/*
===============
idAudioHardwareALSA::Flush
===============
*/
bool idAudioHardwareALSA::Flush( void ) {
int ret;
snd_pcm_state_t state;
state = id_snd_pcm_state( m_pcm_handle );
if ( state != SND_PCM_STATE_RUNNING && state != SND_PCM_STATE_PREPARED ) {
if ( ( ret = id_snd_pcm_prepare( m_pcm_handle ) ) < 0 ) {
Sys_Printf( "failed to recover from SND_PCM_STATE_XRUN: %s\n", id_snd_strerror( ret ) );
cvarSystem->SetCVarBool( "s_noSound", true );
return false;
}
Sys_Printf( "preparing audio device for output\n" );
}
Write( true );
}
/*
===============
idAudioHardwareALSA::Write
rely on m_freeWriteChunks which has been set in Flush() before engine did the mixing for this MIXBUFFER_SAMPLE
===============
*/
void idAudioHardwareALSA::Write( bool flushing ) {
if ( !flushing && m_remainingFrames ) {
// if we write after a new mixing loop, we should have m_writeChunk == 0
// otherwise that last remaining chunk that was never flushed out to the audio device has just been overwritten
Sys_Printf( "idAudioHardwareALSA::Write: %d frames overflowed and dropped\n", m_remainingFrames );
}
if ( !flushing ) {
// if running after the mix loop, then we have a full buffer to write out
m_remainingFrames = MIXBUFFER_SAMPLES;
}
if ( m_remainingFrames == 0 ) {
return;
}
// write the max frames you can in one shot - we need to write it all out in Flush() calls before the next Write() happens
int pos = (int)m_buffer + ( MIXBUFFER_SAMPLES - m_remainingFrames ) * m_channels * 2;
snd_pcm_sframes_t frames = id_snd_pcm_writei( m_pcm_handle, (void*)pos, m_remainingFrames );
if ( frames < 0 ) {
if ( frames != -EAGAIN ) {
Sys_Printf( "snd_pcm_writei %d frames failed: %s\n", m_remainingFrames, id_snd_strerror( frames ) );
}
return;
}
m_remainingFrames -= frames;
}
/*
=====================
idAudioHardwareALSA::Initialize
=====================
*/
bool idAudioHardwareALSA::Initialize( void ) {
int err;
common->Printf( "------ Alsa Sound Initialization -----\n" );
if ( !DLOpen() ) {
InitFailed();
return false;
}
if ( ( err = id_snd_pcm_open( &m_pcm_handle, s_alsa_pcm.GetString(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) < 0 ) {
common->Printf( "snd_pcm_open SND_PCM_STREAM_PLAYBACK '%s' failed: %s\n", s_alsa_pcm.GetString(), id_snd_strerror( err ) );
InitFailed();
return false;
}
common->Printf( "opened Alsa PCM device %s for playback\n", s_alsa_pcm.GetString() );
// set hardware parameters ----------------------------------------------------------------------
// init hwparams with the full configuration space
snd_pcm_hw_params_t *hwparams;
// this one is a define
id_snd_pcm_hw_params_alloca( &hwparams );
if ( ( err = id_snd_pcm_hw_params_any( m_pcm_handle, hwparams ) ) < 0 ) {
common->Printf( "cannot configure the PCM device: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
if ( ( err = id_snd_pcm_hw_params_set_access( m_pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 ) {
common->Printf( "SND_PCM_ACCESS_RW_INTERLEAVED failed: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
if ( ( err = id_snd_pcm_hw_params_set_format( m_pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE ) ) < 0 ) {
common->Printf( "SND_PCM_FORMAT_S16_LE failed: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
// channels
// sanity over number of speakers
if ( idSoundSystemLocal::s_numberOfSpeakers.GetInteger() != 6 && idSoundSystemLocal::s_numberOfSpeakers.GetInteger() != 2 ) {
common->Warning( "invalid value for s_numberOfSpeakers. Use either 2 or 6" );
idSoundSystemLocal::s_numberOfSpeakers.SetInteger( 2 );
}
m_channels = idSoundSystemLocal::s_numberOfSpeakers.GetInteger();
if ( ( err = id_snd_pcm_hw_params_set_channels( m_pcm_handle, hwparams, m_channels ) ) < 0 ) {
common->Printf( "error setting %d channels: %s\n", m_channels, id_snd_strerror( err ) );
if ( idSoundSystemLocal::s_numberOfSpeakers.GetInteger() != 2 ) {
// fallback to stereo if that works
m_channels = 2;
if ( ( err = id_snd_pcm_hw_params_set_channels( m_pcm_handle, hwparams, m_channels ) ) < 0 ) {
common->Printf( "fallback to stereo failed: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
} else {
common->Printf( "fallback to stereo\n" );
idSoundSystemLocal::s_numberOfSpeakers.SetInteger( 2 );
}
} else {
InitFailed();
return false;
}
}
// set sample rate (frequency)
if ( ( err = id_snd_pcm_hw_params_set_rate( m_pcm_handle, hwparams, PRIMARYFREQ, 0 ) ) < 0 ) {
common->Printf( "failed to set 44.1KHz rate: %s - try ( +set s_alsa_pcm plughw:0 ? )\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
// have enough space in the input buffer for our MIXBUFFER_SAMPLE feedings and async ticks
snd_pcm_uframes_t frames;
frames = MIXBUFFER_SAMPLES + MIXBUFFER_SAMPLES / 3;
if ( ( err = id_snd_pcm_hw_params_set_buffer_size_min( m_pcm_handle, hwparams, &frames ) ) < 0 ) {
common->Printf( "buffer size select failed: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
// apply parameters
if ( ( err = id_snd_pcm_hw_params( m_pcm_handle, hwparams ) ) < 0 ) {
common->Printf( "snd_pcm_hw_params failed: %s\n", id_snd_strerror( err ) );
InitFailed();
return false;
}
// check the buffer size
if ( ( err = id_snd_pcm_hw_params_get_buffer_size( hwparams, &frames ) ) < 0 ) {
common->Printf( "snd_pcm_hw_params_get_buffer_size failed: %s\n", id_snd_strerror( err ) );
} else {
common->Printf( "device buffer size: %lu frames ( %lu bytes )\n", ( long unsigned int )frames, frames * m_channels * 2 );
}
// TODO: can use swparams to setup the device so it doesn't underrun but rather loops over
// snd_pcm_sw_params_set_stop_threshold
// To get alsa to just loop on underruns. set the swparam stop_threshold to equal buffer size. The sound buffer will just loop and never throw an xrun.
// allocate the final mix buffer
m_buffer_size = MIXBUFFER_SAMPLES * m_channels * 2;
m_buffer = malloc( m_buffer_size );
common->Printf( "allocated a mix buffer of %d bytes\n", m_buffer_size );
#ifdef _DEBUG
// verbose the state
snd_pcm_state_t curstate = id_snd_pcm_state( m_pcm_handle );
assert( curstate == SND_PCM_STATE_PREPARED );
#endif
common->Printf( "--------------------------------------\n" );
return true;
}