X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=dd279cb76171ca1e01abd1dd612844da71741a09;hb=66b2d87574e61f5bf293ffd29235819805d0fd96;hp=4a5b4c16e3d285248acc2c75439285bb3776c262;hpb=1689e7ca103e3a47bf7f7ce18d305f6f478e9bde;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index 4a5b4c16e3..dd279cb761 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -29,6 +29,8 @@ # include "config.h" #endif +#include + #include #include @@ -39,10 +41,8 @@ #include #include /* for vout_Request */ -#ifdef HAVE_ALLOCA_H -# include -#endif #include +#include #include #include "aout_internal.h" @@ -64,8 +64,6 @@ static void ReplayGainSelect( aout_instance_t *, aout_input_t * ); static vout_thread_t *RequestVout( void *, vout_thread_t *, video_format_t *, bool ); -static vout_thread_t *RequestVoutFromFilter( void *, - vout_thread_t *, video_format_t *, bool ); /***************************************************************************** * aout_InputNew : allocate a new input and rework the filter pipeline @@ -83,8 +81,8 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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 ) @@ -98,10 +96,8 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const 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 ); @@ -129,13 +125,6 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); } - /* Look for galaktos plugin */ - if( module_exists( "galaktos" ) ) - { - val.psz_string = (char*)"galaktos"; text.psz_string = (char*)"GaLaktos"; - var_Change( p_aout, "visual", VLC_VAR_ADDCHOICE, &val, &text ); - } - /* Look for libprojectM plugin */ if( module_exists( "projectm" ) ) { @@ -248,6 +237,8 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ /* 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 *psz_next = NULL; @@ -258,7 +249,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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 ) { @@ -293,46 +284,40 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ vlc_object_attach( p_filter , p_aout ); - p_filter->request_vout.pf_request_vout = RequestVoutFromFilter; - p_filter->request_vout.p_private = p_input; - 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", 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 ); + 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 ); @@ -340,12 +325,12 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ /* 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", + "visualization2", psz_parser, true ); } } @@ -366,12 +351,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ } /* 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 ); @@ -387,11 +373,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ } /* 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; } @@ -414,16 +402,16 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ } /* 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++; @@ -431,7 +419,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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; @@ -440,26 +428,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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]; - /* FIXME: suspicious access to psz_object_name */ -#warning Is this right? - if( strcmp( "scaletempo", - vlc_internals(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]; @@ -468,7 +443,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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( @@ -512,7 +487,7 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) 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; } @@ -542,18 +517,18 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* A little trick to avoid loosing our input fifo and properties */ - p_first_byte_to_mix = p_input->p_first_byte_to_mix; - fifo = p_input->fifo; + 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_FifoInit( p_aout, &p_input->fifo, p_aout->mixer.mixer.i_rate ); + 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->p_first_byte_to_mix = p_first_byte_to_mix; - p_input->fifo = fifo; + 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; @@ -571,19 +546,23 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* 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 @@ -591,7 +570,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* 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); @@ -603,7 +582,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * 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() ) @@ -614,21 +593,22 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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_input, p_buffer ); inputResamplingStop( p_input ); @@ -639,41 +619,43 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * 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 ); + 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 : @@ -683,7 +665,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * 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; @@ -707,11 +689,11 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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 @@ -720,15 +702,15 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, (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 @@ -740,13 +722,14 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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; } } @@ -754,25 +737,24 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, /* 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; } @@ -792,7 +774,7 @@ static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input, 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" ); @@ -819,11 +801,10 @@ 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; } } @@ -835,14 +816,19 @@ static vout_thread_t *RequestVout( void *p_private, return vout_Request( p_aout, p_vout, p_fmt ); } -static vout_thread_t *RequestVoutFromFilter( void *p_private, - vout_thread_t *p_vout, video_format_t *p_fmt, bool b_recycle ) +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_private; + 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 && b_recycle ); + p_vout, p_fmt, p_input->b_recycle_vout ); } static int ChangeFiltersString( aout_instance_t * p_aout, const char* psz_variable, @@ -863,7 +849,6 @@ static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, { 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 @@ -872,20 +857,12 @@ static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, { 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 ) ) - { - ChangeFiltersString( p_aout, "audio-visual", "visual", false ); - ChangeFiltersString( p_aout, "audio-visual", "goom", false ); - ChangeFiltersString( p_aout, "audio-visual", "galaktos", true ); - } 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", false ); ChangeFiltersString( p_aout, "audio-visual", "projectm", true ); } else @@ -895,7 +872,6 @@ static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, 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 ); } } @@ -946,7 +922,8 @@ static int ReplayGainCallback( vlc_object_t *p_this, char const *psz_cmd, 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; @@ -960,7 +937,7 @@ static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input ) int i_use; float f_gain; - p_input->f_multiplier = 1.0; + p_input->mixer.multiplier = 1.0; if( !psz_replay_gain ) return; @@ -991,14 +968,14 @@ static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input ) 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 );