#include <vlc_input.h>
#include <vlc_vout.h> /* for vout_Request */
+#include <vlc_modules.h>
#include <vlc_aout.h>
+#include <vlc_filter.h>
#include <libvlc.h>
#include "aout_internal.h"
-#define AOUT_ASSERT_MIXER_LOCKED vlc_assert_locked( &p_aout->mixer_lock )
-#define AOUT_ASSERT_INPUT_LOCKED vlc_assert_locked( &p_input->lock )
+#define AOUT_ASSERT_LOCKED vlc_assert_locked( &p_aout->lock )
-static void inputFailure( aout_instance_t *, aout_input_t *, const char * );
+static void inputFailure( audio_output_t *, aout_input_t *, const char * );
static void inputDrop( aout_input_t *, aout_buffer_t * );
static void inputResamplingStop( aout_input_t *p_input );
vlc_value_t, vlc_value_t, void * );
static int ReplayGainCallback( vlc_object_t *, char const *,
vlc_value_t, vlc_value_t, void * );
-static void ReplayGainSelect( aout_instance_t *, aout_input_t * );
+static void ReplayGainSelect( audio_output_t *, aout_input_t * );
static vout_thread_t *RequestVout( void *,
vout_thread_t *, video_format_t *, bool );
/*****************************************************************************
* aout_InputNew : allocate a new input and rework the filter pipeline
*****************************************************************************/
-int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout )
+int aout_InputNew( audio_output_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout )
{
+ aout_owner_t *owner = aout_owner (p_aout);
audio_sample_format_t chain_input_format;
audio_sample_format_t chain_output_format;
vlc_value_t val, text;
p_input->i_nb_resamplers = p_input->i_nb_filters = 0;
/* Prepare FIFO. */
- aout_FifoInit( p_aout, &p_input->mixer.fifo, p_aout->mixer_format.i_rate );
- p_input->mixer.begin = NULL;
+ aout_FifoInit (p_aout, &p_input->fifo, owner->mixer_format.i_rate);
/* */
if( p_request_vout )
/* Prepare format structure */
chain_input_format = p_input->input;
- chain_output_format = p_aout->mixer_format;
+ chain_output_format = owner->mixer_format;
chain_output_format.i_rate = p_input->input.i_rate;
aout_FormatPrepare( &chain_output_format );
var_Create( p_aout, "audio-replay-gain-peak-protection",
VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
}
- if( var_Type( p_aout, "audio-time-stretch" ) == 0 )
- {
- var_Create( p_aout, "audio-time-stretch",
- VLC_VAR_BOOL | VLC_VAR_DOINHERIT );
- }
psz_filters = var_GetString( p_aout, "audio-filter" );
psz_visual = var_GetString( p_aout, "audio-visual");
- psz_scaletempo = var_GetBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL;
+ psz_scaletempo = var_InheritBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL;
p_input->b_recycle_vout = psz_visual && *psz_visual;
while( psz_parser && *psz_parser )
{
- aout_filter_t * p_filter = NULL;
+ filter_t * p_filter = NULL;
if( p_input->i_nb_filters >= AOUT_MAX_FILTERS )
{
}
/* Create a VLC object */
- static const char typename[] = "audio filter";
p_filter = vlc_custom_create( p_aout, sizeof(*p_filter),
- VLC_OBJECT_GENERIC, typename );
+ "audio filter" );
if( p_filter == NULL )
{
msg_Err( p_aout, "cannot add user filter %s (skipped)",
continue;
}
- vlc_object_attach( p_filter , p_aout );
-
p_filter->p_owner = malloc( sizeof(*p_filter->p_owner) );
p_filter->p_owner->p_aout = p_aout;
p_filter->p_owner->p_input = p_input;
/* request format */
memcpy( &p_filter->fmt_in.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
+ p_filter->fmt_in.i_codec = chain_output_format.i_format;
memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
-
+ p_filter->fmt_out.i_codec = chain_output_format.i_format;
+ p_filter->pf_audio_buffer_new = aout_FilterBufferNew;
/* try to find the requested filter */
if( i_visual == 2 ) /* this can only be a visualization module */
{
- p_filter->p_module = module_need( p_filter, "visualization",
+ p_filter->p_module = module_need( p_filter, "visualization2",
psz_parser, true );
}
else /* this can be a audio filter module as well as a visualization module */
memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
p_filter->p_module = module_need( p_filter,
- "visualization",
+ "visualization2",
psz_parser, true );
}
}
psz_parser );
free( p_filter->p_owner );
- vlc_object_detach( p_filter );
vlc_object_release( p_filter );
psz_parser = psz_next;
}
/* complete the filter chain if necessary */
- if ( !AOUT_FMTS_IDENTICAL( &chain_input_format,
- &p_filter->fmt_in.audio ) )
+ if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
+ &p_input->i_nb_filters,
+ &chain_input_format,
+ &p_filter->fmt_in.audio ) < 0 )
{
- if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
- &p_input->i_nb_filters,
- &chain_input_format,
- &p_filter->fmt_in.audio ) < 0 )
- {
- msg_Err( p_aout, "cannot add user filter %s (skipped)",
- psz_parser );
+ msg_Err( p_aout, "cannot add user filter %s (skipped)",
+ psz_parser );
- module_unneed( p_filter, p_filter->p_module );
- free( p_filter->p_owner );
- vlc_object_detach( p_filter );
- vlc_object_release( p_filter );
+ module_unneed( p_filter, p_filter->p_module );
+ free( p_filter->p_owner );
+ vlc_object_release( p_filter );
- psz_parser = psz_next;
- continue;
- }
+ psz_parser = psz_next;
+ continue;
}
/* success */
- p_filter->b_continuity = false;
p_input->pp_filters[p_input->i_nb_filters++] = p_filter;
memcpy( &chain_input_format, &p_filter->fmt_out.audio,
sizeof( audio_sample_format_t ) );
free( psz_scaletempo );
/* complete the filter chain if necessary */
- if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &chain_output_format ) )
+ if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
+ &p_input->i_nb_filters,
+ &chain_input_format,
+ &chain_output_format ) < 0 )
{
- if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
- &p_input->i_nb_filters,
- &chain_input_format,
- &chain_output_format ) < 0 )
- {
- inputFailure( p_aout, p_input, "couldn't set an input pipeline" );
- return -1;
- }
+ inputFailure( p_aout, p_input, "couldn't set an input pipeline" );
+ return -1;
}
- /* Prepare hints for the buffer allocator. */
- p_input->input_alloc.b_alloc = true;
- p_input->input_alloc.i_bytes_per_sec = -1;
-
/* Create resamplers. */
- if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer_format ) )
+ if (!AOUT_FMT_NON_LINEAR(&owner->mixer_format))
{
chain_output_format.i_rate = (__MAX(p_input->input.i_rate,
- p_aout->mixer_format.i_rate)
+ owner->mixer_format.i_rate)
* (100 + AOUT_MAX_RESAMPLING)) / 100;
- if ( chain_output_format.i_rate == p_aout->mixer_format.i_rate )
+ if ( chain_output_format.i_rate == owner->mixer_format.i_rate )
{
/* Just in case... */
chain_output_format.i_rate++;
}
- if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_resamplers,
- &p_input->i_nb_resamplers,
- &chain_output_format,
- &p_aout->mixer_format ) < 0 )
+ if (aout_FiltersCreatePipeline (p_aout, p_input->pp_resamplers,
+ &p_input->i_nb_resamplers,
+ &chain_output_format,
+ &owner->mixer_format) < 0)
{
inputFailure( p_aout, p_input, "couldn't set a resampler pipeline");
return -1;
}
- aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers,
- p_input->i_nb_resamplers,
- &p_input->input_alloc );
- p_input->input_alloc.b_alloc = true;
-
/* Setup the initial rate of the resampler */
p_input->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->input.i_rate;
}
p_input->p_playback_rate_filter = p_input->pp_resamplers[0];
}
- aout_FiltersHintBuffers( p_aout, p_input->pp_filters,
- p_input->i_nb_filters,
- &p_input->input_alloc );
- p_input->input_alloc.b_alloc = true;
-
- /* i_bytes_per_sec is still == -1 if no filters */
- p_input->input_alloc.i_bytes_per_sec = __MAX(
- p_input->input_alloc.i_bytes_per_sec,
- (int)(p_input->input.i_bytes_per_frame
- * p_input->input.i_rate
- / p_input->input.i_frame_length) );
-
ReplayGainSelect( p_aout, p_input );
/* Success */
p_input->b_error = false;
- p_input->b_restart = false;
p_input->i_last_input_rate = INPUT_RATE_DEFAULT;
return 0;
*****************************************************************************
* This function must be entered with the mixer lock.
*****************************************************************************/
-int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
+int aout_InputDelete( audio_output_t * p_aout, aout_input_t * p_input )
{
- AOUT_ASSERT_MIXER_LOCKED;
+ AOUT_ASSERT_LOCKED;
if ( p_input->b_error )
return 0;
p_input->b_recycle_vout = psz_visual && *psz_visual;
free( psz_visual );
- aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
- p_input->i_nb_filters );
+ aout_FiltersDestroyPipeline( p_input->pp_filters, p_input->i_nb_filters );
p_input->i_nb_filters = 0;
- aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
+ aout_FiltersDestroyPipeline( p_input->pp_resamplers,
p_input->i_nb_resamplers );
p_input->i_nb_resamplers = 0;
- aout_FifoDestroy( p_aout, &p_input->mixer.fifo );
+ aout_FifoDestroy( &p_input->fifo );
return 0;
}
/*****************************************************************************
- * aout_InputPlay : play a buffer
+ * aout_InputCheckAndRestart : restart an input
*****************************************************************************
- * This function must be entered with the input lock.
+ * This function must be entered with the input and mixer lock.
*****************************************************************************/
-/* XXX Do not activate it !! */
-//#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 )
+void aout_InputCheckAndRestart( audio_output_t * p_aout, aout_input_t * p_input )
{
- mtime_t start_date;
- AOUT_ASSERT_INPUT_LOCKED;
+ aout_owner_t *owner = aout_owner (p_aout);
+ AOUT_ASSERT_LOCKED;
- if( p_input->b_restart )
- {
- aout_fifo_t fifo;
- uint8_t *p_first_byte_to_mix;
- bool b_paused;
- mtime_t i_pause_date;
-
- aout_lock_mixer( p_aout );
- aout_lock_input_fifos( p_aout );
+ if( !p_input->b_restart )
+ return;
- /* A little trick to avoid loosing our input fifo and properties */
+ /* A little trick to avoid loosing our input fifo and properties */
- p_first_byte_to_mix = p_input->mixer.begin;
- fifo = p_input->mixer.fifo;
- b_paused = p_input->b_paused;
- i_pause_date = p_input->i_pause_date;
+ aout_fifo_t fifo = p_input->fifo;
+ mtime_t i_pause_date = p_input->i_pause_date;
- aout_FifoInit( p_aout, &p_input->mixer.fifo, p_aout->mixer_format.i_rate );
+ aout_FifoInit (p_aout, &p_input->fifo, owner->mixer_format.i_rate);
- aout_InputDelete( p_aout, p_input );
+ aout_InputDelete( p_aout, p_input );
- aout_InputNew( p_aout, p_input, &p_input->request_vout );
- p_input->mixer.begin = p_first_byte_to_mix;
- p_input->mixer.fifo = fifo;
- p_input->b_paused = b_paused;
- p_input->i_pause_date = i_pause_date;
+ aout_InputNew( p_aout, p_input, &p_input->request_vout );
+ p_input->fifo = fifo;
+ p_input->i_pause_date = i_pause_date;
- aout_unlock_input_fifos( p_aout );
- aout_unlock_mixer( p_aout );
- }
+ p_input->b_restart = false;
+}
+/*****************************************************************************
+ * aout_InputPlay : play a buffer
+ *****************************************************************************
+ * This function must be entered with the input lock.
+ *****************************************************************************/
+/* XXX Do not activate it !! */
+//#define AOUT_PROCESS_BEFORE_CHEKS
+block_t *aout_InputPlay( audio_output_t *p_aout, aout_input_t *p_input,
+ block_t *p_buffer, int i_input_rate )
+{
+ mtime_t start_date;
+ AOUT_ASSERT_LOCKED;
if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
{
inputDrop( p_input, p_buffer );
- return 0;
+ return NULL;
}
#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 NULL;
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
&p_buffer );
}
+ if( !p_buffer )
+ return NULL;
if( p_buffer->i_nb_samples <= 0 )
{
- aout_BufferFree( p_buffer );
- return 0;
+ block_Release( p_buffer );
+ return NULL;
}
#endif
p_input->i_last_input_rate = i_input_rate;
}
+ mtime_t now = mdate();
+
/* 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 );
+ start_date = aout_FifoNextStart( &p_input->fifo );
- if ( start_date != 0 && start_date < mdate() )
+ if ( start_date != VLC_TS_INVALID && start_date < now )
{
/* 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 );
+ "clearing out", now - start_date );
+ aout_FifoReset( &p_input->fifo );
+ aout_OutputFlush( p_aout, false );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
- start_date = 0;
+ p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
+ start_date = VLC_TS_INVALID;
}
- if ( p_buffer->i_pts < mdate() + AOUT_MIN_PREPARE_TIME )
+ if ( p_buffer->i_pts < now + 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 );
-
+ now - p_buffer->i_pts );
inputDrop( p_input, p_buffer );
inputResamplingStop( p_input );
- return 0;
+ return NULL;
}
/* 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( start_date == VLC_TS_INVALID )
+ start_date = p_buffer->i_pts;
+
+ mtime_t drift = start_date - p_buffer->i_pts;
+
+ if( drift < -i_input_rate * 3 * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT )
+ {
+ msg_Warn( p_aout, "buffer way too early (%"PRId64"), clearing queue",
+ drift );
+ aout_FifoReset( &p_input->fifo );
+ aout_OutputFlush( p_aout, false );
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
msg_Warn( p_aout, "timing screwed, stopping resampling" );
inputResamplingStop( p_input );
- start_date = 0;
+ p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY;
+ start_date = p_buffer->i_pts;
+ drift = 0;
}
- else if ( start_date != 0 &&
- ( start_date > p_buffer->i_pts + i_pts_tolerance) )
+ else
+ if( drift > +i_input_rate * 3 * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT )
{
- msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer",
- start_date - p_buffer->i_pts );
+ msg_Warn( p_aout, "buffer way too late (%"PRId64"), dropping buffer",
+ drift );
inputDrop( p_input, p_buffer );
- return 0;
+ return NULL;
}
- if ( start_date == 0 ) start_date = p_buffer->i_pts;
-
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
- aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
- &p_buffer );
+ aout_FiltersPlay( p_input->pp_filters, p_input->i_nb_filters, &p_buffer );
+ if( !p_buffer )
+ return NULL;
#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 ) &&
+ ( drift < -AOUT_MAX_PTS_ADVANCE || drift > +AOUT_MAX_PTS_DELAY ) &&
p_input->i_nb_resamplers > 0 )
{
/* Can happen in several circumstances :
* 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");
+ p_input->i_resamp_start_date = now;
+ p_input->i_resamp_start_drift = (int)-drift;
+ p_input->i_resampling_type = (drift < 0) ? AOUT_RESAMPLING_DOWN
+ : AOUT_RESAMPLING_UP;
+ msg_Warn( p_aout, (drift < 0)
+ ? "buffer too early (%"PRId64"), down-sampling"
+ : "buffer too late (%"PRId64"), up-sampling", drift );
}
if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE )
* 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 */
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,
+ now - p_input->i_resamp_start_date,
p_buffer->i_pts - start_date);
}
else if( abs( (int)(p_buffer->i_pts - start_date) ) <
* 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;
}
}
/* 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,
+ aout_FiltersPlay( p_input->pp_resamplers, p_input->i_nb_resamplers,
&p_buffer );
}
+ if( !p_buffer )
+ return NULL;
if( p_buffer->i_nb_samples <= 0 )
{
- aout_BufferFree( p_buffer );
- return 0;
+ block_Release( p_buffer );
+ return NULL;
}
#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;
+ return p_buffer;
}
/*****************************************************************************
* static functions
*****************************************************************************/
-static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input,
+static void inputFailure( audio_output_t * p_aout, aout_input_t * p_input,
const char * psz_error_message )
{
/* error message */
msg_Err( p_aout, "%s", psz_error_message );
/* clean up */
- aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
- p_input->i_nb_filters );
- aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
+ aout_FiltersDestroyPipeline( p_input->pp_filters, p_input->i_nb_filters );
+ aout_FiltersDestroyPipeline( p_input->pp_resamplers,
p_input->i_nb_resamplers );
- aout_FifoDestroy( p_aout, &p_input->mixer.fifo );
+ aout_FifoDestroy( &p_input->fifo );
var_Destroy( p_aout, "visual" );
var_Destroy( p_aout, "equalizer" );
var_Destroy( p_aout, "audio-filter" );
( p_input->pp_resamplers[0] == p_input->p_playback_rate_filter )
? INPUT_RATE_DEFAULT * p_input->input.i_rate / p_input->i_last_input_rate
: p_input->input.i_rate;
- p_input->pp_resamplers[0]->b_continuity = false;
}
}
static vout_thread_t *RequestVout( void *p_private,
vout_thread_t *p_vout, video_format_t *p_fmt, bool b_recycle )
{
- aout_instance_t *p_aout = p_private;
+ audio_output_t *p_aout = p_private;
VLC_UNUSED(b_recycle);
- return vout_Request( p_aout, p_vout, p_fmt );
+ vout_configuration_t cfg = {
+ .vout = p_vout,
+ .input = NULL,
+ .change_fmt = true,
+ .fmt = p_fmt,
+ .dpb_size = 1,
+ };
+ return vout_Request( p_aout, &cfg );
}
-vout_thread_t *aout_filter_RequestVout( aout_filter_t *p_filter,
+vout_thread_t *aout_filter_RequestVout( filter_t *p_filter,
vout_thread_t *p_vout, video_format_t *p_fmt )
{
aout_input_t *p_input = p_filter->p_owner->p_input;
p_vout, p_fmt, p_input->b_recycle_vout );
}
-static int ChangeFiltersString( aout_instance_t * p_aout, const char* psz_variable,
+static int ChangeFiltersString( audio_output_t * p_aout, const char* psz_variable,
const char *psz_name, bool b_add )
{
- return AoutChangeFilterString( VLC_OBJECT(p_aout), p_aout,
- psz_variable, psz_name, b_add ) ? 1 : 0;
+ return aout_ChangeFilterString( VLC_OBJECT(p_aout), p_aout,
+ psz_variable, psz_name, b_add ) ? 1 : 0;
}
static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd,
vlc_value_t oldval, vlc_value_t newval, void *p_data )
{
- aout_instance_t *p_aout = (aout_instance_t *)p_this;
+ audio_output_t *p_aout = (audio_output_t *)p_this;
char *psz_mode = newval.psz_string;
(void)psz_cmd; (void)oldval; (void)p_data;
static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd,
vlc_value_t oldval, vlc_value_t newval, void *p_data )
{
- aout_instance_t *p_aout = (aout_instance_t *)p_this;
+ audio_output_t *p_aout = (audio_output_t *)p_this;
char *psz_mode = newval.psz_string;
int i_ret;
(void)psz_cmd; (void)oldval; (void)p_data;
{
VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
VLC_UNUSED(newval); VLC_UNUSED(p_data);
- aout_instance_t *p_aout = (aout_instance_t *)p_this;
- int i;
+ audio_output_t *aout = (audio_output_t *)p_this;
+ aout_owner_t *owner = aout_owner (aout);
- aout_lock_mixer( p_aout );
- for( i = 0; i < p_aout->i_nb_inputs; i++ )
- ReplayGainSelect( p_aout, p_aout->pp_inputs[i] );
-
- /* Restart the mixer (a trivial mixer may be in use) */
- if( p_aout->p_mixer )
- aout_MixerMultiplierSet( p_aout, p_aout->mixer_multiplier );
- aout_unlock_mixer( p_aout );
+ aout_lock (aout);
+ if (owner->input != NULL)
+ ReplayGainSelect (aout, owner->input);
+ aout_unlock (aout);
return VLC_SUCCESS;
}
-static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input )
+static void ReplayGainSelect( audio_output_t *p_aout, aout_input_t *p_input )
{
char *psz_replay_gain = var_GetNonEmptyString( p_aout,
"audio-replay-gain-mode" );
int i_use;
float f_gain;
- p_input->mixer.multiplier = 1.0;
+ p_input->multiplier = 1.0;
if( !psz_replay_gain )
return;
f_gain = var_GetFloat( p_aout, "audio-replay-gain-default" );
else
f_gain = 0.0;
- p_input->mixer.multiplier = pow( 10.0, f_gain / 20.0 );
+ p_input->multiplier = pow( 10.0, f_gain / 20.0 );
/* */
if( p_input->replay_gain.pb_peak[i_use] &&
var_GetBool( p_aout, "audio-replay-gain-peak-protection" ) &&
- p_input->replay_gain.pf_peak[i_use] * p_input->mixer.multiplier > 1.0 )
+ p_input->replay_gain.pf_peak[i_use] * p_input->multiplier > 1.0 )
{
- p_input->mixer.multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use];
+ p_input->multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use];
}
free( psz_replay_gain );
}
-
projects / vlc / blobdiff