/*****************************************************************************
* 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 Play (aout_instance_t *aout)
{
aout_sys_t *sys = aout->output.p_sys;
block_t *block;
while ((block = aout_FifoPop(&aout->output.fifo)) != NULL)
{
sys->play (sys->opaque, block->p_buffer, block->i_nb_samples,
block->i_pts);
block_Release (block);
}
}
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;
}
/*****************************************************************************
* 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;
}
/*****************************************************************************
* 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;
}
/*****************************************************************************
* 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;
}
}
}
}
/**
* Dequeues the next audio packet (a.k.a. audio fragment).
* The audio output plugin must first call aout_PacketPlay() to queue the
* decoded audio samples. Typically, audio_output_t.pf_play is set to, or calls
* aout_PacketPlay().
* @note This function is considered legacy. Please do not use this function in
* new audio output plugins.
* @param p_aout audio output instance
* @param start_date expected PTS of the audio packet
*/
block_t *aout_PacketNext (audio_output_t *p_aout, mtime_t start_date)
{
aout_packet_t *p = aout_packet (p_aout);
aout_fifo_t *p_fifo = &p->fifo;
block_t *p_buffer;
const bool b_can_sleek = !AOUT_FMT_LINEAR(&p_aout->format);
const mtime_t now = mdate ();
const mtime_t threshold =
(b_can_sleek ? start_date : now) - AOUT_MAX_PTS_DELAY;
vlc_mutex_lock( &p->lock );
if( p->pause_date != VLC_TS_INVALID )
goto out; /* paused: do not dequeue buffers */
for (;;)
{
p_buffer = p_fifo->p_first;
if (p_buffer == NULL)
goto out; /* nothing to play */
if (p_buffer->i_pts >= threshold)
break;
/* 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. */
msg_Dbg (p_aout, "audio output is too slow (%"PRId64" us): "
" trashing %"PRId64" us", threshold - p_buffer->i_pts,
p_buffer->i_length);
block_Release (aout_FifoPop (p_fifo));
}
mtime_t delta = start_date - p_buffer->i_pts;
/* This assumes that all buffers have the same duration. This is true
* since aout_PacketPlay() (aout_OutputSlice()) is used. */
if (0 >= delta + p_buffer->i_length)
{
if (!p->starving)
{
msg_Dbg (p_aout, "audio output is starving (%"PRId64"), "
"playing silence", delta);
p->starving = true;
}
goto out; /* nothing to play _yet_ */
}
p->starving = false;
p_buffer = aout_FifoPop( p_fifo );
if (!b_can_sleek
&& (delta < -AOUT_MAX_PTS_ADVANCE || AOUT_MAX_PTS_DELAY < delta))
{
msg_Warn (p_aout, "audio output out of sync, "
"adjusting dates (%"PRId64" us)", delta);
aout_FifoMoveDates (&p->partial, delta);
aout_FifoMoveDates (p_fifo, delta);
p->time_report = delta;
}
vlc_mutex_unlock( &p->lock );
return p_buffer;
out:
vlc_mutex_unlock( &p->lock );
return NULL;
}
/**
* Rearranges audio blocks in correct number of samples.
* @note (FIXME) This is left here for historical reasons. It belongs in the
* output code. Besides, this operation should be avoided if possible.
*/
static block_t *aout_OutputSlice (audio_output_t *p_aout)
{
aout_packet_t *p = aout_packet (p_aout);
aout_fifo_t *p_fifo = &p->partial;
const unsigned samples = p->samples;
assert( samples > 0 );
vlc_assert_locked( &p->lock );
/* Retrieve the date of the next buffer. */
date_t exact_start_date = p->fifo.end_date;
mtime_t start_date = date_Get( &exact_start_date );
/* See if we have enough data to prepare a new buffer for the audio output. */
aout_buffer_t *p_buffer = p_fifo->p_first;
if( p_buffer == NULL )
return NULL;
/* Find the earliest start date available. */
if ( start_date == VLC_TS_INVALID )
{
start_date = p_buffer->i_pts;
date_Set( &exact_start_date, start_date );
}
/* Compute the end date for the new buffer. */
mtime_t end_date = date_Increment( &exact_start_date, samples );
/* Check that we have enough samples (TODO merge with next loop). */
for( unsigned available = 0; available < samples; )
{
p_buffer = p_buffer->p_next;
if( p_buffer == NULL )
return NULL;
available += p_buffer->i_nb_samples;
}
if( AOUT_FMT_LINEAR( &p_aout->format ) )
{
const unsigned framesize = p_aout->format.i_bytes_per_frame;
/* Build packet with adequate number of samples */
unsigned needed = samples * framesize;
p_buffer = block_Alloc( needed );
if( unlikely(p_buffer == NULL) )
/* XXX: should free input buffers */
return NULL;
p_buffer->i_nb_samples = samples;
for( uint8_t *p_out = p_buffer->p_buffer; needed > 0; )
{
aout_buffer_t *p_inbuf = p_fifo->p_first;
if( unlikely(p_inbuf == NULL) )
{
msg_Err( p_aout, "packetization error" );
vlc_memset( p_out, 0, needed );
break;
}
const uint8_t *p_in = p_inbuf->p_buffer;
size_t avail = p_inbuf->i_nb_samples * framesize;
if( avail > needed )
{
vlc_memcpy( p_out, p_in, needed );
p_fifo->p_first->p_buffer += needed;
p_fifo->p_first->i_buffer -= needed;
needed /= framesize;
p_fifo->p_first->i_nb_samples -= needed;
mtime_t t = needed * CLOCK_FREQ / p_aout->format.i_rate;
p_fifo->p_first->i_pts += t;
p_fifo->p_first->i_length -= t;
break;
}
vlc_memcpy( p_out, p_in, avail );
needed -= avail;
p_out += avail;
/* Next buffer */
aout_BufferFree( aout_FifoPop( p_fifo ) );
}
}
else
p_buffer = aout_FifoPop( p_fifo );
p_buffer->i_pts = start_date;
p_buffer->i_length = end_date - start_date;
return p_buffer;
}
/*****************************************************************************
* DoWork: mix a new output buffer
*****************************************************************************/
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];
int i_nb_channels = aout_FormatNbChannels( &p_aout->mixer.mixer );
int i_nb_bytes = p_buffer->i_nb_samples * sizeof(int32_t)
* i_nb_channels;
uint8_t * p_in;
uint8_t * p_out;
while ( p_input->b_error )
{
p_input = p_aout->pp_inputs[++i];
/* This can't crash because if no input has b_error == 0, the
* audio mixer cannot run and we can't be here. */
}
p_in = p_input->p_first_byte_to_mix;
p_out = p_buffer->p_buffer;
for ( ; ; )
{
ptrdiff_t i_available_bytes = (p_input->fifo.p_first->p_buffer
- p_in)
+ p_input->fifo.p_first->i_nb_samples
* sizeof(int32_t)
* i_nb_channels;
if ( i_available_bytes < i_nb_bytes )
{
aout_buffer_t * p_old_buffer;
vlc_memcpy( p_out, p_in, i_available_bytes );
i_nb_bytes -= i_available_bytes;
p_out += i_available_bytes;
/* 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 = p_input->fifo.p_first->p_buffer;
}
else
{
vlc_memcpy( p_out, p_in, i_nb_bytes );
p_input->p_first_byte_to_mix = p_in + i_nb_bytes;
break;
}
}
/* 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;
}
}
