# include "config.h"
#endif
+#include <math.h>
+
#include <assert.h>
#include <vlc_common.h>
#include <vlc_aout.h>
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
-int aout_OutputNew( audio_output_t * p_aout,
+int aout_OutputNew( audio_output_t *p_aout,
const audio_sample_format_t * p_format )
{
- vlc_assert_locked( &p_aout->lock );
+ aout_owner_t *owner = aout_owner (p_aout);
+
+ aout_assert_locked( p_aout );
p_aout->format = *p_format;
/* Retrieve user defaults. */
aout_FormatPrepare( &p_aout->format );
/* Find the best output plug-in. */
- p_aout->module = module_need( p_aout, "audio output", "$aout", false );
- if ( p_aout->module == NULL )
+ owner->module = module_need (p_aout, "audio output", "$aout", false);
+ if (owner->module == NULL)
{
msg_Err( p_aout, "no suitable audio output module" );
return -1;
var_TriggerCallback( p_aout, "intf-change" );
aout_FormatPrepare( &p_aout->format );
-
- /* Prepare FIFO. */
- aout_FifoInit( p_aout, &p_aout->fifo, p_aout->format.i_rate );
aout_FormatPrint( p_aout, "output", &p_aout->format );
/* Choose the mixer format. */
- p_aout->mixer_format = p_aout->format;
- if ( AOUT_FMT_NON_LINEAR(&p_aout->format) )
- p_aout->mixer_format.i_format = p_format->i_format;
+ owner->mixer_format = p_aout->format;
+ if (AOUT_FMT_NON_LINEAR(&p_aout->format))
+ owner->mixer_format.i_format = p_format->i_format;
else
/* Most audio filters can only deal with single-precision,
* so lets always use that when hardware supports floating point. */
if( HAVE_FPU )
- p_aout->mixer_format.i_format = VLC_CODEC_FL32;
+ owner->mixer_format.i_format = VLC_CODEC_FL32;
else
/* Otherwise, audio filters will not work. Use fixed-point if the input has
* more than 16-bits depth. */
if( p_format->i_bitspersample > 16 )
- p_aout->mixer_format.i_format = VLC_CODEC_FI32;
+ owner->mixer_format.i_format = VLC_CODEC_FI32;
else
/* Fallback to 16-bits. This avoids pointless conversion to and from
* 32-bits samples for the sole purpose of software mixing. */
- p_aout->mixer_format.i_format = VLC_CODEC_S16N;
+ owner->mixer_format.i_format = VLC_CODEC_S16N;
- aout_FormatPrepare( &p_aout->mixer_format );
- aout_FormatPrint( p_aout, "mixer", &p_aout->mixer_format );
+ aout_FormatPrepare (&owner->mixer_format);
+ aout_FormatPrint (p_aout, "mixer", &owner->mixer_format);
/* Create filters. */
- p_aout->i_nb_filters = 0;
- if ( aout_FiltersCreatePipeline( p_aout, p_aout->pp_filters,
- &p_aout->i_nb_filters,
- &p_aout->mixer_format,
- &p_aout->format ) < 0 )
+ owner->nb_filters = 0;
+ if (aout_FiltersCreatePipeline (p_aout, owner->filters,
+ &owner->nb_filters, &owner->mixer_format,
+ &p_aout->format) < 0)
{
msg_Err( p_aout, "couldn't create audio output pipeline" );
- module_unneed( p_aout, p_aout->module );
- p_aout->module = NULL;
+ module_unneed (p_aout, owner->module);
+ owner->module = NULL;
return -1;
}
return 0;
*****************************************************************************/
void aout_OutputDelete( audio_output_t * p_aout )
{
- vlc_assert_locked( &p_aout->lock );
+ aout_owner_t *owner = aout_owner (p_aout);
- if( p_aout->module == NULL )
+ aout_assert_locked( p_aout );
+
+ if (owner->module == NULL)
return;
- module_unneed( p_aout, p_aout->module );
+ module_unneed (p_aout, owner->module);
aout_VolumeNoneInit( p_aout ); /* clear volume callback */
- p_aout->module = NULL;
- aout_FiltersDestroyPipeline( p_aout->pp_filters, p_aout->i_nb_filters );
- aout_FifoDestroy( &p_aout->fifo );
+ owner->module = NULL;
+ aout_FiltersDestroyPipeline (owner->filters, owner->nb_filters);
}
-static block_t *aout_OutputSlice( audio_output_t *, aout_fifo_t * );
-
/*****************************************************************************
* aout_OutputPlay : play a buffer
*****************************************************************************
* This function is entered with the mixer lock.
*****************************************************************************/
-void aout_OutputPlay( audio_output_t * p_aout, aout_buffer_t * p_buffer )
+void aout_OutputPlay (audio_output_t *aout, block_t *block)
{
- vlc_assert_locked( &p_aout->lock );
+ aout_owner_t *owner = aout_owner (aout);
- aout_FiltersPlay( p_aout->pp_filters, p_aout->i_nb_filters, &p_buffer );
- if( !p_buffer )
+ aout_assert_locked (aout);
+
+ aout_FiltersPlay (owner->filters, owner->nb_filters, &block);
+ if (block == NULL)
return;
- if( p_buffer->i_buffer == 0 )
+ if (block->i_buffer == 0)
{
- block_Release( p_buffer );
+ block_Release (block);
return;
}
- aout_fifo_t *fifo = &p_aout->p_input->fifo;
- /* XXX: cleanup */
- aout_FifoPush( fifo, p_buffer );
-
- while( (p_buffer = aout_OutputSlice( p_aout, fifo ) ) != NULL )
- {
- aout_FifoPush( &p_aout->fifo, p_buffer );
- p_aout->pf_play( p_aout );
- }
+ aout->pf_play (aout, block);
}
/**
*/
void aout_OutputPause( audio_output_t *aout, bool pause, mtime_t date )
{
- vlc_assert_locked( &aout->lock );
-
+ aout_assert_locked( aout );
if( aout->pf_pause != NULL )
aout->pf_pause( aout, pause, date );
- if( !pause )
- {
- mtime_t duration = date - aout->p_input->i_pause_date;
- /* XXX: ^ onk onk! gruik! ^ */
- aout_FifoMoveDates( &aout->fifo, duration );
- }
}
/**
*/
void aout_OutputFlush( audio_output_t *aout, bool wait )
{
- vlc_assert_locked( &aout->lock );
+ aout_assert_locked( aout );
if( aout->pf_flush != NULL )
aout->pf_flush( aout, wait );
- aout_FifoReset( &aout->fifo );
}
{
/* aout_New() -safely- calls this function without the lock, before any
* other thread knows of this audio output instance.
- vlc_assert_locked (&aout->lock); */
+ aout_assert_locked (aout); */
aout->pf_volume_set = aout_VolumeNoneSet;
+ var_Destroy (aout, "volume");
+ var_Destroy (aout, "mute");
}
/**
*/
static int aout_VolumeSoftSet (audio_output_t *aout, float volume, bool mute)
{
- vlc_assert_locked (&aout->lock);
+ aout_owner_t *owner = aout_owner (aout);
+
+ aout_assert_locked (aout);
/* Cubic mapping from software volume to amplification factor.
* This provides a good tradeoff between low and high volume ranges.
else
volume = 0.;
- aout->mixer_multiplier = volume;
+ owner->volume.multiplier = volume;
return 0;
}
audio_volume_t volume = var_InheritInteger (aout, "volume");
bool mute = var_InheritBool (aout, "mute");
- vlc_assert_locked (&aout->lock);
+ aout_assert_locked (aout);
aout->pf_volume_set = aout_VolumeSoftSet;
aout_VolumeSoftSet (aout, volume / (float)AOUT_VOLUME_DEFAULT, mute);
}
*/
void aout_VolumeHardInit (audio_output_t *aout, aout_volume_cb setter)
{
- vlc_assert_locked (&aout->lock);
+ aout_assert_locked (aout);
aout->pf_volume_set = setter;
+ var_Create (aout, "volume", VLC_VAR_INTEGER|VLC_VAR_DOINHERIT);
+ var_Create (aout, "mute", VLC_VAR_BOOL|VLC_VAR_DOINHERIT);
}
/**
* @param setter volume setter callback
* @param volume current custom volume
* @param mute current mute flag
- * @note Audio output plugins that cannot apply the volume
- * should call this function during activation.
+ *
+ * @warning The caller (i.e. the audio output plug-in) is responsible for
+ * interlocking and synchronizing call to this function and to the
+ * audio_output_t.pf_volume_set callback. This ensures that VLC gets correct
+ * volume information (possibly with a latency).
*/
void aout_VolumeHardSet (audio_output_t *aout, float volume, bool mute)
{
-#warning FIXME
- /* REVISIT: This is tricky. We cannot acquire the volume lock as this gets
- * called from the audio output (it would cause a lock inversion).
- * We also should not override the input manager volume, but only the
- * volume of the current audio output... FIXME */
- msg_Err (aout, "%s(%f, %u)", __func__, volume, (unsigned)mute);
+ audio_volume_t vol = lroundf (volume * (float)AOUT_VOLUME_DEFAULT);
+
+ /* We cannot acquire the volume lock as this gets called from the audio
+ * output plug-in (it would cause a lock inversion). */
+ var_SetInteger (aout, "volume", vol);
+ var_SetBool (aout, "mute", mute);
+ var_TriggerCallback (aout, "intf-change");
+}
+
+
+/*** Packet-oriented audio output support ***/
+
+static inline aout_packet_t *aout_packet (audio_output_t *aout)
+{
+ return (aout_packet_t *)(aout->sys);
+}
+
+void aout_PacketInit (audio_output_t *aout, aout_packet_t *p, unsigned samples)
+{
+ assert (p == aout_packet (aout));
+
+ vlc_mutex_init (&p->lock);
+ aout_FifoInit (aout, &p->partial, aout->format.i_rate);
+ aout_FifoInit (aout, &p->fifo, aout->format.i_rate);
+ p->pause_date = VLC_TS_INVALID;
+ p->time_report = VLC_TS_INVALID;
+ p->samples = samples;
+ p->starving = true;
+}
+
+void aout_PacketDestroy (audio_output_t *aout)
+{
+ aout_packet_t *p = aout_packet (aout);
+
+ aout_FifoDestroy (&p->partial);
+ aout_FifoDestroy (&p->fifo);
+ vlc_mutex_destroy (&p->lock);
+}
+
+static block_t *aout_OutputSlice (audio_output_t *);
+
+void aout_PacketPlay (audio_output_t *aout, block_t *block)
+{
+ aout_packet_t *p = aout_packet (aout);
+ mtime_t time_report;
+
+ vlc_mutex_lock (&p->lock);
+ aout_FifoPush (&p->partial, block);
+ while ((block = aout_OutputSlice (aout)) != NULL)
+ aout_FifoPush (&p->fifo, block);
+
+ time_report = p->time_report;
+ p->time_report = VLC_TS_INVALID;
+ vlc_mutex_unlock (&p->lock);
+
+ if (time_report != VLC_TS_INVALID)
+ aout_TimeReport (aout, mdate () + time_report);
+}
+
+void aout_PacketPause (audio_output_t *aout, bool pause, mtime_t date)
+{
+ aout_packet_t *p = aout_packet (aout);
+
+ if (pause)
+ {
+ assert (p->pause_date == VLC_TS_INVALID);
+ p->pause_date = date;
+ }
+ else
+ {
+ assert (p->pause_date != VLC_TS_INVALID);
+
+ mtime_t duration = date - p->pause_date;
+
+ p->pause_date = VLC_TS_INVALID;
+ vlc_mutex_lock (&p->lock);
+ aout_FifoMoveDates (&p->partial, duration);
+ aout_FifoMoveDates (&p->fifo, duration);
+ vlc_mutex_unlock (&p->lock);
+ }
}
+void aout_PacketFlush (audio_output_t *aout, bool drain)
+{
+ aout_packet_t *p = aout_packet (aout);
+
+ vlc_mutex_lock (&p->lock);
+ aout_FifoReset (&p->partial);
+ aout_FifoReset (&p->fifo);
+ vlc_mutex_unlock (&p->lock);
+
+ (void) drain; /* TODO */
+}
-/*** Buffer management ***/
/**
* 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_fifo_t *p_fifo)
+static block_t *aout_OutputSlice (audio_output_t *p_aout)
{
- const unsigned samples = p_aout->i_nb_samples;
- /* FIXME: Remove this silly constraint. Just pass buffers as they come to
- * "smart" audio outputs. */
+ 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_aout->lock );
+ vlc_assert_locked( &p->lock );
/* Retrieve the date of the next buffer. */
- date_t exact_start_date = p_aout->fifo.end_date;
+ 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. */
if( delta < 0 )
{
/* Is it really the best way to do it ? */
- aout_FifoReset( &p_aout->fifo );
+ aout_FifoReset (&p->fifo);
return NULL;
}
if( delta > 0 )
return p_buffer;
}
-/*****************************************************************************
- * 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( audio_output_t * p_aout,
- mtime_t start_date,
- bool b_can_sleek )
+/**
+ * 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_fifo_t *p_fifo = &p_aout->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 )
+ 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_NON_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 (;;)
{
- 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 ) );
- }
+ p_buffer = p_fifo->p_first;
+ if (p_buffer == NULL)
+ goto out; /* nothing to play */
- 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_FifoReset( &p_aout->fifo );
- if ( !p_aout->b_starving )
- msg_Dbg( p_aout,
- "audio output is starving (no input), playing silence" );
- p_aout->b_starving = true;
-#endif
- goto out;
+ 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;
- /* Here we suppose that all buffers have the same duration - this is
- * generally true, and anyway if it's wrong it won't be a disaster.
- */
- if ( 0 > delta + p_buffer->i_length )
+ /* 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_aout->b_starving )
- msg_Dbg( p_aout, "audio output is starving (%"PRId64"), "
- "playing silence", -delta );
- p_aout->b_starving = true;
- p_buffer = NULL;
- goto out;
+ 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_aout->b_starving = false;
+ p->starving = false;
p_buffer = aout_FifoPop( p_fifo );
- if( !b_can_sleek
- && ( delta > AOUT_MAX_PTS_DELAY || delta < -AOUT_MAX_PTS_ADVANCE ) )
+ if (!b_can_sleek
+ && (delta < -AOUT_MAX_PTS_ADVANCE || AOUT_MAX_PTS_DELAY < delta))
{
- /* Try to compensate the drift by doing some resampling. */
- msg_Warn( p_aout, "output date isn't PTS date, requesting "
- "resampling (%"PRId64")", delta );
-
- aout_FifoMoveDates( &p_aout->p_input->fifo, delta );
- aout_FifoMoveDates( p_fifo, 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;
}
-out:
- aout_unlock( p_aout );
+ vlc_mutex_unlock( &p->lock );
return p_buffer;
+out:
+ vlc_mutex_unlock( &p->lock );
+ return NULL;
}
projects / vlc / blobdiff