X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=a179080fa7bea9dd70bdff7af170e096ad7c7da6;hb=5e1e150cc50494f679ddcd9ad8534bf2177a1fd7;hp=66bc0b383b8f08467bd49ae9489afe346d775533;hpb=eff24974c62da8740d381ae45694474ac1ee2c05;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index 66bc0b383b..a179080fa7 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -29,6 +29,8 @@ # include "config.h" #endif +#include + #include #include @@ -36,23 +38,20 @@ #include #include -#include /* for input_thread_t and i_pts_delay */ +#include +#include /* for vout_Request */ -#ifdef HAVE_ALLOCA_H -# include -#endif #include +#include +#include #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 *, @@ -62,10 +61,14 @@ static int EqualizerCallback( 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; @@ -78,14 +81,23 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_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; + 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 ); @@ -107,22 +119,22 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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 ); @@ -217,17 +229,18 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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_GetBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL; + p_input->b_recycle_vout = psz_visual && *psz_visual; + /* parse user filter lists */ - for( i_visual = 0; i_visual < 2; i_visual++ ) + 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]; @@ -236,7 +249,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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 ) { @@ -271,52 +284,53 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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 ); } } @@ -328,7 +342,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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; @@ -336,18 +350,19 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) } /* 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; @@ -356,11 +371,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) } /* 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; } @@ -383,16 +400,16 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) } /* 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++; @@ -400,7 +417,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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; @@ -409,23 +426,13 @@ 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 ); - 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]; @@ -434,7 +441,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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( @@ -447,7 +454,6 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) /* Success */ p_input->b_error = false; - p_input->b_restart = false; p_input->i_last_input_rate = INPUT_RATE_DEFAULT; return 0; @@ -461,7 +467,16 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input ) 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 ); @@ -469,11 +484,47 @@ 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; } +/***************************************************************************** + * 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 ***************************************************************************** @@ -487,31 +538,9 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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; } @@ -519,19 +548,23 @@ 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 @@ -539,7 +572,7 @@ 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); @@ -551,7 +584,7 @@ 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() ) @@ -562,23 +595,24 @@ 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_aout, p_input, p_buffer ); + inputDrop( p_input, p_buffer ); inputResamplingStop( p_input ); return 0; } @@ -587,41 +621,43 @@ 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 ); - 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 : @@ -631,7 +667,7 @@ 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; @@ -655,11 +691,11 @@ 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 @@ -668,15 +704,15 @@ 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 @@ -688,13 +724,14 @@ 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; } } @@ -702,25 +739,24 @@ 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; } @@ -740,7 +776,7 @@ 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" ); @@ -755,16 +791,11 @@ static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input, 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 ) @@ -772,14 +803,43 @@ 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 ) { @@ -792,14 +852,13 @@ 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 { @@ -807,23 +866,22 @@ 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 ) ) + 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 ); } } @@ -838,7 +896,6 @@ static int EqualizerCallback( 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; int i_ret; (void)psz_cmd; (void)oldval; (void)p_data; @@ -849,12 +906,10 @@ static int EqualizerCallback( vlc_object_t *p_this, char const *psz_cmd, } 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 */ @@ -876,7 +931,8 @@ 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; @@ -890,7 +946,7 @@ 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; @@ -921,14 +977,14 @@ 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 );