/*****************************************************************************
* input.c : internal management of input streams for the audio output
*****************************************************************************
* Copyright (C) 2002-2007 the VideoLAN team
* $Id$
*
* 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
* (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.
*
* 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.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "vlc_common.h"
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <assert.h>
#include "vlc_input.h" /* for input_thread_t and i_pts_delay */
#ifdef HAVE_ALLOCA_H
# include <alloca.h>
#endif
#include "vlc_aout.h"
#include "aout_internal.h"
/** FIXME: Ugly but needed to access the counters */
#include "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 inputResamplingStop( aout_input_t *p_input );
static int VisualizationCallback( vlc_object_t *, char const *,
vlc_value_t, vlc_value_t, void * );
static int EqualizerCallback( vlc_object_t *, char const *,
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 * );
/*****************************************************************************
* aout_InputNew : allocate a new input and rework the filter pipeline
*****************************************************************************/
int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
{
audio_sample_format_t chain_input_format;
audio_sample_format_t chain_output_format;
vlc_value_t val, text;
char * psz_filters, *psz_visual;
int i_visual;
aout_FormatPrint( p_aout, "input", &p_input->input );
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;
/* 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_output_format.i_rate = p_input->input.i_rate;
aout_FormatPrepare( &chain_output_format );
/* Now add user filters */
if( var_Type( p_aout, "visual" ) == 0 )
{
var_Create( p_aout, "visual", VLC_VAR_STRING | VLC_VAR_HASCHOICE );
text.psz_string = _("Visualizations");
var_Change( p_aout, "visual", VLC_VAR_SETTEXT, &text, NULL );
val.psz_string = (char*)""; text.psz_string = _("Disable");
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
val.psz_string = (char*)"spectrometer"; text.psz_string = _("Spectrometer");
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
val.psz_string = (char*)"scope"; text.psz_string = _("Scope");
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
val.psz_string = (char*)"spectrum"; text.psz_string = _("Spectrum");
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
val.psz_string = (char*)"vuMeter"; text.psz_string = _("Vu meter");
var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text );
/* Look for goom plugin */
if( module_Exists( VLC_OBJECT(p_aout), "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" ) )
{
val.psz_string = (char*)"galaktos"; text.psz_string = (char*)"GaLaktos";
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 );
free( val.psz_string );
}
var_AddCallback( p_aout, "visual", VisualizationCallback, NULL );
}
if( var_Type( p_aout, "equalizer" ) == 0 )
{
module_config_t *p_config;
int i;
p_config = config_FindConfig( VLC_OBJECT(p_aout), "equalizer-preset" );
if( p_config && p_config->i_list )
{
var_Create( p_aout, "equalizer",
VLC_VAR_STRING | VLC_VAR_HASCHOICE );
text.psz_string = _("Equalizer");
var_Change( p_aout, "equalizer", VLC_VAR_SETTEXT, &text, NULL );
val.psz_string = (char*)""; text.psz_string = _("Disable");
var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE, &val, &text );
for( i = 0; i < p_config->i_list; i++ )
{
val.psz_string = (char *)p_config->ppsz_list[i];
text.psz_string = (char *)p_config->ppsz_list_text[i];
var_Change( p_aout, "equalizer", VLC_VAR_ADDCHOICE,
&val, &text );
}
var_AddCallback( p_aout, "equalizer", EqualizerCallback, NULL );
}
}
if( var_Type( p_aout, "audio-filter" ) == 0 )
{
var_Create( p_aout, "audio-filter",
VLC_VAR_STRING | VLC_VAR_DOINHERIT );
text.psz_string = _("Audio filters");
var_Change( p_aout, "audio-filter", VLC_VAR_SETTEXT, &text, NULL );
}
if( var_Type( p_aout, "audio-visual" ) == 0 )
{
var_Create( p_aout, "audio-visual",
VLC_VAR_STRING | VLC_VAR_DOINHERIT );
text.psz_string = _("Audio visualizations");
var_Change( p_aout, "audio-visual", VLC_VAR_SETTEXT, &text, NULL );
}
if( var_Type( p_aout, "audio-replay-gain-mode" ) == 0 )
{
module_config_t *p_config;
int i;
p_config = config_FindConfig( VLC_OBJECT(p_aout), "audio-replay-gain-mode" );
if( p_config && p_config->i_list )
{
var_Create( p_aout, "audio-replay-gain-mode",
VLC_VAR_STRING | VLC_VAR_DOINHERIT );
text.psz_string = _("Replay gain");
var_Change( p_aout, "audio-replay-gain-mode", VLC_VAR_SETTEXT, &text, NULL );
for( i = 0; i < p_config->i_list; i++ )
{
val.psz_string = (char *)p_config->ppsz_list[i];
text.psz_string = (char *)p_config->ppsz_list_text[i];
var_Change( p_aout, "audio-replay-gain-mode", VLC_VAR_ADDCHOICE,
&val, &text );
}
var_AddCallback( p_aout, "audio-replay-gain-mode", ReplayGainCallback, NULL );
}
}
if( var_Type( p_aout, "audio-replay-gain-preamp" ) == 0 )
{
var_Create( p_aout, "audio-replay-gain-preamp",
VLC_VAR_FLOAT | VLC_VAR_DOINHERIT );
}
if( var_Type( p_aout, "audio-replay-gain-default" ) == 0 )
{
var_Create( p_aout, "audio-replay-gain-default",
VLC_VAR_FLOAT | VLC_VAR_DOINHERIT );
}
if( var_Type( p_aout, "audio-replay-gain-peak-protection" ) == 0 )
{
var_Create( p_aout, "audio-replay-gain-peak-protection",
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;
/* parse user filter lists */
for( i_visual = 0; i_visual < 2; i_visual++ )
{
char *psz_next = NULL;
char *psz_parser = i_visual ? psz_visual : psz_filters;
if( psz_parser == NULL || !*psz_parser )
continue;
while( psz_parser && *psz_parser )
{
aout_filter_t * p_filter = NULL;
if( p_input->i_nb_filters >= AOUT_MAX_FILTERS )
{
msg_Dbg( p_aout, "max filters reached (%d)", AOUT_MAX_FILTERS );
break;
}
while( *psz_parser == ' ' && *psz_parser == ':' )
{
psz_parser++;
}
if( ( psz_next = strchr( psz_parser , ':' ) ) )
{
*psz_next++ = '\0';
}
if( *psz_parser =='\0' )
{
break;
}
/* Create a VLC object */
static const char typename[] = "audio filter";
p_filter = vlc_custom_create( p_aout, sizeof(*p_filter),
VLC_OBJECT_GENERIC, typename );
if( p_filter == NULL )
{
msg_Err( p_aout, "cannot add user filter %s (skipped)",
psz_parser );
psz_parser = psz_next;
continue;
}
vlc_object_attach( p_filter , p_aout );
/* try to find the requested filter */
if( i_visual == 1 ) /* 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",
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",
psz_parser, true );
if ( p_filter->p_module == NULL )
{
/* if the filter requested a special format, retry */
if ( !( AOUT_FMTS_IDENTICAL( &p_filter->input,
&chain_input_format )
&& AOUT_FMTS_IDENTICAL( &p_filter->output,
&chain_output_format ) ) )
{
aout_FormatPrepare( &p_filter->input );
aout_FormatPrepare( &p_filter->output );
p_filter->p_module = module_Need( p_filter,
"audio filter",
psz_parser, true );
}
/* try visual filters */
else
{
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",
psz_parser, true );
}
}
}
/* failure */
if ( p_filter->p_module == NULL )
{
msg_Err( p_aout, "cannot add user filter %s (skipped)",
psz_parser );
vlc_object_detach( p_filter );
vlc_object_release( p_filter );
psz_parser = psz_next;
continue;
}
/* complete the filter chain if necessary */
if ( !AOUT_FMTS_IDENTICAL( &chain_input_format, &p_filter->input ) )
{
if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters,
&p_input->i_nb_filters,
&chain_input_format,
&p_filter->input ) < 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 );
vlc_object_release( p_filter );
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->output,
sizeof( audio_sample_format_t ) );
/* next filter if any */
psz_parser = psz_next;
}
}
free( psz_filters );
free( psz_visual );
/* 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 )
{
inputFailure( p_aout, p_input, "couldn't set an input pipeline" );
return -1;
}
}
/* Prepare hints for the buffer allocator. */
p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP;
p_input->input_alloc.i_bytes_per_sec = -1;
/* Create resamplers. */
if ( !AOUT_FMT_NON_LINEAR( &p_aout->mixer.mixer ) )
{
chain_output_format.i_rate = (__MAX(p_input->input.i_rate,
p_aout->mixer.mixer.i_rate)
* (100 + AOUT_MAX_RESAMPLING)) / 100;
if ( chain_output_format.i_rate == p_aout->mixer.mixer.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 )
{
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;
/* Setup the initial rate of the resampler */
p_input->pp_resamplers[0]->input.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;
/* 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;
}
/*****************************************************************************
* aout_InputDelete : delete an input
*****************************************************************************
* This function must be entered with the mixer lock.
*****************************************************************************/
int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
{
AOUT_ASSERT_MIXER_LOCKED;
if ( p_input->b_error ) return 0;
aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters,
p_input->i_nb_filters );
p_input->i_nb_filters = 0;
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 );
return 0;
}
/*****************************************************************************
* aout_InputPlay : play a buffer
*****************************************************************************
* This function must be entered with the input 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 )
{
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 );
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 );
/* Actually run the resampler now. */
if ( p_input->i_nb_resamplers > 0 )
{
const mtime_t i_date = p_buffer->start_date;
aout_FiltersPlay( p_aout, p_input->pp_resamplers,
p_input->i_nb_resamplers,
&p_buffer );
}
if( p_buffer->i_nb_samples <= 0 )
{
aout_BufferFree( 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;
#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->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->fifo, 0 );
p_input->p_first_byte_to_mix = 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 );
start_date = 0;
}
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 (%"PRId64"), dropping buffer",
mdate() - p_buffer->start_date );
inputDrop( p_aout, 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->start_date - i_pts_tolerance ) )
{
msg_Warn( p_aout, "audio drift is too big (%"PRId64"), clearing out",
start_date - p_buffer->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_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 );
start_date = 0;
}
else if ( start_date != 0 &&
( start_date > p_buffer->start_date + 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 );
return 0;
}
if ( start_date == 0 ) start_date = p_buffer->start_date;
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* Run pre-filters. */
aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters,
&p_buffer );
#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 ) &&
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 %"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]->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 */
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]->input.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);
}
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" );
inputResamplingStop( p_input );
}
}
#ifndef AOUT_PROCESS_BEFORE_CHEKS
/* 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 );
}
if( p_buffer->i_nb_samples <= 0 )
{
aout_BufferFree( 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;
aout_lock_input_fifos( p_aout );
aout_FifoPush( p_aout, &p_input->fifo, p_buffer );
aout_unlock_input_fifos( p_aout );
return 0;
}
/*****************************************************************************
* static functions
*****************************************************************************/
static void inputFailure( aout_instance_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,
p_input->i_nb_resamplers );
aout_FifoDestroy( p_aout, &p_input->fifo );
var_Destroy( p_aout, "visual" );
var_Destroy( p_aout, "equalizer" );
var_Destroy( p_aout, "audio-filter" );
var_Destroy( p_aout, "audio-visual" );
var_Destroy( p_aout, "audio-replay-gain-mode" );
var_Destroy( p_aout, "audio-replay-gain-default" );
var_Destroy( p_aout, "audio-replay-gain-preamp" );
var_Destroy( p_aout, "audio-replay-gain-peak-protection" );
/* error flag */
p_input->b_error = 1;
}
static void inputDrop( aout_instance_t *p_aout, 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);
}
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] == 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 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;
}
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;
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 );
}
else
{
if( !strcmp( "goom", psz_mode ) )
{
ChangeFiltersString( p_aout, "audio-visual", "visual", false );
ChangeFiltersString( p_aout, "audio-visual", "goom", true );
ChangeFiltersString( p_aout, "audio-visual", "galaktos", false);
}
else if( !strcmp( "galaktos", psz_mode ) )
{
ChangeFiltersString( p_aout, "audio-visual", "visual", false );
ChangeFiltersString( p_aout, "audio-visual", "goom", false );
ChangeFiltersString( p_aout, "audio-visual", "galaktos", true );
}
else
{
val.psz_string = psz_mode;
var_Create( p_aout, "effect-list", VLC_VAR_STRING );
var_Set( p_aout, "effect-list", val );
ChangeFiltersString( p_aout, "audio-visual", "goom", false );
ChangeFiltersString( p_aout, "audio-visual", "visual", true );
ChangeFiltersString( p_aout, "audio-visual", "galaktos", false);
}
}
/* That sucks */
AoutInputsMarkToRestart( p_aout );
return VLC_SUCCESS;
}
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;
char *psz_mode = newval.psz_string;
vlc_value_t val;
int i_ret;
(void)psz_cmd; (void)oldval; (void)p_data;
if( !psz_mode || !*psz_mode )
{
i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer",
false );
}
else
{
val.psz_string = psz_mode;
var_Create( p_aout, "equalizer-preset", VLC_VAR_STRING );
var_Set( p_aout, "equalizer-preset", val );
i_ret = ChangeFiltersString( p_aout, "audio-filter", "equalizer",
true );
}
/* That sucks */
if( i_ret == 1 )
AoutInputsMarkToRestart( p_aout );
return VLC_SUCCESS;
}
static int ReplayGainCallback( vlc_object_t *p_this, char const *psz_cmd,
vlc_value_t oldval, vlc_value_t newval, 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;
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) */
aout_MixerMultiplierSet( p_aout, p_aout->mixer.f_multiplier );
aout_unlock_mixer( p_aout );
return VLC_SUCCESS;
}
static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input )
{
char *psz_replay_gain = var_GetNonEmptyString( p_aout,
"audio-replay-gain-mode" );
int i_mode;
int i_use;
float f_gain;
p_input->f_multiplier = 1.0;
if( !psz_replay_gain )
return;
/* Find select mode */
if( !strcmp( psz_replay_gain, "track" ) )
i_mode = AUDIO_REPLAY_GAIN_TRACK;
else if( !strcmp( psz_replay_gain, "album" ) )
i_mode = AUDIO_REPLAY_GAIN_ALBUM;
else
i_mode = AUDIO_REPLAY_GAIN_MAX;
/* If the select mode is not available, prefer the other one */
i_use = i_mode;
if( i_use != AUDIO_REPLAY_GAIN_MAX && !p_input->replay_gain.pb_gain[i_use] )
{
for( i_use = 0; i_use < AUDIO_REPLAY_GAIN_MAX; i_use++ )
{
if( p_input->replay_gain.pb_gain[i_use] )
break;
}
}
/* */
if( i_use != AUDIO_REPLAY_GAIN_MAX )
f_gain = p_input->replay_gain.pf_gain[i_use] + var_GetFloat( p_aout, "audio-replay-gain-preamp" );
else if( i_mode != AUDIO_REPLAY_GAIN_MAX )
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 );
/* */
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->f_multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use];
}
free( psz_replay_gain );
}
