void
ca_Play(audio_output_t * p_aout, block_t * p_block)
{
struct aout_sys_common *p_sys = (struct aout_sys_common *) p_aout->sys;
/* Do the channel reordering */
if (p_sys->chans_to_reorder)
aout_ChannelReorder(p_block->p_buffer, p_block->i_buffer,
p_sys->chans_to_reorder, p_sys->chan_table,
VLC_CODEC_FL32);
/* move data to buffer */
while (!TPCircularBufferProduceBytes(&p_sys->circular_buffer,
p_block->p_buffer, p_block->i_buffer))
{
if (atomic_load_explicit(&p_sys->b_paused, memory_order_relaxed))
{
msg_Warn(p_aout, "dropping block because the circular buffer is "
"full and paused");
break;
}
/* Try to play what we can */
int32_t i_avalaible_bytes;
TPCircularBufferHead(&p_sys->circular_buffer, &i_avalaible_bytes);
assert(i_avalaible_bytes >= 0);
if (unlikely((size_t) i_avalaible_bytes >= p_block->i_buffer))
continue;
bool ret =
TPCircularBufferProduceBytes(&p_sys->circular_buffer,
p_block->p_buffer, i_avalaible_bytes);
assert(ret == true);
p_block->p_buffer += i_avalaible_bytes;
p_block->i_buffer -= i_avalaible_bytes;
/* Wait for the render buffer to play the remaining data */
const mtime_t i_frame_us =
FramesToUs(p_sys, BytesToFrames(p_sys, p_block->i_buffer));
msleep(i_frame_us / 2);
}
unsigned i_underrun_size = atomic_exchange(&p_sys->i_underrun_size, 0);
if (i_underrun_size > 0)
msg_Warn(p_aout, "underrun of %u bytes", i_underrun_size);
block_Release(p_block);
}
/* 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;
}
/*****************************************************************************
* Demux: read packet and send them to decoders
*****************************************************************************
* Returns -1 in case of error, 0 in case of EOF, 1 otherwise
*****************************************************************************/
static int Demux( demux_t *p_demux )
{
demux_sys_t *p_sys = p_demux->p_sys;
block_t *p_block;
const int64_t i_pos = stream_Tell( p_demux->s );
if( p_sys->i_data_size > 0 &&
i_pos >= p_sys->i_data_pos + p_sys->i_data_size )
{
/* EOF */
return 0;
}
if( ( p_block = stream_Block( p_demux->s, p_sys->i_frame_size ) ) == NULL )
{
msg_Warn( p_demux, "cannot read data" );
return 0;
}
p_block->i_dts =
p_block->i_pts = VLC_TS_0 + date_Get( &p_sys->pts );
/* set PCR */
es_out_Control( p_demux->out, ES_OUT_SET_PCR, p_block->i_pts );
/* Do the channel reordering */
if( p_sys->i_chans_to_reorder )
aout_ChannelReorder( p_block->p_buffer, p_block->i_buffer,
p_sys->fmt.audio.i_channels,
p_sys->pi_chan_table, p_sys->fmt.i_codec );
es_out_Send( p_demux->out, p_sys->p_es, p_block );
date_Increment( &p_sys->pts, p_sys->i_frame_samples );
return 1;
}
/* 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;
}
/*****************************************************************************
* DecodeFrame: decodes an lpcm frame.
****************************************************************************
* Beware, this function must be fed with complete frames (PES packet).
*****************************************************************************/
static block_t *DecodeFrame( decoder_t *p_dec, block_t **pp_block )
{
decoder_sys_t *p_sys = p_dec->p_sys;
block_t *p_block;
unsigned int i_rate = 0, i_original_channels = 0, i_channels = 0, i_bits = 0;
int i_frame_length;
if( !pp_block || !*pp_block ) return NULL;
p_block = *pp_block;
*pp_block = NULL; /* So the packet doesn't get re-sent */
/* Date management */
if( p_block->i_pts > VLC_TS_INVALID &&
p_block->i_pts != date_Get( &p_sys->end_date ) )
{
date_Set( &p_sys->end_date, p_block->i_pts );
}
if( !date_Get( &p_sys->end_date ) )
{
/* We've just started the stream, wait for the first PTS. */
block_Release( p_block );
return NULL;
}
if( p_block->i_buffer <= p_sys->i_header_size )
{
msg_Err(p_dec, "frame is too short");
block_Release( p_block );
return NULL;
}
int i_ret;
unsigned i_channels_padding = 0;
unsigned i_padding = 0;
aob_group_t p_aob_group[2];
switch( p_sys->i_type )
{
case LPCM_VOB:
i_ret = VobHeader( &i_rate, &i_channels, &i_original_channels, &i_bits,
p_block->p_buffer );
break;
case LPCM_AOB:
i_ret = AobHeader( &i_rate, &i_channels, &i_original_channels, &i_bits, &i_padding,
p_aob_group,
p_block->p_buffer );
break;
case LPCM_BD:
i_ret = BdHeader( p_sys, &i_rate, &i_channels, &i_channels_padding, &i_original_channels, &i_bits,
p_block->p_buffer );
break;
default:
abort();
}
if( i_ret || p_block->i_buffer <= p_sys->i_header_size + i_padding )
{
msg_Warn( p_dec, "no frame sync or too small frame" );
block_Release( p_block );
return NULL;
}
/* Set output properties */
if( p_dec->fmt_out.audio.i_rate != i_rate )
{
date_Init( &p_sys->end_date, i_rate, 1 );
date_Set( &p_sys->end_date, p_block->i_pts );
}
p_dec->fmt_out.audio.i_rate = i_rate;
p_dec->fmt_out.audio.i_channels = i_channels;
p_dec->fmt_out.audio.i_original_channels = i_original_channels;
p_dec->fmt_out.audio.i_physical_channels = i_original_channels;
i_frame_length = (p_block->i_buffer - p_sys->i_header_size - i_padding) /
(i_channels + i_channels_padding) * 8 / i_bits;
if( p_sys->b_packetizer )
{
p_block->i_pts = p_block->i_dts = date_Get( &p_sys->end_date );
p_block->i_length =
date_Increment( &p_sys->end_date, i_frame_length ) -
p_block->i_pts;
/* Just pass on the incoming frame */
return p_block;
}
else
{
/* */
if( i_bits == 16 )
{
p_dec->fmt_out.i_codec = VLC_CODEC_S16N;
p_dec->fmt_out.audio.i_bitspersample = 16;
}
else
{
p_dec->fmt_out.i_codec = VLC_CODEC_S32N;
p_dec->fmt_out.audio.i_bitspersample = 32;
}
/* */
block_t *p_aout_buffer;
p_aout_buffer = decoder_NewAudioBuffer( p_dec, i_frame_length );
if( !p_aout_buffer )
return NULL;
p_aout_buffer->i_pts = date_Get( &p_sys->end_date );
p_aout_buffer->i_length =
date_Increment( &p_sys->end_date, i_frame_length )
- p_aout_buffer->i_pts;
p_block->p_buffer += p_sys->i_header_size + i_padding;
p_block->i_buffer -= p_sys->i_header_size + i_padding;
if( p_sys->i_chans_to_reorder )
{
aout_ChannelReorder( p_block->p_buffer, p_block->i_buffer,
p_sys->i_chans_to_reorder, p_sys->pi_chan_table,
p_dec->fmt_out.i_codec );
}
switch( p_sys->i_type )
{
case LPCM_VOB:
VobExtract( p_aout_buffer, p_block, i_bits );
break;
case LPCM_AOB:
AobExtract( p_aout_buffer, p_block, i_bits, p_aob_group );
break;
default:
assert(0);
case LPCM_BD:
BdExtract( p_aout_buffer, p_block, i_frame_length, i_channels, i_channels_padding, i_bits );
break;
}
block_Release( p_block );
return p_aout_buffer;
}
}
/**
* Fills in one of the DirectSound frame buffers.
*
* @return VLC_SUCCESS on success.
*/
static HRESULT FillBuffer( vlc_object_t *obj, aout_stream_sys_t *p_sys,
block_t *p_buffer )
{
size_t towrite = (p_buffer != NULL) ? p_buffer->i_buffer : DS_BUF_SIZE;
void *p_write_position, *p_wrap_around;
unsigned long l_bytes1, l_bytes2;
HRESULT dsresult;
vlc_mutex_lock( &p_sys->lock );
/* Before copying anything, we have to lock the buffer */
dsresult = IDirectSoundBuffer_Lock(
p_sys->p_dsbuffer, /* DS buffer */
p_sys->i_write, /* Start offset */
towrite, /* Number of bytes */
&p_write_position, /* Address of lock start */
&l_bytes1, /* Count of bytes locked before wrap around */
&p_wrap_around, /* Buffer address (if wrap around) */
&l_bytes2, /* Count of bytes after wrap around */
0 ); /* Flags: DSBLOCK_FROMWRITECURSOR is buggy */
if( dsresult == DSERR_BUFFERLOST )
{
IDirectSoundBuffer_Restore( p_sys->p_dsbuffer );
dsresult = IDirectSoundBuffer_Lock(
p_sys->p_dsbuffer,
p_sys->i_write,
towrite,
&p_write_position,
&l_bytes1,
&p_wrap_around,
&l_bytes2,
0 );
}
if( dsresult != DS_OK )
{
msg_Warn( obj, "cannot lock buffer" );
if( p_buffer != NULL )
block_Release( p_buffer );
vlc_mutex_unlock( &p_sys->lock );
return dsresult;
}
if( p_buffer == NULL )
{
memset( p_write_position, 0, l_bytes1 );
memset( p_wrap_around, 0, l_bytes2 );
}
else
{
if( p_sys->chans_to_reorder ) /* Do the channel reordering here */
aout_ChannelReorder( p_buffer->p_buffer, p_buffer->i_buffer,
p_sys->chans_to_reorder, p_sys->chan_table,
p_sys->format );
memcpy( p_write_position, p_buffer->p_buffer, l_bytes1 );
if( p_wrap_around && l_bytes2 )
memcpy( p_wrap_around, p_buffer->p_buffer + l_bytes1, l_bytes2 );
if( unlikely( ( l_bytes1 + l_bytes2 ) < p_buffer->i_buffer ) )
msg_Err( obj, "Buffer overrun");
block_Release( p_buffer );
}
/* Now the data has been copied, unlock the buffer */
IDirectSoundBuffer_Unlock( p_sys->p_dsbuffer, p_write_position, l_bytes1,
p_wrap_around, l_bytes2 );
p_sys->i_write += towrite;
p_sys->i_write %= DS_BUF_SIZE;
p_sys->i_data += towrite;
vlc_mutex_unlock( &p_sys->lock );
return DS_OK;
}
/*****************************************************************************
* WaveOutThread: this thread will capture play notification events.
*****************************************************************************
* We use this thread to feed new audio samples to the sound card because
* we are not authorized to use waveOutWrite() directly in the waveout
* callback.
*****************************************************************************/
static void WaveOutThread( notification_thread_t *p_notif )
{
aout_instance_t *p_aout = p_notif->p_aout;
aout_sys_t *p_sys = p_aout->output.p_sys;
aout_buffer_t *p_buffer = NULL;
WAVEHDR *p_waveheader = p_sys->waveheader;
int i, i_queued_frames;
vlc_bool_t b_sleek;
/* We don't want any resampling when using S/PDIF */
b_sleek = p_aout->output.output.i_format == VLC_FOURCC('s','p','d','i');
while( 1 )
{
WaitForSingleObject( p_sys->event, INFINITE );
/* Cleanup and find out the current latency */
i_queued_frames = 0;
for( i = 0; i < FRAMES_NUM; i++ )
{
if( (p_waveheader[i].dwFlags & WHDR_DONE) &&
p_waveheader[i].dwUser )
{
/* Unprepare and free the buffers which has just been played */
waveOutUnprepareHeader( p_sys->h_waveout, &p_waveheader[i],
sizeof(WAVEHDR) );
if( p_waveheader[i].dwUser != 1 )
aout_BufferFree( (aout_buffer_t *)p_waveheader[i].dwUser );
p_waveheader[i].dwUser = 0;
}
/* Check if frame buf is available */
if( !(p_waveheader[i].dwFlags & WHDR_DONE) )
{
i_queued_frames++;
}
}
if( p_aout->b_die ) return;
/* Try to fill in as many frame buffers as possible */
for( i = 0; i < FRAMES_NUM; i++ )
{
/* Check if frame buf is available */
if( p_waveheader[i].dwFlags & WHDR_DONE )
{
/* Take into account the latency */
p_buffer = aout_OutputNextBuffer( p_aout,
mdate() + 1000000 * i_queued_frames /
p_aout->output.output.i_rate * p_aout->output.i_nb_samples,
b_sleek );
if( !p_buffer && i_queued_frames )
{
/* We aren't late so no need to play a blank sample */
break;
}
/* Do the channel reordering */
if( p_buffer && p_sys->b_chan_reorder )
{
aout_ChannelReorder( p_buffer->p_buffer,
p_buffer->i_nb_bytes,
p_sys->waveformat.Format.nChannels,
p_sys->pi_chan_table,
p_sys->waveformat.Format.wBitsPerSample );
}
PlayWaveOut( p_aout, p_sys->h_waveout,
&p_waveheader[i], p_buffer );
i_queued_frames++;
}
}
}
}
