# include "config.h"
#endif
+#include <assert.h>
+
#include <vlc_common.h>
#include <stdio.h>
#include <math.h>
#include <assert.h>
-#include <vlc_input.h> /* for input_thread_t and i_pts_delay */
+#include <vlc_input.h>
+#include <vlc_vout.h> /* for vout_Request */
-#ifdef HAVE_ALLOCA_H
-# include <alloca.h>
-#endif
#include <vlc_aout.h>
+#include <vlc_filter.h>
+#include <libvlc.h>
#include "aout_internal.h"
-/** FIXME: Ugly but needed to access the counters */
-#include "input/input_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 )
static void inputFailure( aout_instance_t *, aout_input_t *, const char * );
-static void inputDrop( aout_instance_t *, aout_input_t *, aout_buffer_t * );
+static void inputDrop( aout_input_t *, aout_buffer_t * );
static void inputResamplingStop( aout_input_t *p_input );
static int VisualizationCallback( vlc_object_t *, char const *,
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 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 )
+int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout )
{
audio_sample_format_t chain_input_format;
audio_sample_format_t chain_output_format;
p_input->i_nb_resamplers = p_input->i_nb_filters = 0;
/* Prepare FIFO. */
- aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate );
- p_input->p_first_byte_to_mix = NULL;
+ aout_FifoInit( p_aout, &p_input->mixer.fifo, p_aout->mixer_format.i_rate );
+ p_input->mixer.begin = NULL;
+
+ /* */
+ if( p_request_vout )
+ {
+ p_input->request_vout = *p_request_vout;
+ }
+ else
+ {
+ p_input->request_vout.pf_request_vout = RequestVout;
+ p_input->request_vout.p_private = p_aout;
+ }
/* Prepare format structure */
- memcpy( &chain_input_format, &p_input->input,
- sizeof(audio_sample_format_t) );
- memcpy( &chain_output_format, &p_aout->mixer.mixer,
- sizeof(audio_sample_format_t) );
+ chain_input_format = p_input->input;
+ chain_output_format = p_aout->mixer_format;
chain_output_format.i_rate = p_input->input.i_rate;
aout_FormatPrepare( &chain_output_format );
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
/* Look for goom plugin */
- if( module_Exists( VLC_OBJECT(p_aout), "goom" ) )
+ if( module_exists( "goom" ) )
{
val.psz_string = (char*)"goom"; text.psz_string = (char*)"Goom";
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
}
- /* Look for galaktos plugin */
- if( module_Exists( VLC_OBJECT(p_aout), "galaktos" ) )
+ /* Look for libprojectM plugin */
+ if( module_exists( "projectm" ) )
{
- val.psz_string = (char*)"galaktos"; text.psz_string = (char*)"GaLaktos";
+ val.psz_string = (char*)"projectm"; text.psz_string = (char*)"projectM";
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
}
if( var_Get( p_aout, "effect-list", &val ) == VLC_SUCCESS )
{
- var_Set( p_aout, "visual", val );
+ var_SetString( p_aout, "visual", val.psz_string );
free( val.psz_string );
}
var_AddCallback( p_aout, "visual", VisualizationCallback, NULL );
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 );
- }
- var_Get( p_aout, "audio-filter", &val );
- psz_filters = val.psz_string;
- var_Get( p_aout, "audio-visual", &val );
- psz_visual = val.psz_string;
+ psz_filters = var_GetString( p_aout, "audio-filter" );
+ psz_visual = var_GetString( p_aout, "audio-visual");
+ psz_scaletempo = var_InheritBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL;
- psz_scaletempo = var_GetBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL;
+ p_input->b_recycle_vout = psz_visual && *psz_visual;
/* parse user filter lists */
+ char *const ppsz_array[] = { psz_scaletempo, psz_filters, psz_visual };
+ p_input->p_playback_rate_filter = NULL;
+
for( i_visual = 0; i_visual < 3 && !AOUT_FMT_NON_LINEAR(&chain_output_format); i_visual++ )
{
- char *ppsz_array[] = { psz_scaletempo, psz_filters, psz_visual };
char *psz_next = NULL;
char *psz_parser = ppsz_array[i_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 )
{
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 */
{
- /* request format */
- memcpy( &p_filter->input, &chain_output_format,
- sizeof(audio_sample_format_t) );
- memcpy( &p_filter->output, &chain_output_format,
- sizeof(audio_sample_format_t) );
-
- 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 */
{
- /* request format */
- memcpy( &p_filter->input, &chain_input_format,
- sizeof(audio_sample_format_t) );
- memcpy( &p_filter->output, &chain_output_format,
- sizeof(audio_sample_format_t) );
-
- p_filter->p_module = module_Need( p_filter, "audio filter",
+ p_filter->p_module = module_need( p_filter, "audio filter",
psz_parser, true );
if ( p_filter->p_module == NULL )
{
/* if the filter requested a special format, retry */
- if ( !( AOUT_FMTS_IDENTICAL( &p_filter->input,
+ if ( !( AOUT_FMTS_IDENTICAL( &p_filter->fmt_in.audio,
&chain_input_format )
- && AOUT_FMTS_IDENTICAL( &p_filter->output,
+ && AOUT_FMTS_IDENTICAL( &p_filter->fmt_out.audio,
&chain_output_format ) ) )
{
- aout_FormatPrepare( &p_filter->input );
- aout_FormatPrepare( &p_filter->output );
- p_filter->p_module = module_Need( p_filter,
+ aout_FormatPrepare( &p_filter->fmt_in.audio );
+ aout_FormatPrepare( &p_filter->fmt_out.audio );
+ p_filter->p_module = module_need( p_filter,
"audio filter",
psz_parser, true );
}
/* try visual filters */
else
{
- memcpy( &p_filter->input, &chain_output_format,
+ memcpy( &p_filter->fmt_in.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
- memcpy( &p_filter->output, &chain_output_format,
+ memcpy( &p_filter->fmt_out.audio, &chain_output_format,
sizeof(audio_sample_format_t) );
- p_filter->p_module = module_Need( p_filter,
- "visualization",
+ p_filter->p_module = module_need( p_filter,
+ "visualization2",
psz_parser, true );
}
}
msg_Err( p_aout, "cannot add user filter %s (skipped)",
psz_parser );
- vlc_object_detach( p_filter );
+ free( p_filter->p_owner );
vlc_object_release( p_filter );
psz_parser = psz_next;
}
/* complete the filter chain if necessary */
- if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &p_filter->input ) )
+ 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->input ) < 0 )
+ &p_filter->fmt_in.audio ) < 0 )
{
msg_Err( p_aout, "cannot add user filter %s (skipped)",
psz_parser );
- module_Unneed( p_filter, p_filter->p_module );
- vlc_object_detach( p_filter );
+ module_unneed( p_filter, p_filter->p_module );
+ free( p_filter->p_owner );
vlc_object_release( p_filter );
psz_parser = psz_next;
}
/* success */
- p_filter->b_continuity = false;
p_input->pp_filters[p_input->i_nb_filters++] = p_filter;
- memcpy( &chain_input_format, &p_filter->output,
+ memcpy( &chain_input_format, &p_filter->fmt_out.audio,
sizeof( audio_sample_format_t ) );
+ if( i_visual == 0 ) /* scaletempo */
+ p_input->p_playback_rate_filter = p_filter;
+
/* next filter if any */
psz_parser = psz_next;
}
}
/* Prepare hints for the buffer allocator. */
- p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
+ 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.mixer ) )
+ if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer_format ) )
{
chain_output_format.i_rate = (__MAX(p_input->input.i_rate,
- p_aout->mixer.mixer.i_rate)
+ p_aout->mixer_format.i_rate)
* (100 + AOUT_MAX_RESAMPLING)) / 100;
- if ( chain_output_format.i_rate == p_aout->mixer.mixer.i_rate )
+ if ( chain_output_format.i_rate == p_aout->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.mixer ) < 0 )
+ &p_aout->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.i_alloc_type = AOUT_ALLOC_HEAP;
+ p_input->input_alloc.b_alloc = true;
/* Setup the initial rate of the resampler */
- p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate;
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->input.i_rate;
}
p_input->i_resampling_type = AOUT_RESAMPLING_NONE;
- p_input->p_playback_rate_filter = NULL;
- for( int i = 0; i < p_input->i_nb_filters; i++ )
- {
- aout_filter_t *p_filter = p_input->pp_filters[i];
- if( strcmp( "scaletempo", p_filter->psz_object_name ) == 0 )
- {
- p_input->p_playback_rate_filter = p_filter;
- break;
- }
- }
if( ! p_input->p_playback_rate_filter && p_input->i_nb_resamplers > 0 )
{
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.i_alloc_type = AOUT_ALLOC_HEAP;
+ 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(
/* Success */
p_input->b_error = false;
- p_input->b_restart = false;
p_input->i_last_input_rate = INPUT_RATE_DEFAULT;
return 0;
int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
{
AOUT_ASSERT_MIXER_LOCKED;
- if ( p_input->b_error ) return 0;
+ if ( p_input->b_error )
+ return 0;
+
+ /* XXX We need to update b_recycle_vout before calling aout_FiltersDestroyPipeline.
+ * FIXME They can be a race condition if audio-visual is updated between
+ * aout_InputDelete and aout_InputNew.
+ */
+ char *psz_visual = var_GetString( p_aout, "audio-visual");
+ 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_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers );
p_input->i_nb_resamplers = 0;
- aout_FifoDestroy( p_aout, &p_input->fifo );
+ aout_FifoDestroy( p_aout, &p_input->mixer.fifo );
return 0;
}
+/*****************************************************************************
+ * aout_InputCheckAndRestart : restart an input
+ *****************************************************************************
+ * This function must be entered with the input and mixer lock.
+ *****************************************************************************/
+void aout_InputCheckAndRestart( aout_instance_t * p_aout, aout_input_t * p_input )
+{
+ AOUT_ASSERT_MIXER_LOCKED;
+ AOUT_ASSERT_INPUT_LOCKED;
+
+ if( !p_input->b_restart )
+ return;
+
+ aout_lock_input_fifos( p_aout );
+
+ /* A little trick to avoid loosing our input fifo and properties */
+
+ uint8_t *p_first_byte_to_mix = p_input->mixer.begin;
+ aout_fifo_t fifo = p_input->mixer.fifo;
+ bool b_paused = p_input->b_paused;
+ 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_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;
+
+ p_input->b_restart = false;
+
+ aout_unlock_input_fifos( p_aout );
+}
/*****************************************************************************
* aout_InputPlay : play a buffer
*****************************************************************************
mtime_t start_date;
AOUT_ASSERT_INPUT_LOCKED;
- if( p_input->b_restart )
- {
- aout_fifo_t fifo, dummy_fifo;
- uint8_t *p_first_byte_to_mix;
-
- aout_lock_mixer( p_aout );
- aout_lock_input_fifos( p_aout );
-
- /* 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;
-
- aout_unlock_input_fifos( p_aout );
- aout_unlock_mixer( p_aout );
- }
-
if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL )
{
- inputDrop( p_aout, p_input, p_buffer );
+ inputDrop( p_input, p_buffer );
return 0;
}
/* 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->start_date;
+ 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 )
{
- aout_BufferFree( p_buffer );
+ 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->input.i_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);
* 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->fifo );
+ start_date = aout_FifoNextStart( p_aout, &p_input->mixer.fifo );
aout_unlock_input_fifos( p_aout );
if ( start_date != 0 && start_date < mdate() )
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->fifo, 0 );
- p_input->p_first_byte_to_mix = NULL;
+ 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->start_date < mdate() + AOUT_MIN_PREPARE_TIME )
+ 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->start_date );
+ mdate() - p_buffer->i_pts );
- inputDrop( p_aout, p_input, p_buffer );
+ inputDrop( p_input, p_buffer );
inputResamplingStop( p_input );
return 0;
}
* 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->start_date - i_pts_tolerance ) )
+ ( 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->start_date );
+ start_date - p_buffer->i_pts );
aout_lock_input_fifos( p_aout );
- aout_FifoSet( p_aout, &p_input->fifo, 0 );
- p_input->p_first_byte_to_mix = NULL;
+ 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->start_date + i_pts_tolerance) )
+ ( 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->start_date );
- inputDrop( p_aout, p_input, p_buffer );
+ start_date - p_buffer->i_pts );
+ inputDrop( p_input, p_buffer );
return 0;
}
- if ( start_date == 0 ) start_date = p_buffer->start_date;
+ 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 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->start_date - AOUT_PTS_TOLERANCE
- || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) &&
+ ( 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 :
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
- mtime_t drift = p_buffer->start_date - start_date;
+ 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( p_input->i_resampling_type == AOUT_RESAMPLING_UP )
{
- p_input->pp_resamplers[0]->input.i_rate += 2; /* Hz */
+ p_input->pp_resamplers[0]->fmt_in.audio.i_rate += 2; /* Hz */
}
else
{
- p_input->pp_resamplers[0]->input.i_rate -= 2; /* Hz */
+ 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
(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]->input.i_rate == i_nominal_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->start_date - start_date);
+ p_buffer->i_pts - start_date);
}
- else if( abs( (int)(p_buffer->start_date - 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
p_input->i_resamp_start_drift = 0;
}
else if( p_input->i_resamp_start_drift &&
- ( abs( (int)(p_buffer->start_date - start_date) ) >
+ ( 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;
}
}
/* 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 0;
if( p_buffer->i_nb_samples <= 0 )
{
- aout_BufferFree( p_buffer );
+ block_Release( p_buffer );
return 0;
}
#endif
/* 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;
+ p_buffer->i_pts = start_date;
aout_lock_input_fifos( p_aout );
- aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
+ aout_FifoPush( p_aout, &p_input->mixer.fifo, p_buffer );
aout_unlock_input_fifos( p_aout );
return 0;
}
p_input->i_nb_filters );
aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers );
- aout_FifoDestroy( p_aout, &p_input->fifo );
+ aout_FifoDestroy( p_aout, &p_input->mixer.fifo );
var_Destroy( p_aout, "visual" );
var_Destroy( p_aout, "equalizer" );
var_Destroy( p_aout, "audio-filter" );
p_input->b_error = 1;
}
-static void inputDrop( aout_instance_t *p_aout, aout_input_t *p_input, aout_buffer_t *p_buffer )
+static void inputDrop( aout_input_t *p_input, aout_buffer_t *p_buffer )
{
aout_BufferFree( p_buffer );
- if( !p_input->p_input_thread )
- return;
-
- vlc_mutex_lock( &p_input->p_input_thread->p->counters.counters_lock);
- stats_UpdateInteger( p_aout, p_input->p_input_thread->p->counters.p_lost_abuffers, 1, NULL );
- vlc_mutex_unlock( &p_input->p_input_thread->p->counters.counters_lock);
+ p_input->i_buffer_lost++;
}
static void inputResamplingStop( aout_input_t *p_input )
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->pp_resamplers[0]->fmt_in.audio.i_rate =
( 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;
+ VLC_UNUSED(b_recycle);
+ 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( 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;
+ aout_request_vout_t *p_request = &p_input->request_vout;
+
+ /* XXX: this only works from audio input */
+ /* If you want to use visualization filters from another place, you will
+ * need to add a new pf_aout_request_vout callback or store a pointer
+ * to aout_request_vout_t inside filter_t (i.e. a level of indirection). */
+
+ return p_request->pf_request_vout( p_request->p_private,
+ p_vout, p_fmt, p_input->b_recycle_vout );
+}
+
static int ChangeFiltersString( aout_instance_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,
{
aout_instance_t *p_aout = (aout_instance_t *)p_this;
char *psz_mode = newval.psz_string;
- vlc_value_t val;
(void)psz_cmd; (void)oldval; (void)p_data;
if( !psz_mode || !*psz_mode )
{
ChangeFiltersString( p_aout, "audio-visual", "goom", false );
ChangeFiltersString( p_aout, "audio-visual", "visual", false );
- ChangeFiltersString( p_aout, "audio-visual", "galaktos", false );
+ ChangeFiltersString( p_aout, "audio-visual", "projectm", false );
}
else
{
{
ChangeFiltersString( p_aout, "audio-visual", "visual", false );
ChangeFiltersString( p_aout, "audio-visual", "goom", true );
- ChangeFiltersString( p_aout, "audio-visual", "galaktos", false);
+ ChangeFiltersString( p_aout, "audio-visual", "projectm", false );
}
- else if( !strcmp( "galaktos", psz_mode ) )
+ else if( !strcmp( "projectm", psz_mode ) )
{
ChangeFiltersString( p_aout, "audio-visual", "visual", false );
ChangeFiltersString( p_aout, "audio-visual", "goom", false );
- ChangeFiltersString( p_aout, "audio-visual", "galaktos", true );
+ ChangeFiltersString( p_aout, "audio-visual", "projectm", true );
}
else
{
- val.psz_string = psz_mode;
var_Create( p_aout, "effect-list", VLC_VAR_STRING );
- var_Set( p_aout, "effect-list", val );
+ var_SetString( p_aout, "effect-list", psz_mode );
ChangeFiltersString( p_aout, "audio-visual", "goom", false );
ChangeFiltersString( p_aout, "audio-visual", "visual", true );
- ChangeFiltersString( p_aout, "audio-visual", "galaktos", false);
+ ChangeFiltersString( p_aout, "audio-visual", "projectm", false );
}
}
{
aout_instance_t *p_aout = (aout_instance_t *)p_this;
char *psz_mode = newval.psz_string;
- vlc_value_t val;
int i_ret;
(void)psz_cmd; (void)oldval; (void)p_data;
}
else
{
- val.psz_string = psz_mode;
var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING );
- var_Set( p_aout, "equalizer-preset", val );
+ var_SetString( p_aout, "equalizer-preset", psz_mode );
i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer",
true );
-
}
/* That sucks */
ReplayGainSelect( p_aout, p_aout->pp_inputs[i] );
/* Restart the mixer (a trivial mixer may be in use) */
- aout_MixerMultiplierSet( p_aout, p_aout->mixer.f_multiplier );
+ if( p_aout->p_mixer )
+ aout_MixerMultiplierSet( p_aout, p_aout->mixer_multiplier );
aout_unlock_mixer( p_aout );
return VLC_SUCCESS;
int i_use;
float f_gain;
- p_input->f_multiplier = 1.0;
+ p_input->mixer.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->f_multiplier = pow( 10.0, f_gain / 20.0 );
+ p_input->mixer.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->f_multiplier > 1.0 )
+ p_input->replay_gain.pf_peak[i_use] * p_input->mixer.multiplier > 1.0 )
{
- p_input->f_multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use];
+ p_input->mixer.multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use];
}
free( psz_replay_gain );
projects / vlc / blobdiff