channels;
}
if (!inchans && !outchans)
goto blewit;
for (iodev = 0; iodev < alsa_nindev; iodev++)
snd_pcm_prepare(alsa_indev[iodev].a_handle);
for (iodev = 0; iodev < alsa_noutdev; iodev++)
snd_pcm_prepare(alsa_outdev[iodev].a_handle);
/* if duplex we can link the channels so they start together */
for (iodev = 0; iodev < alsa_nindev; iodev++)
for (dev2 = 0; dev2 < alsa_noutdev; dev2++)
{
if (alsa_indev[iodev].a_devno == alsa_outdev[iodev].a_devno)
{
snd_pcm_link(alsa_indev[iodev].a_handle,
alsa_outdev[iodev].a_handle);
}
}
/* allocate the status variables */
if (!alsa_status)
{
err = snd_pcm_status_malloc(&alsa_status);
check_error(err, -1, "snd_pcm_status_malloc");
}
/* fill the buffer with silence and prime the output FIFOs. This
should automatically start the output devices. */
memset(alsa_snd_buf, 0, alsa_snd_bufsize);
if (outchans)
{
i = (frag_size * nfrags)/DEFDACBLKSIZE + 1;
while (i--)
{
for (iodev = 0; iodev < alsa_noutdev; iodev++)
snd_pcm_writei(alsa_outdev[iodev].a_handle, alsa_snd_buf,
DEFDACBLKSIZE);
}
}
if (inchans)
{
/* some of the ADC devices might already have been started by
starting the outputs above, but others might still need it. */
for (iodev = 0; iodev < alsa_nindev; iodev++)
if (snd_pcm_state(alsa_indev[iodev].a_handle)
!= SND_PCM_STATE_RUNNING)
if ((err = snd_pcm_start(alsa_indev[iodev].a_handle)) < 0)
check_error(err, -1, "input start failed");
}
return (0);
blewit:
STUFF->st_inchannels = 0;
STUFF->st_outchannels = 0;
alsa_close_audio();
return (1);
}
bool QAudioDeviceInfoInternal::testSettings(const QAudioFormat& format) const
{
// Set nearest to closest settings that do work.
// See if what is in settings will work (return value).
int err = 0;
snd_pcm_t* handle;
snd_pcm_hw_params_t *params;
QString dev = device;
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 = device;
#else
int idx = 0;
char *name;
QString shortName = device.mid(device.indexOf(QLatin1String("="),0)+1);
while(snd_card_get_name(idx,&name) == 0) {
if(shortName.compare(QLatin1String(name)) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
#endif
}
if(mode == QAudio::AudioOutput) {
err=snd_pcm_open( &handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_PLAYBACK,0);
} else {
err=snd_pcm_open( &handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_CAPTURE,0);
}
if(err < 0) {
handle = 0;
return false;
}
bool testChannel = false;
bool testCodec = false;
bool testFreq = false;
bool testType = false;
bool testSize = false;
int dir = 0;
snd_pcm_nonblock( handle, 0 );
snd_pcm_hw_params_alloca( ¶ms );
snd_pcm_hw_params_any( handle, params );
// set the values!
snd_pcm_hw_params_set_channels(handle,params,format.channels());
snd_pcm_hw_params_set_rate(handle,params,format.frequency(),dir);
err = -1;
switch(format.sampleSize()) {
case 8:
if(format.sampleType() == QAudioFormat::SignedInt)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S8);
else if(format.sampleType() == QAudioFormat::UnSignedInt)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U8);
break;
case 16:
if(format.sampleType() == QAudioFormat::SignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S16_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S16_BE);
} else if(format.sampleType() == QAudioFormat::UnSignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U16_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U16_BE);
}
break;
case 32:
if(format.sampleType() == QAudioFormat::SignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S32_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S32_BE);
} else if(format.sampleType() == QAudioFormat::UnSignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U32_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U32_BE);
}
}
// For now, just accept only audio/pcm codec
if(!format.codec().startsWith(QLatin1String("audio/pcm"))) {
err=-1;
} else
testCodec = true;
if(err>=0 && format.channels() != -1) {
err = snd_pcm_hw_params_test_channels(handle,params,format.channels());
if(err>=0)
err = snd_pcm_hw_params_set_channels(handle,params,format.channels());
if(err>=0)
testChannel = true;
}
if(err>=0 && format.frequency() != -1) {
err = snd_pcm_hw_params_test_rate(handle,params,format.frequency(),0);
if(err>=0)
err = snd_pcm_hw_params_set_rate(handle,params,format.frequency(),dir);
if(err>=0)
testFreq = true;
}
if((err>=0 && format.sampleSize() != -1) &&
(format.sampleType() != QAudioFormat::Unknown)) {
switch(format.sampleSize()) {
case 8:
if(format.sampleType() == QAudioFormat::SignedInt)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S8);
else if(format.sampleType() == QAudioFormat::UnSignedInt)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U8);
break;
case 16:
if(format.sampleType() == QAudioFormat::SignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S16_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S16_BE);
} else if(format.sampleType() == QAudioFormat::UnSignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U16_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U16_BE);
}
break;
case 32:
if(format.sampleType() == QAudioFormat::SignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S32_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_S32_BE);
} else if(format.sampleType() == QAudioFormat::UnSignedInt) {
if(format.byteOrder() == QAudioFormat::LittleEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U32_LE);
else if(format.byteOrder() == QAudioFormat::BigEndian)
err = snd_pcm_hw_params_set_format(handle,params,SND_PCM_FORMAT_U32_BE);
}
}
if(err>=0) {
testSize = true;
testType = true;
}
}
if(err>=0)
err = snd_pcm_hw_params(handle, params);
if(err == 0) {
// settings work
// close()
if(handle)
snd_pcm_close(handle);
return true;
}
if(handle)
snd_pcm_close(handle);
return false;
}
bool QAudioInputPrivate::open( QObject *input )
{
// Open the Alsa capture device.
bool rc = true;
int err;
unsigned int freakuency = frequency;
if ((err = snd_pcm_open(&handle,
m_device.constData(), //"plughw:0,0"
SND_PCM_STREAM_CAPTURE,
0/*SND_PCM_ASYNC*/)) < 0) {
qWarning( "QAudioInput: snd_pcm_open: error %d", err);
rc = false;
}
else {
snd_pcm_hw_params_t *hwparams;
// We want non-blocking mode.
snd_pcm_nonblock(handle, 1);
// Set the desired parameters.
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_any: err %d", err);
}
err = snd_pcm_hw_params_set_access(handle, hwparams,
access);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_access: err %d",err);
}
err = snd_pcm_hw_params_set_format(handle, hwparams,format);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_format: err %d",err);
}
err = snd_pcm_hw_params_set_channels(handle,hwparams,(unsigned int)channels);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_channels: err %d",err);
}
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &freakuency, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_rate_near: err %d",err);
}
if(freakuency > 1.05 * frequency || freakuency < 0.95 * frequency) {
qWarning("QAudioInput: warning, sample rate %i not supported by the hardware, using %u", frequency, freakuency);
}
if ( samplesPerBlock != -1 ) {
// Set buffer and period sizes based on the supplied block size.
sample_size = (snd_pcm_uframes_t)( samplesPerBlock * channels / 8 );
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &freakuency, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_rate_near: err %d",err);
}
if(freakuency > 1.05 * frequency || freakuency < 0.95 * frequency) {
qWarning( "QAudioInput: warning, sample rate %i not supported by the hardware, using %u", frequency, freakuency);
}
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_buffer_time_near: err %d",err);
}
period_time = 1000000 * 256 / frequency;
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_period_time_near: err %d",err);
}
} else {
// Use the largest buffer and period sizes we can.
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_buffer_time_near: err %d",err);
}
period_time = 1000000 * 256 / frequency;
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params_set_period_time_near: err %d",err);
}
}
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_hw_params: err %d",err);
}
int dir;
unsigned int vval, vval2;
snd_pcm_access_t aval;
snd_pcm_format_t fval;
snd_pcm_subformat_t sval;
qLog(QAudioInput) << "PCM handle name = " << snd_pcm_name(handle);
qLog(QAudioInput) << "PCM state = " << snd_pcm_state_name(snd_pcm_state(handle));
snd_pcm_hw_params_get_access(hwparams,&aval);
vval = (unsigned int)aval;
if ( (int)vval != (int)access ) {
qLog(QAudioInput) << QString("access type not set, want %1 got %2")
.arg(snd_pcm_access_name((snd_pcm_access_t)access))
.arg(snd_pcm_access_name((snd_pcm_access_t)vval));
access = (snd_pcm_access_t)vval;
}
qLog(QAudioInput) << "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 ) {
qLog(QAudioInput) << QString("format type not set, want %1 got %2")
.arg(snd_pcm_format_name((snd_pcm_format_t)format))
.arg(snd_pcm_format_name((snd_pcm_format_t)vval));
format = (snd_pcm_format_t)vval;
}
qLog(QAudioInput) << 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(QAudioInput) << 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 ) {
qLog(QAudioInput) << QString("channels type not set, want %1 got %2").arg(channels).arg(vval);
channels = vval;
}
qLog(QAudioInput) << "channels = " << vval;
snd_pcm_hw_params_get_rate(hwparams, &vval, &dir);
if ( (int)vval != (int)frequency ) {
qLog(QAudioInput) << QString("frequency type not set, want %1 got %2").arg(frequency).arg(vval);
frequency = vval;
}
qLog(QAudioInput) << "rate =" << vval << " bps";
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
qLog(QAudioInput) << "period time =" << period_time << " us";
snd_pcm_hw_params_get_period_size(hwparams,&period_size, &dir);
qLog(QAudioInput) << "period size =" << (int)period_size;
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
qLog(QAudioInput) << "buffer time =" << buffer_time;
snd_pcm_hw_params_get_buffer_size(hwparams,(snd_pcm_uframes_t *) &buffer_size);
qLog(QAudioInput) << "buffer size =" << (int)buffer_size;
snd_pcm_hw_params_get_periods(hwparams, &vval, &dir);
qLog(QAudioInput) << "periods per buffer =" << vval;
snd_pcm_hw_params_get_rate_numden(hwparams, &vval, &vval2);
qLog(QAudioInput) << QString("exact rate = %1/%2 bps").arg(vval).arg(vval2).toLatin1().constData();
vval = snd_pcm_hw_params_get_sbits(hwparams);
qLog(QAudioInput) << "significant bits =" << vval;
snd_pcm_hw_params_get_tick_time(hwparams,&vval, &dir);
qLog(QAudioInput) << "tick time =" << vval;
vval = snd_pcm_hw_params_is_batch(hwparams);
qLog(QAudioInput) << "is batch =" << vval;
vval = snd_pcm_hw_params_is_block_transfer(hwparams);
qLog(QAudioInput) << "is block transfer =" << vval;
vval = snd_pcm_hw_params_is_double(hwparams);
qLog(QAudioInput) << "is double =" << vval;
vval = snd_pcm_hw_params_is_half_duplex(hwparams);
qLog(QAudioInput) << "is half duplex =" << vval;
vval = snd_pcm_hw_params_is_joint_duplex(hwparams);
qLog(QAudioInput) << "is joint duplex =" << vval;
vval = snd_pcm_hw_params_can_overrange(hwparams);
qLog(QAudioInput) << "can overrange =" << vval;
vval = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams);
qLog(QAudioInput) << "can mmap =" << vval;
vval = snd_pcm_hw_params_can_pause(hwparams);
qLog(QAudioInput) << "can pause =" << vval;
vval = snd_pcm_hw_params_can_resume(hwparams);
qLog(QAudioInput) << "can resume =" << vval;
vval = snd_pcm_hw_params_can_sync_start(hwparams);
qLog(QAudioInput) << "can sync start =" << vval;
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_sw_params_current(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_current: err %d",err);
}
err = snd_pcm_sw_params_set_start_threshold(handle,swparams,period_size);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_start_threshold: err %d",err);
}
err = snd_pcm_sw_params_set_avail_min(handle, swparams,period_size);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_avail_min: err %d",err);
}
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params_set_xfer_align: err %d",err);
}
err = snd_pcm_sw_params(handle, swparams);
if ( err < 0 ) {
qWarning( "QAudioInput: snd_pcm_sw_params: err %d",err);
}
snd_pcm_prepare(handle);
snd_pcm_start(handle);
int count = snd_pcm_poll_descriptors_count(handle);
pollfd *pfds = new pollfd[count];
snd_pcm_poll_descriptors(handle, pfds, count);
for (int i = 0; i < count; ++i)
{
if ((pfds[i].events & POLLIN) != 0) {
notifier = new QSocketNotifier(pfds[i].fd, QSocketNotifier::Read);
QObject::connect(notifier,
SIGNAL(activated(int)),
input,
SIGNAL(readyRead()));
break;
}
}
if (notifier == NULL) {
rc = false;
}
delete pfds;
}
return rc;
}
s32 Init()
{
//fprintf(stderr,"* SPU2-X: Initing Alsa\n");
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_status_t *status;
int pchannels = 2;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
handle = NULL;
pcm_callback = NULL;
pspeed = SAMPLE_RATE;
// buffer time and period time are in microseconds...
// (don't simplify the equation below -- it'll just cause integer rounding errors.
period_time = (SndOutPacketSize * 1000) / (SampleRate / 1000);
buffer_time = period_time * NumBuffers;
int err;
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_ASYNC /*| SND_PCM_NONBLOCK*/);
if (err < 0) {
fprintf(stderr, "Audio open error: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_nonblock(handle, 0);
if (err < 0) {
fprintf(stderr, "Can't set blocking mode: %s\n", snd_strerror(err));
return -1;
}
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));
return -1;
}
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
fprintf(stderr, "Access type not available: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_format(handle, hwparams, format);
if (err < 0) {
fprintf(stderr, "Sample format not available: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, pchannels);
if (err < 0) {
fprintf(stderr, "Channels count not available: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &pspeed, 0);
if (err < 0) {
fprintf(stderr, "Rate not available: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0);
if (err < 0) {
fprintf(stderr, "Buffer time error: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0);
if (err < 0) {
fprintf(stderr, "Period time error: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) {
fprintf(stderr, "Unable to install hw params: %s\n", snd_strerror(err));
return -1;
}
snd_pcm_status_alloca(&status);
err = snd_pcm_status(handle, status);
if (err < 0) {
fprintf(stderr, "Unable to get status: %s\n", snd_strerror(err));
return -1;
}
// Bind our asynchronous callback magic:
if (handle == NULL)
fprintf(stderr, "No handle.");
//fprintf(stderr,"* SPU2-X:Iz setting your internal callback.\n");
// The external handler never seems to get called after this.
snd_async_add_pcm_handler(&pcm_callback, handle, ExternalCallback, this);
err = snd_pcm_start(handle);
if (err < 0) {
fprintf(stderr, "Pcm start failed: %s\n", snd_strerror(err));
return -1;
}
// Diagnostic code:
//buffer_size = snd_pcm_status_get_avail(status);
//fprintf(stderr,"All set up.\n");
return 0;
}
static int
setparams(int32 sps, snd_pcm_t * handle)
{
snd_pcm_hw_params_t *hwparams;
unsigned int out_sps, buffer_time, period_time;
int err;
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0) {
fprintf(stderr, "Can not configure this PCM device: %s\n",
snd_strerror(err));
return -1;
}
err =
snd_pcm_hw_params_set_access(handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
fprintf(stderr,
"Failed to set PCM device to interleaved: %s\n",
snd_strerror(err));
return -1;
}
err =
snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16);
if (err < 0) {
fprintf(stderr,
"Failed to set PCM device to 16-bit signed PCM: %s\n",
snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
if (err < 0) {
fprintf(stderr, "Failed to set PCM device to mono: %s\n",
snd_strerror(err));
return -1;
}
out_sps = sps;
err =
snd_pcm_hw_params_set_rate_near(handle, hwparams, &out_sps, NULL);
if (err < 0) {
fprintf(stderr, "Failed to set sampling rate: %s\n",
snd_strerror(err));
return -1;
}
if (abs(out_sps - sps) > SPS_EPSILON) {
fprintf(stderr,
"Available samping rate %d is too far from requested %d\n",
out_sps, sps);
return -1;
}
/* Set buffer time to the maximum. */
err = snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time, 0);
period_time = buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams,
&period_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set period time to %u: %s\n",
period_time, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams,
&buffer_time, 0);
if (err < 0) {
fprintf(stderr, "Failed to set buffer time to %u: %s\n",
buffer_time, snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) {
fprintf(stderr, "Failed to set hwparams: %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_nonblock(handle, 1);
if (err < 0) {
fprintf(stderr, "Failed to set non-blocking mode: %s\n",
snd_strerror(err));
return -1;
}
return 0;
}
Error AudioDriverALSA::init() {
active=false;
thread_exited=false;
exit_thread=false;
pcm_open = false;
samples_in = NULL;
samples_out = NULL;
mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
output_format = OUTPUT_STEREO;
channels = 2;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
#define CHECK_FAIL(m_cond)\
if (m_cond) {\
fprintf(stderr,"ALSA ERR: %s\n",snd_strerror(status));\
snd_pcm_close(pcm_handle);\
ERR_FAIL_COND_V(m_cond,ERR_CANT_OPEN);\
}
//todo, add
//6 chans - "plug:surround51"
//4 chans - "plug:surround40";
status = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
ERR_FAIL_COND_V( status<0, ERR_CANT_OPEN );
snd_pcm_hw_params_alloca(&hwparams);
status = snd_pcm_hw_params_any(pcm_handle, hwparams);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
CHECK_FAIL( status<0 );
//not interested in anything else
status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
CHECK_FAIL( status<0 );
//todo: support 4 and 6
status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, NULL);
CHECK_FAIL( status<0 );
int latency = GLOBAL_DEF("audio/output_latency",25);
buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
// set buffer size from project settings
status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
CHECK_FAIL( status<0 );
// make period size 1/8
period_size = buffer_size >> 3;
status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
CHECK_FAIL( status<0 );
unsigned int periods=2;
status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, NULL);
CHECK_FAIL( status<0 );
status = snd_pcm_hw_params(pcm_handle,hwparams);
CHECK_FAIL( status<0 );
//snd_pcm_hw_params_free(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
status = snd_pcm_sw_params_current(pcm_handle, swparams);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
CHECK_FAIL( status<0 );
status = snd_pcm_sw_params(pcm_handle, swparams);
CHECK_FAIL( status<0 );
samples_in = memnew_arr(int32_t, period_size*channels);
samples_out = memnew_arr(int16_t, period_size*channels);
snd_pcm_nonblock(pcm_handle, 0);
mutex=Mutex::create();
thread = Thread::create(AudioDriverALSA::thread_func, this);
return OK;
};
int snd_pcm_generic_nonblock(snd_pcm_t *pcm, int nonblock)
{
snd_pcm_generic_t *generic = pcm->private_data;
return snd_pcm_nonblock(generic->slave, nonblock);
}
int
cubeb_stream_init(cubeb * context, cubeb_stream ** stream, char const * stream_name,
cubeb_stream_params stream_params, unsigned int latency,
cubeb_data_callback data_callback, cubeb_state_callback state_callback,
void * user_ptr)
{
cubeb_stream * stm;
int r;
snd_pcm_format_t format;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
assert(context);
assert(stream);
if (stream_params.rate < 1 || stream_params.rate > 192000 ||
stream_params.channels < 1 || stream_params.channels > 32 ||
latency < 1 || latency > 2000) {
return CUBEB_ERROR_INVALID_FORMAT;
}
switch (stream_params.format) {
case CUBEB_SAMPLE_S16LE:
format = SND_PCM_FORMAT_S16_LE;
break;
case CUBEB_SAMPLE_S16BE:
format = SND_PCM_FORMAT_S16_BE;
break;
case CUBEB_SAMPLE_FLOAT32LE:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case CUBEB_SAMPLE_FLOAT32BE:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
stm = calloc(1, sizeof(*stm));
assert(stm);
stm->context = context;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->params = stream_params;
r = pthread_mutex_init(&stm->lock, NULL);
assert(r == 0);
r = pthread_cond_init(&stm->cond, NULL);
assert(r == 0);
r = cubeb_locked_pcm_open(&stm->pcm, SND_PCM_STREAM_PLAYBACK);
if (r < 0) {
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
r = snd_pcm_nonblock(stm->pcm, 1);
assert(r == 0);
r = snd_pcm_set_params(stm->pcm, format, SND_PCM_ACCESS_RW_INTERLEAVED,
stm->params.channels, stm->params.rate, 1,
latency * 1000);
if (r < 0) {
/* XXX: return format error if necessary */
cubeb_stream_destroy(stm);
return CUBEB_ERROR;
}
r = snd_pcm_get_params(stm->pcm, &buffer_size, &period_size);
assert(r == 0);
fprintf(stderr, "b=%u p=%u\n", buffer_size, period_size);
/* set up poll infrastructure */
stm->n_descriptors = snd_pcm_poll_descriptors_count(stm->pcm);
assert(stm->n_descriptors > 0);
stm->descriptors = calloc(stm->n_descriptors, sizeof(*stm->descriptors));
assert(stm->descriptors);
r = snd_pcm_poll_descriptors(stm->pcm, stm->descriptors, stm->n_descriptors);
assert(r == stm->n_descriptors);
r = snd_pcm_pause(stm->pcm, 1);
#if 0
assert(r == 0);
#endif
stm->state = CUBEB_STREAM_STATE_INACTIVE;
*stream = stm;
return CUBEB_OK;
}
static int
alsa_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name,
cubeb_stream_params stream_params, unsigned int latency,
cubeb_data_callback data_callback, cubeb_state_callback state_callback,
void * user_ptr)
{
cubeb_stream * stm;
int r;
snd_pcm_format_t format;
assert(ctx && stream);
*stream = NULL;
switch (stream_params.format) {
case CUBEB_SAMPLE_S16LE:
format = SND_PCM_FORMAT_S16_LE;
break;
case CUBEB_SAMPLE_S16BE:
format = SND_PCM_FORMAT_S16_BE;
break;
case CUBEB_SAMPLE_FLOAT32LE:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case CUBEB_SAMPLE_FLOAT32BE:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
pthread_mutex_lock(&ctx->mutex);
if (ctx->active_streams >= CUBEB_STREAM_MAX) {
pthread_mutex_unlock(&ctx->mutex);
return CUBEB_ERROR;
}
ctx->active_streams += 1;
pthread_mutex_unlock(&ctx->mutex);
stm = calloc(1, sizeof(*stm));
assert(stm);
stm->context = ctx;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->params = stream_params;
stm->state = INACTIVE;
r = pthread_mutex_init(&stm->mutex, NULL);
assert(r == 0);
r = alsa_locked_pcm_open(&stm->pcm, SND_PCM_STREAM_PLAYBACK, ctx->local_config);
if (r < 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR;
}
r = snd_pcm_nonblock(stm->pcm, 1);
assert(r == 0);
/* Ugly hack: the PA ALSA plugin allows buffer configurations that can't
possibly work. See https://bugzilla.mozilla.org/show_bug.cgi?id=761274.
Only resort to this hack if the handle_underrun workaround failed. */
if (!ctx->local_config && ctx->is_pa) {
latency = latency < 500 ? 500 : latency;
}
r = snd_pcm_set_params(stm->pcm, format, SND_PCM_ACCESS_RW_INTERLEAVED,
stm->params.channels, stm->params.rate, 1,
latency * 1000);
if (r < 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR_INVALID_FORMAT;
}
r = snd_pcm_get_params(stm->pcm, &stm->buffer_size, &stm->period_size);
assert(r == 0);
stm->nfds = snd_pcm_poll_descriptors_count(stm->pcm);
assert(stm->nfds > 0);
stm->saved_fds = calloc(stm->nfds, sizeof(struct pollfd));
assert(stm->saved_fds);
r = snd_pcm_poll_descriptors(stm->pcm, stm->saved_fds, stm->nfds);
assert((nfds_t) r == stm->nfds);
r = pthread_cond_init(&stm->cond, NULL);
assert(r == 0);
if (alsa_register_stream(ctx, stm) != 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR;
}
*stream = stm;
return CUBEB_OK;
}
/*****************************************************************
* \brief init audio device
* \return STREAM_OK if success, STREAM_ERROR otherwise
*/
static int init_audio(radio_priv_t *priv)
{
int is_oss=1;
int seconds=2;
char* tmp;
if (priv->audio_initialized) return 1;
/* do_capture==0 mplayer was not started with capture keyword, so disabling capture*/
if(!priv->do_capture)
return STREAM_OK;
if (!priv->radio_param->adevice) {
priv->do_capture=0;
return STREAM_OK;
}
priv->do_capture=1;
mp_tmsg(MSGT_RADIO,MSGL_V,"[radio] Starting capture stuff.\n");
#ifdef CONFIG_ALSA
while ((tmp = strrchr(priv->radio_param->adevice, '='))) {
tmp[0] = ':';
//adevice option looks like ALSA device name. Switching to ALSA
is_oss=0;
}
while ((tmp = strrchr(priv->radio_param->adevice, '.')))
tmp[0] = ',';
#endif
if(audio_in_init(&priv->audio_in, is_oss?AUDIO_IN_OSS:AUDIO_IN_ALSA)<0) {
mp_tmsg(MSGT_RADIO, MSGL_ERR, "[radio] audio_in_init failed.\n");
}
audio_in_set_device(&priv->audio_in, priv->radio_param->adevice);
audio_in_set_channels(&priv->audio_in, priv->radio_param->achannels);
audio_in_set_samplerate(&priv->audio_in, priv->radio_param->arate);
if (audio_in_setup(&priv->audio_in) < 0) {
mp_tmsg(MSGT_RADIO, MSGL_ERR, "[radio] audio_in_setup call failed: %s\n", strerror(errno));
return STREAM_ERROR;
}
#ifdef CONFIG_OSS_AUDIO
if(is_oss)
ioctl(priv->audio_in.oss.audio_fd, SNDCTL_DSP_NONBLOCK, 0);
#endif
#ifdef CONFIG_ALSA
if(!is_oss)
snd_pcm_nonblock(priv->audio_in.alsa.handle,1);
#endif
priv->audio_buffer_size = seconds*priv->audio_in.samplerate*priv->audio_in.channels*
priv->audio_in.bytes_per_sample+priv->audio_in.blocksize;
if (priv->audio_buffer_size < 256*priv->audio_in.blocksize)
priv->audio_buffer_size = 256*priv->audio_in.blocksize;
mp_tmsg(MSGT_RADIO, MSGL_V, "[radio] Audio capture - buffer=%d bytes (block=%d bytes).\n",
priv->audio_buffer_size,priv->audio_in.blocksize);
/* start capture */
priv->audio_ringbuffer = calloc(1, priv->audio_buffer_size);
if (!priv->audio_ringbuffer) {
mp_tmsg(MSGT_RADIO, MSGL_ERR, "[radio] cannot allocate audio buffer (block=%d,buf=%d): %s\n",priv->audio_in.blocksize, priv->audio_buffer_size, strerror(errno));
return STREAM_ERROR;
}
priv->audio_head = 0;
priv->audio_tail = 0;
priv->audio_cnt = 0;
priv->audio_drop = 0;
audio_in_start_capture(&priv->audio_in);
priv->audio_initialized = 1;
return STREAM_OK;
}
int main(int argc, char *argv[])
{
struct option long_option[] =
{
{"help", 0, NULL, 'h'},
{"pdevice", 1, NULL, 'P'},
{"cdevice", 1, NULL, 'C'},
{"min", 1, NULL, 'm'},
{"max", 1, NULL, 'M'},
{"frames", 1, NULL, 'F'},
{"format", 1, NULL, 'f'},
{"channels", 1, NULL, 'c'},
{"rate", 1, NULL, 'r'},
{"seconds", 1, NULL, 's'},
{"block", 0, NULL, 'b'},
{"time", 1, NULL, 't'},
{"poll", 0, NULL, 'p'},
{"effect", 0, NULL, 'e'},
{NULL, 0, NULL, 0},
};
snd_pcm_t *phandle, *chandle;
char *buffer;
int err, latency, morehelp;
int ok;
snd_timestamp_t p_tstamp, c_tstamp;
ssize_t r;
size_t frames_in, frames_out, in_max;
int effect = 0;
morehelp = 0;
while (1) {
int c;
if ((c = getopt_long(argc, argv, "hP:C:m:M:F:f:c:r:s:bt:pe", long_option, NULL)) < 0)
break;
switch (c) {
case 'h':
morehelp++;
break;
case 'P':
pdevice = strdup(optarg);
break;
case 'C':
cdevice = strdup(optarg);
break;
case 'm':
err = atoi(optarg) / 2;
latency_min = err >= 4 ? err : 4;
if (latency_max < latency_min)
latency_max = latency_min;
break;
case 'M':
err = atoi(optarg) / 2;
latency_max = latency_min > err ? latency_min : err;
break;
case 'f':
format = snd_pcm_format_value(optarg);
if (format == SND_PCM_FORMAT_UNKNOWN) {
printf("Unknown format, setting to default S16_LE\n");
format = SND_PCM_FORMAT_S16_LE;
}
break;
case 'c':
err = atoi(optarg);
channels = err >= 1 && err < 1024 ? err : 1;
break;
case 'r':
err = atoi(optarg);
rate = err >= 4000 && err < 200000 ? err : 44100;
break;
case 's':
err = atoi(optarg);
loop_sec = err >= 1 && err <= 100000 ? err : 30;
break;
case 'b':
block = 1;
break;
case 't':
tick_time = atoi(optarg);
tick_time = tick_time < 0 ? 0 : tick_time;
break;
case 'p':
use_poll = 1;
break;
case 'e':
effect = 1;
break;
}
}
if (morehelp) {
help();
return 0;
}
err = snd_output_stdio_attach(&output, stdout, 0);
if (err < 0) {
printf("Output failed: %s\n", snd_strerror(err));
return 0;
}
loop_limit = loop_sec * rate;
latency = latency_min - 4;
buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2);
setscheduler();
printf("Playback device is %s\n", pdevice);
printf("Capture device is %s\n", cdevice);
printf("Parameters are %iHz, %s, %i channels, %s mode\n", rate, snd_pcm_format_name(format), channels, block ? "blocking" : "non-blocking");
printf("Wanted tick time: %ius, poll mode: %s\n", tick_time, use_poll ? "yes" : "no");
printf("Loop limit is %li frames, minimum latency = %i, maximum latency = %i\n", loop_limit, latency_min * 2, latency_max * 2);
if ((err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, block ? 0 : SND_PCM_NONBLOCK)) < 0) {
printf("Playback open error: %s\n", snd_strerror(err));
return 0;
}
if ((err = snd_pcm_open(&chandle, cdevice, SND_PCM_STREAM_CAPTURE, block ? 0 : SND_PCM_NONBLOCK)) < 0) {
printf("Record open error: %s\n", snd_strerror(err));
return 0;
}
/* initialize the filter sweep variables */
if (effect) {
fs = (float) rate;
BW = FILTER_BANDWIDTH;
lfo = 0;
dlfo = 2.*M_PI*FILTERSWEEP_LFO_FREQ/fs;
x[0] = (float*) malloc(channels*sizeof(float));
x[1] = (float*) malloc(channels*sizeof(float));
x[2] = (float*) malloc(channels*sizeof(float));
y[0] = (float*) malloc(channels*sizeof(float));
y[1] = (float*) malloc(channels*sizeof(float));
y[2] = (float*) malloc(channels*sizeof(float));
}
while (1) {
frames_in = frames_out = 0;
if (setparams(phandle, chandle, &latency) < 0)
break;
showlatency(latency);
if (tick_time_ok)
printf("Using tick time %ius\n", tick_time_ok);
if ((err = snd_pcm_link(chandle, phandle)) < 0) {
printf("Streams link error: %s\n", snd_strerror(err));
exit(0);
}
if (snd_pcm_format_set_silence(format, buffer, latency*channels) < 0) {
fprintf(stderr, "silence error\n");
break;
}
if (writebuf(phandle, buffer, latency, &frames_out) < 0) {
fprintf(stderr, "write error\n");
break;
}
if (writebuf(phandle, buffer, latency, &frames_out) < 0) {
fprintf(stderr, "write error\n");
break;
}
if ((err = snd_pcm_start(chandle)) < 0) {
printf("Go error: %s\n", snd_strerror(err));
exit(0);
}
gettimestamp(phandle, &p_tstamp);
gettimestamp(chandle, &c_tstamp);
#if 0
printf("Playback:\n");
showstat(phandle, frames_out);
printf("Capture:\n");
showstat(chandle, frames_in);
#endif
ok = 1;
in_max = 0;
while (ok && frames_in < loop_limit) {
if (use_poll) {
/* use poll to wait for next event */
snd_pcm_wait(chandle, 1000);
}
if ((r = readbuf(chandle, buffer, latency, &frames_in, &in_max)) < 0)
ok = 0;
else {
if (effect)
applyeffect(buffer,r);
if (writebuf(phandle, buffer, r, &frames_out) < 0)
ok = 0;
}
}
if (ok)
printf("Success\n");
else
printf("Failure\n");
printf("Playback:\n");
showstat(phandle, frames_out);
printf("Capture:\n");
showstat(chandle, frames_in);
showinmax(in_max);
if (p_tstamp.tv_sec == p_tstamp.tv_sec &&
p_tstamp.tv_usec == c_tstamp.tv_usec)
printf("Hardware sync\n");
snd_pcm_drop(chandle);
snd_pcm_nonblock(phandle, 0);
snd_pcm_drain(phandle);
snd_pcm_nonblock(phandle, !block ? 1 : 0);
if (ok) {
#if 1
printf("Playback time = %li.%i, Record time = %li.%i, diff = %li\n",
p_tstamp.tv_sec,
(int)p_tstamp.tv_usec,
c_tstamp.tv_sec,
(int)c_tstamp.tv_usec,
timediff(p_tstamp, c_tstamp));
#endif
break;
}
snd_pcm_unlink(chandle);
snd_pcm_hw_free(phandle);
snd_pcm_hw_free(chandle);
}
snd_pcm_close(phandle);
snd_pcm_close(chandle);
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 = QString(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());
#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 ) {
unsigned int maxBufferTime = 0;
unsigned int minBufferTime = 0;
unsigned int maxPeriodTime = 0;
unsigned int minPeriodTime = 0;
err = snd_pcm_hw_params_get_buffer_time_max(hwparams, &maxBufferTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_buffer_time_min(hwparams, &minBufferTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_period_time_max(hwparams, &maxPeriodTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_period_time_min(hwparams, &minPeriodTime, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: buffer/period min and max: err = %1").arg(err);
} else {
if (maxBufferTime < buffer_time || buffer_time < minBufferTime || maxPeriodTime < period_time || minPeriodTime > period_time) {
#ifdef DEBUG_AUDIO
qDebug()<<"defaults out of range";
qDebug()<<"pmin="<<minPeriodTime<<", pmax="<<maxPeriodTime<<", bmin="<<minBufferTime<<", bmax="<<maxBufferTime;
#endif
period_time = minPeriodTime;
if (period_time*4 <= maxBufferTime) {
// Use 4 periods if possible
buffer_time = period_time*4;
chunks = 4;
} else if (period_time*2 <= maxBufferTime) {
// Use 2 periods if possible
buffer_time = period_time*2;
chunks = 2;
} else {
qWarning()<<"QAudioOutput: alsa only supports single period!";
fatal = true;
}
#ifdef DEBUG_AUDIO
qDebug()<<"used: buffer_time="<<buffer_time<<", period_time="<<period_time;
#endif
}
}
}
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 int
alsa_stream_init(cubeb * ctx, cubeb_stream ** stream, char const * stream_name,
cubeb_devid input_device,
cubeb_stream_params * input_stream_params,
cubeb_devid output_device,
cubeb_stream_params * output_stream_params,
unsigned int latency_frames,
cubeb_data_callback data_callback, cubeb_state_callback state_callback,
void * user_ptr)
{
(void)stream_name;
cubeb_stream * stm;
int r;
snd_pcm_format_t format;
snd_pcm_uframes_t period_size;
int latency_us = 0;
assert(ctx && stream);
if (input_stream_params) {
/* Capture support not yet implemented. */
return CUBEB_ERROR_NOT_SUPPORTED;
}
if (input_device || output_device) {
/* Device selection not yet implemented. */
return CUBEB_ERROR_DEVICE_UNAVAILABLE;
}
*stream = NULL;
switch (output_stream_params->format) {
case CUBEB_SAMPLE_S16LE:
format = SND_PCM_FORMAT_S16_LE;
break;
case CUBEB_SAMPLE_S16BE:
format = SND_PCM_FORMAT_S16_BE;
break;
case CUBEB_SAMPLE_FLOAT32LE:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case CUBEB_SAMPLE_FLOAT32BE:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
return CUBEB_ERROR_INVALID_FORMAT;
}
pthread_mutex_lock(&ctx->mutex);
if (ctx->active_streams >= CUBEB_STREAM_MAX) {
pthread_mutex_unlock(&ctx->mutex);
return CUBEB_ERROR;
}
ctx->active_streams += 1;
pthread_mutex_unlock(&ctx->mutex);
stm = calloc(1, sizeof(*stm));
assert(stm);
stm->context = ctx;
stm->data_callback = data_callback;
stm->state_callback = state_callback;
stm->user_ptr = user_ptr;
stm->params = *output_stream_params;
stm->state = INACTIVE;
stm->volume = 1.0;
r = pthread_mutex_init(&stm->mutex, NULL);
assert(r == 0);
r = alsa_locked_pcm_open(&stm->pcm, SND_PCM_STREAM_PLAYBACK, ctx->local_config);
if (r < 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR;
}
r = snd_pcm_nonblock(stm->pcm, 1);
assert(r == 0);
latency_us = latency_frames * 1e6 / stm->params.rate;
/* Ugly hack: the PA ALSA plugin allows buffer configurations that can't
possibly work. See https://bugzilla.mozilla.org/show_bug.cgi?id=761274.
Only resort to this hack if the handle_underrun workaround failed. */
if (!ctx->local_config && ctx->is_pa) {
const int min_latency = 5e5;
latency_us = latency_us < min_latency ? min_latency: latency_us;
}
r = snd_pcm_set_params(stm->pcm, format, SND_PCM_ACCESS_RW_INTERLEAVED,
stm->params.channels, stm->params.rate, 1,
latency_us);
if (r < 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR_INVALID_FORMAT;
}
r = snd_pcm_get_params(stm->pcm, &stm->buffer_size, &period_size);
assert(r == 0);
stm->nfds = snd_pcm_poll_descriptors_count(stm->pcm);
assert(stm->nfds > 0);
stm->saved_fds = calloc(stm->nfds, sizeof(struct pollfd));
assert(stm->saved_fds);
r = snd_pcm_poll_descriptors(stm->pcm, stm->saved_fds, stm->nfds);
assert((nfds_t) r == stm->nfds);
r = pthread_cond_init(&stm->cond, NULL);
assert(r == 0);
if (alsa_register_stream(ctx, stm) != 0) {
alsa_stream_destroy(stm);
return CUBEB_ERROR;
}
*stream = stm;
return CUBEB_OK;
}