jamulus/linux/sound.cpp
2009-03-28 09:05:56 +00:00

582 lines
17 KiB
C++
Executable file

/******************************************************************************\
* Copyright (c) 2004-2009
*
* Author(s):
* Volker Fischer, Alexander Kurpiers
*
* This code is based on the Open-Source sound interface implementation of
* the Dream DRM Receiver project and on the simple_client example of the
* Jack audio interface.
*
\******************************************************************************/
#include "sound.h"
#ifdef WITH_SOUND
# if USE_JACK
void CSound::OpenJack()
{
jack_status_t JackStatus;
// try to become a client of the JACK server
pJackClient = jack_client_open ( "llcon", JackNullOption, &JackStatus );
if ( pJackClient == NULL )
{
throw CGenErr ( "Jack server not running" );
}
// tell the JACK server to call "process()" whenever
// there is work to be done
jack_set_process_callback ( pJackClient, process, this );
// register a "buffer size changed" callback function
jack_set_buffer_size_callback ( pJackClient, bufferSizeCallback, this );
// register shutdown callback function
jack_on_shutdown ( pJackClient, shutdownCallback, this );
// TEST check sample rate, if not correct, just fire error
if ( jack_get_sample_rate ( pJackClient ) != SND_CRD_SAMPLE_RATE )
{
throw CGenErr ( "Jack server sample rate is different from "
"required one" );
}
}
void CSound::CloseJack()
{
// close client connection to jack server
jack_client_close ( pJackClient );
}
void CSound::Start()
{
const char** ports;
// create four ports (two for input, two for output -> stereo)
input_port_left = jack_port_register ( pJackClient, "input left",
JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
input_port_right = jack_port_register ( pJackClient, "input right",
JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
output_port_left = jack_port_register ( pJackClient, "output left",
JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
output_port_right = jack_port_register ( pJackClient, "output right",
JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
// tell the JACK server that we are ready to roll
if ( jack_activate ( pJackClient ) )
{
throw CGenErr ( "Cannot activate client" );
}
// connect the ports, note: you cannot do this before
// the client is activated, because we cannot allow
// connections to be made to clients that are not
// running
if ( ( ports = jack_get_ports ( pJackClient, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput ) ) == NULL )
{
throw CGenErr ( "Cannot find any physical capture ports" );
}
if ( !ports[1] )
{
throw CGenErr ( "Cannot find enough physical capture ports" );
}
if ( jack_connect ( pJackClient, ports[0], jack_port_name ( input_port_left ) ) )
{
throw CGenErr ( "Cannot connect input ports" );
}
if ( jack_connect ( pJackClient, ports[1], jack_port_name ( input_port_right ) ) )
{
throw CGenErr ( "Cannot connect input ports" );
}
free ( ports );
if ( ( ports = jack_get_ports ( pJackClient, NULL, NULL,
JackPortIsPhysical | JackPortIsInput ) ) == NULL )
{
throw CGenErr ( "Cannot find any physical playback ports" );
}
if ( !ports[1] )
{
throw CGenErr ( "Cannot find enough physical playback ports" );
}
if ( jack_connect ( pJackClient, jack_port_name ( output_port_left ), ports[0] ) )
{
throw CGenErr ( "Cannot connect output ports" );
}
if ( jack_connect ( pJackClient, jack_port_name ( output_port_right ), ports[1] ) )
{
throw CGenErr ( "Cannot connect output ports" );
}
free ( ports );
// call base class
CSoundBase::Start();
}
void CSound::Stop()
{
// deactivate client
jack_deactivate ( pJackClient );
// unregister ports
jack_port_unregister ( pJackClient, input_port_left );
jack_port_unregister ( pJackClient, input_port_right );
jack_port_unregister ( pJackClient, output_port_left );
jack_port_unregister ( pJackClient, output_port_right );
// call base class
CSoundBase::Stop();
}
int CSound::Init ( const int iNewPrefMonoBufferSize )
{
// try setting buffer size
// TODO seems not to work! -> no audio after this operation!
//jack_set_buffer_size ( pJackClient, iNewPrefMonoBufferSize );
// get actual buffer size
iJACKBufferSizeMono = jack_get_buffer_size ( pJackClient );
// init base clasee
CSoundBase::Init ( iJACKBufferSizeMono );
// set internal buffer size value and calculate stereo buffer size
iJACKBufferSizeStero = 2 * iJACKBufferSizeMono;
// create memory for intermediate audio buffer
vecsTmpAudioSndCrdStereo.Init ( iJACKBufferSizeStero );
return iJACKBufferSizeMono;
}
// JACK callbacks --------------------------------------------------------------
int CSound::process ( jack_nframes_t nframes, void* arg )
{
int i;
CSound* pSound = reinterpret_cast<CSound*> ( arg );
// get input data pointer
jack_default_audio_sample_t* in_left =
(jack_default_audio_sample_t*) jack_port_get_buffer (
pSound->input_port_left, nframes );
jack_default_audio_sample_t* in_right =
(jack_default_audio_sample_t*) jack_port_get_buffer (
pSound->input_port_right, nframes );
// copy input data
for ( i = 0; i < pSound->iJACKBufferSizeMono; i++ )
{
pSound->vecsTmpAudioSndCrdStereo[2 * i] = (short) ( in_left[i] * _MAXSHORT );
pSound->vecsTmpAudioSndCrdStereo[2 * i + 1] = (short) ( in_right[i] * _MAXSHORT );
}
// call processing callback function
pSound->ProcessCallback ( pSound->vecsTmpAudioSndCrdStereo );
// get output data pointer
jack_default_audio_sample_t* out_left =
(jack_default_audio_sample_t*) jack_port_get_buffer (
pSound->output_port_left, nframes );
jack_default_audio_sample_t* out_right =
(jack_default_audio_sample_t*) jack_port_get_buffer (
pSound->output_port_right, nframes );
// copy output data
for ( i = 0; i < pSound->iJACKBufferSizeMono; i++ )
{
out_left[i] = (jack_default_audio_sample_t)
pSound->vecsTmpAudioSndCrdStereo[2 * i] / _MAXSHORT;
out_right[i] = (jack_default_audio_sample_t)
pSound->vecsTmpAudioSndCrdStereo[2 * i + 1] / _MAXSHORT;
}
return 0; // zero on success, non-zero on error
}
int CSound::bufferSizeCallback ( jack_nframes_t nframes, void *arg )
{
CSound* pSound = reinterpret_cast<CSound*> ( arg );
pSound->EmitReinitRequestSignal();
return 0; // zero on success, non-zero on error
}
void CSound::shutdownCallback ( void *arg )
{
// without a Jack server, our software makes no sense to run, throw
// error message
throw CGenErr ( "Jack server was shut down" );
}
# else
// Wave in *********************************************************************
void CSound::InitRecording()
{
int err;
// if recording device was already open, close it first
if ( rhandle != NULL )
{
snd_pcm_close ( rhandle );
}
/* record device: The most important ALSA interfaces to the PCM devices are
the "plughw" and the "hw" interface. If you use the "plughw" interface,
you need not care much about the sound hardware. If your soundcard does
not support the sample rate or sample format you specify, your data will
be automatically converted. This also applies to the access type and the
number of channels. With the "hw" interface, you have to check whether
your hardware supports the configuration you would like to use */
// either "hw:0,0" or "plughw:0,0"
if ( err = snd_pcm_open ( &rhandle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0 ) != 0 )
{
qDebug ( "open error: %s", snd_strerror ( err ) );
}
// recording should be blocking
if ( err = snd_pcm_nonblock ( rhandle, FALSE ) != 0 )
{
qDebug ( "cannot set blocking: %s", snd_strerror ( err ) );
}
// set hardware parameters
SetHWParams ( rhandle, iBufferSizeIn, iCurPeriodSizeIn );
// start record
snd_pcm_reset ( rhandle );
snd_pcm_start ( rhandle );
qDebug ( "alsa init record done" );
}
bool CSound::Read ( CVector<short>& psData )
{
int ret;
// check if device must be opened or reinitialized
if ( bChangParamIn == true )
{
InitRecording();
// reset flag
bChangParamIn = false;
}
ret = snd_pcm_readi ( rhandle, &psData[0], iBufferSizeIn );
if ( ret < 0 )
{
if ( ret == -EPIPE )
{
// under-run
qDebug ( "rprepare" );
ret = snd_pcm_prepare ( rhandle );
if ( ret < 0 )
{
qDebug ( "Can't recover from underrun, prepare failed: %s", snd_strerror ( ret ) );
}
ret = snd_pcm_start ( rhandle );
if ( ret < 0 )
{
qDebug ( "Can't recover from underrun, start failed: %s", snd_strerror ( ret ) );
}
return true;
}
else if ( ret == -ESTRPIPE )
{
qDebug ( "strpipe" );
// wait until the suspend flag is released
while ( ( ret = snd_pcm_resume ( rhandle ) ) == -EAGAIN )
{
sleep ( 1 );
}
if ( ret < 0 )
{
ret = snd_pcm_prepare ( rhandle );
if ( ret < 0 )
{
qDebug ( "Can't recover from suspend, prepare failed: %s", snd_strerror ( ret ) );
}
throw CGenErr ( "CSound:Read" );
}
return true;
}
else
{
qDebug ( "CSound::Read: %s", snd_strerror ( ret ) );
throw CGenErr ( "CSound:Read" );
}
}
else
{
return false;
}
}
// Wave out ********************************************************************
void CSound::InitPlayback()
{
int err;
// if playback device was already open, close it first
if ( phandle != NULL )
{
snd_pcm_close ( phandle );
}
// playback device (either "hw:0,0" or "plughw:0,0")
if ( err = snd_pcm_open ( &phandle, "hw:0,0",
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) != 0 )
{
qDebug ( "open error: %s", snd_strerror ( err ) );
}
// non-blocking playback
if ( err = snd_pcm_nonblock ( phandle, TRUE ) != 0 )
{
qDebug ( "cannot set blocking: %s", snd_strerror ( err ) );
}
// set hardware parameters
SetHWParams ( phandle, iBufferSizeOut, iCurPeriodSizeOut );
// start playback
snd_pcm_start ( phandle );
qDebug ( "alsa init playback done" );
}
bool CSound::Write ( CVector<short>& psData )
{
int size = iBufferSizeOut;
int start = 0;
int ret;
// check if device must be opened or reinitialized
if ( bChangParamOut == true )
{
InitPlayback();
// reset flag
bChangParamOut = false;
}
while ( size )
{
ret = snd_pcm_writei ( phandle, &psData[start], size );
if ( ret < 0 )
{
if ( ret == -EPIPE )
{
// under-run
qDebug ( "wunderrun" );
ret = snd_pcm_prepare ( phandle );
if ( ret < 0 )
{
qDebug ( "Can't recover from underrun, prepare failed: %s", snd_strerror ( ret ) );
}
continue;
}
else if ( ret == -EAGAIN )
{
if ( ( ret = snd_pcm_wait ( phandle, 1000 ) ) < 0 )
{
qDebug ( "poll failed (%s)", snd_strerror ( ret ) );
break;
}
continue;
}
else if ( ret == -ESTRPIPE )
{
qDebug ( "wstrpipe" );
// wait until the suspend flag is released
while ( ( ret = snd_pcm_resume ( phandle ) ) == -EAGAIN )
{
sleep ( 1 );
}
if ( ret < 0 )
{
ret = snd_pcm_prepare ( phandle );
if ( ret < 0 )
{
qDebug ( "Can't recover from suspend, prepare failed: %s", snd_strerror ( ret ) );
}
}
continue;
}
else
{
qDebug ( "Write error: %s", snd_strerror ( ret ) );
}
break; // skip one period
}
if ( ret > 0 )
{
size -= ret;
start += ret;
}
}
return false;
}
// Common ***********************************************************************
bool CSound::SetHWParams ( snd_pcm_t* handle, const int iDesiredBufferSize,
const int iNumPeriodBlocks )
{
int err;
snd_pcm_hw_params_t* hwparams;
// allocate an invalid snd_pcm_hw_params_t using standard malloc
if ( err = snd_pcm_hw_params_malloc ( &hwparams ) < 0 )
{
qDebug ( "cannot allocate hardware parameter structure (%s)\n", snd_strerror ( err ) );
return true;
}
// fill params with a full configuration space for a PCM
if ( err = snd_pcm_hw_params_any ( handle, hwparams ) < 0 )
{
qDebug ( "cannot initialize hardware parameter structure (%s)\n", snd_strerror ( err ) );
return true;
}
// restrict a configuration space to contain only one access type:
// set the interleaved read/write format
if ( err = snd_pcm_hw_params_set_access ( handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED ) < 0 )
{
qDebug ( "Access type not available : %s", snd_strerror ( err ) );
return true;
}
// restrict a configuration space to contain only one format:
// set the sample format PCM, 16 bit
if ( err = snd_pcm_hw_params_set_format ( handle, hwparams, SND_PCM_FORMAT_S16 ) < 0 )
{
qDebug ( "Sample format not available : %s", snd_strerror ( err ) );
return true;
}
// restrict a configuration space to contain only one channels count:
// set the count of channels (usually stereo, 2 channels)
if ( err = snd_pcm_hw_params_set_channels ( handle, hwparams, NUM_IN_OUT_CHANNELS ) < 0 )
{
qDebug ( "Channels count (%i) not available s: %s", NUM_IN_OUT_CHANNELS, snd_strerror ( err ) );
return true;
}
// restrict a configuration space to have rate nearest to a target:
// set the sample-rate
unsigned int rrate = SND_CRD_SAMPLE_RATE;
if ( err = snd_pcm_hw_params_set_rate_near ( handle, hwparams, &rrate, 0 ) < 0 )
{
qDebug ( "Rate %iHz not available : %s", rrate, snd_strerror ( err ) );
return true;
}
if ( rrate != SND_CRD_SAMPLE_RATE ) // check if rate is possible
{
qDebug ( "Rate doesn't match (requested %iHz, get %iHz)", rrate, err );
return true;
}
// set the period size
snd_pcm_uframes_t PeriodSize = iDesiredBufferSize;
if ( err = snd_pcm_hw_params_set_period_size_near ( handle, hwparams, &PeriodSize, 0 ) < 0 )
{
qDebug ( "cannot set period size (%s)\n", snd_strerror ( err ) );
return true;
}
// set the buffer size and period size
snd_pcm_uframes_t BufferFrames = iDesiredBufferSize * iNumPeriodBlocks;
if ( err = snd_pcm_hw_params_set_buffer_size_near ( handle, hwparams, &BufferFrames ) < 0 )
{
qDebug ( "cannot set buffer size (%s)\n", snd_strerror ( err ) );
return true;
}
// check period and buffer size
snd_pcm_uframes_t period_size;
err = snd_pcm_hw_params_get_period_size ( hwparams, &period_size, 0 );
if ( err < 0 )
{
qDebug ( "Unable to get period size: %s\n", snd_strerror ( err ) );
}
qDebug ( "frame size: %d (desired: %d)", (int) period_size, iDesiredBufferSize );
snd_pcm_uframes_t buffer_size;
if ( err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size ) < 0 )
{
qDebug ( "Unable to get buffer size: %s\n", snd_strerror ( err ) );
}
qDebug ( "buffer size: %d (desired: %d)", (int) buffer_size, iDesiredBufferSize * iNumPeriodBlocks );
// write the parameters to device
if ( err = snd_pcm_hw_params ( handle, hwparams ) < 0 )
{
qDebug("Unable to set hw params : %s", snd_strerror(err));
return true;
}
// clean-up
snd_pcm_hw_params_free ( hwparams );
return false;
}
void CSound::Close()
{
// read
if ( rhandle != NULL )
{
snd_pcm_close ( rhandle );
}
rhandle = NULL;
// playback
if ( phandle != NULL )
{
snd_pcm_close ( phandle );
}
phandle = NULL;
}
# endif // USE_JACK
#endif // WITH_SOUND