X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=dcd3d7a0b867972c172962d5d97daf9fae29b1cf;hb=4f7824b2e24b2c3a4c04e229ff33005b08489943;hp=b394eab2d9e85d3a2e7883415da90cc4693e9e36;hpb=6b15eba41eb55231031a8325248065a11793f7ef;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index b394eab2d9..dcd3d7a0b8 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -29,6 +29,8 @@ # include "config.h" #endif +#include + #include #include @@ -38,19 +40,17 @@ #include #include /* for vout_Request */ +#include -#ifdef HAVE_ALLOCA_H -# include -#endif #include +#include #include #include "aout_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 ) +#define AOUT_ASSERT_LOCKED vlc_assert_locked( &p_aout->lock ) -static void inputFailure( aout_instance_t *, aout_input_t *, const char * ); +static void inputFailure( audio_output_t *, aout_input_t *, const char * ); static void inputDrop( aout_input_t *, aout_buffer_t * ); static void inputResamplingStop( aout_input_t *p_input ); @@ -60,17 +60,15 @@ 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 * ); +static void ReplayGainSelect( audio_output_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 *****************************************************************************/ -int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_request_vout_t *p_request_vout ) +int aout_InputNew( audio_output_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; @@ -83,8 +81,7 @@ 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->fifo, p_aout->mixer_format.i_rate ); /* */ if( p_request_vout ) @@ -98,10 +95,8 @@ 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 +124,6 @@ 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" ) ) { @@ -234,15 +222,10 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ var_Create( p_aout, "audio-replay-gain-peak-protection", VLC_VAR_BOOL | VLC_VAR_DOINHERIT ); } - if( var_Type( p_aout, "audio-time-stretch" ) == 0 ) - { - var_Create( p_aout, "audio-time-stretch", - VLC_VAR_BOOL | VLC_VAR_DOINHERIT ); - } 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; + psz_scaletempo = var_InheritBool( p_aout, "audio-time-stretch" ) ? strdup( "scaletempo" ) : NULL; p_input->b_recycle_vout = psz_visual && *psz_visual; @@ -260,7 +243,7 @@ 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 ) { @@ -282,9 +265,8 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ } /* Create a VLC object */ - static const char typename[] = "audio filter"; p_filter = vlc_custom_create( p_aout, sizeof(*p_filter), - VLC_OBJECT_GENERIC, typename ); + "audio filter" ); if( p_filter == NULL ) { msg_Err( p_aout, "cannot add user filter %s (skipped)", @@ -293,27 +275,23 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ continue; } - 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->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, + 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 */ { - - 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 */ @@ -324,13 +302,13 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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 ); @@ -338,12 +316,12 @@ 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 ); } } @@ -356,7 +334,6 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ psz_parser ); free( p_filter->p_owner ); - vlc_object_detach( p_filter ); vlc_object_release( p_filter ); psz_parser = psz_next; @@ -364,30 +341,25 @@ 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_FiltersCreatePipeline( p_aout, p_input->pp_filters, + &p_input->i_nb_filters, + &chain_input_format, + &p_filter->fmt_in.audio ) < 0 ) { - 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 ); + msg_Err( p_aout, "cannot add user filter %s (skipped)", + psz_parser ); - module_unneed( p_filter, p_filter->p_module ); - free( p_filter->p_owner ); - vlc_object_detach( p_filter ); - vlc_object_release( p_filter ); + module_unneed( p_filter, p_filter->p_module ); + free( p_filter->p_owner ); + vlc_object_release( p_filter ); - psz_parser = psz_next; - continue; - } + 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, + memcpy( &chain_input_format, &p_filter->fmt_out.audio, sizeof( audio_sample_format_t ) ); if( i_visual == 0 ) /* scaletempo */ @@ -402,29 +374,22 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ free( psz_scaletempo ); /* 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 ) { - 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; - } + 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 ) ) + 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++; @@ -432,19 +397,14 @@ 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; } - 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->pp_resamplers[0]->fmt_in.audio.i_rate = p_input->input.i_rate; } p_input->i_resampling_type = AOUT_RESAMPLING_NONE; @@ -453,23 +413,10 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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; @@ -480,9 +427,9 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ ***************************************************************************** * This function must be entered with the mixer lock. *****************************************************************************/ -int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) +int aout_InputDelete( audio_output_t * p_aout, aout_input_t * p_input ) { - AOUT_ASSERT_MIXER_LOCKED; + AOUT_ASSERT_LOCKED; if ( p_input->b_error ) return 0; @@ -494,92 +441,93 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) p_input->b_recycle_vout = psz_visual && *psz_visual; free( psz_visual ); - aout_FiltersDestroyPipeline( p_aout, p_input->pp_filters, - p_input->i_nb_filters ); + aout_FiltersDestroyPipeline( p_input->pp_filters, p_input->i_nb_filters ); p_input->i_nb_filters = 0; - aout_FiltersDestroyPipeline( p_aout, p_input->pp_resamplers, + aout_FiltersDestroyPipeline( 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_input->fifo ); return 0; } /***************************************************************************** - * aout_InputPlay : play a buffer + * aout_InputCheckAndRestart : restart an input ***************************************************************************** - * This function must be entered with the input lock. + * This function must be entered with the input and mixer 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 ) +void aout_InputCheckAndRestart( audio_output_t * p_aout, aout_input_t * p_input ) { - mtime_t start_date; - AOUT_ASSERT_INPUT_LOCKED; + AOUT_ASSERT_LOCKED; - if( p_input->b_restart ) - { - aout_fifo_t fifo; - uint8_t *p_first_byte_to_mix; - bool b_paused; - mtime_t i_pause_date; - - aout_lock_mixer( p_aout ); - aout_lock_input_fifos( p_aout ); + if( !p_input->b_restart ) + return; - /* A little trick to avoid loosing our input fifo and properties */ + /* 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; - b_paused = p_input->b_paused; - i_pause_date = p_input->i_pause_date; + aout_fifo_t fifo = p_input->fifo; + bool b_paused = p_input->b_paused; + mtime_t 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->fifo, p_aout->mixer_format.i_rate ); - aout_InputDelete( p_aout, p_input ); + 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->b_paused = b_paused; - p_input->i_pause_date = i_pause_date; + aout_InputNew( p_aout, p_input, &p_input->request_vout ); + p_input->fifo = fifo; + p_input->b_paused = b_paused; + p_input->i_pause_date = i_pause_date; - aout_unlock_input_fifos( p_aout ); - aout_unlock_mixer( p_aout ); - } + p_input->b_restart = false; +} +/***************************************************************************** + * aout_InputPlay : play a buffer + ***************************************************************************** + * This function must be entered with the input lock. + *****************************************************************************/ +/* XXX Do not activate it !! */ +//#define AOUT_PROCESS_BEFORE_CHEKS +void aout_InputPlay( audio_output_t * p_aout, aout_input_t * p_input, + aout_buffer_t * p_buffer, int i_input_rate ) +{ + mtime_t start_date; + AOUT_ASSERT_LOCKED; if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL ) { inputDrop( p_input, p_buffer ); - return 0; + return; } #ifdef AOUT_PROCESS_BEFORE_CHEKS /* Run pre-filters. */ aout_FiltersPlay( p_aout, p_input->pp_filters, p_input->i_nb_filters, &p_buffer ); + if( !p_buffer ) + return; /* 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; if( p_buffer->i_nb_samples <= 0 ) { - aout_BufferFree( p_buffer ); - return 0; + block_Release( p_buffer ); + return; } #endif /* Handle input rate change, but keep drift correction */ if( i_input_rate != p_input->i_last_input_rate ) { - unsigned int * const pi_rate = &p_input->p_playback_rate_filter->input.i_rate; + unsigned int * const pi_rate = &p_input->p_playback_rate_filter->fmt_in.audio.i_rate; #define F(r,ir) ( INPUT_RATE_DEFAULT * (r) / (ir) ) const int i_delta = *pi_rate - F(p_input->input.i_rate,p_input->i_last_input_rate); *pi_rate = F(p_input->input.i_rate + i_delta, i_input_rate); @@ -587,81 +535,79 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, p_input->i_last_input_rate = i_input_rate; } + mtime_t now = mdate(); + /* 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 ); + start_date = aout_FifoNextStart( &p_input->fifo ); - if ( start_date != 0 && start_date < mdate() ) + if ( start_date != VLC_TS_INVALID && start_date < now ) { /* 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 ); + "clearing out", now - start_date ); + aout_FifoReset( &p_input->fifo ); + aout_FifoReset( &p_aout->fifo ); if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) msg_Warn( p_aout, "timing screwed, stopping resampling" ); inputResamplingStop( p_input ); - start_date = 0; + p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY; + start_date = VLC_TS_INVALID; } - if ( p_buffer->start_date < mdate() + AOUT_MIN_PREPARE_TIME ) + if ( p_buffer->i_pts < now + 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 ); - + now - p_buffer->i_pts ); inputDrop( p_input, p_buffer ); inputResamplingStop( p_input ); - return 0; + return; } /* 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( start_date == VLC_TS_INVALID ) + start_date = p_buffer->i_pts; + + mtime_t drift = start_date - p_buffer->i_pts; + + if( drift < -i_input_rate * 3 * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT ) + { + msg_Warn( p_aout, "buffer way too early (%"PRId64"), clearing queue", + drift ); + aout_FifoReset( &p_input->fifo ); if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) msg_Warn( p_aout, "timing screwed, stopping resampling" ); inputResamplingStop( p_input ); - start_date = 0; + p_buffer->i_flags |= BLOCK_FLAG_DISCONTINUITY; + start_date = p_buffer->i_pts; + drift = 0; } - else if ( start_date != 0 && - ( start_date > p_buffer->start_date + i_pts_tolerance) ) + else + if( drift > +i_input_rate * 3 * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT ) { - msg_Warn( p_aout, "audio drift is too big (%"PRId64"), dropping buffer", - start_date - p_buffer->start_date ); + msg_Warn( p_aout, "buffer way too late (%"PRId64"), dropping buffer", + drift ); inputDrop( p_input, p_buffer ); - return 0; + return; } - 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 ); + aout_FiltersPlay( p_input->pp_filters, p_input->i_nb_filters, &p_buffer ); + if( !p_buffer ) + return; #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 ) && + ( drift < -AOUT_MAX_PTS_ADVANCE || drift > +AOUT_MAX_PTS_DELAY ) && p_input->i_nb_resamplers > 0 ) { /* Can happen in several circumstances : @@ -671,20 +617,13 @@ 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; - - 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"); + p_input->i_resamp_start_date = now; + p_input->i_resamp_start_drift = (int)-drift; + p_input->i_resampling_type = (drift < 0) ? AOUT_RESAMPLING_DOWN + : AOUT_RESAMPLING_UP; + msg_Warn( p_aout, (drift < 0) + ? "buffer too early (%"PRId64"), down-sampling" + : "buffer too late (%"PRId64"), up-sampling", drift ); } if ( p_input->i_resampling_type != AOUT_RESAMPLING_NONE ) @@ -694,13 +633,9 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * 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 */ - } + 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 * resampling */ @@ -708,15 +643,15 @@ 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); + now - p_input->i_resamp_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 @@ -728,13 +663,14 @@ 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; } } @@ -742,45 +678,39 @@ 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; if( p_buffer->i_nb_samples <= 0 ) { - aout_BufferFree( p_buffer ); - return 0; + block_Release( p_buffer ); + return; } #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; + p_buffer->i_pts = start_date; + aout_FifoPush( &p_input->fifo, p_buffer ); } /***************************************************************************** * static functions *****************************************************************************/ -static void inputFailure( aout_instance_t * p_aout, aout_input_t * p_input, +static void inputFailure( audio_output_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, + aout_FiltersDestroyPipeline( p_input->pp_filters, p_input->i_nb_filters ); + aout_FiltersDestroyPipeline( p_input->pp_resamplers, p_input->i_nb_resamplers ); - aout_FifoDestroy( p_aout, &p_input->fifo ); + aout_FifoDestroy( &p_input->fifo ); var_Destroy( p_aout, "visual" ); var_Destroy( p_aout, "equalizer" ); var_Destroy( p_aout, "audio-filter" ); @@ -807,43 +737,54 @@ 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; + audio_output_t *p_aout = p_private; VLC_UNUSED(b_recycle); - return vout_Request( p_aout, p_vout, p_fmt ); + vout_configuration_t cfg = { + .vout = p_vout, + .input = NULL, + .change_fmt = true, + .fmt = p_fmt, + .dpb_size = 1, + }; + return vout_Request( p_aout, &cfg ); } -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, +static int ChangeFiltersString( audio_output_t * p_aout, const char* psz_variable, const char *psz_name, bool b_add ) { - return AoutChangeFilterString( VLC_OBJECT(p_aout), p_aout, - psz_variable, psz_name, b_add ) ? 1 : 0; + return aout_ChangeFilterString( VLC_OBJECT(p_aout), p_aout, + psz_variable, psz_name, b_add ) ? 1 : 0; } static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, vlc_value_t oldval, vlc_value_t newval, void *p_data ) { - aout_instance_t *p_aout = (aout_instance_t *)p_this; + audio_output_t *p_aout = (audio_output_t *)p_this; char *psz_mode = newval.psz_string; (void)psz_cmd; (void)oldval; (void)p_data; @@ -851,7 +792,6 @@ 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 @@ -860,20 +800,12 @@ 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 @@ -883,7 +815,6 @@ 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 ); } } @@ -897,7 +828,7 @@ static int VisualizationCallback( vlc_object_t *p_this, char const *psz_cmd, 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; + audio_output_t *p_aout = (audio_output_t *)p_this; char *psz_mode = newval.psz_string; int i_ret; (void)psz_cmd; (void)oldval; (void)p_data; @@ -926,21 +857,17 @@ static int ReplayGainCallback( vlc_object_t *p_this, char const *psz_cmd, { 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; + audio_output_t *p_aout = (audio_output_t *)p_this; - 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 ); + aout_lock( p_aout ); + if( p_aout->p_input != NULL ) + ReplayGainSelect( p_aout, p_aout->p_input ); + aout_unlock( p_aout ); return VLC_SUCCESS; } -static void ReplayGainSelect( aout_instance_t *p_aout, aout_input_t *p_input ) +static void ReplayGainSelect( audio_output_t *p_aout, aout_input_t *p_input ) { char *psz_replay_gain = var_GetNonEmptyString( p_aout, "audio-replay-gain-mode" ); @@ -948,7 +875,7 @@ 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->multiplier = 1.0; if( !psz_replay_gain ) return; @@ -979,16 +906,15 @@ 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->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->multiplier > 1.0 ) { - p_input->f_multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use]; + p_input->multiplier = 1.0f / p_input->replay_gain.pf_peak[i_use]; } free( psz_replay_gain ); } -