X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Finput.c;h=8e4075624b02cd3748daea934eef426c0d815dfb;hb=80158557a6793a52dac40fa2a75c627b3366ca9c;hp=79916b574269739477eeb6677050793250d76c22;hpb=9b4469b28e0e41ebe99a379fe3252fdc6bec22f0;p=vlc diff --git a/src/audio_output/input.c b/src/audio_output/input.c index 79916b5742..8e4075624b 100644 --- a/src/audio_output/input.c +++ b/src/audio_output/input.c @@ -42,6 +42,7 @@ #include /* for vout_Request */ #include +#include #include #include "aout_internal.h" @@ -248,7 +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 ) { @@ -288,16 +289,18 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const 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 */ @@ -308,13 +311,13 @@ 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 ); @@ -322,12 +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 ); } } @@ -348,12 +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 ); @@ -369,9 +373,8 @@ 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 */ @@ -428,7 +431,7 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ 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; @@ -453,7 +456,6 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input, const aout_ /* Success */ p_input->b_error = false; - p_input->b_restart = false; p_input->i_last_input_rate = INPUT_RATE_DEFAULT; return 0; @@ -490,48 +492,53 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input ) } /***************************************************************************** - * 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( aout_instance_t * p_aout, aout_input_t * p_input ) { - mtime_t start_date; + AOUT_ASSERT_MIXER_LOCKED; AOUT_ASSERT_INPUT_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; + if( !p_input->b_restart ) + return; - aout_lock_mixer( p_aout ); - aout_lock_input_fifos( p_aout ); + aout_lock_input_fifos( p_aout ); - /* 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->mixer.begin; - fifo = p_input->mixer.fifo; - b_paused = p_input->b_paused; - i_pause_date = p_input->i_pause_date; + 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_FifoInit( p_aout, &p_input->mixer.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->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; + 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; - aout_unlock_input_fifos( p_aout ); - aout_unlock_mixer( p_aout ); - } + p_input->b_restart = false; + + aout_unlock_input_fifos( p_aout ); +} +/***************************************************************************** + * aout_InputPlay : play a buffer + ***************************************************************************** + * This function must be entered with the input lock. + *****************************************************************************/ +/* XXX Do not activate it !! */ +//#define AOUT_PROCESS_BEFORE_CHEKS +int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, + aout_buffer_t * p_buffer, int i_input_rate ) +{ + mtime_t start_date; + AOUT_ASSERT_INPUT_LOCKED; if( i_input_rate != INPUT_RATE_DEFAULT && p_input->p_playback_rate_filter == NULL ) { @@ -543,6 +550,8 @@ 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 ) @@ -553,9 +562,11 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, &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 @@ -563,7 +574,7 @@ 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); @@ -592,6 +603,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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; } @@ -622,6 +634,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, 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 && @@ -637,8 +650,9 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, #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. @@ -679,11 +693,11 @@ 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 @@ -692,7 +706,7 @@ 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 " @@ -719,6 +733,7 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input, * 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; } } @@ -726,14 +741,15 @@ 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 @@ -789,11 +805,10 @@ 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; } } @@ -805,12 +820,17 @@ static vout_thread_t *RequestVout( void *p_private, return vout_Request( p_aout, p_vout, p_fmt ); } -vout_thread_t *aout_filter_RequestVout( aout_filter_t *p_filter, +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 ); }