X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=8d6c7a050ee78f20200ba439e9d15cb95ecc21d1;hb=11aaa50ff15c954163b1d03a314dad116f0b1984;hp=ac4ff722a6a68f650658aa3b66dd77e851b2e3f4;hpb=51fa2629c7ed5183f2a4ca37e0a6d88d5ac46ca3;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index ac4ff722a6..8d6c7a050e 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -1,8 +1,8 @@ /***************************************************************************** * input.c : internal management of input streams for the audio output ***************************************************************************** - * Copyright (C) 2002 VideoLAN - * $Id: input.c,v 1.16 2002/10/09 22:54:22 massiot Exp $ + * Copyright (C) 2002-2004 VideoLAN + * $Id$ * * Authors: Christophe Massiot * @@ -10,7 +10,7 @@ * 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 @@ -28,6 +28,7 @@ #include #include +#include /* for input_thread_t and i_pts_delay */ #ifdef HAVE_ALLOCA_H # include @@ -36,33 +37,221 @@ #include "audio_output.h" #include "aout_internal.h" +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 aout_filter_t * allocateUserChannelMixer( aout_instance_t *, + audio_sample_format_t *, + audio_sample_format_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 intermediate_format; + audio_sample_format_t user_filter_format; + audio_sample_format_t intermediate_format;/* input of resampler */ + vlc_value_t val, text; + char * psz_filters; + aout_filter_t * p_user_channel_mixer; + + aout_FormatPrint( p_aout, "input", &p_input->input ); /* Prepare FIFO. */ aout_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate ); p_input->p_first_byte_to_mix = NULL; - /* Create filters. */ + /* Prepare format structure */ memcpy( &intermediate_format, &p_aout->mixer.mixer, sizeof(audio_sample_format_t) ); intermediate_format.i_rate = p_input->input.i_rate; + + /* Try to use the channel mixer chosen by the user */ + memcpy ( &user_filter_format, &intermediate_format, + sizeof(audio_sample_format_t) ); + user_filter_format.i_physical_channels = p_input->input.i_physical_channels; + user_filter_format.i_original_channels = p_input->input.i_original_channels; + user_filter_format.i_bytes_per_frame = user_filter_format.i_bytes_per_frame + * aout_FormatNbChannels( &user_filter_format ) + / aout_FormatNbChannels( &intermediate_format ); + p_user_channel_mixer = allocateUserChannelMixer( p_aout, &user_filter_format, + &intermediate_format ); + /* If it failed, let the main pipeline do channel mixing */ + if ( ! p_user_channel_mixer ) + { + memcpy ( &user_filter_format, &intermediate_format, + sizeof(audio_sample_format_t) ); + } + + /* Create filters. */ if ( aout_FiltersCreatePipeline( p_aout, p_input->pp_filters, - &p_input->i_nb_filters, &p_input->input, - &intermediate_format ) < 0 ) + &p_input->i_nb_filters, + &p_input->input, + &user_filter_format + ) < 0 ) { msg_Err( p_aout, "couldn't set an input pipeline" ); aout_FifoDestroy( p_aout, &p_input->fifo ); p_input->b_error = 1; - return -1; } + /* Now add user filters */ + if( var_Type( p_aout, "visual" ) == 0 ) + { + module_t *p_module; + 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 = ""; text.psz_string = _("Disable"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = "random"; text.psz_string = _("Random"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = "scope"; text.psz_string = _("Scope"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + val.psz_string = "spectrum"; text.psz_string = _("Spectrum"); + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + + /* Look for goom plugin */ + p_module = config_FindModule( VLC_OBJECT(p_aout), "goom" ); + if( p_module ) + { + val.psz_string = "goom"; text.psz_string = "Goom"; + var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); + } + + /* Look for galaktos plugin */ + p_module = config_FindModule( VLC_OBJECT(p_aout), "galaktos" ); + if( p_module ) + { + val.psz_string = "galaktos"; text.psz_string = "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 ); + if( val.psz_string ) 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 = ""; 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 = p_config->ppsz_list[i]; + text.psz_string = 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 ); + } + + var_Get( p_aout, "audio-filter", &val ); + psz_filters = val.psz_string; + if( psz_filters && *psz_filters ) + { + char *psz_parser = psz_filters; + char *psz_next; + + while( psz_parser && *psz_parser ) + { + aout_filter_t * p_filter; + + if( p_input->i_nb_filters >= AOUT_MAX_FILTERS ) + { + msg_Dbg( p_aout, "max filter 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; + } + + msg_Dbg( p_aout, "user filter \"%s\"", psz_parser ); + + /* Create a VLC object */ + p_filter = vlc_object_create( p_aout, sizeof(aout_filter_t) ); + 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 ); + memcpy( &p_filter->input, &user_filter_format, + sizeof(audio_sample_format_t) ); + memcpy( &p_filter->output, &user_filter_format, + sizeof(audio_sample_format_t) ); + + p_filter->p_module = + module_Need( p_filter,"audio filter", psz_parser, VLC_TRUE ); + + 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_destroy( p_filter ); + psz_parser = psz_next; + continue; + + } + p_filter->b_continuity = VLC_FALSE; + + p_input->pp_filters[p_input->i_nb_filters++] = p_filter; + + /* next filter if any */ + psz_parser = psz_next; + } + } + if( psz_filters ) free( psz_filters ); + + /* Attach the user channel mixer */ + if ( p_user_channel_mixer ) + { + p_input->pp_filters[p_input->i_nb_filters++] = p_user_channel_mixer; + } + /* 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; @@ -74,7 +263,8 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) else { /* Create resamplers. */ - intermediate_format.i_rate = (p_input->input.i_rate + intermediate_format.i_rate = (__MAX(p_input->input.i_rate, + p_aout->mixer.mixer.i_rate) * (100 + AOUT_MAX_RESAMPLING)) / 100; if ( intermediate_format.i_rate == p_aout->mixer.mixer.i_rate ) { @@ -91,6 +281,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters, p_input->i_nb_filters ); aout_FifoDestroy( p_aout, &p_input->fifo ); + var_Destroy( p_aout, "visual" ); p_input->b_error = 1; return -1; @@ -99,10 +290,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) aout_FiltersHintBuffers( p_aout, p_input->pp_resamplers, p_input->i_nb_resamplers, &p_input->input_alloc ); + + /* 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->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; - p_input->input_alloc.i_bytes_per_sec = -1; aout_FiltersHintBuffers( p_aout, p_input->pp_filters, p_input->i_nb_filters, &p_input->input_alloc ); @@ -110,13 +304,14 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) /* 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, - p_input->input.i_bytes_per_frame + (int)(p_input->input.i_bytes_per_frame * p_input->input.i_rate - / p_input->input.i_frame_length ); + / p_input->input.i_frame_length) ); /* Allocate in the heap, it is more convenient for the decoder. */ p_input->input_alloc.i_alloc_type = AOUT_ALLOC_HEAP; - p_input->b_error = 0; + p_input->b_error = VLC_FALSE; + p_input->b_restart = VLC_FALSE; return 0; } @@ -147,7 +342,27 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, aout_buffer_t * p_buffer ) { - mtime_t start_date, duration; + mtime_t start_date; + + if( p_input->b_restart ) + { + aout_fifo_t fifo, dummy_fifo; + byte_t *p_first_byte_to_mix; + + vlc_mutex_lock( &p_aout->mixer_lock ); + + /* 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; + + vlc_mutex_unlock( &p_aout->mixer_lock ); + } /* 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 @@ -161,33 +376,82 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* 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 (%lld), clearing out", - start_date ); + msg_Warn( p_aout, "computed PTS is out of range ("I64Fd"), " + "clearing out", mdate() - start_date ); vlc_mutex_lock( &p_aout->input_fifos_lock ); aout_FifoSet( p_aout, &p_input->fifo, 0 ); + p_input->p_first_byte_to_mix = NULL; vlc_mutex_unlock( &p_aout->input_fifos_lock ); + if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) + msg_Warn( p_aout, "timing screwed, stopping resampling" ); + 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->input.i_rate; + p_input->pp_resamplers[0]->b_continuity = VLC_FALSE; + } 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 (%lld), dropping buffer", + msg_Warn( p_aout, "PTS is out of range ("I64Fd"), dropping buffer", mdate() - p_buffer->start_date ); aout_BufferFree( p_buffer ); + 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->input.i_rate; + p_input->pp_resamplers[0]->b_continuity = VLC_FALSE; + } + return 0; + } + /* If the audio drift is too big then it's not worth trying to resample + * the audio. */ + if ( start_date != 0 && + ( start_date < p_buffer->start_date - 3 * AOUT_PTS_TOLERANCE ) ) + { + msg_Warn( p_aout, "audio drift is too big ("I64Fd"), clearing out", + start_date - p_buffer->start_date ); + vlc_mutex_lock( &p_aout->input_fifos_lock ); + aout_FifoSet( p_aout, &p_input->fifo, 0 ); + p_input->p_first_byte_to_mix = NULL; + vlc_mutex_unlock( &p_aout->input_fifos_lock ); + if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) + msg_Warn( p_aout, "timing screwed, stopping resampling" ); + 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->input.i_rate; + p_input->pp_resamplers[0]->b_continuity = VLC_FALSE; + } + start_date = 0; + } + else if ( start_date != 0 && + ( start_date > p_buffer->start_date + 3 * AOUT_PTS_TOLERANCE ) ) + { + msg_Warn( p_aout, "audio drift is too big ("I64Fd"), dropping buffer", + start_date - p_buffer->start_date ); + aout_BufferFree( p_buffer ); return 0; } if ( start_date == 0 ) start_date = p_buffer->start_date; /* Run pre-filters. */ + aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters, &p_buffer ); - if ( start_date < p_buffer->start_date - AOUT_PTS_TOLERANCE - || start_date > p_buffer->start_date + AOUT_PTS_TOLERANCE ) + /* 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) @@ -196,54 +460,261 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * synchronization * Solution : resample the buffer to avoid a scratch. */ - int i_ratio; - mtime_t old_duration; mtime_t drift = p_buffer->start_date - start_date; - msg_Warn( p_aout, "buffer is %lld %s, resampling", - drift > 0 ? drift : -drift, - drift > 0 ? "in advance" : "late" ); - old_duration = p_buffer->end_date - p_buffer->start_date; - duration = p_buffer->end_date - start_date; - i_ratio = (duration * 100) / old_duration; - /* If the ratio is too != 100, the sound quality will be awful. */ - if ( i_ratio < 100 - AOUT_MAX_RESAMPLING /* % */ ) + 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 "I64Fd" %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 ) { - duration = (old_duration * (100 - AOUT_MAX_RESAMPLING)) / 100; + p_input->pp_resamplers[0]->input.i_rate += 1; /* Hz */ } - if ( i_ratio > 100 + AOUT_MAX_RESAMPLING /* % */ ) + else { - duration = (old_duration * (100 + AOUT_MAX_RESAMPLING)) / 100; + p_input->pp_resamplers[0]->input.i_rate -= 1; /* Hz */ } - p_input->pp_resamplers[0]->input.i_rate - = (p_input->input.i_rate * old_duration) / duration; + /* Check if everything is back to normal, in which case we can stop the + * resampling */ + if( p_input->pp_resamplers[0]->input.i_rate == + p_input->input.i_rate ) + { + p_input->i_resampling_type = AOUT_RESAMPLING_NONE; + msg_Warn( p_aout, "resampling stopped after "I64Fi" usec " + "(drift: "I64Fi")", + 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" ); + p_input->i_resampling_type = AOUT_RESAMPLING_NONE; + p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate; + } + } + + /* 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; + + /* 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 ); } + + vlc_mutex_lock( &p_aout->input_fifos_lock ); + aout_FifoPush( p_aout, &p_input->fifo, p_buffer ); + vlc_mutex_unlock( &p_aout->input_fifos_lock ); + + return 0; +} + +static int ChangeFiltersString( aout_instance_t * p_aout, + char *psz_name, vlc_bool_t b_add ) +{ + vlc_value_t val; + char *psz_parser; + + var_Get( p_aout, "audio-filter", &val ); + + if( !val.psz_string ) val.psz_string = strdup(""); + + psz_parser = strstr( val.psz_string, psz_name ); + + if( b_add ) + { + if( !psz_parser ) + { + psz_parser = val.psz_string; + asprintf( &val.psz_string, (*val.psz_string) ? "%s,%s" : "%s%s", + val.psz_string, psz_name ); + free( psz_parser ); + } + else + { + return 0; + } + } else { - duration = p_buffer->end_date - p_buffer->start_date; + if( psz_parser ) + { + memmove( psz_parser, psz_parser + strlen(psz_name) + + (*(psz_parser + strlen(psz_name)) == ',' ? 1 : 0 ), + strlen(psz_parser + strlen(psz_name)) + 1 ); + } + else + { + free( val.psz_string ); + return 0; + } + } - if ( p_input->input.i_rate != p_aout->mixer.mixer.i_rate ) + var_Set( p_aout, "audio-filter", val ); + free( val.psz_string ); + return 1; +} + +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; + int i; + + if( !psz_mode || !*psz_mode ) + { + ChangeFiltersString( p_aout, "goom", VLC_FALSE ); + ChangeFiltersString( p_aout, "visual", VLC_FALSE ); + ChangeFiltersString( p_aout, "galaktos", VLC_FALSE ); + } + else + { + if( !strcmp( "goom", psz_mode ) ) { - /* Standard resampling is needed ! */ - p_input->pp_resamplers[0]->input.i_rate = p_input->input.i_rate; + ChangeFiltersString( p_aout, "visual", VLC_FALSE ); + ChangeFiltersString( p_aout, "goom", VLC_TRUE ); + ChangeFiltersString( p_aout, "galaktos", VLC_FALSE ); + } + else if( !strcmp( "galaktos", psz_mode ) ) + { + ChangeFiltersString( p_aout, "visual", VLC_FALSE ); + ChangeFiltersString( p_aout, "goom", VLC_FALSE ); + ChangeFiltersString( p_aout, "galaktos", VLC_TRUE ); + } + else + { + val.psz_string = psz_mode; + var_Create( p_aout, "effect-list", VLC_VAR_STRING ); + var_Set( p_aout, "effect-list", val ); - aout_FiltersPlay( p_aout, p_input->pp_resamplers, - p_input->i_nb_resamplers, - &p_buffer ); + ChangeFiltersString( p_aout, "goom", VLC_FALSE ); + ChangeFiltersString( p_aout, "visual", VLC_TRUE ); + ChangeFiltersString( p_aout, "galaktos", VLC_FALSE ); } } - /* Adding the start date will be managed by aout_FifoPush(). */ - p_buffer->start_date = start_date; - p_buffer->end_date = start_date + duration; + /* That sucks */ + for( i = 0; i < p_aout->i_nb_inputs; i++ ) + { + p_aout->pp_inputs[i]->b_restart = VLC_TRUE; + } - vlc_mutex_lock( &p_aout->input_fifos_lock ); - aout_FifoPush( p_aout, &p_input->fifo, p_buffer ); - vlc_mutex_unlock( &p_aout->input_fifos_lock ); + return VLC_SUCCESS; +} - return 0; +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; + int i_ret; + + if( !psz_mode || !*psz_mode ) + { + i_ret = ChangeFiltersString( p_aout, "equalizer", VLC_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, "equalizer", VLC_TRUE ); + + } + + /* That sucks */ + if( i_ret == 1 ) + { + for( i = 0; i < p_aout->i_nb_inputs; i++ ) + { + p_aout->pp_inputs[i]->b_restart = VLC_TRUE; + } + } + + return VLC_SUCCESS; +} + +static aout_filter_t * allocateUserChannelMixer( aout_instance_t * p_aout, + audio_sample_format_t * p_input_format, + audio_sample_format_t * p_output_format ) +{ + aout_filter_t * p_channel_mixer; + + /* Retreive user preferred channel mixer */ + char * psz_name = config_GetPsz( p_aout, "audio-channel-mixer" ); + + /* Not specified => let the main pipeline do the mixing */ + if ( ! psz_name ) return NULL; + + /* Debug information */ + aout_FormatsPrint( p_aout, "channel mixer", p_input_format, + p_output_format ); + + /* Create a VLC object */ + p_channel_mixer = vlc_object_create( p_aout, sizeof(aout_filter_t) ); + if( p_channel_mixer == NULL ) + { + msg_Err( p_aout, "cannot add user channel mixer %s", psz_name ); + return NULL; + } + vlc_object_attach( p_channel_mixer , p_aout ); + + /* Attach the suitable module */ + memcpy( &p_channel_mixer->input, p_input_format, + sizeof(audio_sample_format_t) ); + memcpy( &p_channel_mixer->output, p_output_format, + sizeof(audio_sample_format_t) ); + p_channel_mixer->p_module = + module_Need( p_channel_mixer,"audio filter", psz_name, VLC_TRUE ); + if( p_channel_mixer->p_module== NULL ) + { + msg_Err( p_aout, "cannot add user channel mixer %s", psz_name ); + vlc_object_detach( p_channel_mixer ); + vlc_object_destroy( p_channel_mixer ); + return NULL; + } + p_channel_mixer->b_continuity = VLC_FALSE; + + /* Ok */ + return p_channel_mixer; }