// Initialize the BlockSound class
BlockSound::BlockSound() {
sample_size = 0;
#ifdef __APPLE__
remaining = 0;
UInt32 size = sizeof(device);
if (AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
&size, (void *)&device) != noErr) return;
size = sizeof(format);
if (AudioDeviceGetProperty(device, 0, false, kAudioDevicePropertyStreamFormat,
&size, &format) != noErr) return;
// Set up a format we like...
format.mSampleRate = 44100.0; // 44.1kHz
format.mChannelsPerFrame = 2; // stereo
if (AudioDeviceSetProperty(device, NULL, 0, false,
kAudioDevicePropertyStreamFormat,
sizeof(format), &format) != noErr) return;
// Check we got linear pcm - what to do if we did not ???
if (format.mFormatID != kAudioFormatLinearPCM) return;
// Attach the callback and start the device
# if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_5
if (AudioDeviceCreateIOProcID(device, audio_cb, (void *)this, &audio_proc_id) != noErr) return;
AudioDeviceStart(device, audio_proc_id);
# else
if (AudioDeviceAddIOProc(device, audio_cb, (void *)this) != noErr) return;
AudioDeviceStart(device, audio_cb);
# endif
sample_size = (int)format.mSampleRate;
#elif defined(WIN32)
WAVEFORMATEX format;
memset(&format, 0, sizeof(format));
format.cbSize = sizeof(format);
format.wFormatTag = WAVE_FORMAT_PCM;
format.nChannels = 2;
format.nSamplesPerSec = 44100;
format.nAvgBytesPerSec = 44100 * 4;
format.nBlockAlign = 4;
format.wBitsPerSample = 16;
data_handle = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, format.nSamplesPerSec * 4);
if (!data_handle) return;
data_ptr = (LPSTR)GlobalLock(data_handle);
header_handle = GlobalAlloc(GMEM_MOVEABLE | GMEM_SHARE, sizeof(WAVEHDR));
if (!header_handle) return;
header_ptr = (WAVEHDR *)GlobalLock(header_handle);
header_ptr->lpData = data_ptr;
header_ptr->dwFlags = 0;
header_ptr->dwLoops = 0;
if (waveOutOpen(&device, WAVE_MAPPER, &format, 0, 0, WAVE_ALLOWSYNC)
!= MMSYSERR_NOERROR) return;
sample_size = format.nSamplesPerSec;
#else
# ifdef HAVE_ALSA_ASOUNDLIB_H
handle = NULL;
if (snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0) >= 0) {
// Initialize PCM sound stuff...
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16);
snd_pcm_hw_params_set_channels(handle, params, 2);
unsigned rate = 44100;
int dir;
snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir);
snd_pcm_uframes_t period = (int)rate;
snd_pcm_hw_params_set_period_size_near(handle, params, &period, &dir);
sample_size = rate;
if (snd_pcm_hw_params(handle, params) < 0) {
sample_size = 0;
snd_pcm_close(handle);
handle = NULL;
}
}
# endif // HAVE_ALSA_ASOUNDLIB_H
#endif // __APPLE__
if (sample_size) {
// Make an explosion sound by passing white noise through a low pass
// filter with a decreasing frequency...
sample_data = new short[2 * sample_size];
short *sample_ptr = sample_data;
int max_sample = 2 * sample_size - 2;
*sample_ptr++ = 0;
*sample_ptr++ = 0;
for (int j = max_sample; j > 0; j --, sample_ptr ++) {
float freq = (float)j / (float)max_sample;
float volume = 32767.0 * (0.5 * sqrt(freq) + 0.5);
float sample = 0.0001 * ((rand() % 20001) - 10000);
*sample_ptr = (int)(volume * freq * sample +
(1.0 - freq) * sample_ptr[-2]);
}
}
}
int Aisound::Open_Pcm()
{
BEGINFUNC_USING_BY_VOICE
int err;
int dir;
int size;
char *buffer;
unsigned int val;
snd_pcm_uframes_t frames;
snd_pcm_hw_params_t* params;
int channels = 0;
if ( (err = snd_pcm_open(&handle, "plug:cmd", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
dbug_out("cannot open audio device (%s)\n", snd_strerror (err));
goto OpenPcmFail;
}
dbug_out("After snd_pcm_open for SND_PCM_STREAM_PLAYBACK");
snd_pcm_hw_params_alloca(¶ms);//为å�‚æ•°å�˜é‡�分é…�空间
if( (err = snd_pcm_hw_params_any(handle, params)) < 0)//�数�始化
{
dbug_out("snd_pcm_hw_params_any failed! err = %d\n", err);
goto OpenPcmFail;
}
if( (err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)//设置为交错模�
{
dbug_out("snd_pcm_hw_params_set_access failed! err = %d\n", err);
goto OpenPcmFail;
}
if( (err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE)) < 0)//使用16ä½�æ ·æœ¬
{
dbug_out("snd_pcm_hw_params_set_format failed! err = %d\n", err);
goto OpenPcmFail;
}
channels = val = 1;
if( (err = snd_pcm_hw_params_set_channels(handle, params, val)) < 0)//设置为立体声
{
dbug_out("snd_pcm_hw_params_set_channels failed! err = %d\n", err);
goto OpenPcmFail;
}
val = 16000;
if( (err = snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir)) < 0)//è®¾ç½®é‡‡æ ·çŽ‡ä¸º44.1KHz
{
dbug_out("snd_pcm_hw_params_set_rate_near failed! err = %d\n", err);
goto OpenPcmFail;
}
#if 1
frames = 32;
if( (err = snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir)) < 0)
{
dbug_out("snd_pcm_hw_params_set_period_size_near failed! err = %d\n", err);
goto OpenPcmFail;
}
if ( (err = snd_pcm_hw_params(handle, params)) < 0)
{
dbug_out("snd_pcm_hw_params-failed! err = %d err: %s\n", err, snd_strerror(err));
goto OpenPcmFail;
}
if( (err = snd_pcm_hw_params_get_period_size(params, &frames, &dir)) < 0)
{
dbug_out("snd_pcm_hw_params_get_period_size failed! err = %d\n", err);
goto OpenPcmFail;
}
size = frames * channels * 2;
buffer = (char *)malloc(sizeof(char) * size);
if( (err = snd_pcm_hw_params_get_period_time(params, &val, &dir)) < 0)
{
dbug_out("snd_pcm_hw_params_get_period_time failed! err = %d\n", err);
goto OpenPcmFail;
}
return 0;
OpenPcmFail:
return -1;
#endif
ENDFUNC_USING_BY_VOICE
}
bool CAESinkALSA::InitializeHW(AEAudioFormat &format)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca(&hw_params);
memset(hw_params, 0, snd_pcm_hw_params_sizeof());
snd_pcm_hw_params_any(m_pcm, hw_params);
snd_pcm_hw_params_set_access(m_pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
unsigned int sampleRate = format.m_sampleRate;
#if defined(HAS_AMLPLAYER) || defined(HAS_LIBAMCODEC)
// alsa/kernel lies, so map everything to 44100 or 48000
switch(sampleRate)
{
case 11025:
case 22050:
case 88200:
case 176400:
sampleRate = 44100;
break;
case 8000:
case 16000:
case 24000:
case 32000:
case 96000:
case 192000:
case 384000:
sampleRate = 48000;
break;
}
#endif
unsigned int channelCount = format.m_channelLayout.Count();
snd_pcm_hw_params_set_rate_near (m_pcm, hw_params, &sampleRate, NULL);
snd_pcm_hw_params_set_channels_near(m_pcm, hw_params, &channelCount);
/* ensure we opened X channels or more */
if (format.m_channelLayout.Count() > channelCount)
{
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Unable to open the required number of channels");
}
/* update the channelLayout to what we managed to open */
format.m_channelLayout.Reset();
for (unsigned int i = 0; i < channelCount; ++i)
format.m_channelLayout += ALSAChannelMap[i];
snd_pcm_format_t fmt = AEFormatToALSAFormat(format.m_dataFormat);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
/* if we dont support the requested format, fallback to float */
format.m_dataFormat = AE_FMT_FLOAT;
fmt = SND_PCM_FORMAT_FLOAT;
}
/* try the data format */
if (snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
/* if the chosen format is not supported, try each one in decending order */
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Your hardware does not support %s, trying other formats", CAEUtil::DataFormatToStr(format.m_dataFormat));
for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
{
if (AE_IS_RAW(i) || i == AE_FMT_MAX)
continue;
if (m_passthrough && i != AE_FMT_S16BE && i != AE_FMT_S16LE)
continue;
fmt = AEFormatToALSAFormat(i);
if (fmt == SND_PCM_FORMAT_UNKNOWN || snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
fmt = SND_PCM_FORMAT_UNKNOWN;
continue;
}
int fmtBits = CAEUtil::DataFormatToBits(i);
int bits = snd_pcm_hw_params_get_sbits(hw_params);
if (bits != fmtBits)
{
/* if we opened in 32bit and only have 24bits, pack into 24 */
if (fmtBits == 32 && bits == 24)
i = AE_FMT_S24NE4;
else
continue;
}
/* record that the format fell back to X */
format.m_dataFormat = i;
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Using data format %s", CAEUtil::DataFormatToStr(format.m_dataFormat));
break;
}
/* if we failed to find a valid output format */
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to find a suitable output format");
return false;
}
}
snd_pcm_uframes_t periodSize, bufferSize;
snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufferSize);
snd_pcm_hw_params_get_period_size_max(hw_params, &periodSize, NULL);
/*
We want to make sure, that we have max 200 ms Buffer with
a periodSize of approx 50 ms. Choosing a higher bufferSize
will cause problems with menu sounds. Buffer will be increased
after those are fixed.
*/
periodSize = std::min(periodSize, (snd_pcm_uframes_t) sampleRate / 20);
bufferSize = std::min(bufferSize, (snd_pcm_uframes_t) sampleRate / 5);
#if defined(HAS_AMLPLAYER) || defined(HAS_LIBAMCODEC)
// must be pot for pivos.
bufferSize = CheckNP2(bufferSize);
#endif
/*
According to upstream we should set buffer size first - so make sure it is always at least
4x period size to not get underruns (some systems seem to have issues with only 2 periods)
*/
periodSize = std::min(periodSize, bufferSize / 4);
#if defined(HAS_AMLPLAYER) || defined(HAS_LIBAMCODEC)
// must be pot for pivos.
periodSize = CheckNP2(periodSize);
#endif
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Request: periodSize %lu, bufferSize %lu", periodSize, bufferSize);
snd_pcm_hw_params_t *hw_params_copy;
snd_pcm_hw_params_alloca(&hw_params_copy);
snd_pcm_hw_params_copy(hw_params_copy, hw_params); // copy what we have and is already working
// first trying bufferSize, PeriodSize
// for more info see here:
// http://mailman.alsa-project.org/pipermail/alsa-devel/2009-September/021069.html
// the last three tries are done as within pulseaudio
// backup periodSize and bufferSize first. Restore them after every failed try
snd_pcm_uframes_t periodSizeTemp, bufferSizeTemp;
periodSizeTemp = periodSize;
bufferSizeTemp = bufferSize;
if (snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize) != 0
|| snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL) != 0
|| snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
bufferSize = bufferSizeTemp;
periodSize = periodSizeTemp;
// retry with PeriodSize, bufferSize
snd_pcm_hw_params_copy(hw_params_copy, hw_params); // restore working copy
if (snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL) != 0
|| snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize) != 0
|| snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
// try only periodSize
periodSize = periodSizeTemp;
snd_pcm_hw_params_copy(hw_params_copy, hw_params); // restore working copy
if(snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL) != 0
|| snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
// try only BufferSize
bufferSize = bufferSizeTemp;
snd_pcm_hw_params_copy(hw_params_copy, hw_params); // restory working copy
if (snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize) != 0
|| snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
// set default that Alsa would choose
CLog::Log(LOGWARNING, "CAESinkAlsa::IntializeHW - Using default alsa values - set failed");
if (snd_pcm_hw_params(m_pcm, hw_params) != 0)
{
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Could not init a valid sink");
return false;
}
}
}
// reread values when alsa default was kept
snd_pcm_get_params(m_pcm, &bufferSize, &periodSize);
}
}
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Got: periodSize %lu, bufferSize %lu", periodSize, bufferSize);
/* set the format parameters */
format.m_sampleRate = sampleRate;
format.m_frames = periodSize;
format.m_frameSamples = periodSize * format.m_channelLayout.Count();
format.m_frameSize = snd_pcm_frames_to_bytes(m_pcm, 1);
m_bufferSize = (unsigned int)bufferSize;
m_timeout = std::ceil((double)(bufferSize * 1000) / (double)sampleRate);
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Setting timeout to %d ms", m_timeout);
return true;
}
int main()
{
int rc;
snd_pcm_t* handle;
snd_pcm_hw_params_t* params;
unsigned int val;
unsigned int val2;
int dir;
snd_pcm_uframes_t frames;
if ( (rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
std::cerr << "unable to open pcm devices: " << snd_strerror(rc) << std::endl;
exit(1);
}
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(handle, params, 2);
val = 44100;
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
if ( (rc = snd_pcm_hw_params(handle, params)) < 0)
{
std::cerr << "unable to set hw parameters: " << snd_strerror(rc) << std::endl;
exit(1);
}
std::cout << "PCM handle name = " << snd_pcm_name(handle) << std::endl;
std::cout << "PCM state = " << snd_pcm_state_name(snd_pcm_state(handle)) << std::endl;
snd_pcm_hw_params_get_access(params, (snd_pcm_access_t *)&val);
std::cout << "access type = " << snd_pcm_access_name((snd_pcm_access_t)val) << std::endl;
snd_pcm_hw_params_get_format(params, (snd_pcm_format_t*)(&val));
std::cout << "format = '" << snd_pcm_format_name((snd_pcm_format_t)val) << "' (" << snd_pcm_format_description((snd_pcm_format_t)val) << ")" << std::endl;
snd_pcm_hw_params_get_subformat(params, (snd_pcm_subformat_t *)&val);
std::cout << "subformat = '" <<
snd_pcm_subformat_name((snd_pcm_subformat_t)val) << "' (" << snd_pcm_subformat_description((snd_pcm_subformat_t)val) << ")" << std::endl;
snd_pcm_hw_params_get_channels(params, &val);
std::cout << "channels = " << val << std::endl;
snd_pcm_hw_params_get_rate(params, &val, &dir);
std::cout << "rate = " << val << " bps" << std::endl;
snd_pcm_hw_params_get_period_time(params, &val, &dir);
std::cout << "period time = " << val << " us" << std::endl;
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
std::cout << "period size = " << static_cast<int>(frames) << " frames" << std::endl;
snd_pcm_hw_params_get_buffer_time(params, &val, &dir);
std::cout << "buffer time = " << val << " us" << std::endl;
snd_pcm_hw_params_get_buffer_size(params, (snd_pcm_uframes_t *) &val);
std::cout << "buffer size = " << val << " frames" << std::endl;
snd_pcm_hw_params_get_periods(params, &val, &dir);
std::cout << "periods per buffer = " << val << " frames" << std::endl;
snd_pcm_hw_params_get_rate_numden(params, &val, &val2);
std::cout << "exact rate = " << val/val2 << " bps" << std::endl;
val = snd_pcm_hw_params_get_sbits(params);
std::cout << "significant bits = " << val << std::endl;
snd_pcm_hw_params_get_tick_time(params, &val, &dir);
std::cout << "tick time = " << val << " us" << std::endl;
val = snd_pcm_hw_params_is_batch(params);
std::cout << "is batch = " << val << std::endl;
val = snd_pcm_hw_params_is_block_transfer(params);
std::cout << "is block transfer = " << val << std::endl;
val = snd_pcm_hw_params_is_double(params);
std::cout << "is double = " << val << std::endl;
val = snd_pcm_hw_params_is_half_duplex(params);
std::cout << "is half duplex = " << val << std::endl;
val = snd_pcm_hw_params_is_joint_duplex(params);
std::cout << "is joint duplex = " << val << std::endl;
val = snd_pcm_hw_params_can_overrange(params);
std::cout << "can overrange = " << val << std::endl;
val = snd_pcm_hw_params_can_mmap_sample_resolution(params);
std::cout << "can mmap = " << val << std::endl;
val = snd_pcm_hw_params_can_pause(params);
std::cout << "can pause = " << val << std::endl;
val = snd_pcm_hw_params_can_resume(params);
std::cout << "can resume = " << val << std::endl;
val = snd_pcm_hw_params_can_sync_start(params);
std::cout << "can sync start = " << val << std::endl;
snd_pcm_close(handle);
return 0;
}
bool ALSAWriter::processParams(bool *paramsCorrected)
{
const unsigned chn = getParam("chn").toUInt();
const unsigned rate = getParam("rate").toUInt();
const bool resetAudio = channels != chn || sample_rate != rate;
channels = chn;
sample_rate = rate;
if (resetAudio || err)
{
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(¶ms);
close();
QString chosenDevName = devName;
if (autoFindMultichannelDevice && channels > 2)
{
bool mustAutoFind = true, forceStereo = false;
if (!snd_pcm_open(&snd, chosenDevName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0))
{
if (snd_pcm_type(snd) == SND_PCM_TYPE_HW)
{
unsigned max_chn = 0;
snd_pcm_hw_params_any(snd, params);
mustAutoFind = snd_pcm_hw_params_get_channels_max(params, &max_chn) || max_chn < channels;
}
#ifdef HAVE_CHMAP
else if (paramsCorrected)
{
snd_pcm_chmap_query_t **chmaps = snd_pcm_query_chmaps(snd);
if (chmaps)
snd_pcm_free_chmaps(chmaps);
else
forceStereo = true;
}
#endif
snd_pcm_close(snd);
snd = nullptr;
}
if (mustAutoFind)
{
QString newDevName;
if (channels <= 4)
newDevName = "surround40";
else if (channels <= 6)
newDevName = "surround51";
else
newDevName = "surround71";
if (!newDevName.isEmpty() && newDevName != chosenDevName)
{
if (ALSACommon::getDevices().first.contains(newDevName))
chosenDevName = newDevName;
else if (forceStereo)
{
channels = 2;
*paramsCorrected = true;
}
}
}
}
if (!chosenDevName.isEmpty())
{
bool sndOpen = !snd_pcm_open(&snd, chosenDevName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0);
if (devName != chosenDevName)
{
if (sndOpen)
QMPlay2Core.logInfo("ALSA :: " + devName + "\" -> \"" + chosenDevName + "\"");
else
{
sndOpen = !snd_pcm_open(&snd, devName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0);
QMPlay2Core.logInfo("ALSA :: " + tr("Cannot open") + " \"" + chosenDevName + "\", " + tr("back to") + " \"" + devName + "\"");
}
}
if (sndOpen)
{
snd_pcm_hw_params_any(snd, params);
snd_pcm_format_t fmt = SND_PCM_FORMAT_UNKNOWN;
if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S32))
{
fmt = SND_PCM_FORMAT_S32;
sample_size = 4;
}
else if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S16))
{
fmt = SND_PCM_FORMAT_S16;
sample_size = 2;
}
else if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S8))
{
fmt = SND_PCM_FORMAT_S8;
sample_size = 1;
}
unsigned delay_us = round(delay * 1000000.0);
if (fmt != SND_PCM_FORMAT_UNKNOWN && set_snd_pcm_hw_params(snd, params, fmt, channels, sample_rate, delay_us))
{
bool err2 = false;
if (channels != chn || sample_rate != rate)
{
if (paramsCorrected)
*paramsCorrected = true;
else
err2 = true;
}
if (!err2)
{
err2 = snd_pcm_hw_params(snd, params);
if (err2 && paramsCorrected) //jakiś błąd, próba zmiany sample_rate
{
snd_pcm_hw_params_any(snd, params);
err2 = snd_pcm_hw_params_set_rate_resample(snd, params, false) || !set_snd_pcm_hw_params(snd, params, fmt, channels, sample_rate, delay_us) || snd_pcm_hw_params(snd, params);
if (!err2)
*paramsCorrected = true;
}
if (!err2)
{
modParam("delay", delay_us / 1000000.0);
if (paramsCorrected && *paramsCorrected)
{
modParam("chn", channels);
modParam("rate", sample_rate);
}
canPause = snd_pcm_hw_params_can_pause(params) && snd_pcm_hw_params_can_resume(params);
mustSwapChn = channels == 6 || channels == 8;
#ifdef HAVE_CHMAP
if (mustSwapChn)
{
snd_pcm_chmap_query_t **chmaps = snd_pcm_query_chmaps(snd);
if (chmaps)
{
for (int i = 0; chmaps[i]; ++i)
{
if (chmaps[i]->map.channels >= channels)
{
for (uint j = 0; j < channels; ++j)
{
mustSwapChn &= chmaps[i]->map.pos[j] == j + SND_CHMAP_FL;
if (!mustSwapChn)
break;
}
break;
}
}
snd_pcm_free_chmaps(chmaps);
}
}
#endif
return true;
}
}
}
}
}
err = true;
QMPlay2Core.logError("ALSA :: " + tr("Cannot open audio output stream"));
}
return readyWrite();
}
void play(int sec, int freq, char *file)
{
long loops;
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
char *buffer;
int fd;
/* Check ranges */
if (sec < 1 || sec > 4000) {
printf("WARNING: Incorrect time to play range [1,4000] s\n");
printf("\tSetting time to play to 5s...\n");
sec = 5;
}
if (freq < 1000 || freq > 100000) {
printf("ERROR: Incorrect frequency range [1000,100000] Hz\n");
printf("\tSetting frequency to 44.1 kHz...\n");
freq = 44100;
}
/* Open file */
fd = open(file, O_RDONLY);
if (fd < 0) {
/* There was an error opening the file */
printf("ERROR: Couldn't open file to play\n");
printf("\tPlease make sure file exists\n");
} else {
/* Print that the file is playing with its parameters */
printf("Playing file %s for %d seconds", file, sec);
printf(" and frequency %d...\n", freq);
/* Open PCM device for playback */
rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK,
0);
if (rc < 0) {
fprintf(stderr, "unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values */
snd_pcm_hw_params_any(handle, params);
/* Set hardware parameters */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 2);
/* freq bits/second sampling rate (CD quality) */
val = freq;
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
/* Set period size to 32 frames */
frames = 32;
snd_pcm_hw_params_set_period_size_near(handle, params,
&frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr, "unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
size = frames * 4; /* 2 bytes/sample, 2 channels */
buffer = (char *) malloc(size);
/* We want to loop for sec seconds */
snd_pcm_hw_params_get_period_time(params, &val, &dir);
/* sec seconds in microseconds divided by period time */
loops = sec*1000000 / val;
while (loops > 0) {
loops--;
rc = read(fd, buffer, size);
if (rc == 0) {
fprintf(stderr, "end of file on input\n");
break;
} else if (rc != size) {
fprintf(stderr, "short read: read %d bytes\n",
rc);
}
rc = snd_pcm_writei(handle, buffer, frames);
if (rc == -EPIPE) {
/* EPIPE means underrun */
fprintf(stderr, "underrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr, "error from writei: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr,
"short write, write %d frames\n", rc);
}
}
/*Close handle and free buffer*/
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
}
}
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;
};
static int init_audio_device(common_data_t *p_common_data)
{
int ret;
int dir = 0;
unsigned int val;
snd_pcm_hw_params_t *p_params;
snd_pcm_uframes_t buffer_size;
unsigned int buffer_time;
unsigned int period_time;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
size_t bits_per_sample;
size_t bits_per_frame;
ret = snd_pcm_open(&p_common_data->handle, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0) {
dbg_alsa("unable to open pcm device: %s\n", snd_strerror(ret));
return -1;
}
snd_pcm_hw_params_alloca(&p_params);
snd_pcm_hw_params_any(p_common_data->handle, p_params);
snd_pcm_hw_params_set_access(p_common_data->handle, p_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(p_common_data->handle, p_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(p_common_data->handle, p_params, 2);
val = 44100;
snd_pcm_hw_params_set_rate_near(p_common_data->handle, p_params, &val, &dir);
snd_pcm_hw_params_get_buffer_time_max(p_params, &buffer_time, 0);
// if (buffer_time > 500000)
// buffer_time = 500000;
//
// period_time = buffer_time / 4;
//
// snd_pcm_hw_params_set_period_time_near(p_common_data->handle, p_params, &period_time, 0);
// snd_pcm_hw_params_set_buffer_time_near(p_common_data->handle, p_params, &buffer_time, 0);
p_common_data->period_size = 2048;
snd_pcm_hw_params_set_period_size_near(p_common_data->handle, p_params, &p_common_data->period_size, &dir);
ret = snd_pcm_hw_params(p_common_data->handle, p_params);
if (ret < 0){
dbg_alsa("unable to set hw parameters: %s\n", snd_strerror(ret));
exit(1);
}
ret = snd_pcm_state(p_common_data->handle);
dbg("state is %d\n", ret);
snd_pcm_hw_params_get_period_size(p_params, &p_common_data->period_size, &dir);
snd_pcm_hw_params_get_buffer_size(p_params, &buffer_size);
if (p_common_data->period_size == buffer_size) {
dbg_alsa("Can't use period size equal to buffer size (%lu == %lu)\n", p_common_data->period_size, buffer_size);
}
dbg("period size is : %lu frames\n", p_common_data->period_size);
dbg("buffer size is : %lu frames\n", buffer_size);
bits_per_sample = snd_pcm_format_physical_width(format);
bits_per_frame = bits_per_sample * 2;
p_common_data->chunk_bytes = p_common_data->period_size * bits_per_frame / 8;
//g_buf = (msg_t *)malloc(sizeof(msg_t)+(char)p_common_data->chunk_bytes);
dbg("sample rate is %d\n", val);
dbg("bits_per_sample %d\n", bits_per_sample);
dbg("bits_per_frame %d\n", bits_per_frame);
dbg("chunk_bytes %d\n", p_common_data->chunk_bytes);
dbg("PCM handle name = '%s'\n",snd_pcm_name(p_common_data->handle));
return 0;
}
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.frequency() <= 0) {
qWarning("QAudioOutput: open error, invalid sample rate (%d).",
settings.frequency());
} else {
err = -1;
}
if (err == 0) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
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)
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_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
if(audioBuffer == 0)
audioBuffer = new char[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;
}
bool CAESinkALSA::InitializeHW(AEAudioFormat &format)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca(&hw_params);
memset(hw_params, 0, snd_pcm_hw_params_sizeof());
snd_pcm_hw_params_any(m_pcm, hw_params);
snd_pcm_hw_params_set_access(m_pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
unsigned int sampleRate = format.m_sampleRate;
unsigned int channelCount = format.m_channelLayout.Count();
snd_pcm_hw_params_set_rate_near (m_pcm, hw_params, &sampleRate, NULL);
snd_pcm_hw_params_set_channels_near(m_pcm, hw_params, &channelCount);
/* ensure we opened X channels or more */
if (format.m_channelLayout.Count() > channelCount)
{
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Unable to open the required number of channels");
}
/* update the channelLayout to what we managed to open */
format.m_channelLayout.Reset();
for (unsigned int i = 0; i < channelCount; ++i)
format.m_channelLayout += ALSAChannelMap[i];
snd_pcm_format_t fmt = AEFormatToALSAFormat(format.m_dataFormat);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
/* if we dont support the requested format, fallback to float */
format.m_dataFormat = AE_FMT_FLOAT;
fmt = SND_PCM_FORMAT_FLOAT;
}
/* try the data format */
if (snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
/* if the chosen format is not supported, try each one in decending order */
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Your hardware does not support %s, trying other formats", CAEUtil::DataFormatToStr(format.m_dataFormat));
for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
{
if (AE_IS_RAW(i) || i == AE_FMT_MAX)
continue;
if (m_passthrough && i != AE_FMT_S16BE && i != AE_FMT_S16LE)
continue;
fmt = AEFormatToALSAFormat(i);
if (fmt == SND_PCM_FORMAT_UNKNOWN || snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
fmt = SND_PCM_FORMAT_UNKNOWN;
continue;
}
int fmtBits = CAEUtil::DataFormatToBits(i);
int bits = snd_pcm_hw_params_get_sbits(hw_params);
if (bits != fmtBits)
{
/* if we opened in 32bit and only have 24bits, pack into 24 */
if (fmtBits == 32 && bits == 24)
i = AE_FMT_S24NE4;
else
continue;
}
/* record that the format fell back to X */
format.m_dataFormat = i;
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Using data format %s", CAEUtil::DataFormatToStr(format.m_dataFormat));
break;
}
/* if we failed to find a valid output format */
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to find a suitable output format");
return false;
}
}
unsigned int periods;
snd_pcm_uframes_t periodSize, bufferSize;
snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufferSize);
bufferSize = std::min(bufferSize, (snd_pcm_uframes_t)8192);
periodSize = bufferSize / ALSA_PERIODS;
periods = ALSA_PERIODS;
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Request: periodSize %lu, periods %u, bufferSize %lu", periodSize, periods, bufferSize);
/* work on a copy of the hw params */
snd_pcm_hw_params_t *hw_params_copy;
snd_pcm_hw_params_alloca(&hw_params_copy);
/* try to set the buffer size then the period size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to set the period size then the buffer size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to just set the buffer size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to just set the period size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Failed to set the parameters");
return false;
}
}
}
}
snd_pcm_hw_params_get_period_size(hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_get_buffer_size(hw_params_copy, &bufferSize);
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Got: periodSize %lu, periods %u, bufferSize %lu", periodSize, periods, bufferSize);
/* set the format parameters */
format.m_sampleRate = sampleRate;
format.m_frames = periodSize;
format.m_frameSamples = periodSize * format.m_channelLayout.Count();
format.m_frameSize = snd_pcm_frames_to_bytes(m_pcm, 1);
m_bufferSize = (unsigned int)bufferSize;
m_timeout = std::ceil((double)(bufferSize * 1000) / (double)sampleRate);
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Setting timeout to %d ms", m_timeout);
return true;
}
/* Initialise audio devices. */
int digi_init()
{
int err, tmp;
char *device = "plughw:0,0";
snd_pcm_hw_params_t *params;
pthread_attr_t attr;
pthread_mutexattr_t mutexattr;
//added on 980905 by adb to init sound kill system
memset(SampleHandles, 255, sizeof(SampleHandles));
//end edit by adb
/* Open the ALSA sound device */
if ((err = snd_pcm_open(&snd_devhandle,device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
con_printf(CON_CRITICAL, "open failed: %s\n", snd_strerror( err ));
return -1;
}
snd_pcm_hw_params_alloca(¶ms);
err = snd_pcm_hw_params_any(snd_devhandle, params);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_access(snd_devhandle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_format(snd_devhandle, params, SND_PCM_FORMAT_U8);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
err = snd_pcm_hw_params_set_channels(snd_devhandle, params, 2);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
tmp = 11025;
err = snd_pcm_hw_params_set_rate_near(snd_devhandle, params, &tmp, NULL);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
snd_pcm_hw_params_set_periods(snd_devhandle, params, 3, 0);
snd_pcm_hw_params_set_buffer_size(snd_devhandle,params, (SOUND_BUFFER_SIZE*3)/2);
err = snd_pcm_hw_params(snd_devhandle, params);
if (err < 0) {
con_printf(CON_CRITICAL,"ALSA: Error %s\n", snd_strerror(err));
return -1;
}
/* Start the mixer thread */
/* We really should check the results of these */
pthread_mutexattr_init(&mutexattr);
pthread_mutex_init(&mutex,&mutexattr);
pthread_mutexattr_destroy(&mutexattr);
if (pthread_attr_init(&attr) != 0) {
con_printf(CON_CRITICAL, "failed to init attr\n");
snd_pcm_close( snd_devhandle );
return -1;
}
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
pthread_create(&thread_id,&attr,mixer_thread,NULL);
pthread_attr_destroy(&attr);
digi_initialised = 1;
return 0;
}
void
ags_devout_alsa_init(AgsDevout *devout,
GError **error)
{
static unsigned int period_time = 100000;
static snd_pcm_format_t format = SND_PCM_FORMAT_S16;
int rc;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hwparams;
unsigned int val;
snd_pcm_uframes_t frames;
unsigned int rate;
unsigned int rrate;
unsigned int channels;
snd_pcm_uframes_t size;
snd_pcm_sframes_t buffer_size;
snd_pcm_sframes_t period_size;
snd_pcm_sw_params_t *swparams;
int period_event = 0;
int err, dir;
/* Open PCM device for playback. */
if ((err = snd_pcm_open(&handle, devout->out.alsa.device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
printf("Playback open error: %s\n", snd_strerror(err));
return;
}
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
/* choose all parameters */
err = snd_pcm_hw_params_any(handle, hwparams);
if (err < 0) {
printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
return;
}
/* set hardware resampling */
err = snd_pcm_hw_params_set_rate_resample(handle, hwparams, 1);
if (err < 0) {
printf("Resampling setup failed for playback: %s\n", snd_strerror(err));
return;
}
/* set the interleaved read/write format */
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
printf("Access type not available for playback: %s\n", snd_strerror(err));
return;
}
/* set the sample format */
err = snd_pcm_hw_params_set_format(handle, hwparams, format);
if (err < 0) {
printf("Sample format not available for playback: %s\n", snd_strerror(err));
return;
}
/* set the count of channels */
channels = devout->dsp_channels;
err = snd_pcm_hw_params_set_channels(handle, hwparams, channels);
if (err < 0) {
printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err));
return;
}
/* set the stream rate */
rate = devout->frequency;
rrate = rate;
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rrate, 0);
if (err < 0) {
printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
return;
}
if (rrate != rate) {
printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err);
exit(-EINVAL);
}
/* set the buffer size */
size = devout->buffer_size;
err = snd_pcm_hw_params_set_buffer_size(handle, hwparams, size);
if (err < 0) {
printf("Unable to set buffer size %i for playback: %s\n", size, snd_strerror(err));
return;
}
buffer_size = size;
/* set the period time */
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir);
if (err < 0) {
printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err));
return;
}
err = snd_pcm_hw_params_get_period_size(hwparams, &size, &dir);
if (err < 0) {
printf("Unable to get period size for playback: %s\n", snd_strerror(err));
return;
}
period_size = size;
/* write the parameters to device */
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) {
printf("Unable to set hw params for playback: %s\n", snd_strerror(err));
return;
}
/* get the current swparams */
err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
printf("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
return;
}
/* start the transfer when the buffer is almost full: */
/* (buffer_size / avail_min) * avail_min */
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, (buffer_size / period_size) * period_size);
if (err < 0) {
printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
return;
}
/* allow the transfer when at least period_size samples can be processed */
/* or disable this mechanism when period event is enabled (aka interrupt like style processing) */
err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_event ? buffer_size : period_size);
if (err < 0) {
printf("Unable to set avail min for playback: %s\n", snd_strerror(err));
return;
}
/* write the parameters to the playback device */
err = snd_pcm_sw_params(handle, swparams);
if (err < 0) {
printf("Unable to set sw params for playback: %s\n", snd_strerror(err));
return;
}
/* */
devout->out.alsa.handle = handle;
devout->delay_counter = 0;
}
void
ags_devout_pcm_info(char *card_id,
guint *channels_min, guint *channels_max,
guint *rate_min, guint *rate_max,
guint *buffer_size_min, guint *buffer_size_max,
GError **error)
{
int rc;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
int err;
/* Open PCM device for playback. */
handle = NULL;
rc = snd_pcm_open(&handle, card_id, SND_PCM_STREAM_PLAYBACK, 0);
if(rc < 0) {
g_message("unable to open pcm device: %s\n\0", snd_strerror(rc));
g_set_error(error,
AGS_DEVOUT_ERROR,
AGS_DEVOUT_ERROR_LOCKED_SOUNDCARD,
"unable to open pcm device: %s\n\0",
snd_strerror(rc));
return;
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* channels */
snd_pcm_hw_params_get_channels_min(params, &val);
*channels_min = val;
snd_pcm_hw_params_get_channels_max(params, &val);
*channels_max = val;
/* samplerate */
dir = 0;
snd_pcm_hw_params_get_rate_min(params, &val, &dir);
*rate_min = val;
dir = 0;
snd_pcm_hw_params_get_rate_max(params, &val, &dir);
*rate_max = val;
/* buffer size */
dir = 0;
snd_pcm_hw_params_get_buffer_size_min(params, &frames);
*buffer_size_min = frames;
dir = 0;
snd_pcm_hw_params_get_buffer_size_max(params, &frames);
*buffer_size_max = frames;
snd_pcm_close(handle);
}
/**
* Create and initialize Alsa audio output device with given parameters.
*
* @param Parameters - optional parameters
* @throws AudioOutputException if output device cannot be opened
*/
AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, true, 1, 0) {
pcm_handle = NULL;
stream = SND_PCM_STREAM_PLAYBACK;
this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["CHANNELS"])->ValueAsInt();
this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["SAMPLERATE"])->ValueAsInt();
this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["FRAGMENTSIZE"])->ValueAsInt();
uint Fragments = ((DeviceCreationParameterInt*)Parameters["FRAGMENTS"])->ValueAsInt();
String Card = ((DeviceCreationParameterString*)Parameters["CARD"])->ValueAsString();
dmsg(2,("Checking if hw parameters supported...\n"));
if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
pcm_name = "hw:" + Card;
}
else {
fprintf(stderr, "Warning: your soundcard doesn't support chosen hardware parameters; ");
fprintf(stderr, "trying to compensate support lack with plughw...");
fflush(stdout);
pcm_name = "plughw:" + Card;
}
dmsg(2,("HW check completed.\n"));
int err;
snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
/* Open PCM. The last parameter of this function is the mode. */
/* If this is set to 0, the standard mode is used. Possible */
/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
/* If SND_PCM_NONBLOCK is used, read / write access to the */
/* PCM device will return immediately. If SND_PCM_ASYNC is */
/* specified, SIGIO will be emitted whenever a period has */
/* been completely processed by the soundcard. */
if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
}
/* Set access type. This can be either */
/* SND_PCM_ACCESS_RW_INTERLEAVED or */
/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
}
/* Set sample format */
#if WORDS_BIGENDIAN
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
#else // little endian
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
}
int dir = 0;
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
#if ALSA_MAJOR > 0
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
#else
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
}
/* Set number of periods. Periods used to be called fragments. */
if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
throw AudioOutputException(String("Error setting number of periods: ") + snd_strerror(err));
}
/* Set buffer size (in frames). The resulting latency is given by */
/* latency = periodsize * periods / (rate * bytes_per_frame) */
if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
}
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_malloc(&swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
}
if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
}
if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
}
// allocate Alsa output buffer
pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
// create audio channels for this audio device to which the sampler engines can write to
for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(i, FragmentSize));
if (((DeviceCreationParameterBool*)Parameters["ACTIVE"])->ValueAsBool()) {
Play();
}
}
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);
}
int alsa_init(alsa_dev_t *dev, void *ptr, void (*callback)(snd_async_handler_t *))
{
snd_pcm_t *phandle;
snd_pcm_t *chandle;
int err;
snd_pcm_hw_params_t *hwparams_capture, *hwparams_playback;
snd_pcm_sw_params_t *swparams_capture, *swparams_playback;
struct sched_param param;
err = sched_getparam(0, ¶m);
if (err < 0) {
perror("sched_getparam():");
exit(-1);
}
param.sched_priority = sched_get_priority_max(SCHED_FIFO);
err = sched_setscheduler(0, SCHED_FIFO, ¶m);
if (err < 0) {
perror("sched_setscheduler():");
exit(-1);
}
snd_pcm_hw_params_alloca(&hwparams_capture);
snd_pcm_sw_params_alloca(&swparams_capture);
snd_pcm_hw_params_alloca(&hwparams_playback);
snd_pcm_sw_params_alloca(&swparams_playback);
err = snd_output_stdio_attach(&dev->output, stdout, 0);
if (err < 0) {
printf("Output failed: %s\n", snd_strerror(err));
exit(-1);
}
printf("Playback device is %s\n", dev->pdevice);
printf("Stream parameters are %dHz, %s, %d channels\n", dev->rate, snd_pcm_format_name(dev->format), dev->channels);
if ((err = snd_pcm_open(&phandle, dev->pdevice, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
printf("Playback open error: %s\n", snd_strerror(err));
exit(-1);
}
if ((err = alsa_set_hwparams(dev, phandle, hwparams_playback, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0) {
printf("Setting of hwparams_playback failed: %s\n", snd_strerror(err));
exit(-1);
}
if ((err = alsa_set_swparams(dev, phandle, swparams_playback)) < 0) {
printf("Setting of swparams_playback failed: %s\n", snd_strerror(err));
exit(-1);
}
snd_pcm_dump(phandle, dev->output);
printf("Capture device is %s\n", dev->cdevice);
if ((err = snd_pcm_open(&chandle, dev->cdevice, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
printf("Capture open error: %s\n", snd_strerror(err));
exit(-1);
}
if ((err = alsa_set_hwparams(dev, chandle, hwparams_capture, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0) {
printf("Setting of hwparams_capture failed: %s\n", snd_strerror(err));
exit(-1);
}
if ((err = alsa_set_swparams(dev, chandle, swparams_capture)) < 0) {
printf("Setting of swparams_capture failed: %s\n", snd_strerror(err));
exit(-1);
}
snd_pcm_dump(chandle, dev->output);
if ((err = snd_pcm_link(phandle, chandle)) < 0)
{
printf("snd_pcm_link() failed: %s\n", snd_strerror(err));
exit(-1);
}
dev->chandle = chandle;
dev->phandle = phandle;
err = alsa_async_direct_loop(dev, ptr, callback);
if (err < 0)
printf("Transfer failed: %s\n", snd_strerror(err));
return 0;
}
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;
}
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);
}
/*
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
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;
}
int main() {
int rc;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val, val2;
int dir;
snd_pcm_uframes_t frames;
/* Open PCM device for playback. */
rc = snd_pcm_open(&handle, "default",
SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 2);
/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
snd_pcm_hw_params_set_rate_near(handle,
params, &val, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
/* Display information about the PCM interface */
printf("PCM handle name = '%s'\n",
snd_pcm_name(handle));
printf("PCM state = %s\n",
snd_pcm_state_name(snd_pcm_state(handle)));
snd_pcm_hw_params_get_access(params,
(snd_pcm_access_t *) &val);
printf("access type = %s\n",
snd_pcm_access_name((snd_pcm_access_t)val));
snd_pcm_hw_params_get_format(params, &val);
printf("format = '%s' (%s)\n",
snd_pcm_format_name((snd_pcm_format_t)val),
snd_pcm_format_description(
(snd_pcm_format_t)val));
snd_pcm_hw_params_get_subformat(params,
(snd_pcm_subformat_t *)&val);
printf("subformat = '%s' (%s)\n",
snd_pcm_subformat_name((snd_pcm_subformat_t)val),
snd_pcm_subformat_description(
(snd_pcm_subformat_t)val));
snd_pcm_hw_params_get_channels(params, &val);
printf("channels = %d\n", val);
snd_pcm_hw_params_get_rate(params, &val, &dir);
printf("rate = %d bps\n", val);
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
printf("period time = %d us\n", val);
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
printf("period size = %d frames\n", (int)frames);
snd_pcm_hw_params_get_buffer_time(params,
&val, &dir);
printf("buffer time = %d us\n", val);
snd_pcm_hw_params_get_buffer_size(params,
(snd_pcm_uframes_t *) &val);
printf("buffer size = %d frames\n", val);
snd_pcm_hw_params_get_periods(params, &val, &dir);
printf("periods per buffer = %d frames\n", val);
snd_pcm_hw_params_get_rate_numden(params,
&val, &val2);
printf("exact rate = %d/%d bps\n", val, val2);
val = snd_pcm_hw_params_get_sbits(params);
printf("significant bits = %d\n", val);
snd_pcm_hw_params_get_tick_time(params,
&val, &dir);
printf("tick time = %d us\n", val);
val = snd_pcm_hw_params_is_batch(params);
printf("is batch = %d\n", val);
val = snd_pcm_hw_params_is_block_transfer(params);
printf("is block transfer = %d\n", val);
val = snd_pcm_hw_params_is_double(params);
printf("is double = %d\n", val);
val = snd_pcm_hw_params_is_half_duplex(params);
printf("is half duplex = %d\n", val);
val = snd_pcm_hw_params_is_joint_duplex(params);
printf("is joint duplex = %d\n", val);
val = snd_pcm_hw_params_can_overrange(params);
printf("can overrange = %d\n", val);
val = snd_pcm_hw_params_can_mmap_sample_resolution(params);
printf("can mmap = %d\n", val);
val = snd_pcm_hw_params_can_pause(params);
printf("can pause = %d\n", val);
val = snd_pcm_hw_params_can_resume(params);
printf("can resume = %d\n", val);
val = snd_pcm_hw_params_can_sync_start(params);
printf("can sync start = %d\n", val);
snd_pcm_close(handle);
return 0;
}
/* setup alsa data format and buffer geometry */
static inline int alsa_set_hwparams(ao_device *device,
ao_sample_format *format)
{
ao_alsa_internal *internal = (ao_alsa_internal *) device->internal;
snd_pcm_hw_params_t *params;
int err;
unsigned int rate = internal->sample_rate = format->rate;
/* allocate the hardware parameter structure */
snd_pcm_hw_params_alloca(¶ms);
/* fetch all possible hardware parameters */
err = snd_pcm_hw_params_any(internal->pcm_handle, params);
if (err < 0){
adebug("snd_pcm_hw_params_any() failed.\n"
" Device exists but no matching hardware?\n");
return err;
}
/* set the access type */
err = snd_pcm_hw_params_set_access(internal->pcm_handle,
params, internal->access_mask);
if (err < 0){
adebug("snd_pcm_hw_params_set_access() failed.\n");
return err;
}
/* set the sample bitformat */
err = snd_pcm_hw_params_set_format(internal->pcm_handle,
params, internal->bitformat);
if (err < 0){
/* the device may support a greater bit-depth than the one
requested. */
switch(internal->bitformat){
case SND_PCM_FORMAT_U8:
if (!snd_pcm_hw_params_set_format(internal->pcm_handle,
params, SND_PCM_FORMAT_S16)){
adebug("snd_pcm_hw_params_set_format() unable to open %d bit playback.\n",format->bits);
adebug("snd_pcm_hw_params_set_format() using 16 bit playback instead.\n");
format->bits = 16;
break;
}
case SND_PCM_FORMAT_S16:
if (!snd_pcm_hw_params_set_format(internal->pcm_handle,
params, SND_PCM_FORMAT_S24)){
adebug("snd_pcm_hw_params_set_format() unable to open %d bit playback.\n",format->bits);
adebug("snd_pcm_hw_params_set_format() using 24 bit playback instead.\n");
format->bits = 24;
break;
}
case SND_PCM_FORMAT_S24:
if (!snd_pcm_hw_params_set_format(internal->pcm_handle,
params, SND_PCM_FORMAT_S32)){
adebug("snd_pcm_hw_params_set_format() unable to open %d bit playback.\n",format->bits);
adebug("snd_pcm_hw_params_set_format() using 32 bit playback instead.\n");
format->bits = 32;
break;
}
case SND_PCM_FORMAT_S32:
adebug("snd_pcm_hw_params_set_format() failed.\n");
return err;
}
}
/* set the number of channels */
err = snd_pcm_hw_params_set_channels(internal->pcm_handle,
params, (unsigned int)device->output_channels);
if (err < 0){
adebug("snd_pcm_hw_params_set_channels() failed.\n");
return err;
}
/* set the sample rate */
err = snd_pcm_hw_params_set_rate_near(internal->pcm_handle,
params, &rate, 0);
if (err < 0){
adebug("snd_pcm_hw_params_set_rate_near() failed.\n");
return err;
}
if (rate > 1.05 * format->rate || rate < 0.95 * format->rate) {
awarn("sample rate %i not supported "
"by the hardware, using %u\n", format->rate, rate);
}
/* set the time per hardware sample transfer */
if(internal->period_time==0)
internal->period_time=internal->buffer_time/4;
err = snd_pcm_hw_params_set_period_time_near(internal->pcm_handle,
params, &(internal->period_time), 0);
if (err < 0){
adebug("snd_pcm_hw_params_set_period_time_near() failed.\n");
return err;
}
/* set the length of the hardware sample buffer in microseconds */
/* some plug devices have very high minimum periods; don't
allow a buffer size small enough that it's ~ guaranteed to
skip */
if(internal->buffer_time<internal->period_time*3)
internal->buffer_time=internal->period_time*3;
err = snd_pcm_hw_params_set_buffer_time_near(internal->pcm_handle,
params, &(internal->buffer_time), 0);
if (err < 0){
adebug("snd_pcm_hw_params_set_buffer_time_near() failed.\n");
return err;
}
/* commit the params structure to the hardware via ALSA */
err = snd_pcm_hw_params(internal->pcm_handle, params);
if (err < 0){
adebug("snd_pcm_hw_params() failed.\n");
return err;
}
/* save the period size in frames for posterity */
err = snd_pcm_hw_params_get_period_size(params,
&(internal->period_size), 0);
if (err < 0){
adebug("snd_pcm_hw_params_get_period_size() failed.\n");
return err;
}
return 1;
}
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 = 0;
int count=0;
unsigned int freakuency=settings.frequency();
if (!settings.isValid()) {
qWarning("QAudioOutput: open error, invalid format.");
} else if (settings.frequency() <= 0) {
qWarning("QAudioOutput: open error, invalid sample rate (%d).",
settings.frequency());
} else {
err = -1;
}
if (err == 0) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
return false;
}
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;
}
int main(int argc, char **argv){
int i;
int aver,val,val2;
int16_t buf[BUFSIZE];
double d_buffer[BUFSIZE];
double pitch = 0.0;
struct timespec before,after;
struct pitch_tracker_params *settings;
snd_pcm_uframes_t period_size = PERIOD_SIZE;
//ALSA PCM configuration
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
int dir,rc;
snd_pcm_uframes_t frames;
/* Open PCM device for capture */
rc = snd_pcm_open( &handle, "default" , SND_PCM_STREAM_CAPTURE , 0);
if(rc < 0 ){
fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED);
/* unsigned 16-bit format */
snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16);
snd_pcm_hw_params_set_channels(handle, params, 1);
/* 44100 bits/second sampling rate */
val = 44100;
snd_pcm_hw_params_set_rate_near(handle,params, &val, &dir);
/* set size time*/
snd_pcm_hw_params_set_period_size_near(handle, params, &period_size , &dir);
/* write configuration */
rc = snd_pcm_hw_params(handle,params);
/*Display info*/
printf("PCM handle name = '%s'\n",
snd_pcm_name(handle));
printf("PCM state = %s\n",
snd_pcm_state_name(snd_pcm_state(handle)));
snd_pcm_hw_params_get_access(params,
(snd_pcm_access_t *) &val);
printf("access type = %s\n",
snd_pcm_access_name((snd_pcm_access_t)val));
snd_pcm_hw_params_get_format(params, &val);
printf("format = '%s' (%s)\n",
snd_pcm_format_name((snd_pcm_format_t)val),
snd_pcm_format_description(
(snd_pcm_format_t)val));
snd_pcm_hw_params_get_subformat(params,
(snd_pcm_subformat_t *)&val);
printf("subformat = '%s' (%s)\n",
snd_pcm_subformat_name((snd_pcm_subformat_t)val),
snd_pcm_subformat_description(
(snd_pcm_subformat_t)val));
snd_pcm_hw_params_get_channels(params, &val);
printf("channels = %d\n", val);
snd_pcm_hw_params_get_rate(params, &val, &dir);
printf("rate = %d bps\n", val);
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
printf("period time = %d us\n", val);
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
printf("period size = %d frames\n", (int)frames);
snd_pcm_hw_params_get_buffer_time(params,
&val, &dir);
printf("buffer time = %d us\n", val);
snd_pcm_hw_params_get_buffer_size(params,
(snd_pcm_uframes_t *) &val);
printf("buffer size = %d frames\n", val);
snd_pcm_hw_params_get_periods(params, &val, &dir);
printf("periods per buffer = %d frames\n", val);
snd_pcm_hw_params_get_rate_numden(params,
&val, &val2);
printf("exact rate = %d/%d bps\n", val, val2);
val = snd_pcm_hw_params_get_sbits(params);
printf("significant bits = %d\n", val);
val = snd_pcm_hw_params_is_batch(params);
printf("is batch = %d\n", val);
val = snd_pcm_hw_params_is_block_transfer(params);
printf("is block transfer = %d\n", val);
val = snd_pcm_hw_params_is_double(params);
printf("is double = %d\n", val);
val = snd_pcm_hw_params_is_half_duplex(params);
printf("is half duplex = %d\n", val);
val = snd_pcm_hw_params_is_joint_duplex(params);
printf("is joint duplex = %d\n", val);
val = snd_pcm_hw_params_can_overrange(params);
printf("can overrange = %d\n", val);
val = snd_pcm_hw_params_can_mmap_sample_resolution(params);
printf("can mmap = %d\n", val);
val = snd_pcm_hw_params_can_pause(params);
printf("can pause = %d\n", val);
val = snd_pcm_hw_params_can_resume(params);
printf("can resume = %d\n", val);
val = snd_pcm_hw_params_can_sync_start(params);
printf("can sync start = %d\n", val);
settings = open_pitch_tracker();
while(1){
rc = snd_pcm_readi(handle, buf, frames);
if (rc == -EPIPE) {
/* EPIPE means overrun */
fprintf(stderr, "overrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,
"error from read: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
for( i = 0 ; i< BUFSIZE ; i++){
d_buffer[i] = (double) buf[i];
}
pitch = compute_pitch(d_buffer, BUFSIZE, S16, settings ,ACCURACY);
if( pitch != 0.0 )
printf("frequency -> %f\n",pitch);
memset(buf,0,BUFSIZE);
}
close_pitch_tracker(&settings);
snd_pcm_drain(handle);
snd_pcm_close(handle);
return 0;
}
int main(int argc , char *argv[])
{
int socket_desc , client_sock , c , *new_sock;
struct sockaddr_in server , client;
pthread_t sniffer_thread1;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
int rcm;
//Prepare the sockaddr_in structure
server.sin_family = AF_INET;
server.sin_addr.s_addr = htonl(INADDR_ANY);
server.sin_port = htons( 2008 );
//Create socket
socket_desc = socket(AF_INET , SOCK_STREAM , 0);
if (socket_desc == -1)
{
printf("Could not create socket");
}
//Bind
if( bind(socket_desc,(struct sockaddr *)&server , sizeof(server)) < 0)
{
//print the error message
perror("bind failed. Error");
return 1;
}
puts("bind done");
//Listen
listen(socket_desc , 3);
//Accept and incoming connection
puts("Waiting for incoming connections...");
c = sizeof(struct sockaddr_in);
snd_pcm_open(&handle, "plughw:1,0",
SND_PCM_STREAM_CAPTURE, 0);
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_U8);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 1);
/* 44100 bits/second sampling rate (CD quality) */
val = 32768;
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);
/* Set period size to 32 frames. */
frames = SIZE;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);
/* Write the parameters to the driver */
rcm = snd_pcm_hw_params(handle, params);
if (rcm < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rcm));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
/* 2 bytes/sample, 2 channels */
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
while(1){
pthread_t sniffer_thread;
if(client_sock = accept(socket_desc, (struct sockaddr *)&client, (socklen_t*)&c))
{
puts("Connection accepted");
new_sock = malloc(4);
*new_sock = client_sock;
}
if( pthread_create(&sniffer_thread , NULL , connection_handler , (void*) new_sock) < 0)
{
perror("could not create thread");
return 1;
}
//Now join the thread , so that we dont terminate before the thread
//pthread_join( sniffer_thread , NULL);
puts("Handler assigned");
if (client_sock < 0)
{
perror("accept failed");
{
perror("accept failed");
return 1;
}
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
return 0;
}
static pj_status_t open_capture (struct alsa_stream* stream,
const pjmedia_aud_param *param)
{
snd_pcm_hw_params_t* params;
snd_pcm_format_t format;
int result;
unsigned int rate;
snd_pcm_uframes_t tmp_buf_size;
snd_pcm_uframes_t tmp_period_size;
if (param->rec_id < 0 || param->rec_id >= stream->af->dev_cnt)
return PJMEDIA_EAUD_INVDEV;
/* Open PCM for capture */
PJ_LOG (5,(THIS_FILE, "open_capture: Open capture device '%s'",
stream->af->devs[param->rec_id].name));
result = snd_pcm_open (&stream->ca_pcm,
stream->af->devs[param->rec_id].name,
SND_PCM_STREAM_CAPTURE,
0);
if (result < 0)
return PJMEDIA_EAUD_SYSERR;
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca (¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any (stream->ca_pcm, params);
/* Set interleaved mode */
snd_pcm_hw_params_set_access (stream->ca_pcm, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Set format */
switch (param->bits_per_sample) {
case 8:
TRACE_((THIS_FILE, "open_capture: set format SND_PCM_FORMAT_S8"));
format = SND_PCM_FORMAT_S8;
break;
case 16:
TRACE_((THIS_FILE, "open_capture: set format SND_PCM_FORMAT_S16_LE"));
format = SND_PCM_FORMAT_S16_LE;
break;
case 24:
TRACE_((THIS_FILE, "open_capture: set format SND_PCM_FORMAT_S24_LE"));
format = SND_PCM_FORMAT_S24_LE;
break;
case 32:
TRACE_((THIS_FILE, "open_capture: set format SND_PCM_FORMAT_S32_LE"));
format = SND_PCM_FORMAT_S32_LE;
break;
default:
TRACE_((THIS_FILE, "open_capture: set format SND_PCM_FORMAT_S16_LE"));
format = SND_PCM_FORMAT_S16_LE;
break;
}
snd_pcm_hw_params_set_format (stream->ca_pcm, params, format);
/* Set number of channels */
TRACE_((THIS_FILE, "open_capture: set channels: %d",
param->channel_count));
snd_pcm_hw_params_set_channels (stream->ca_pcm, params,
param->channel_count);
/* Set clock rate */
rate = param->clock_rate;
TRACE_((THIS_FILE, "open_capture: set clock rate: %d", rate));
snd_pcm_hw_params_set_rate_near (stream->ca_pcm, params, &rate, NULL);
TRACE_((THIS_FILE, "open_capture: clock rate set to: %d", rate));
/* Set period size to samples_per_frame frames. */
stream->ca_frames = (snd_pcm_uframes_t) param->samples_per_frame /
param->channel_count;
TRACE_((THIS_FILE, "open_capture: set period size: %d",
stream->ca_frames));
tmp_period_size = stream->ca_frames;
snd_pcm_hw_params_set_period_size_near (stream->ca_pcm, params,
&tmp_period_size, NULL);
TRACE_((THIS_FILE, "open_capture: period size set to: %d",
tmp_period_size));
/* Set the sound device buffer size and latency */
if (param->flags & PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY)
tmp_buf_size = (rate / 1000) * param->input_latency_ms;
else
tmp_buf_size = (rate / 1000) * PJMEDIA_SND_DEFAULT_REC_LATENCY;
snd_pcm_hw_params_set_buffer_size_near (stream->ca_pcm, params,
&tmp_buf_size);
stream->param.input_latency_ms = tmp_buf_size / (rate / 1000);
/* Set our buffer */
stream->ca_buf_size = stream->ca_frames * param->channel_count *
(param->bits_per_sample/8);
stream->ca_buf = (char*) pj_pool_alloc (stream->pool, stream->ca_buf_size);
TRACE_((THIS_FILE, "open_capture: buffer size set to: %d",
(int)tmp_buf_size));
TRACE_((THIS_FILE, "open_capture: capture_latency set to: %d ms",
(int)stream->param.input_latency_ms));
/* Activate the parameters */
result = snd_pcm_hw_params (stream->ca_pcm, params);
if (result < 0) {
snd_pcm_close (stream->ca_pcm);
return PJMEDIA_EAUD_SYSERR;
}
PJ_LOG (5,(THIS_FILE, "Opened device alsa(%s) for capture, sample rate=%d"
", ch=%d, bits=%d, period size=%d frames, latency=%d ms",
stream->af->devs[param->rec_id].name,
rate, param->channel_count,
param->bits_per_sample, stream->ca_frames,
(int)stream->param.input_latency_ms));
return PJ_SUCCESS;
}
static bool audio_capture_configure(uint32_t samplerate)
{
assert(acapt_hdl != NULL);
snd_pcm_hw_params_t *hw_params = NULL;
snd_pcm_hw_params_alloca(&hw_params);
int err = 0;
/* set hardware params */
if ((err = snd_pcm_hw_params_any(acapt_hdl, hw_params)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params_any() has failed - %s\n",
snd_strerror(err));
return false;
}
/* interleaved samples */
if ((err = snd_pcm_hw_params_set_access(acapt_hdl, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params_set_access() has "
"failed - %s\n", snd_strerror(err));
return false;
}
/* 1 channel (mono) capture */
if ((err = snd_pcm_hw_params_set_channels(acapt_hdl, hw_params, 1)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params_set_channels() has "
"failed - %s\n", snd_strerror(err));
return false;
}
/* 16 bit PCM samples */
if ((err = snd_pcm_hw_params_set_format(acapt_hdl, hw_params,
SND_PCM_FORMAT_S16_LE)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params_set_format() has "
"failed - %s\n", snd_strerror(err));
return false;
}
/* set samplerate */
if ((err = snd_pcm_hw_params_set_rate(acapt_hdl, hw_params, samplerate, 0)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params_set_rate() has "
"failed - %s\n", snd_strerror(err));
return false;
}
/* apply hardware parameters */
if ((err = snd_pcm_hw_params(acapt_hdl, hw_params)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_hw_params() has failed - %s\n\n",
snd_strerror(err));
return false;
}
/* set software parameters */
snd_pcm_sw_params_t *sw_params = NULL;
snd_pcm_sw_params_alloca(&sw_params);
if ((err = snd_pcm_sw_params_current(acapt_hdl, sw_params)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_sw_params_current() has failed - %s\n",
snd_strerror(err));
return false;
}
if ((err = snd_pcm_sw_params_set_start_threshold(acapt_hdl,
sw_params, 0U)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_sw_params_set_start_threshold() has "
"failed - %s\n", snd_strerror(err));
return false;
}
/* apply software parameters */
if ((err = snd_pcm_sw_params(acapt_hdl, sw_params)) < 0) {
fprintf(stderr, "ERROR: snd_pcm_sw_params() has failed - %s\n",
snd_strerror(err));
return false;
}
return true;
}
void *audiothread(void *v) {
int policy;
sched_param sched;
int err;
alsaaudio *dev;
pthread_getschedparam(pthread_self(),&policy,&sched);
sched.sched_priority++;//policy=SCHED_RR;
pthread_setschedparam(pthread_self(),policy,&sched);
dev=(alsaaudio*)v;
unsigned int val;
snd_pcm_t *fd;
snd_pcm_uframes_t periodsize;
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
int output_rate;
int channels;
int fragment_size;
int fragment_count;
err=snd_pcm_open(&fd, strdup("default"), SND_PCM_STREAM_PLAYBACK, 0);
if (err<0) return -1;
fragment_size=LINUXFRAG; //overall buffer size
fragment_count=2; //2 - 16 fragment count - 2 minimum, the lower it is potentially the lower the latency
//configure device
if (snd_pcm_hw_params_any(fd, hwparams) < 0) {
//printf("linuxaudio failed at params any\n");
return -1;
}
if (snd_pcm_hw_params_set_access(fd, hwparams,SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
//printf("linuxaudio failed at set access\n");
return -1;
}
if (snd_pcm_hw_params_set_format(fd, hwparams,SND_PCM_FORMAT_S16_LE) < 0) {
//printf("linuxaudio failed at set format\n");
return -1;
}
val = 44100;
if (snd_pcm_hw_params_set_rate_near(fd, hwparams,&val, 0) < 0) {
// Try 48KHZ too
//printf("linuxaudio - %d HZ not available, trying 48000HZ\n", output_rate);
val = 48000;
if (snd_pcm_hw_params_set_rate_near(fd, hwparams,&val, 0) < 0) {
//printf("linuxaudio failed at setting output rate (%d)\n", output_rate);
return -1;
}
dev->mix->freq=val;
}
channels=2;
if (snd_pcm_hw_params_set_channels(fd, hwparams, channels) < 0) {
//printf("linuxaudio failed at set channels (%d)\n", channels);
return -1;
}
periodsize = (fragment_size) / 4; // bytes -> frames for 16-bit,stereo - should be a minimum of 512
if (snd_pcm_hw_params_set_period_size_near(fd, hwparams,&periodsize, 0) < 0) {
//printf("linuxaudio failed at set period size (%d)\n", (int)periodsize);
return -1;
}
val = fragment_count;
if (snd_pcm_hw_params_set_periods_near(fd, hwparams,&val, 0) < 0) {
//printf("linuxaudio failed at set periods (%d)\n", val);
//should attempt a one by one period increase up to 16?
return -1;
}
if (snd_pcm_hw_params(fd, hwparams) < 0) {
//printf("linuxaudio failed at installing hw params\n");
return -1;
}
//loop while playing sound
dev->playing=1;
while (dev->playing)
{
dev->mix->mix16(dev->buffer);
if ((snd_pcm_writei (fd, dev->buffer,LINUXFRAG/2)) < 0) { //Half buffer for two channels?
//printf ("linuxaudio warning: buffer underrun occurred\n");
if (snd_pcm_prepare(fd) < 0) {
//printf ("linuxaudio failed at preparing pcm\n");
dev->playing=0; //die gracefully
}
}
}
snd_pcm_drop(fd);
snd_pcm_close (fd);
return 0;
}
void CAESinkALSA::EnumerateDevice(AEDeviceInfoList &list, const std::string &device, const std::string &description, snd_config_t *config)
{
snd_pcm_t *pcmhandle = NULL;
if (!OpenPCMDevice(device, "", ALSA_MAX_CHANNELS, &pcmhandle, config))
return;
snd_pcm_info_t *pcminfo;
snd_pcm_info_alloca(&pcminfo);
memset(pcminfo, 0, snd_pcm_info_sizeof());
int err = snd_pcm_info(pcmhandle, pcminfo);
if (err < 0)
{
CLog::Log(LOGINFO, "CAESinkALSA - Unable to get pcm_info for \"%s\"", device.c_str());
snd_pcm_close(pcmhandle);
}
int cardNr = snd_pcm_info_get_card(pcminfo);
CAEDeviceInfo info;
info.m_deviceName = device;
info.m_deviceType = AEDeviceTypeFromName(device);
if (cardNr >= 0)
{
/* "HDA NVidia", "HDA Intel", "HDA ATI HDMI", "SB Live! 24-bit External", ... */
char *cardName;
if (snd_card_get_name(cardNr, &cardName) == 0)
info.m_displayName = cardName;
if (info.m_deviceType == AE_DEVTYPE_HDMI && info.m_displayName.size() > 5 &&
info.m_displayName.substr(info.m_displayName.size()-5) == " HDMI")
{
/* We already know this is HDMI, strip it */
info.m_displayName.erase(info.m_displayName.size()-5);
}
/* "CONEXANT Analog", "USB Audio", "HDMI 0", "ALC889 Digital" ... */
std::string pcminfoName = snd_pcm_info_get_name(pcminfo);
/*
* Filter "USB Audio", in those cases snd_card_get_name() is more
* meaningful already
*/
if (pcminfoName != "USB Audio")
info.m_displayNameExtra = pcminfoName;
if (info.m_deviceType == AE_DEVTYPE_HDMI)
{
/* replace, this was likely "HDMI 0" */
info.m_displayNameExtra = "HDMI";
int dev = snd_pcm_info_get_device(pcminfo);
if (dev >= 0)
{
/* lets see if we can get ELD info */
snd_ctl_t *ctlhandle;
std::stringstream sstr;
sstr << "hw:" << cardNr;
std::string strHwName = sstr.str();
if (snd_ctl_open_lconf(&ctlhandle, strHwName.c_str(), 0, config) == 0)
{
snd_hctl_t *hctl;
if (snd_hctl_open_ctl(&hctl, ctlhandle) == 0)
{
snd_hctl_load(hctl);
bool badHDMI = false;
if (!GetELD(hctl, dev, info, badHDMI))
CLog::Log(LOGDEBUG, "CAESinkALSA - Unable to obtain ELD information for device \"%s\" (not supported by device, or kernel older than 3.2)",
device.c_str());
/* snd_hctl_close also closes ctlhandle */
snd_hctl_close(hctl);
if (badHDMI)
{
/*
* Warn about disconnected devices, but keep them enabled
* Detection can go wrong on Intel, Nvidia and on all
* AMD (fglrx) hardware, so it is not safe to close those
* handles
*/
CLog::Log(LOGDEBUG, "CAESinkALSA - HDMI device \"%s\" may be unconnected (no ELD data)", device.c_str());
}
}
else
{
snd_ctl_close(ctlhandle);
}
}
}
}
else if (info.m_deviceType == AE_DEVTYPE_IEC958)
{
/* append instead of replace, pcminfoName is useful for S/PDIF */
if (!info.m_displayNameExtra.empty())
info.m_displayNameExtra += ' ';
info.m_displayNameExtra += "S/PDIF";
}
else if (info.m_displayNameExtra.empty())
{
/* for USB audio, it gets a bit confusing as there is
* - "SB Live! 24-bit External"
* - "SB Live! 24-bit External, S/PDIF"
* so add "Analog" qualifier to the first one */
info.m_displayNameExtra = "Analog";
}
/* "default" is a device that will be used for all inputs, while
* "@" will be mangled to front/default/surroundXX as necessary */
if (device == "@" || device == "default")
{
/* Make it "Default (whatever)" */
info.m_displayName = "Default (" + info.m_displayName + (info.m_displayNameExtra.empty() ? "" : " " + info.m_displayNameExtra + ")");
info.m_displayNameExtra = "";
}
}
else
{
/* virtual devices: "default", "pulse", ... */
/* description can be e.g. "PulseAudio Sound Server" - for hw devices it is
* normally uninteresting, like "HDMI Audio Output" or "Default Audio Device",
* so we only use it for virtual devices that have no better display name */
info.m_displayName = description;
}
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
memset(hwparams, 0, snd_pcm_hw_params_sizeof());
/* ensure we can get a playback configuration for the device */
if (snd_pcm_hw_params_any(pcmhandle, hwparams) < 0)
{
CLog::Log(LOGINFO, "CAESinkALSA - No playback configurations available for device \"%s\"", device.c_str());
snd_pcm_close(pcmhandle);
return;
}
/* detect the available sample rates */
for (unsigned int *rate = ALSASampleRateList; *rate != 0; ++rate)
if (snd_pcm_hw_params_test_rate(pcmhandle, hwparams, *rate, 0) >= 0)
info.m_sampleRates.push_back(*rate);
/* detect the channels available */
int channels = 0;
for (int i = ALSA_MAX_CHANNELS; i >= 1; --i)
{
/* Reopen the device if needed on the special "surroundXX" cases */
if (info.m_deviceType == AE_DEVTYPE_PCM && (i == 8 || i == 6 || i == 4))
OpenPCMDevice(device, "", i, &pcmhandle, config);
if (snd_pcm_hw_params_test_channels(pcmhandle, hwparams, i) >= 0)
{
channels = i;
break;
}
}
if (device == "default" && channels == 2)
{
/* This looks like the ALSA standard default stereo dmix device, we
* probably want to use "@" instead to get surroundXX. */
snd_pcm_close(pcmhandle);
EnumerateDevice(list, "@", description, config);
return;
}
CAEChannelInfo alsaChannels;
for (int i = 0; i < channels; ++i)
{
if (!info.m_channels.HasChannel(ALSAChannelMap[i]))
info.m_channels += ALSAChannelMap[i];
alsaChannels += ALSAChannelMap[i];
}
/* remove the channels from m_channels that we cant use */
info.m_channels.ResolveChannels(alsaChannels);
/* detect the PCM sample formats that are available */
for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
{
if (AE_IS_RAW(i) || i == AE_FMT_MAX)
continue;
snd_pcm_format_t fmt = AEFormatToALSAFormat(i);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
continue;
if (snd_pcm_hw_params_test_format(pcmhandle, hwparams, fmt) >= 0)
info.m_dataFormats.push_back(i);
}
snd_pcm_close(pcmhandle);
list.push_back(info);
}
input_module_t *alsa_open_module(module_param_t *params)
{
input_module_t *mod = calloc(1, sizeof(input_module_t));
im_alsa_state *s;
module_param_t *current;
char *device = "plughw:0,0"; /* default device */
int format = AFMT_S16_LE;
int channels, rate;
int use_metadata = 1; /* Default to on */
unsigned int buffered_time;
snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE;
snd_pcm_hw_params_t *hwparams;
int err;
mod->type = ICES_INPUT_PCM;
mod->subtype = INPUT_PCM_LE_16;
mod->getdata = alsa_read;
mod->handle_event = event_handler;
mod->metadata_update = metadata_update;
mod->internal = calloc(1, sizeof(im_alsa_state));
s = mod->internal;
s->fd = NULL; /* Set it to something invalid, for now */
s->rate = 44100; /* Defaults */
s->channels = 2;
thread_mutex_create(&s->metadatalock);
current = params;
while(current)
{
if(!strcmp(current->name, "rate"))
s->rate = atoi(current->value);
else if(!strcmp(current->name, "channels"))
s->channels = atoi(current->value);
else if(!strcmp(current->name, "device"))
device = current->value;
else if(!strcmp(current->name, "metadata"))
use_metadata = atoi(current->value);
else if(!strcmp(current->name, "metadatafilename"))
ices_config->metadata_filename = current->value;
else
LOG_WARN1("Unknown parameter %s for alsa module", current->name);
current = current->next;
}
snd_pcm_hw_params_alloca(&hwparams);
if ((err = snd_pcm_open(&s->fd, device, stream, 0)) < 0)
{
LOG_ERROR2("Failed to open audio device %s: %s", device, snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_any(s->fd, hwparams)) < 0)
{
LOG_ERROR1("Failed to initialize hwparams: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_set_access(s->fd, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
LOG_ERROR1("Error setting access: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_set_format(s->fd, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
{
LOG_ERROR1("Couldn't set sample format to SND_PCM_FORMAT_S16_LE: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_set_rate_near(s->fd, hwparams, &s->rate, 0)) < 0)
{
LOG_ERROR1("Error setting rate: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_set_channels(s->fd, hwparams, s->channels)) < 0)
{
LOG_ERROR1("Error setting channels: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params_set_periods(s->fd, hwparams, 2, 0)) < 0)
{
LOG_ERROR1("Error setting periods: %s", snd_strerror(err));
goto fail;
}
buffered_time = 500000;
if ((err = snd_pcm_hw_params_set_buffer_time_near(s->fd, hwparams, &buffered_time, 0)) < 0)
{
LOG_ERROR1("Error setting buffersize: %s", snd_strerror(err));
goto fail;
}
if ((err = snd_pcm_hw_params(s->fd, hwparams)) < 0)
{
LOG_ERROR1("Error setting HW params: %s", snd_strerror(err));
goto fail;
}
/* We're done, and we didn't fail! */
LOG_INFO3("Opened audio device %s at %d channel(s), %d Hz",
device, s->channels, s->rate);
if(use_metadata)
{
if(ices_config->metadata_filename)
thread_create("im_alsa-metadata", metadata_thread_signal, mod, 1);
else
thread_create("im_alsa-metadata", metadata_thread_stdin, mod, 1);
LOG_INFO0("Started metadata update thread");
}
return mod;
fail:
close_module(mod); /* safe, this checks for valid contents */
return NULL;
}
