/*****************************************************************************
* dec.c : audio output API towards decoders
*****************************************************************************
- * Copyright (C) 2002-2007 the VideoLAN team
+ * Copyright (C) 2002-2007 VLC authors and VideoLAN
* $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser General Public License for more details.
*
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
#include <assert.h>
#include <vlc_common.h>
-
#include <vlc_aout.h>
#include <vlc_input.h>
#include "aout_internal.h"
+#include "libvlc.h"
-#undef aout_DecNew
/**
* Creates an audio output
*/
-aout_input_t *aout_DecNew( aout_instance_t *p_aout,
- audio_sample_format_t *p_format,
- const audio_replay_gain_t *p_replay_gain,
- const aout_request_vout_t *p_request_vout )
+int aout_DecNew( audio_output_t *p_aout,
+ const audio_sample_format_t *p_format,
+ const audio_replay_gain_t *p_replay_gain,
+ const aout_request_vout_t *p_request_vout )
{
- aout_input_t * p_input;
-
/* Sanitize audio format */
- if( p_format->i_channels > 32 )
- {
- msg_Err( p_aout, "too many audio channels (%u)",
- p_format->i_channels );
- return NULL;
- }
- if( p_format->i_channels <= 0 )
- {
- msg_Err( p_aout, "no audio channels" );
- return NULL;
- }
if( p_format->i_channels != aout_FormatNbChannels( p_format ) )
{
msg_Err( p_aout, "incompatible audio channels count with layout mask" );
- return NULL;
+ return -1;
}
- if( p_format->i_rate > 192000 )
+ if( p_format->i_rate > 352800 )
{
msg_Err( p_aout, "excessive audio sample frequency (%u)",
p_format->i_rate );
- return NULL;
+ return -1;
}
if( p_format->i_rate < 4000 )
{
msg_Err( p_aout, "too low audio sample frequency (%u)",
p_format->i_rate );
- return NULL;
+ return -1;
}
- /* We can only be called by the decoder, so no need to lock
- * p_input->lock. */
- aout_lock_mixer( p_aout );
- assert( p_aout->i_nb_inputs == 0 );
-
- p_input = calloc( 1, sizeof(aout_input_t));
- if( !p_input )
- goto error;
+ aout_owner_t *owner = aout_owner(p_aout);
- vlc_mutex_init( &p_input->lock );
+ /* TODO: reduce lock scope depending on decoder's real need */
+ aout_OutputLock (p_aout);
- p_input->b_changed = false;
- p_input->b_error = true;
- p_input->b_paused = false;
- p_input->i_pause_date = 0;
+ var_Destroy( p_aout, "stereo-mode" );
- aout_FormatPrepare( p_format );
+ /* Create the audio output stream */
+ owner->volume = aout_volume_New (p_aout, p_replay_gain);
- memcpy( &p_input->input, p_format,
- sizeof(audio_sample_format_t) );
- if( p_replay_gain )
- p_input->replay_gain = *p_replay_gain;
+ atomic_store (&owner->restart, 0);
+ owner->input_format = *p_format;
+ owner->mixer_format = owner->input_format;
+ owner->request_vout = *p_request_vout;
- aout_lock_input_fifos( p_aout );
- p_aout->pp_inputs[p_aout->i_nb_inputs] = p_input;
- p_aout->i_nb_inputs++;
+ if (aout_OutputNew (p_aout, &owner->mixer_format))
+ goto error;
+ aout_volume_SetFormat (owner->volume, owner->mixer_format.i_format);
- if ( !p_aout->p_mixer )
+ /* Create the audio filtering "input" pipeline */
+ owner->filters = aout_FiltersNew (p_aout, p_format, &owner->mixer_format,
+ &owner->request_vout);
+ if (owner->filters == NULL)
{
- int i;
-
- var_Destroy( p_aout, "audio-device" );
- var_Destroy( p_aout, "audio-channels" );
+ aout_OutputDelete (p_aout);
+error:
+ aout_volume_Delete (owner->volume);
+ aout_OutputUnlock (p_aout);
+ return -1;
+ }
- /* Recreate the output using the new format. */
- if ( aout_OutputNew( p_aout, p_format ) < 0 )
- {
- for ( i = 0; i < p_aout->i_nb_inputs - 1; i++ )
- {
- aout_lock_input( p_aout, p_aout->pp_inputs[i] );
- aout_InputDelete( p_aout, p_aout->pp_inputs[i] );
- aout_unlock_input( p_aout, p_aout->pp_inputs[i] );
- }
- aout_unlock_input_fifos( p_aout );
- aout_unlock_mixer( p_aout );
- return p_input;
- }
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ owner->sync.discontinuity = true;
+ aout_OutputUnlock (p_aout);
- /* Create other input streams. */
- for ( i = 0; i < p_aout->i_nb_inputs - 1; i++ )
- {
- aout_input_t *p_input = p_aout->pp_inputs[i];
+ atomic_init (&owner->buffers_lost, 0);
+ return 0;
+}
- aout_lock_input( p_aout, p_input );
- aout_InputDelete( p_aout, p_input );
- aout_InputNew( p_aout, p_input, &p_input->request_vout );
- aout_unlock_input( p_aout, p_input );
- }
- }
- else
- {
- aout_MixerDelete( p_aout );
- }
+/**
+ * Stops all plugins involved in the audio output.
+ */
+void aout_DecDelete (audio_output_t *aout)
+{
+ aout_owner_t *owner = aout_owner (aout);
- if ( aout_MixerNew( p_aout ) == -1 )
+ aout_OutputLock (aout);
+ if (owner->mixer_format.i_format)
{
- aout_OutputDelete( p_aout );
- aout_unlock_input_fifos( p_aout );
- goto error;
+ aout_FiltersDelete (aout, owner->filters);
+ aout_OutputDelete (aout);
}
-
- aout_InputNew( p_aout, p_input, p_request_vout );
- aout_unlock_input_fifos( p_aout );
-
- aout_unlock_mixer( p_aout );
-
- return p_input;
-
-error:
- aout_unlock_mixer( p_aout );
- return NULL;
+ aout_volume_Delete (owner->volume);
+ aout_OutputUnlock (aout);
+ var_Destroy (aout, "stereo-mode");
}
-/*****************************************************************************
- * aout_DecDelete : delete a decoder
- *****************************************************************************/
-int aout_DecDelete( aout_instance_t * p_aout, aout_input_t * p_input )
+static int aout_CheckReady (audio_output_t *aout)
{
- int i_input;
-
- /* This function can only be called by the decoder itself, so no need
- * to lock p_input->lock. */
- aout_lock_mixer( p_aout );
+ aout_owner_t *owner = aout_owner (aout);
- for ( i_input = 0; i_input < p_aout->i_nb_inputs; i_input++ )
+ int restart = atomic_exchange (&owner->restart, 0);
+ if (unlikely(restart))
{
- if ( p_aout->pp_inputs[i_input] == p_input )
- {
- break;
+ if (owner->mixer_format.i_format)
+ aout_FiltersDelete (aout, owner->filters);
+
+ if (restart & AOUT_RESTART_OUTPUT)
+ { /* Reinitializes the output */
+ msg_Dbg (aout, "restarting output...");
+ if (owner->mixer_format.i_format)
+ aout_OutputDelete (aout);
+ owner->mixer_format = owner->input_format;
+ if (aout_OutputNew (aout, &owner->mixer_format))
+ owner->mixer_format.i_format = 0;
+ aout_volume_SetFormat (owner->volume,
+ owner->mixer_format.i_format);
}
- }
-
- if ( i_input == p_aout->i_nb_inputs )
- {
- msg_Err( p_aout, "cannot find an input to delete" );
- aout_unlock_mixer( p_aout );
- return -1;
- }
-
- /* Remove the input from the list. */
- p_aout->i_nb_inputs--;
- assert( p_aout->i_nb_inputs == 0 );
- aout_InputDelete( p_aout, p_input );
+ msg_Dbg (aout, "restarting filters...");
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
- vlc_mutex_destroy( &p_input->lock );
- free( p_input );
-
- if ( !p_aout->i_nb_inputs )
- {
- aout_OutputDelete( p_aout );
- aout_MixerDelete( p_aout );
- var_Destroy( p_aout, "audio-device" );
- var_Destroy( p_aout, "audio-channels" );
+ if (owner->mixer_format.i_format)
+ {
+ owner->filters = aout_FiltersNew (aout, &owner->input_format,
+ &owner->mixer_format,
+ &owner->request_vout);
+ if (owner->filters == NULL)
+ {
+ aout_OutputDelete (aout);
+ owner->mixer_format.i_format = 0;
+ }
+ }
+ /* TODO: This would be a good time to call clean up any video output
+ * left over by an audio visualization:
+ input_resource_TerminatVout(MAGIC HERE); */
}
-
- aout_unlock_mixer( p_aout );
-
- return 0;
+ return (owner->mixer_format.i_format) ? 0 : -1;
}
+/**
+ * Marks the audio output for restart, to update any parameter of the output
+ * plug-in (e.g. output device or channel mapping).
+ */
+void aout_RequestRestart (audio_output_t *aout, unsigned mode)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ atomic_fetch_or (&owner->restart, mode);
+ msg_Dbg (aout, "restart requested (%u)", mode);
+}
/*
* Buffer management
*/
-/*****************************************************************************
- * aout_DecNewBuffer : ask for a new empty buffer
- *****************************************************************************/
-aout_buffer_t * aout_DecNewBuffer( aout_input_t * p_input,
- size_t i_nb_samples )
+static void aout_StopResampling (audio_output_t *aout)
+{
+ aout_owner_t *owner = aout_owner (aout);
+
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ aout_FiltersAdjustResampling (owner->filters, 0);
+}
+
+static void aout_DecSilence (audio_output_t *aout, mtime_t length, mtime_t pts)
{
+ aout_owner_t *owner = aout_owner (aout);
+ const audio_sample_format_t *fmt = &owner->mixer_format;
+ size_t frames = (fmt->i_rate * length) / CLOCK_FREQ;
block_t *block;
- size_t length;
- aout_lock_input( NULL, p_input );
+ if (AOUT_FMT_SPDIF(fmt))
+ block = block_Alloc (4 * frames);
+ else
+ block = block_Alloc (frames * fmt->i_bytes_per_frame);
+ if (unlikely(block == NULL))
+ return; /* uho! */
+
+ msg_Dbg (aout, "inserting %zu zeroes", frames);
+ memset (block->p_buffer, 0, block->i_buffer);
+ block->i_nb_samples = frames;
+ block->i_pts = pts;
+ block->i_dts = pts;
+ block->i_length = length;
+ aout_OutputPlay (aout, block);
+}
- if ( p_input->b_error )
+static void aout_DecSynchronize (audio_output_t *aout, mtime_t dec_pts,
+ int input_rate)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ mtime_t drift;
+
+ /**
+ * Depending on the drift between the actual and intended playback times,
+ * the audio core may ignore the drift, trigger upsampling or downsampling,
+ * insert silence or even discard samples.
+ * Future VLC versions may instead adjust the input rate.
+ *
+ * The audio output plugin is responsible for estimating its actual
+ * playback time, or rather the estimated time when the next sample will
+ * be played. (The actual playback time is always the current time, that is
+ * to say mdate(). It is not an useful statistic.)
+ *
+ * Most audio output plugins can estimate the delay until playback of
+ * the next sample to be written to the buffer, or equally the time until
+ * all samples in the buffer will have been played. Then:
+ * pts = mdate() + delay
+ */
+ if (aout_OutputTimeGet (aout, &drift) != 0)
+ return; /* nothing can be done if timing is unknown */
+ drift += mdate () - dec_pts;
+
+ /* Late audio output.
+ * This can happen due to insufficient caching, scheduling jitter
+ * or bug in the decoder. Ideally, the output would seek backward. But that
+ * is not portable, not supported by some hardware and often unsafe/buggy
+ * where supported. The other alternative is to flush the buffers
+ * completely. */
+ if (drift > (owner->sync.discontinuity ? 0
+ : +3 * input_rate * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT))
{
- aout_unlock_input( NULL, p_input );
- return NULL;
+ if (!owner->sync.discontinuity)
+ msg_Warn (aout, "playback way too late (%"PRId64"): "
+ "flushing buffers", drift);
+ else
+ msg_Dbg (aout, "playback too late (%"PRId64"): "
+ "flushing buffers", drift);
+ aout_OutputFlush (aout, false);
+
+ aout_StopResampling (aout);
+ owner->sync.end = VLC_TS_INVALID;
+ owner->sync.discontinuity = true;
+
+ /* Now the output might be too early... Recheck. */
+ if (aout_OutputTimeGet (aout, &drift) != 0)
+ return; /* nothing can be done if timing is unknown */
+ drift += mdate () - dec_pts;
}
- length = i_nb_samples * p_input->input.i_bytes_per_frame
- / p_input->input.i_frame_length;
- block = block_Alloc( length );
+ /* Early audio output.
+ * This is rare except at startup when the buffers are still empty. */
+ if (drift < (owner->sync.discontinuity ? 0
+ : -3 * input_rate * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT))
+ {
+ if (!owner->sync.discontinuity)
+ msg_Warn (aout, "playback way too early (%"PRId64"): "
+ "playing silence", drift);
+ aout_DecSilence (aout, -drift, dec_pts);
+
+ aout_StopResampling (aout);
+ owner->sync.discontinuity = true;
+ drift = 0;
+ }
- /* Suppose the decoder doesn't have more than one buffered buffer */
- p_input->b_changed = false;
+ /* Resampling */
+ if (drift > +AOUT_MAX_PTS_DELAY
+ && owner->sync.resamp_type != AOUT_RESAMPLING_UP)
+ {
+ msg_Warn (aout, "playback too late (%"PRId64"): up-sampling",
+ drift);
+ owner->sync.resamp_type = AOUT_RESAMPLING_UP;
+ owner->sync.resamp_start_drift = +drift;
+ }
+ if (drift < -AOUT_MAX_PTS_ADVANCE
+ && owner->sync.resamp_type != AOUT_RESAMPLING_DOWN)
+ {
+ msg_Warn (aout, "playback too early (%"PRId64"): down-sampling",
+ drift);
+ owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
+ owner->sync.resamp_start_drift = -drift;
+ }
- aout_unlock_input( NULL, p_input );
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
+ return; /* Everything is fine. Nothing to do. */
- if( likely(block != NULL) )
- {
- block->i_nb_samples = i_nb_samples;
- block->i_pts = block->i_length = 0;
+ if (llabs (drift) > 2 * owner->sync.resamp_start_drift)
+ { /* If the drift is ever increasing, then something is seriously wrong.
+ * Cease resampling and hope for the best. */
+ msg_Warn (aout, "timing screwed (drift: %"PRId64" us): "
+ "stopping resampling", drift);
+ aout_StopResampling (aout);
+ return;
}
- return block;
-}
-/*****************************************************************************
- * aout_DecDeleteBuffer : destroy an undecoded buffer
- *****************************************************************************/
-void aout_DecDeleteBuffer( aout_instance_t * p_aout, aout_input_t * p_input,
- aout_buffer_t * p_buffer )
-{
- (void)p_aout; (void)p_input;
- aout_BufferFree( p_buffer );
+ /* Resampling has been triggered earlier. This checks if it needs to be
+ * increased or decreased. Resampling rate changes must be kept slow for
+ * the comfort of listeners. */
+ int adj = (owner->sync.resamp_type == AOUT_RESAMPLING_UP) ? +2 : -2;
+
+ if (2 * llabs (drift) <= owner->sync.resamp_start_drift)
+ /* If the drift has been reduced from more than half its initial
+ * value, then it is time to switch back the resampling direction. */
+ adj *= -1;
+
+ if (!aout_FiltersAdjustResampling (owner->filters, adj))
+ { /* Everything is back to normal: stop resampling. */
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ msg_Dbg (aout, "resampling stopped (drift: %"PRId64" us)", drift);
+ }
}
/*****************************************************************************
* aout_DecPlay : filter & mix the decoded buffer
*****************************************************************************/
-int aout_DecPlay( aout_instance_t * p_aout, aout_input_t * p_input,
- aout_buffer_t * p_buffer, int i_input_rate )
+int aout_DecPlay (audio_output_t *aout, block_t *block, int input_rate)
{
- 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
- / p_input->input.i_rate;
-
- aout_lock_mixer( p_aout );
- aout_lock_input( p_aout, p_input );
-
- if( p_input->b_error )
- {
- aout_unlock_input( p_aout, p_input );
- aout_unlock_mixer( p_aout );
-
- aout_BufferFree( p_buffer );
- return -1;
+ aout_owner_t *owner = aout_owner (aout);
+
+ assert (input_rate >= INPUT_RATE_DEFAULT / AOUT_MAX_INPUT_RATE);
+ assert (input_rate <= INPUT_RATE_DEFAULT * AOUT_MAX_INPUT_RATE);
+ assert (block->i_pts >= VLC_TS_0);
+
+ block->i_length = CLOCK_FREQ * block->i_nb_samples
+ / owner->input_format.i_rate;
+
+ aout_OutputLock (aout);
+ if (unlikely(aout_CheckReady (aout)))
+ goto drop; /* Pipeline is unrecoverably broken :-( */
+
+ const mtime_t now = mdate (), advance = block->i_pts - now;
+ if (advance < -AOUT_MAX_PTS_DELAY)
+ { /* Late buffer can be caused by bugs in the decoder, by scheduling
+ * latency spikes (excessive load, SIGSTOP, etc.) or if buffering is
+ * insufficient. We assume the PTS is wrong and play the buffer anyway:
+ * Hopefully video has encountered a similar PTS problem as audio. */
+ msg_Warn (aout, "buffer too late (%"PRId64" us): dropped", advance);
+ goto drop;
}
-
- if( p_input->b_changed )
- {
- /* Maybe the allocation size has changed. Re-allocate a buffer. */
- aout_buffer_t * p_new_buffer;
- mtime_t duration = (1000000 * (mtime_t)p_buffer->i_nb_samples)
- / p_input->input.i_rate;
-
- p_new_buffer = aout_BufferAlloc( &p_input->input_alloc, duration, NULL);
- vlc_memcpy( p_new_buffer->p_buffer, p_buffer->p_buffer,
- p_buffer->i_buffer );
- p_new_buffer->i_nb_samples = p_buffer->i_nb_samples;
- p_new_buffer->i_buffer = p_buffer->i_buffer;
- p_new_buffer->i_pts = p_buffer->i_pts;
- p_new_buffer->i_length = p_buffer->i_length;
- aout_BufferFree( p_buffer );
- p_buffer = p_new_buffer;
- p_input->b_changed = false;
+ if (advance > AOUT_MAX_ADVANCE_TIME)
+ { /* Early buffers can only be caused by bugs in the decoder. */
+ msg_Err (aout, "buffer too early (%"PRId64" us): dropped", advance);
+ goto drop;
}
+ if (block->i_flags & BLOCK_FLAG_DISCONTINUITY)
+ owner->sync.discontinuity = true;
- aout_InputCheckAndRestart( p_aout, p_input );
- aout_unlock_mixer( p_aout );
+ block = aout_FiltersPlay (owner->filters, block, input_rate);
+ if (block == NULL)
+ goto lost;
- int i_ret = aout_InputPlay( p_aout, p_input, p_buffer, i_input_rate );
+ /* Software volume */
+ aout_volume_Amplify (owner->volume, block);
- aout_unlock_input( p_aout, p_input );
-
- if( i_ret == -1 )
- return -1;
-
- /* Run the mixer if it is able to run. */
- aout_lock_mixer( p_aout );
-
- aout_MixerRun( p_aout );
-
- aout_unlock_mixer( p_aout );
+ /* Drift correction */
+ aout_DecSynchronize (aout, block->i_pts, input_rate);
+ /* Output */
+ owner->sync.end = block->i_pts + block->i_length + 1;
+ owner->sync.discontinuity = false;
+ aout_OutputPlay (aout, block);
+out:
+ aout_OutputUnlock (aout);
return 0;
+drop:
+ owner->sync.discontinuity = true;
+ block_Release (block);
+lost:
+ atomic_fetch_add(&owner->buffers_lost, 1);
+ goto out;
}
-int aout_DecGetResetLost( aout_instance_t *p_aout, aout_input_t *p_input )
+int aout_DecGetResetLost (audio_output_t *aout)
{
- aout_lock_input( p_aout, p_input );
- int i_value = p_input->i_buffer_lost;
- p_input->i_buffer_lost = 0;
- aout_unlock_input( p_aout, p_input );
-
- return i_value;
+ aout_owner_t *owner = aout_owner (aout);
+ return atomic_exchange(&owner->buffers_lost, 0);
}
-void aout_DecChangePause( aout_instance_t *p_aout, aout_input_t *p_input, bool b_paused, mtime_t i_date )
+void aout_DecChangePause (audio_output_t *aout, bool paused, mtime_t date)
{
- mtime_t i_duration = 0;
- aout_lock_input( p_aout, p_input );
- assert( !p_input->b_paused || !b_paused );
- if( p_input->b_paused )
- {
- i_duration = i_date - p_input->i_pause_date;
- }
- p_input->b_paused = b_paused;
- p_input->i_pause_date = i_date;
- aout_unlock_input( p_aout, p_input );
+ aout_owner_t *owner = aout_owner (aout);
- if( i_duration != 0 )
+ aout_OutputLock (aout);
+ if (owner->sync.end != VLC_TS_INVALID)
{
- aout_lock_mixer( p_aout );
- for( aout_buffer_t *p = p_input->mixer.fifo.p_first; p != NULL; p = p->p_next )
- {
- p->i_pts += i_duration;
- }
- aout_unlock_mixer( p_aout );
+ if (paused)
+ owner->sync.end -= date;
+ else
+ owner->sync.end += date;
}
+ if (owner->mixer_format.i_format)
+ aout_OutputPause (aout, paused, date);
+ aout_OutputUnlock (aout);
}
-void aout_DecFlush( aout_instance_t *p_aout, aout_input_t *p_input )
+void aout_DecFlush (audio_output_t *aout, bool wait)
{
- aout_lock_input_fifos( p_aout );
+ aout_owner_t *owner = aout_owner (aout);
- aout_FifoSet( p_aout, &p_input->mixer.fifo, 0 );
- p_input->mixer.begin = NULL;
-
- aout_unlock_input_fifos( p_aout );
+ aout_OutputLock (aout);
+ owner->sync.end = VLC_TS_INVALID;
+ if (owner->mixer_format.i_format)
+ aout_OutputFlush (aout, wait);
+ aout_OutputUnlock (aout);
}
+bool aout_DecIsEmpty (audio_output_t *aout)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ mtime_t now = mdate ();
+ bool empty = true;
+
+ aout_OutputLock (aout);
+ if (owner->sync.end != VLC_TS_INVALID)
+ empty = owner->sync.end <= now;
+ if (empty && owner->mixer_format.i_format)
+ /* The last PTS has elapsed already. So the underlying audio output
+ * buffer should be empty or almost. Thus draining should be fast
+ * and will not block the caller too long. */
+ aout_OutputFlush (aout, true);
+ aout_OutputUnlock (aout);
+ return empty;
+}
projects / vlc / blobdiff