void AudioQueueCallback(void * inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
audio_output_t * p_aout = (audio_output_t *)inUserData;
aout_buffer_t * p_buffer = NULL;
if (p_aout) {
struct aout_sys_t * p_sys = p_aout->sys;
aout_packet_t * packet = &p_sys->packet;
if (packet)
{
vlc_mutex_lock( &packet->lock );
p_buffer = aout_FifoPop2( &packet->fifo );
vlc_mutex_unlock( &packet->lock );
}
}
if ( p_buffer != NULL ) {
vlc_memcpy( inBuffer->mAudioData, p_buffer->p_buffer, p_buffer->i_buffer );
inBuffer->mAudioDataByteSize = p_buffer->i_buffer;
aout_BufferFree( p_buffer );
} else {
vlc_memset( inBuffer->mAudioData, 0, inBuffer->mAudioDataBytesCapacity );
inBuffer->mAudioDataByteSize = inBuffer->mAudioDataBytesCapacity;
}
AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
}
/*****************************************************************************
* Play
*****************************************************************************/
static void Play( void * _p_aout, void * _p_buffer, size_t i_size,
const media_raw_audio_format &format )
{
aout_instance_t * p_aout = (aout_instance_t*) _p_aout;
float * p_buffer = (float*) _p_buffer;
aout_sys_t * p_sys = (aout_sys_t*) p_aout->output.p_sys;
aout_buffer_t * p_aout_buffer;
p_aout_buffer = aout_OutputNextBuffer( p_aout,
mdate() + p_sys->latency,
VLC_FALSE );
if( p_aout_buffer != NULL )
{
p_aout->p_vlc->pf_memcpy( p_buffer, p_aout_buffer->p_buffer,
MIN( i_size, p_aout_buffer->i_nb_bytes ) );
if( p_aout_buffer->i_nb_bytes < i_size )
{
p_aout->p_vlc->pf_memset( p_buffer + p_aout_buffer->i_nb_bytes,
0, i_size - p_aout_buffer->i_nb_bytes );
}
aout_BufferFree( p_aout_buffer );
}
else
{
p_aout->p_vlc->pf_memset( p_buffer, 0, i_size );
}
}
/*****************************************************************************
* DoWork: mix a new output buffer - this does nothing, indeed
*****************************************************************************/
static void DoWork( aout_mixer_t * p_mixer, aout_buffer_t * p_buffer )
{
VLC_UNUSED( p_buffer );
unsigned i = 0;
aout_mixer_input_t * p_input = p_mixer->input[i];
while ( p_input->is_invalid )
p_input = p_mixer->input[++i];
aout_buffer_t * p_old_buffer = aout_FifoPop( NULL, &p_input->fifo );
/* We don't free the old buffer because,
* The aout core use a hack to avoid useless memcpy: the buffer in which
* to mix is the same as the one in the first active input fifo.
* So the ownership of that buffer belongs to our caller */
assert( p_old_buffer == p_buffer );
/* Empty other FIFOs to avoid a memory leak. */
for ( i++; i < p_mixer->input_count; i++ )
{
p_input = p_mixer->input[i];
if ( p_input->is_invalid )
continue;
while ((p_old_buffer = aout_FifoPop( NULL, &p_input->fifo )))
aout_BufferFree( p_old_buffer );
}
}
/*****************************************************************************
* DoWork: mix a new output buffer - this does nothing, indeed
*****************************************************************************/
static void DoWork( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
{
int i = 0;
aout_input_t * p_input = p_aout->pp_inputs[i];
while ( p_input->b_error )
{
p_input = p_aout->pp_inputs[++i];
}
aout_FifoPop( p_aout, &p_input->fifo );
/* Empty other FIFOs to avoid a memory leak. */
for ( i++; i < p_aout->i_nb_inputs; i++ )
{
aout_fifo_t * p_fifo;
aout_buffer_t * p_deleted;
p_input = p_aout->pp_inputs[i];
if ( p_input->b_error ) continue;
p_fifo = &p_input->fifo;
p_deleted = p_fifo->p_first;
while ( p_deleted != NULL )
{
aout_buffer_t * p_next = p_deleted->p_next;
aout_BufferFree( p_deleted );
p_deleted = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
}
static void OSSThreadCleanup( void *data )
{
oss_thread_ctx_t *p_ctx = data;
if( p_ctx->p_buffer )
aout_BufferFree( p_ctx->p_buffer );
else
free( p_ctx->p_bytes );
}
/*****************************************************************************
* Process: callback for JACK
*****************************************************************************/
int Process( jack_nframes_t i_frames, void *p_arg )
{
unsigned int i, j, i_nb_samples = 0;
audio_output_t *p_aout = (audio_output_t*) p_arg;
struct aout_sys_t *p_sys = p_aout->sys;
jack_sample_t *p_src = NULL;
jack_nframes_t dframes = p_sys->latency
- jack_frames_since_cycle_start( p_sys->p_jack_client );
jack_time_t dtime = dframes * 1000 * 1000 / jack_get_sample_rate( p_sys->p_jack_client );
mtime_t play_date = mdate() + (mtime_t) ( dtime );
/* Get the next audio data buffer */
aout_buffer_t *p_buffer = aout_PacketNext( p_aout, play_date );
if( p_buffer != NULL )
{
p_src = (jack_sample_t *)p_buffer->p_buffer;
i_nb_samples = p_buffer->i_nb_samples;
}
/* Get the JACK buffers to write to */
for( i = 0; i < p_sys->i_channels; i++ )
{
p_sys->p_jack_buffers[i] = jack_port_get_buffer( p_sys->p_jack_ports[i],
i_frames );
}
/* Copy in the audio data */
for( j = 0; j < i_nb_samples; j++ )
{
for( i = 0; i < p_sys->i_channels; i++ )
{
jack_sample_t *p_dst = p_sys->p_jack_buffers[i];
p_dst[j] = *p_src;
p_src++;
}
}
/* Fill any remaining buffer with silence */
if( i_nb_samples < i_frames )
{
for( i = 0; i < p_sys->i_channels; i++ )
{
memset( p_sys->p_jack_buffers[i] + i_nb_samples, 0,
sizeof( jack_sample_t ) * (i_frames - i_nb_samples) );
}
}
if( p_buffer )
{
aout_BufferFree( p_buffer );
}
return 0;
}
static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
VLC_UNUSED( s );
aout_instance_t *p_aout = (aout_instance_t *)userdata;
struct aout_sys_t * p_sys = (struct aout_sys_t *) p_aout->output.p_sys;
mtime_t next_date;
assert(s);
assert(p_sys);
size_t buffer_size = p_sys->buffer_size;
PULSE_DEBUG( "Pulse stream request %d", length);
do {
aout_buffer_t * p_buffer = NULL;
if(p_sys->started) {
pa_usec_t latency;
int negative;
if(pa_stream_get_latency(p_sys->stream, &latency, &negative)<0) {
if (pa_context_errno(p_sys->context) != PA_ERR_NODATA) {
msg_Err(p_aout, "pa_stream_get_latency() failed: %s", pa_strerror(pa_context_errno(p_sys->context)));
}
latency = 0;
}
PULSE_DEBUG( "Pulse stream request latency=%"PRId64"", latency);
next_date = mdate() + latency;
if(p_sys->start_date < next_date + AOUT_PTS_TOLERANCE ) {
p_buffer = aout_OutputNextBuffer( p_aout, next_date, 0);
}
}
if ( p_buffer != NULL )
{
PULSE_DEBUG( "Pulse stream request write buffer %d", p_buffer->i_buffer);
pa_stream_write(p_sys->stream, p_buffer->p_buffer, p_buffer->i_buffer, NULL, 0, PA_SEEK_RELATIVE);
length -= p_buffer->i_buffer;
aout_BufferFree( p_buffer );
}
else
{
PULSE_DEBUG( "Pulse stream request write zeroes");
void *data = pa_xmalloc(buffer_size);
bzero(data, buffer_size);
pa_stream_write(p_sys->stream, data, buffer_size, pa_xfree, 0, PA_SEEK_RELATIVE);
length -= buffer_size;
}
} while(length > buffer_size);
pa_threaded_mainloop_signal(p_sys->mainloop, 0);
}
static void Play(aout_instance_t *p_aout) {
struct aout_sys_t *p_sys = p_aout->output.p_sys;
int length;
aout_buffer_t *p_buffer;
while ((p_buffer = aout_FifoPop(&p_aout->output.fifo)) != NULL) {
length = 0;
while (length < p_buffer->i_buffer) {
length += at_write(p_sys->AudioTrack, (char*)(p_buffer->p_buffer) + length, p_buffer->i_buffer - length);
}
aout_BufferFree(p_buffer);
}
}
/*****************************************************************************
* Process: callback for JACK
*****************************************************************************/
int Process( jack_nframes_t i_frames, void *p_arg )
{
unsigned int i, j, i_nb_samples = 0;
aout_instance_t *p_aout = (aout_instance_t*) p_arg;
struct aout_sys_t *p_sys = p_aout->output.p_sys;
jack_sample_t *p_src = NULL;
/* Get the next audio data buffer */
aout_buffer_t *p_buffer = aout_FifoPop( p_aout, &p_aout->output.fifo );
if( p_buffer != NULL )
{
p_src = (jack_sample_t *)p_buffer->p_buffer;
i_nb_samples = p_buffer->i_nb_samples;
}
/* Get the JACK buffers to write to */
for( i = 0; i < p_sys->i_channels; i++ )
{
p_sys->p_jack_buffers[i] = jack_port_get_buffer( p_sys->p_jack_ports[i],
i_frames );
}
/* Copy in the audio data */
for( j = 0; j < i_nb_samples; j++ )
{
for( i = 0; i < p_sys->i_channels; i++ )
{
jack_sample_t *p_dst = p_sys->p_jack_buffers[i];
p_dst[j] = *p_src;
p_src++;
}
}
/* Fill any remaining buffer with silence */
if( i_nb_samples < i_frames )
{
for( i = 0; i < p_sys->i_channels; i++ )
{
memset( p_sys->p_jack_buffers[i] + i_nb_samples, 0,
sizeof( jack_sample_t ) * (i_frames - i_nb_samples) );
}
}
if( p_buffer )
{
aout_BufferFree( p_buffer );
}
return 0;
}
/*****************************************************************************
* aout_FifoReset: trash all buffers
*****************************************************************************/
void aout_FifoReset( aout_fifo_t * p_fifo )
{
aout_buffer_t * p_buffer;
date_Set( &p_fifo->end_date, VLC_TS_INVALID );
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
/*****************************************************************************
* aout_FifoSet : set end_date and trash all buffers (because they aren't
* properly dated)
*****************************************************************************/
void aout_FifoSet( aout_fifo_t * p_fifo, mtime_t date )
{
aout_buffer_t * p_buffer;
date_Set( &p_fifo->end_date, date );
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
/*****************************************************************************
* aout_FifoDestroy : destroy a FIFO and its buffers
*****************************************************************************/
void aout_FifoDestroy( aout_instance_t * p_aout, aout_fifo_t * p_fifo )
{
aout_buffer_t * p_buffer;
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
/*****************************************************************************
* aout_DecPlay : filter & mix the decoded buffer
*****************************************************************************/
int aout_DecPlay (audio_output_t *p_aout, block_t *p_buffer, int i_input_rate)
{
aout_owner_t *owner = aout_owner (p_aout);
aout_input_t *input;
assert( i_input_rate >= INPUT_RATE_DEFAULT / AOUT_MAX_INPUT_RATE &&
i_input_rate <= INPUT_RATE_DEFAULT * AOUT_MAX_INPUT_RATE );
assert( p_buffer->i_pts > 0 );
p_buffer->i_length = (mtime_t)p_buffer->i_nb_samples * 1000000
/ owner->input_format.i_rate;
aout_lock( p_aout );
aout_CheckRestart( p_aout );
input = owner->input;
if (unlikely(input == NULL)) /* can happen due to restart */
{
aout_unlock( p_aout );
aout_BufferFree( p_buffer );
return -1;
}
/* Input */
p_buffer = aout_InputPlay (p_aout, input, p_buffer, i_input_rate,
&owner->sync.date);
if( p_buffer != NULL )
{
date_Increment (&owner->sync.date, p_buffer->i_nb_samples);
/* Mixer */
if (owner->volume.mixer != NULL)
{
float amp = owner->volume.multiplier
* vlc_atomic_getf (&owner->gain.multiplier);
aout_MixerRun (owner->volume.mixer, p_buffer, amp);
}
/* Output */
aout_OutputPlay( p_aout, p_buffer );
}
aout_unlock( p_aout );
#ifdef __unix__
sched_yield();
#endif
return 0;
}
/*****************************************************************************
* aout_OutputPlay : play a buffer
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
void aout_OutputPlay( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
{
aout_FiltersPlay( p_aout, p_aout->output.pp_filters,
p_aout->output.i_nb_filters,
&p_buffer );
if( p_buffer->i_nb_bytes == 0 )
{
aout_BufferFree( p_buffer );
return;
}
aout_lock_output_fifo( p_aout );
aout_FifoPush( p_aout, &p_aout->output.fifo, p_buffer );
p_aout->output.pf_play( p_aout );
aout_unlock_output_fifo( p_aout );
}
/*****************************************************************************
* aout_FifoDestroy : destroy a FIFO and its buffers
*****************************************************************************/
void aout_FifoDestroy( aout_instance_t * p_aout, aout_fifo_t * p_fifo )
{
aout_buffer_t * p_buffer;
(void)p_aout;
AOUT_ASSERT_FIFO_LOCKED;
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
/*****************************************************************************
* aout_FifoSet : set end_date and trash all buffers (because they aren't
* properly dated)
*****************************************************************************/
void aout_FifoSet( aout_instance_t * p_aout, aout_fifo_t * p_fifo,
mtime_t date )
{
aout_buffer_t * p_buffer;
(void)p_aout;
AOUT_ASSERT_FIFO_LOCKED;
date_Set( &p_fifo->end_date, date );
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
p_fifo->p_first = NULL;
p_fifo->pp_last = &p_fifo->p_first;
}
/*****************************************************************************
* aout_OutputNextBuffer : give the audio output plug-in the right buffer
*****************************************************************************
* If b_can_sleek is 1, the aout core functions won't try to resample
* new buffers to catch up - that is we suppose that the output plug-in can
* compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
* This function is entered with no lock at all :-).
*****************************************************************************/
aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
mtime_t start_date,
bool b_can_sleek )
{
aout_buffer_t * p_buffer;
aout_lock_output_fifo( p_aout );
p_buffer = p_aout->output.fifo.p_first;
/* Drop the audio sample if the audio output is really late.
* In the case of b_can_sleek, we don't use a resampler so we need to be
* a lot more severe. */
while ( p_buffer && p_buffer->i_pts <
(b_can_sleek ? start_date : mdate()) - AOUT_PTS_TOLERANCE )
{
msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
"trashing %"PRId64"us", mdate() - p_buffer->i_pts,
p_buffer->i_length );
p_buffer = p_buffer->p_next;
aout_BufferFree( p_aout->output.fifo.p_first );
p_aout->output.fifo.p_first = p_buffer;
}
if ( p_buffer == NULL )
{
p_aout->output.fifo.pp_last = &p_aout->output.fifo.p_first;
#if 0 /* This is bad because the audio output might just be trying to fill
* in its internal buffers. And anyway, it's up to the audio output
* to deal with this kind of starvation. */
/* Set date to 0, to allow the mixer to send a new buffer ASAP */
aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
if ( !p_aout->output.b_starving )
msg_Dbg( p_aout,
"audio output is starving (no input), playing silence" );
p_aout->output.b_starving = 1;
#endif
aout_unlock_output_fifo( p_aout );
return NULL;
}
/* This routine will be called by the PortAudio engine when audio is needed.
* It may called at interrupt level on some machines so don't do anything
* that could mess up the system like calling malloc() or free().
*/
static int paCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *paDate,
PaStreamCallbackFlags statusFlags, void *p_cookie )
{
VLC_UNUSED( inputBuffer ); VLC_UNUSED( statusFlags );
struct aout_sys_t *p_sys = (struct aout_sys_t*) p_cookie;
aout_instance_t *p_aout = p_sys->p_aout;
aout_buffer_t *p_buffer;
mtime_t out_date;
out_date = mdate() + (mtime_t) ( 1000000 *
( paDate->outputBufferDacTime - paDate->currentTime ) );
p_buffer = aout_OutputNextBuffer( p_aout, out_date, true );
if ( p_buffer != NULL )
{
if( p_sys->b_chan_reorder )
{
/* Do the channel reordering here */
aout_ChannelReorder( p_buffer->p_buffer, p_buffer->i_nb_bytes,
p_sys->i_channels, p_sys->pi_chan_table,
p_sys->i_bits_per_sample );
}
vlc_memcpy( outputBuffer, p_buffer->p_buffer,
framesPerBuffer * p_sys->i_sample_size );
/* aout_BufferFree may be dangereous here, but then so is
* aout_OutputNextBuffer (calls aout_BufferFree internally).
* one solution would be to link the no longer useful buffers
* in a second fifo (in aout_OutputNextBuffer too) and to
* wait until we are in Play to do the actual free.
*/
aout_BufferFree( p_buffer );
}
else
/* Audio output buffer shortage -> stop the fill process and wait */
{
vlc_memset( outputBuffer, 0, framesPerBuffer * p_sys->i_sample_size );
}
return 0;
}
/*****************************************************************************
* Thread: thread used to DMA the data to the device
*****************************************************************************/
static void* Thread( vlc_object_t *p_this )
{
aout_instance_t * p_aout = (aout_instance_t*)p_this;
aout_buffer_t * p_buffer;
struct aout_sys_t * p_sys = p_aout->output.p_sys;
PCMAudioPlayer * pPlayer = p_sys->pPlayer;
int canc = vlc_savecancel ();
while( vlc_object_alive (p_aout) )
{
pPlayer->WaitForBuffer();
vlc_mutex_lock( &p_aout->output_fifo_lock );
p_buffer = aout_FifoPop( p_aout, &p_aout->output.fifo );
vlc_mutex_unlock( &p_aout->output_fifo_lock );
#define i p_sys->nNextBufferIndex
if( p_buffer == NULL )
{
vlc_memset( p_aout, p_sys->ppBuffers[ i ], 0,
p_sys->nBufferSize );
}
else
{
InterleaveS16( (int16_t *)p_buffer->p_buffer,
(int16_t *)p_sys->ppBuffers[ i ] );
aout_BufferFree( p_buffer );
}
if( !pPlayer->QueueBuffer( (s16 *)p_sys->ppBuffers[ i ],
p_sys->nBufferSize / 2 ) )
{
msg_Err( p_aout, "QueueBuffer failed" );
}
i = (i + 1) % p_sys->nBuffers;
#undef i
}
vlc_restorecancel (canc);
return NULL;
}
/*****************************************************************************
* Play: pretend to play a sound
*****************************************************************************/
static void Play( aout_instance_t * p_aout )
{
aout_buffer_t * p_buffer;
p_buffer = aout_FifoPop( p_aout, &p_aout->output.fifo );
if( fwrite( p_buffer->p_buffer, p_buffer->i_buffer, 1,
p_aout->output.p_sys->p_file ) != 1 )
{
msg_Err( p_aout, "write error (%m)" );
}
if( p_aout->output.p_sys->b_add_wav_header )
{
/* Update Wave Header */
p_aout->output.p_sys->waveh.DataLength += p_buffer->i_buffer;
}
aout_BufferFree( p_buffer );
}
void AudioQueueCallback(void * inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
aout_instance_t * p_aout = (aout_instance_t *)inUserData;
aout_buffer_t * p_buffer = NULL;
if (p_aout) {
vlc_mutex_lock( &p_aout->output_fifo_lock );
p_buffer = aout_FifoPop( p_aout, &p_aout->output.fifo );
vlc_mutex_unlock( &p_aout->output_fifo_lock );
}
if ( p_buffer != NULL ) {
vlc_memcpy( inBuffer->mAudioData, p_buffer->p_buffer, p_buffer->i_buffer );
inBuffer->mAudioDataByteSize = p_buffer->i_buffer;
aout_BufferFree( p_buffer );
} else {
vlc_memset( inBuffer->mAudioData, 0, inBuffer->mAudioDataBytesCapacity );
inBuffer->mAudioDataByteSize = inBuffer->mAudioDataBytesCapacity;
}
AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
}
/*****************************************************************************
* SDLCallback: what to do once SDL has played sound samples
*****************************************************************************/
static void SDLCallback( void * _p_aout, uint8_t * p_stream, int i_len )
{
aout_instance_t * p_aout = (aout_instance_t *)_p_aout;
aout_buffer_t * p_buffer;
/* SDL is unable to call us at regular times, or tell us its current
* hardware latency, or the buffer state. So we just pop data and throw
* it at SDL's face. Nah. */
vlc_mutex_lock( &p_aout->output_fifo_lock );
p_buffer = aout_FifoPop( p_aout, &p_aout->output.fifo );
vlc_mutex_unlock( &p_aout->output_fifo_lock );
if ( p_buffer != NULL )
{
vlc_memcpy( p_stream, p_buffer->p_buffer, i_len );
aout_BufferFree( p_buffer );
}
else
{
vlc_memset( p_stream, 0, i_len );
}
}
/*****************************************************************************
* aout_OutputNextBuffer : give the audio output plug-in the right buffer
*****************************************************************************
* If b_can_sleek is 1, the aout core functions won't try to resample
* new buffers to catch up - that is we suppose that the output plug-in can
* compensate it by itself. S/PDIF outputs should always set b_can_sleek = 1.
* This function is entered with no lock at all :-).
*****************************************************************************/
aout_buffer_t * aout_OutputNextBuffer( aout_instance_t * p_aout,
mtime_t start_date,
bool b_can_sleek )
{
aout_fifo_t *p_fifo = &p_aout->output.fifo;
aout_buffer_t * p_buffer;
mtime_t now = mdate();
aout_lock( p_aout );
/* Drop the audio sample if the audio output is really late.
* In the case of b_can_sleek, we don't use a resampler so we need to be
* a lot more severe. */
while( ((p_buffer = p_fifo->p_first) != NULL)
&& p_buffer->i_pts < (b_can_sleek ? start_date : now) - AOUT_MAX_PTS_DELAY )
{
msg_Dbg( p_aout, "audio output is too slow (%"PRId64"), "
"trashing %"PRId64"us", now - p_buffer->i_pts,
p_buffer->i_length );
aout_BufferFree( aout_FifoPop( p_fifo ) );
}
if( p_buffer == NULL )
{
#if 0 /* This is bad because the audio output might just be trying to fill
* in its internal buffers. And anyway, it's up to the audio output
* to deal with this kind of starvation. */
/* Set date to 0, to allow the mixer to send a new buffer ASAP */
aout_FifoSet( &p_aout->output.fifo, 0 );
if ( !p_aout->output.b_starving )
msg_Dbg( p_aout,
"audio output is starving (no input), playing silence" );
p_aout->output.b_starving = true;
#endif
goto out;
}
/* This routine will be called by the PortAudio engine when audio is needed.
* It may called at interrupt level on some machines so don't do anything
* that could mess up the system like calling malloc() or free().
*/
static int paCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo *paDate,
PaStreamCallbackFlags statusFlags, void *p_cookie )
{
VLC_UNUSED( inputBuffer ); VLC_UNUSED( statusFlags );
struct aout_sys_t *p_sys = (struct aout_sys_t*) p_cookie;
audio_output_t *p_aout = p_sys->p_aout;
aout_buffer_t *p_buffer;
mtime_t out_date;
out_date = mdate() + (mtime_t) ( 1000000 *
( paDate->outputBufferDacTime - paDate->currentTime ) );
p_buffer = aout_PacketNext( p_aout, out_date );
if ( p_buffer != NULL )
{
if( p_sys->b_chan_reorder )
{
/* Do the channel reordering here */
aout_ChannelReorder( p_buffer->p_buffer, p_buffer->i_buffer,
p_sys->i_channels, p_sys->pi_chan_table,
p_sys->i_bits_per_sample );
}
vlc_memcpy( outputBuffer, p_buffer->p_buffer,
framesPerBuffer * p_sys->i_sample_size );
aout_BufferFree( p_buffer );
}
else
/* Audio output buffer shortage -> stop the fill process and wait */
{
vlc_memset( outputBuffer, 0, framesPerBuffer * p_sys->i_sample_size );
}
return 0;
}
/*****************************************************************************
* aout_FiltersPlay: play a buffer
*****************************************************************************/
void aout_FiltersPlay( aout_instance_t * p_aout,
aout_filter_t ** pp_filters,
int i_nb_filters, aout_buffer_t ** pp_input_buffer )
{
int i;
for ( i = 0; i < i_nb_filters; i++ )
{
aout_filter_t * p_filter = pp_filters[i];
aout_buffer_t * p_output_buffer;
/* Resamplers can produce slightly more samples than (i_in_nb *
* p_filter->output.i_rate / p_filter->input.i_rate) so we need
* slightly bigger buffers. */
aout_BufferAlloc( &p_filter->output_alloc,
((mtime_t)(*pp_input_buffer)->i_nb_samples + 2)
* 1000000 / p_filter->input.i_rate,
*pp_input_buffer, p_output_buffer );
if ( p_output_buffer == NULL )
{
msg_Err( p_aout, "out of memory" );
return;
}
/* Please note that p_output_buffer->i_nb_samples & i_nb_bytes
* shall be set by the filter plug-in. */
p_filter->pf_do_work( p_aout, p_filter, *pp_input_buffer,
p_output_buffer );
if ( !p_filter->b_in_place )
{
aout_BufferFree( *pp_input_buffer );
*pp_input_buffer = p_output_buffer;
}
}
}
/*****************************************************************************
* Play: pretend to play a sound
*****************************************************************************/
static void Play( aout_instance_t * p_aout )
{
aout_buffer_t * p_buffer = aout_FifoPop( &p_aout->output.fifo );
aout_BufferFree( p_buffer );
}
/*****************************************************************************
* DoWork: mix a new output buffer
*****************************************************************************
* Terminology : in this function a word designates a single float32, eg.
* a stereo sample is consituted of two words.
*****************************************************************************/
static void DoWork( aout_instance_t * p_aout, aout_buffer_t * p_buffer )
{
const int i_nb_inputs = p_aout->i_nb_inputs;
const float f_multiplier_global = p_aout->mixer.f_multiplier;
const int i_nb_channels = aout_FormatNbChannels( &p_aout->mixer.mixer );
int i_input;
for ( i_input = 0; i_input < i_nb_inputs; i_input++ )
{
int i_nb_words = p_buffer->i_nb_samples * i_nb_channels;
aout_input_t * p_input = p_aout->pp_inputs[i_input];
float f_multiplier = f_multiplier_global * p_input->f_multiplier;
float * p_out = (float *)p_buffer->p_buffer;
float * p_in = (float *)p_input->p_first_byte_to_mix;
if ( p_input->b_error ) continue;
for ( ; ; )
{
ptrdiff_t i_available_words = (
(float *)p_input->fifo.p_first->p_buffer - p_in)
+ p_input->fifo.p_first->i_nb_samples
* i_nb_channels;
if ( i_available_words < i_nb_words )
{
aout_buffer_t * p_old_buffer;
if ( i_available_words > 0 )
{
if ( !i_input )
{
ScaleWords( p_out, p_in, i_available_words,
i_nb_inputs, f_multiplier );
}
else
{
MeanWords( p_out, p_in, i_available_words,
i_nb_inputs, f_multiplier );
}
}
i_nb_words -= i_available_words;
p_out += i_available_words;
/* Next buffer */
p_old_buffer = aout_FifoPop( p_aout, &p_input->fifo );
aout_BufferFree( p_old_buffer );
if ( p_input->fifo.p_first == NULL )
{
msg_Err( p_aout, "internal amix error" );
return;
}
p_in = (float *)p_input->fifo.p_first->p_buffer;
}
else
{
if ( i_nb_words > 0 )
{
if ( !i_input )
{
ScaleWords( p_out, p_in, i_nb_words, i_nb_inputs,
f_multiplier );
}
else
{
MeanWords( p_out, p_in, i_nb_words, i_nb_inputs,
f_multiplier );
}
}
p_input->p_first_byte_to_mix = (void *)(p_in
+ i_nb_words);
break;
}
}
}
}
/*****************************************************************************
* MixBuffer: try to prepare one output buffer
*****************************************************************************
* Please note that you must hold the mixer lock.
*****************************************************************************/
static int MixBuffer( aout_instance_t * p_aout )
{
int i, i_first_input = 0;
aout_buffer_t * p_output_buffer;
mtime_t start_date, end_date;
audio_date_t exact_start_date;
if ( p_aout->mixer.b_error )
{
/* Free all incoming buffers. */
vlc_mutex_lock( &p_aout->input_fifos_lock );
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
aout_input_t * p_input = p_aout->pp_inputs[i];
aout_buffer_t * p_buffer = p_input->fifo.p_first;
if ( p_input->b_error ) continue;
while ( p_buffer != NULL )
{
aout_buffer_t * p_next = p_buffer->p_next;
aout_BufferFree( p_buffer );
p_buffer = p_next;
}
}
vlc_mutex_unlock( &p_aout->input_fifos_lock );
return -1;
}
vlc_mutex_lock( &p_aout->output_fifo_lock );
vlc_mutex_lock( &p_aout->input_fifos_lock );
/* Retrieve the date of the next buffer. */
memcpy( &exact_start_date, &p_aout->output.fifo.end_date,
sizeof(audio_date_t) );
start_date = aout_DateGet( &exact_start_date );
if ( start_date != 0 && start_date < mdate() )
{
/* The output is _very_ late. This can only happen if the user
* pauses the stream (or if the decoder is buggy, which cannot
* happen :). */
msg_Warn( p_aout, "output PTS is out of range ("I64Fd"), clearing out",
mdate() - start_date );
aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
aout_DateSet( &exact_start_date, 0 );
start_date = 0;
}
vlc_mutex_unlock( &p_aout->output_fifo_lock );
/* See if we have enough data to prepare a new buffer for the audio
* output. First : start date. */
if ( !start_date )
{
/* Find the latest start date available. */
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
aout_input_t * p_input = p_aout->pp_inputs[i];
aout_fifo_t * p_fifo = &p_input->fifo;
aout_buffer_t * p_buffer;
if ( p_input->b_error ) continue;
p_buffer = p_fifo->p_first;
while ( p_buffer != NULL && p_buffer->start_date < mdate() )
{
msg_Warn( p_aout, "input PTS is out of range ("I64Fd"), "
"trashing", mdate() - p_buffer->start_date );
p_buffer = aout_FifoPop( p_aout, p_fifo );
aout_BufferFree( p_buffer );
p_buffer = p_fifo->p_first;
p_input->p_first_byte_to_mix = NULL;
}
if ( p_buffer == NULL )
{
break;
}
if ( !start_date || start_date < p_buffer->start_date )
{
aout_DateSet( &exact_start_date, p_buffer->start_date );
start_date = p_buffer->start_date;
}
}
if ( i < p_aout->i_nb_inputs )
{
/* Interrupted before the end... We can't run. */
vlc_mutex_unlock( &p_aout->input_fifos_lock );
return -1;
}
}
aout_DateIncrement( &exact_start_date, p_aout->output.i_nb_samples );
end_date = aout_DateGet( &exact_start_date );
/* Check that start_date and end_date are available for all input
* streams. */
for ( i = 0; i < p_aout->i_nb_inputs; i++ )
{
aout_input_t * p_input = p_aout->pp_inputs[i];
aout_fifo_t * p_fifo = &p_input->fifo;
aout_buffer_t * p_buffer;
mtime_t prev_date;
vlc_bool_t b_drop_buffers;
if ( p_input->b_error )
{
if ( i_first_input == i ) i_first_input++;
continue;
}
p_buffer = p_fifo->p_first;
if ( p_buffer == NULL )
{
break;
}
/* Check for the continuity of start_date */
while ( p_buffer != NULL && p_buffer->end_date < start_date - 1 )
{
/* We authorize a +-1 because rounding errors get compensated
* regularly. */
aout_buffer_t * p_next = p_buffer->p_next;
msg_Warn( p_aout, "the mixer got a packet in the past ("I64Fd")",
start_date - p_buffer->end_date );
aout_BufferFree( p_buffer );
p_fifo->p_first = p_buffer = p_next;
p_input->p_first_byte_to_mix = NULL;
}
if ( p_buffer == NULL )
{
p_fifo->pp_last = &p_fifo->p_first;
break;
}
/* Check that we have enough samples. */
for ( ; ; )
{
p_buffer = p_fifo->p_first;
if ( p_buffer == NULL ) break;
if ( p_buffer->end_date >= end_date ) break;
/* Check that all buffers are contiguous. */
prev_date = p_fifo->p_first->end_date;
p_buffer = p_buffer->p_next;
b_drop_buffers = 0;
for ( ; p_buffer != NULL; p_buffer = p_buffer->p_next )
{
if ( prev_date != p_buffer->start_date )
{
msg_Warn( p_aout,
"buffer hole, dropping packets ("I64Fd")",
p_buffer->start_date - prev_date );
b_drop_buffers = 1;
break;
}
if ( p_buffer->end_date >= end_date ) break;
prev_date = p_buffer->end_date;
}
if ( b_drop_buffers )
{
aout_buffer_t * p_deleted = p_fifo->p_first;
while ( p_deleted != NULL && p_deleted != p_buffer )
{
aout_buffer_t * p_next = p_deleted->p_next;
aout_BufferFree( p_deleted );
p_deleted = p_next;
}
p_fifo->p_first = p_deleted; /* == p_buffer */
}
else break;
}
if ( p_buffer == NULL ) break;
p_buffer = p_fifo->p_first;
if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) )
{
/* Additionally check that p_first_byte_to_mix is well
* located. */
mtime_t i_nb_bytes = (start_date - p_buffer->start_date)
* p_aout->mixer.mixer.i_bytes_per_frame
* p_aout->mixer.mixer.i_rate
/ p_aout->mixer.mixer.i_frame_length
/ 1000000;
ptrdiff_t mixer_nb_bytes;
if ( p_input->p_first_byte_to_mix == NULL )
{
p_input->p_first_byte_to_mix = p_buffer->p_buffer;
}
mixer_nb_bytes = p_input->p_first_byte_to_mix - p_buffer->p_buffer;
if ( !((i_nb_bytes + p_aout->mixer.mixer.i_bytes_per_frame
> mixer_nb_bytes) &&
(i_nb_bytes < p_aout->mixer.mixer.i_bytes_per_frame
+ mixer_nb_bytes)) )
{
msg_Warn( p_aout, "mixer start isn't output start ("I64Fd")",
i_nb_bytes - mixer_nb_bytes );
/* Round to the nearest multiple */
i_nb_bytes /= p_aout->mixer.mixer.i_bytes_per_frame;
i_nb_bytes *= p_aout->mixer.mixer.i_bytes_per_frame;
if( i_nb_bytes < 0 )
{
/* Is it really the best way to do it ? */
aout_FifoSet( p_aout, &p_aout->output.fifo, 0 );
aout_DateSet( &exact_start_date, 0 );
break;
}
p_input->p_first_byte_to_mix = p_buffer->p_buffer + i_nb_bytes;
}
}
}
if ( i < p_aout->i_nb_inputs || i_first_input == p_aout->i_nb_inputs )
{
/* Interrupted before the end... We can't run. */
vlc_mutex_unlock( &p_aout->input_fifos_lock );
return -1;
}
/* Run the mixer. */
aout_BufferAlloc( &p_aout->mixer.output_alloc,
((uint64_t)p_aout->output.i_nb_samples * 1000000)
/ p_aout->output.output.i_rate,
/* This is a bit kludgy, but is actually only used
* for the S/PDIF dummy mixer : */
p_aout->pp_inputs[i_first_input]->fifo.p_first,
p_output_buffer );
if ( p_output_buffer == NULL )
{
msg_Err( p_aout, "out of memory" );
vlc_mutex_unlock( &p_aout->input_fifos_lock );
return -1;
}
/* This is again a bit kludgy - for the S/PDIF mixer. */
if ( p_aout->mixer.output_alloc.i_alloc_type != AOUT_ALLOC_NONE )
{
p_output_buffer->i_nb_samples = p_aout->output.i_nb_samples;
p_output_buffer->i_nb_bytes = p_aout->output.i_nb_samples
* p_aout->mixer.mixer.i_bytes_per_frame
/ p_aout->mixer.mixer.i_frame_length;
}
p_output_buffer->start_date = start_date;
p_output_buffer->end_date = end_date;
p_aout->mixer.pf_do_work( p_aout, p_output_buffer );
vlc_mutex_unlock( &p_aout->input_fifos_lock );
aout_OutputPlay( p_aout, p_output_buffer );
return 0;
}
/*****************************************************************************
* QNXaoutThread: asynchronous thread used to DMA the data to the device
*****************************************************************************/
static int QNXaoutThread( aout_instance_t * p_aout )
{
struct aout_sys_t * p_sys = p_aout->output.p_sys;
while ( !p_aout->b_die )
{
aout_buffer_t * p_buffer;
int i_tmp, i_size;
byte_t * p_bytes;
if ( p_aout->output.output.i_format != VLC_FOURCC('s','p','d','i') )
{
mtime_t next_date = 0;
/* Get the presentation date of the next write() operation. It
* is equal to the current date + duration of buffered samples.
* Order is important here, since GetBufInfo is believed to take
* more time than mdate(). */
next_date = (mtime_t)GetBufInfo( p_aout ) * 1000000
/ p_aout->output.output.i_bytes_per_frame
/ p_aout->output.output.i_rate
* p_aout->output.output.i_frame_length;
next_date += mdate();
p_buffer = aout_OutputNextBuffer( p_aout, next_date, VLC_FALSE );
}
else
{
p_buffer = aout_OutputNextBuffer( p_aout, 0, VLC_TRUE );
}
if ( p_buffer != NULL )
{
p_bytes = p_buffer->p_buffer;
i_size = p_buffer->i_nb_bytes;
}
else
{
i_size = DEFAULT_FRAME_SIZE / p_aout->output.output.i_frame_length
* p_aout->output.output.i_bytes_per_frame;
p_bytes = p_aout->output.p_sys->p_silent_buffer;
memset( p_bytes, 0, i_size );
}
i_tmp = snd_pcm_plugin_write( p_aout->output.p_sys->p_pcm_handle,
(void *) p_bytes,
(size_t) i_size );
if( i_tmp < 0 )
{
msg_Err( p_aout, "write failed (%s)", strerror(errno) );
}
if ( p_buffer != NULL )
{
aout_BufferFree( p_buffer );
}
}
return 0;
}
static void inputDrop( aout_input_t *p_input, aout_buffer_t *p_buffer )
{
aout_BufferFree( p_buffer );
p_input->i_buffer_lost++;
}
