/*****************************************************************************
* 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;
}
/*****************************************************************************
* 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;
}
//#define AOUT_PROCESS_BEFORE_CHEKS
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
aout_buffer_t * p_buffer, int i_input_rate )
{
mtime_t start_date;
AOUT_ASSERT_INPUT_LOCKED;
if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
{
inputDrop( p_input, p_buffer );
return 0;
}
#ifdef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
if( !p_buffer )
return 0;
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
const mtime_t i_date = p_buffer->i_pts;
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
if( !p_buffer )
return 0;
if( p_buffer->i_nb_samples <= 0 )
{
block_Release( p_buffer );
return 0;
}
#endif
/* Handle input rate change, but keep drift correction */
if( i_input_rate != p_input->i_last_input_rate )
{
unsigned int * const pi_rate = &p_input->p_playback_rate_filter->fmt_in.audio.i_rate;
#define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) )
const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate);
*pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate);
#undef F
p_input->i_last_input_rate = i_input_rate;
}
/* We don't care if someone changes the start date behind our back after
* this. We'll deal with that when pushing the buffer, and compensate
* with the next incoming buffer. */
aout_lock_input_fifos( p_aout );
start_date = aout_FifoNextStart( p_aout, &p_input->mixer.fifo );
aout_unlock_input_fifos( p_aout );
if ( start_date != 0 && start_date < mdate() )
{
/* The decoder 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, "computed PTS is out of range (%"PRId64"), "
"clearing out", mdate() - start_date );
aout_lock_input_fifos( p_aout );
aout_FifoSet( p_aout, &p_input->mixer.fifo, 0 );
p_input->mixer.begin = NULL;
aout_unlock_input_fifos( p_aout );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
start_date = 0;
}
if ( p_buffer->i_pts < mdate() + AOUT_MIN_PREPARE_TIME )
{
/* The decoder gives us f*cked up PTS. It's its business, but we
* can't present it anyway, so drop the buffer. */
msg_Warn( p_aout, "PTS is out of range (%"PRId64"), dropping buffer",
mdate() - p_buffer->i_pts );
inputDrop( p_input, p_buffer );
inputResamplingStop( p_input );
return 0;
}
/* If the audio drift is too big then it's not worth trying to resample
* the audio. */
mtime_t i_pts_tolerance = 3 * AOUT_PTS_TOLERANCE * i_input_rate / INPUT_RATE_DEFAULT;
if ( start_date != 0 &&
( start_date < p_buffer->i_pts - i_pts_tolerance ) )
{
msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
start_date - p_buffer->i_pts );
aout_lock_input_fifos( p_aout );
aout_FifoSet( p_aout, &p_input->mixer.fifo, 0 );
p_input->mixer.begin = NULL;
aout_unlock_input_fifos( p_aout );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
start_date = 0;
}
else if ( start_date != 0 &&
( start_date > p_buffer->i_pts + i_pts_tolerance) )
{
msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer",
start_date - p_buffer->i_pts );
inputDrop( p_input, p_buffer );
return 0;
}
if ( start_date == 0 ) start_date = p_buffer->i_pts;
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
aout_FiltersPlay( p_input->pp_filters, p_input->i_nb_filters, &p_buffer );
if( !p_buffer )
return 0;
#endif
/* Run the resampler if needed.
* We first need to calculate the output rate of this resampler. */
if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
( start_date < p_buffer->i_pts - AOUT_PTS_TOLERANCE
|| start_date > p_buffer->i_pts + AOUT_PTS_TOLERANCE ) &&
p_input->i_nb_resamplers > 0 )
{
/* Can happen in several circumstances :
* 1. A problem at the input (clock drift)
* 2. A small pause triggered by the user
* 3. Some delay in the output stage, causing a loss of lip
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
mtime_t drift = p_buffer->i_pts - start_date;
p_input->i_resamp_start_date = mdate();
p_input->i_resamp_start_drift = (int)drift;
if ( drift > 0 )
p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
else
p_input->i_resampling_type = AOUT_RESAMPLING_UP;
msg_Warn( p_aout, "buffer is %"PRId64" %s, triggering %ssampling",
drift > 0 ? drift : -drift,
drift > 0 ? "in advance" : "late",
drift > 0 ? "down" : "up");
}
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
{
/* Resampling has been triggered previously (because of dates
* mismatch). We want the resampling to happen progressively so
* it isn't too audible to the listener. */
if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
p_input->pp_resamplers[0]->fmt_in.audio.i_rate += 2; /* Hz */
}
else
{
p_input->pp_resamplers[0]->fmt_in.audio.i_rate -= 2; /* Hz */
}
/* Check if everything is back to normal, in which case we can stop the
* resampling */
unsigned int i_nominal_rate =
(p_input->pp_resamplers[0] == p_input->p_playback_rate_filter)
? INPUT_RATE_DEFAULT * p_input->input.i_rate / i_input_rate
: p_input->input.i_rate;
if( p_input->pp_resamplers[0]->fmt_in.audio.i_rate == i_nominal_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
msg_Warn( p_aout, "resampling stopped after %"PRIi64" usec "
"(drift: %"PRIi64")",
mdate() - p_input->i_resamp_start_date,
p_buffer->i_pts - start_date);
}
else if( abs( (int)(p_buffer->i_pts - start_date) ) <
abs( p_input->i_resamp_start_drift ) / 2 )
{
/* if we reduced the drift from half, then it is time to switch
* back the resampling direction. */
if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
else
p_input->i_resampling_type = AOUT_RESAMPLING_UP;
p_input->i_resamp_start_drift = 0;
}
else if( p_input->i_resamp_start_drift &&
( abs( (int)(p_buffer->i_pts - start_date) ) >
abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
{
/* If the drift is increasing and not decreasing, than something
* is bad. We'd better stop the resampling right now. */
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
}
}
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
aout_FiltersPlay( p_input->pp_resamplers, p_input->i_nb_resamplers,
&p_buffer );
}
if( !p_buffer )
return 0;
if( p_buffer->i_nb_samples <= 0 )
{
block_Release( p_buffer );
return 0;
}
#endif
/* Adding the start date will be managed by aout_FifoPush(). */
p_buffer->i_pts = start_date;
aout_lock_input_fifos( p_aout );
aout_FifoPush( p_aout, &p_input->mixer.fifo, p_buffer );
aout_unlock_input_fifos( p_aout );
return 0;
}
/*****************************************************************************
* aout_InputPlay : play a buffer
*****************************************************************************
* This function must be entered with the input lock.
*****************************************************************************/
int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
aout_buffer_t * p_buffer )
{
mtime_t start_date;
if( p_input->b_restart )
{
aout_fifo_t fifo, dummy_fifo;
byte_t *p_first_byte_to_mix;
vlc_mutex_lock( &p_aout->mixer_lock );
/* A little trick to avoid loosing our input fifo */
aout_FifoInit( p_aout, &dummy_fifo, p_aout->mixer.mixer.i_rate );
p_first_byte_to_mix = p_input->p_first_byte_to_mix;
fifo = p_input->fifo;
p_input->fifo = dummy_fifo;
aout_InputDelete( p_aout, p_input );
aout_InputNew( p_aout, p_input );
p_input->p_first_byte_to_mix = p_first_byte_to_mix;
p_input->fifo = fifo;
vlc_mutex_unlock( &p_aout->mixer_lock );
}
/* We don't care if someone changes the start date behind our back after
* this. We'll deal with that when pushing the buffer, and compensate
* with the next incoming buffer. */
vlc_mutex_lock( &p_aout->input_fifos_lock );
start_date = aout_FifoNextStart( p_aout, &p_input->fifo );
vlc_mutex_unlock( &p_aout->input_fifos_lock );
if ( start_date != 0 && start_date < mdate() )
{
/* The decoder 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, "computed PTS is out of range ("I64Fd"), "
"clearing out", mdate() - start_date );
vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoSet( p_aout, &p_input->fifo, 0 );
p_input->p_first_byte_to_mix = NULL;
vlc_mutex_unlock( &p_aout->input_fifos_lock );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
if ( p_input->i_nb_resamplers != 0 )
{
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
}
start_date = 0;
if( p_input->p_input_thread )
{
stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS, 1,
NULL );
}
}
if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
{
/* The decoder gives us f*cked up PTS. It's its business, but we
* can't present it anyway, so drop the buffer. */
msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer",
mdate() - p_buffer->start_date );
if( p_input->p_input_thread )
{
stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS,
1, NULL );
}
aout_BufferFree( p_buffer );
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
if ( p_input->i_nb_resamplers != 0 )
{
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
}
return 0;
}
/* If the audio drift is too big then it's not worth trying to resample
* the audio. */
if ( start_date != 0 &&
( start_date < p_buffer->start_date - 3 * AOUT_PTS_TOLERANCE ) )
{
msg_Warn( p_aout, "audio drift is too big ("I64Fd"), clearing out",
start_date - p_buffer->start_date );
vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoSet( p_aout, &p_input->fifo, 0 );
p_input->p_first_byte_to_mix = NULL;
vlc_mutex_unlock( &p_aout->input_fifos_lock );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
if ( p_input->i_nb_resamplers != 0 )
{
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
p_input->pp_resamplers[0]->b_continuity = VLC_FALSE;
}
start_date = 0;
}
else if ( start_date != 0 &&
( start_date > p_buffer->start_date + 3 * AOUT_PTS_TOLERANCE ) )
{
msg_Warn( p_aout, "audio drift is too big ("I64Fd"), dropping buffer",
start_date - p_buffer->start_date );
aout_BufferFree( p_buffer );
if( p_input->p_input_thread )
{
stats_UpdateInteger( p_input->p_input_thread, STATS_LOST_ABUFFERS,
1, NULL );
}
return 0;
}
if ( start_date == 0 ) start_date = p_buffer->start_date;
/* Run pre-filters. */
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
/* Run the resampler if needed.
* We first need to calculate the output rate of this resampler. */
if ( ( p_input->i_resampling_type == AOUT_RESAMPLING_NONE ) &&
( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE
|| start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) &&
p_input->i_nb_resamplers > 0 )
{
/* Can happen in several circumstances :
* 1. A problem at the input (clock drift)
* 2. A small pause triggered by the user
* 3. Some delay in the output stage, causing a loss of lip
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
mtime_t drift = p_buffer->start_date - start_date;
p_input->i_resamp_start_date = mdate();
p_input->i_resamp_start_drift = (int)drift;
if ( drift > 0 )
p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
else
p_input->i_resampling_type = AOUT_RESAMPLING_UP;
msg_Warn( p_aout, "buffer is "I64Fd" %s, triggering %ssampling",
drift > 0 ? drift : -drift,
drift > 0 ? "in advance" : "late",
drift > 0 ? "down" : "up");
}
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
{
/* Resampling has been triggered previously (because of dates
* mismatch). We want the resampling to happen progressively so
* it isn't too audible to the listener. */
if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */
}
else
{
p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */
}
/* Check if everything is back to normal, in which case we can stop the
* resampling */
if( p_input->pp_resamplers[0]->input.i_rate ==
p_input->input.i_rate )
{
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
msg_Warn( p_aout, "resampling stopped after "I64Fi" usec "
"(drift: "I64Fi")",
mdate() - p_input->i_resamp_start_date,
p_buffer->start_date - start_date);
}
else if( abs( (int)(p_buffer->start_date - start_date) ) <
abs( p_input->i_resamp_start_drift ) / 2 )
{
/* if we reduced the drift from half, then it is time to switch
* back the resampling direction. */
if( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
p_input->i_resampling_type = AOUT_RESAMPLING_DOWN;
else
p_input->i_resampling_type = AOUT_RESAMPLING_UP;
p_input->i_resamp_start_drift = 0;
}
else if( p_input->i_resamp_start_drift &&
( abs( (int)(p_buffer->start_date - start_date) ) >
abs( p_input->i_resamp_start_drift ) * 3 / 2 ) )
{
/* If the drift is increasing and not decreasing, than something
* is bad. We'd better stop the resampling right now. */
msg_Warn( p_aout, "timing screwed, stopping resampling" );
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
}
}
/* Adding the start date will be managed by aout_FifoPush(). */
p_buffer->end_date = start_date +
(p_buffer->end_date - p_buffer->start_date);
p_buffer->start_date = start_date;
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
vlc_mutex_lock( &p_aout->input_fifos_lock );
aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
vlc_mutex_unlock( &p_aout->input_fifos_lock );
return 0;
}
