本文整理汇总了C++中snd_pcm_hw_params_malloc函数的典型用法代码示例。如果您正苦于以下问题:C++ snd_pcm_hw_params_malloc函数的具体用法?C++ snd_pcm_hw_params_malloc怎么用?C++ snd_pcm_hw_params_malloc使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了snd_pcm_hw_params_malloc函数的15个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: BH_TRACE_INIT
void SoundPlayer::main()
{
BH_TRACE_INIT("SoundPlayer");
unsigned i;
for(i = 0; i < retries; ++i)
{
if(snd_pcm_open(&handle, "hw:0", SND_PCM_STREAM_PLAYBACK, 0) >= 0)
break;
Thread::sleep(retryDelay);
}
ASSERT(i < retries);
snd_pcm_hw_params_t* params;
VERIFY(!snd_pcm_hw_params_malloc(¶ms));
VERIFY(!snd_pcm_hw_params_any(handle, params));
VERIFY(!snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED));
VERIFY(!snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE));
VERIFY(!snd_pcm_hw_params_set_rate_near(handle, params, &sampleRate, 0));;
VERIFY(!snd_pcm_hw_params_set_channels(handle, params, 2));
VERIFY(!snd_pcm_hw_params(handle, params));
VERIFY(!snd_pcm_hw_params_get_period_size(params, &periodSize, 0));
snd_pcm_hw_params_free(params);
while(isRunning() && !closing)
{
flush();
VERIFY(sem.wait());
}
VERIFY(!snd_pcm_close(handle));
}
示例2: snd_pcm_stream_t
AudioProvider::AudioProvider()
{
allChannels ? channels = 4 : channels = 2;
int brokenFirst = (theDamageConfigurationHead.audioChannelsDefect[0] ? 1 : 0) + (theDamageConfigurationHead.audioChannelsDefect[1] ? 1 : 0);
int brokenSecond = (theDamageConfigurationHead.audioChannelsDefect[2] ? 1 : 0) + (theDamageConfigurationHead.audioChannelsDefect[3] ? 1 : 0);
unsigned i;
for(i = 0; i < retries; ++i)
{
if(snd_pcm_open(&handle, allChannels ? "4channelsDeinterleaved" : brokenFirst > brokenSecond ? "hw:0,0,1" : "hw:0",
snd_pcm_stream_t(SND_PCM_STREAM_CAPTURE | SND_PCM_NONBLOCK), 0) >= 0)
break;
Thread::sleep(retryDelay);
}
ASSERT(i < retries);
snd_pcm_hw_params_t* params;
VERIFY(!snd_pcm_hw_params_malloc(¶ms));
VERIFY(!snd_pcm_hw_params_any(handle, params));
VERIFY(!snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED));
VERIFY(!snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE));
VERIFY(!snd_pcm_hw_params_set_rate_near(handle, params, &sampleRate, 0));
VERIFY(!snd_pcm_hw_params_set_channels(handle, params, channels));
VERIFY(!snd_pcm_hw_params(handle, params));
snd_pcm_hw_params_free(params);
VERIFY(!snd_pcm_prepare(handle));
ASSERT(channels <= 4);
short buf[4];
VERIFY(snd_pcm_readi(handle, buf, 1) >= 0);
}
示例3: audio_renderer_init
static void audio_renderer_init() {
int rc;
decoder = opus_decoder_create(SAMPLE_RATE, CHANNEL_COUNT, &rc);
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t period_size = FRAME_SIZE * CHANNEL_COUNT * 2;
snd_pcm_uframes_t buffer_size = 12 * period_size;
unsigned int sampleRate = SAMPLE_RATE;
/* Open PCM device for playback. */
CHECK_RETURN(snd_pcm_open(&handle, audio_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))
/* Set hardware parameters */
CHECK_RETURN(snd_pcm_hw_params_malloc(&hw_params));
CHECK_RETURN(snd_pcm_hw_params_any(handle, hw_params));
CHECK_RETURN(snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED));
CHECK_RETURN(snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16_LE));
CHECK_RETURN(snd_pcm_hw_params_set_rate_near(handle, hw_params, &sampleRate, NULL));
CHECK_RETURN(snd_pcm_hw_params_set_channels(handle, hw_params, CHANNEL_COUNT));
CHECK_RETURN(snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffer_size));
CHECK_RETURN(snd_pcm_hw_params_set_period_size_near(handle, hw_params, &period_size, NULL));
CHECK_RETURN(snd_pcm_hw_params(handle, hw_params));
snd_pcm_hw_params_free(hw_params);
/* Set software parameters */
CHECK_RETURN(snd_pcm_sw_params_malloc(&sw_params));
CHECK_RETURN(snd_pcm_sw_params_current(handle, sw_params));
CHECK_RETURN(snd_pcm_sw_params_set_start_threshold(handle, sw_params, buffer_size - period_size));
CHECK_RETURN(snd_pcm_sw_params_set_avail_min(handle, sw_params, period_size));
CHECK_RETURN(snd_pcm_sw_params(handle, sw_params));
snd_pcm_sw_params_free(sw_params);
CHECK_RETURN(snd_pcm_prepare(handle));
}
示例4: ASSERT
AudioProvider::AudioProvider()
{
unsigned i;
for(i = 0; i < retries; ++i)
{
if(snd_pcm_open(&handle, "hw:0", snd_pcm_stream_t(SND_PCM_STREAM_CAPTURE | SND_PCM_NONBLOCK), 0) >= 0)
break;
SystemCall::sleep(retryDelay);
}
ASSERT(i < retries);
snd_pcm_hw_params_t* params;
VERIFY(!snd_pcm_hw_params_malloc(¶ms));
VERIFY(!snd_pcm_hw_params_any(handle, params));
VERIFY(!snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED));
VERIFY(!snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE));
VERIFY(!snd_pcm_hw_params_set_rate_near(handle, params, &sampleRate, 0));
VERIFY(!snd_pcm_hw_params_set_channels(handle, params, channels));
VERIFY(!snd_pcm_hw_params(handle, params));
snd_pcm_hw_params_free(params);
VERIFY(!snd_pcm_prepare(handle));
ASSERT(channels <= 4);
short buf[4];
VERIFY(snd_pcm_readi(handle, buf, 1) >= 0);
}
示例5: alsa_format_supported
RD_BOOL
alsa_format_supported(RD_WAVEFORMATEX * pwfx)
{
#if 0
int err;
snd_pcm_hw_params_t *hwparams = NULL;
if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
{
error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
return False;
}
if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
{
error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
return False;
}
snd_pcm_hw_params_free(hwparams);
#endif
if (pwfx->wFormatTag != WAVE_FORMAT_PCM)
return False;
if ((pwfx->nChannels != 1) && (pwfx->nChannels != 2))
return False;
if ((pwfx->wBitsPerSample != 8) && (pwfx->wBitsPerSample != 16))
return False;
if ((pwfx->nSamplesPerSec != 44100) && (pwfx->nSamplesPerSec != 22050))
return False;
return True;
}
示例6: audin_alsa_set_params
static boolean audin_alsa_set_params(AudinALSADevice* alsa, snd_pcm_t* capture_handle)
{
int error;
snd_pcm_hw_params_t* hw_params;
if ((error = snd_pcm_hw_params_malloc(&hw_params)) < 0)
{
DEBUG_WARN("snd_pcm_hw_params_malloc (%s)",
snd_strerror(error));
return False;
}
snd_pcm_hw_params_any(capture_handle, hw_params);
snd_pcm_hw_params_set_access(capture_handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(capture_handle, hw_params,
alsa->format);
snd_pcm_hw_params_set_rate_near(capture_handle, hw_params,
&alsa->actual_rate, NULL);
snd_pcm_hw_params_set_channels_near(capture_handle, hw_params,
&alsa->actual_channels);
snd_pcm_hw_params(capture_handle, hw_params);
snd_pcm_hw_params_free(hw_params);
snd_pcm_prepare(capture_handle);
if ((alsa->actual_rate != alsa->target_rate) ||
(alsa->actual_channels != alsa->target_channels))
{
DEBUG_DVC("actual rate %d / channel %d is "
"different from target rate %d / channel %d, resampling required.",
alsa->actual_rate, alsa->actual_channels,
alsa->target_rate, alsa->target_channels);
}
return True;
}
示例7: ALSA
ALSA(unsigned channels, unsigned samplerate, const std::string& device = "default") : runnable(true), pcm(nullptr), params(nullptr), fps(samplerate)
{
int rc = snd_pcm_open(&pcm, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
{
runnable = false;
throw DeviceException(General::join("Unable to open PCM device ", snd_strerror(rc)));
}
snd_pcm_format_t fmt = type_to_format(T());
if (snd_pcm_hw_params_malloc(¶ms) < 0)
{
runnable = false;
throw DeviceException("Failed to allocate memory.");
}
runnable = false;
if (
(snd_pcm_hw_params_any(pcm, params) < 0) ||
(snd_pcm_hw_params_set_access(pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) ||
(snd_pcm_hw_params_set_channels(pcm, params, channels) < 0) ||
(snd_pcm_hw_params_set_format(pcm, params, fmt) < 0) ||
(snd_pcm_hw_params_set_rate(pcm, params, samplerate, 0) < 0) ||
((rc = snd_pcm_hw_params(pcm, params)) < 0)
)
throw DeviceException(General::join("Unable to install HW params: ", snd_strerror(rc)));
runnable = true;
}
示例8: configureInitialState
/*
* Set some stuff up.
*/
static int configureInitialState(const char* pathName, AudioState* audioState)
{
#if BUILD_SIM_WITHOUT_AUDIO
return 0;
#else
audioState->handle = NULL;
snd_pcm_open(&audioState->handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if (audioState->handle) {
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_malloc(¶ms);
snd_pcm_hw_params_any(audioState->handle, params);
snd_pcm_hw_params_set_access(audioState->handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(audioState->handle, params, SND_PCM_FORMAT_S16_LE);
unsigned int rate = 44100;
snd_pcm_hw_params_set_rate_near(audioState->handle, params, &rate, NULL);
snd_pcm_hw_params_set_channels(audioState->handle, params, 2);
snd_pcm_hw_params(audioState->handle, params);
snd_pcm_hw_params_free(params);
} else {
wsLog("Couldn't open audio hardware, faking it\n");
}
return 0;
#endif
}
示例9: main
int main(int argc,char *argv[]){
int i = 0;
int err;
char buf[128];
snd_pcm_t *playback_handle;
int rate = 22025;
int channels = 2;
snd_pcm_hw_params_t *hw_params;
if((err = snd_pcm_open(&playback_handle,"default",SND_PCM_STREAM_PLAYBACK,0)) < 0){
fprintf(stderr,"can't open!%s(%s)\n","default",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_malloc(&hw_params) < 0)){
fprintf(stderr,"can't open!(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_any(playback_handle,hw_params)) < 0){
fprintf(stderr,"can't open(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_set_access(playback_handle,hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0){
fprintf(stderr,"can't open(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_set_format(playback_handle,hw_params,SND_PCM_FORMAT_S16_LE)) < 0){
fprintf(stderr,"can't set(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_set_rate_near(playback_handle,hw_params,&rate,0)) < 0){
fprintf(stderr,"can't set(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params_set_channels(playback_handle,hw_params,channels)) < 0){
fprintf(stderr,"can't set(%s)\n",snd_strerror(err));
exit(1);
}
if((err = snd_pcm_hw_params(playback_handle,hw_params)) < 0){
fprintf(stderr,"can't open(%s)\n",snd_strerror(err));
exit(1);
}
snd_pcm_hw_params_free(hw_params);
if((err = snd_pcm_prepare(playback_handle)) < 0){
fprintf(stderr,"can't prepare(%s)\n",snd_strerror(err));
exit(1);
}
i = 0;
while(i < 256){
memset(buf,i,128);
err = snd_pcm_writei(playback_handle,buf,32);
//fprintf(stderr,"write %d\n",err);
if(err < 0){
snd_pcm_prepare(playback_handle);
printf("a");
}
i++;
}
snd_pcm_close(playback_handle);
exit(0);
}
示例10: InitAudioCaptureDevice
static Int32 InitAudioCaptureDevice (Int32 channels, UInt32 sample_rate, Int32 driver_buf_size)
{
snd_pcm_hw_params_t *hw_params;
Int32 err;
if ((err = snd_pcm_open (&capture_handle, ALSA_CAPTURE_DEVICE, SND_PCM_STREAM_CAPTURE, 0)) < 0)
{
fprintf (stderr, "AUDIO >> cannot open audio device plughw:1,0 (%s)\n", snd_strerror (err));
return -1;
}
// printf ("AUDIO >> opened %s device\n", ALSA_CAPTURE_DEVICE);
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot allocate hardware parameter structure (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot initialize hardware parameter structure (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set access type (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set sample format (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &sample_rate, 0)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set sample rate (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, channels)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set channel count (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params_set_buffer_size (capture_handle, hw_params, driver_buf_size)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set buffer size (%s)\n", err, capture_handle);
}
if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot set parameters (%s)\n", err, capture_handle);
}
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_prepare (capture_handle)) < 0)
{
AUD_DEVICE_PRINT_ERROR_AND_RETURN("cannot prepare audio interface for use (%s)\n", err, capture_handle);
}
return 0;
}
示例11: tsmf_alsa_set_format
static BOOL tsmf_alsa_set_format(ITSMFAudioDevice *audio,
UINT32 sample_rate, UINT32 channels, UINT32 bits_per_sample)
{
int error;
snd_pcm_uframes_t frames;
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
TSMFAlsaAudioDevice *alsa = (TSMFAlsaAudioDevice *) audio;
if(!alsa->out_handle)
return FALSE;
snd_pcm_drop(alsa->out_handle);
alsa->actual_rate = alsa->source_rate = sample_rate;
alsa->actual_channels = alsa->source_channels = channels;
alsa->bytes_per_sample = bits_per_sample / 8;
error = snd_pcm_hw_params_malloc(&hw_params);
if(error < 0)
{
WLog_ERR(TAG, "snd_pcm_hw_params_malloc failed");
return FALSE;
}
snd_pcm_hw_params_any(alsa->out_handle, hw_params);
snd_pcm_hw_params_set_access(alsa->out_handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(alsa->out_handle, hw_params,
SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate_near(alsa->out_handle, hw_params,
&alsa->actual_rate, NULL);
snd_pcm_hw_params_set_channels_near(alsa->out_handle, hw_params,
&alsa->actual_channels);
frames = sample_rate;
snd_pcm_hw_params_set_buffer_size_near(alsa->out_handle, hw_params,
&frames);
snd_pcm_hw_params(alsa->out_handle, hw_params);
snd_pcm_hw_params_free(hw_params);
error = snd_pcm_sw_params_malloc(&sw_params);
if(error < 0)
{
WLog_ERR(TAG, "snd_pcm_sw_params_malloc");
return FALSE;
}
snd_pcm_sw_params_current(alsa->out_handle, sw_params);
snd_pcm_sw_params_set_start_threshold(alsa->out_handle, sw_params,
frames / 2);
snd_pcm_sw_params(alsa->out_handle, sw_params);
snd_pcm_sw_params_free(sw_params);
snd_pcm_prepare(alsa->out_handle);
DEBUG_TSMF("sample_rate %d channels %d bits_per_sample %d",
sample_rate, channels, bits_per_sample);
DEBUG_TSMF("hardware buffer %d frames", (int)frames);
if((alsa->actual_rate != alsa->source_rate) ||
(alsa->actual_channels != alsa->source_channels))
{
DEBUG_TSMF("actual rate %d / channel %d is different "
"from source rate %d / channel %d, resampling required.",
alsa->actual_rate, alsa->actual_channels,
alsa->source_rate, alsa->source_channels);
}
return TRUE;
}
示例12: qPrintable
int AudioALSA::init_audio()
{
int err = 0, dir = 1;
unsigned int tmp_sampfreq = sampfreq;
std::cout << qPrintable(tr("initializing audio at ")) << qPrintable(dsp_devicename) << std::endl;
if ((err = snd_pcm_open(&capture_handle, dsp_devicename.toStdString().c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) {
std::cerr << "cannot open audio device " << qPrintable(dsp_devicename) << " (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
std::cerr << "cannot allocate hardware parameter structure (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
std::cerr << "cannot initialize hardware parameter structure (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
std::cerr << "cannot set access type (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_U8)) < 0) {
std::cerr << "cannot set sample format (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &tmp_sampfreq, &dir)) < 0) {
std::cerr << "cannot set sample rate (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
sampfreq = tmp_sampfreq;
if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, 1)) < 0) {
std::cerr << "cannot set channel count (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
std::cerr << "cannot set parameters (" << snd_strerror(err) << ")." << std::endl;
exit (1);
}
snd_pcm_hw_params_free(hw_params);
if ((err = snd_pcm_prepare(capture_handle)) < 0) {
std::cerr << "cannot prepare audio interface for use (" << snd_strerror(err) << std::endl;
exit (1);
}
blksize = 256;
return 1;
}
示例13: malloc
AlsaDevice *alsa_device_sample( const char *device_name, unsigned int rate )
{
int err;
snd_pcm_hw_params_t *hw_params;
static snd_output_t *jcd_out;
AlsaDevice *dev = malloc( sizeof( *dev ) );
if ( !dev )
return NULL;
dev->device_name = malloc( 1 + strlen( device_name ) );
if ( !dev->device_name )
{
free(dev);
return NULL;
}
strcpy(dev->device_name, device_name);
err = snd_output_stdio_attach( &jcd_out, stdout, 0 );
if ( ( err = snd_pcm_open ( &dev->capture_handle, dev->device_name, SND_PCM_STREAM_CAPTURE, 0 ) ) < 0 )
{
rc = 0;
fprintf (stderr, "\033[0;31m[vokoscreen] alsa_device_sample() in alsadevice.c: cannot open audio device %s (%s)\033[0;0m\n", dev->device_name, snd_strerror (err) );
return NULL;
}
else
{
rc = 1;
// fprintf (stderr, "[vokoscreen] alsa_device_sample() in alsadevice.c: open audio device %s (%s)\n", dev->device_name, snd_strerror (err) );
}
if ( ( err = snd_pcm_hw_params_malloc ( &hw_params ) ) < 0 )
{
fprintf (stderr, "[vokoscreen] alsa_device_sample() in alsadevice.c: cannot allocate hardware parameter structure (%s)\n", snd_strerror( err ) );
}
if ( ( err = snd_pcm_hw_params_any( dev->capture_handle, hw_params ) ) < 0 )
{
fprintf (stderr, "[vokoscreen] alsa_device_sample() in alsadevice.c: cannot initialize hardware parameter structure (%s)\n", snd_strerror( err ) );
}
if ( ( err = snd_pcm_hw_params_set_rate_near (dev->capture_handle, hw_params, &rate, 0 ) ) < 0 )
{
fprintf( stderr, "[vokoscreen] alsa_device_sample() in alsadevice.c: cannot set sample rate (%s)\n", snd_strerror( err ) );
rc = 0;
}
else
{
rc = 1;
rcSampleRate = rate;
}
//fprintf ( stderr, "[vokoscreen] alsa_device_sample() in alsadevice.c: Samplerate = %d\n", rate );
snd_pcm_close( dev->capture_handle );
free( dev->device_name );
free( dev );
return dev;
}
示例14: AudioDevice
AudioAlsa::AudioAlsa( bool & _success_ful, Mixer* _mixer ) :
AudioDevice( tLimit<ch_cnt_t>(
ConfigManager::inst()->value( "audioalsa", "channels" ).toInt(),
DEFAULT_CHANNELS, SURROUND_CHANNELS ),
_mixer ),
m_handle( NULL ),
m_hwParams( NULL ),
m_swParams( NULL ),
m_convertEndian( false )
{
_success_ful = false;
int err;
if( ( err = snd_pcm_open( &m_handle,
probeDevice().toLatin1().constData(),
SND_PCM_STREAM_PLAYBACK,
0 ) ) < 0 )
{
printf( "Playback open error: %s\n", snd_strerror( err ) );
return;
}
snd_pcm_hw_params_malloc( &m_hwParams );
snd_pcm_sw_params_malloc( &m_swParams );
if( ( err = setHWParams( channels(),
SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
{
printf( "Setting of hwparams failed: %s\n",
snd_strerror( err ) );
return;
}
if( ( err = setSWParams() ) < 0 )
{
printf( "Setting of swparams failed: %s\n",
snd_strerror( err ) );
return;
}
// set FD_CLOEXEC flag for all file descriptors so forked processes
// do not inherit them
struct pollfd * ufds;
int count = snd_pcm_poll_descriptors_count( m_handle );
ufds = new pollfd[count];
snd_pcm_poll_descriptors( m_handle, ufds, count );
for( int i = 0; i < qMax( 3, count ); ++i )
{
const int fd = ( i >= count ) ? ufds[0].fd+i : ufds[i].fd;
int oldflags = fcntl( fd, F_GETFD, 0 );
if( oldflags < 0 )
continue;
oldflags |= FD_CLOEXEC;
fcntl( fd, F_SETFD, oldflags );
}
delete[] ufds;
_success_ful = true;
}
示例15: alsa_set_hw_params
static void alsa_set_hw_params(struct alsa_dev *dev, snd_pcm_t *handle,
unsigned int rate, int channels, int period)
{
int dir, ret;
snd_pcm_uframes_t period_size;
snd_pcm_uframes_t buffer_size;
snd_pcm_hw_params_t *hw_params;
ret = snd_pcm_hw_params_malloc(&hw_params);
if (ret < 0)
syslog_panic("Cannot allocate hardware parameters: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_any(handle, hw_params);
if (ret < 0)
syslog_panic("Cannot initialize hardware parameters: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_set_access(handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (ret < 0)
syslog_panic("Cannot set access type: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_set_format(handle, hw_params,
SND_PCM_FORMAT_S16_LE);
if (ret < 0)
syslog_panic("Cannot set sample format: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_set_rate_near(handle, hw_params, &rate, 0);
if (ret < 0)
syslog_panic("Cannot set sample rate: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_set_channels(handle, hw_params, channels);
if (ret < 0)
syslog_panic("Cannot set channel number: %s\n",
snd_strerror(ret));
period_size = period;
dir = 0;
ret = snd_pcm_hw_params_set_period_size_near(handle, hw_params,
&period_size, &dir);
if (ret < 0)
syslog_panic("Cannot set period size: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params_set_periods(handle, hw_params, PERIODS, 0);
if (ret < 0)
syslog_panic("Cannot set period number: %s\n",
snd_strerror(ret));
buffer_size = period_size * PERIODS;
dir = 0;
ret = snd_pcm_hw_params_set_buffer_size_near(handle, hw_params,
&buffer_size);
if (ret < 0)
syslog_panic("Cannot set buffer size: %s\n",
snd_strerror(ret));
ret = snd_pcm_hw_params(handle, hw_params);
if (ret < 0)
syslog_panic("Cannot set capture parameters: %s\n",
snd_strerror(ret));
snd_pcm_hw_params_free(hw_params);
}