X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Ffilters.c;h=b41dc55679044c41068c653a2851240ae45c8dfd;hb=c22b7e0ece66ea1d8a0df64076f783b765b4a371;hp=136212379de14c6b5f1bc8c6362cb1226aa2fe29;hpb=a78e273ec53ff8a6c3993f3deda0b893f8dd709a;p=vlc diff --git a/src/audio_output/filters.c b/src/audio_output/filters.c index 136212379d..b41dc55679 100644 --- a/src/audio_output/filters.c +++ b/src/audio_output/filters.c @@ -28,339 +28,193 @@ # include "config.h" #endif -#include -#include +#include -#ifdef HAVE_ALLOCA_H -# include -#endif +#include +#include +#include #include +#include +#include #include "aout_internal.h" +#include /***************************************************************************** * FindFilter: find an audio filter for a specific transformation *****************************************************************************/ -static aout_filter_t * FindFilter( aout_instance_t * p_aout, - const audio_sample_format_t * p_input_format, - const audio_sample_format_t * p_output_format ) +static filter_t * FindFilter( vlc_object_t *obj, + const audio_sample_format_t * p_input_format, + const audio_sample_format_t * p_output_format ) { - aout_filter_t * p_filter = vlc_object_create( p_aout, - sizeof(aout_filter_t) ); + static const char typename[] = "audio filter"; + filter_t * p_filter; + + p_filter = vlc_custom_create( obj, sizeof(*p_filter), typename ); if ( p_filter == NULL ) return NULL; - vlc_object_attach( p_filter, p_aout ); - memcpy( &p_filter->input, p_input_format, sizeof(audio_sample_format_t) ); - memcpy( &p_filter->output, p_output_format, + memcpy( &p_filter->fmt_in.audio, p_input_format, + sizeof(audio_sample_format_t) ); + p_filter->fmt_in.i_codec = p_input_format->i_format; + memcpy( &p_filter->fmt_out.audio, p_output_format, sizeof(audio_sample_format_t) ); - p_filter->p_module = module_Need( p_filter, "audio filter", NULL, 0 ); + p_filter->fmt_out.i_codec = p_output_format->i_format; + p_filter->p_owner = NULL; + + p_filter->p_module = module_need( p_filter, "audio filter", NULL, false ); if ( p_filter->p_module == NULL ) { - vlc_object_detach( p_filter ); vlc_object_release( p_filter ); return NULL; } - p_filter->b_continuity = VLC_FALSE; - + assert( p_filter->pf_audio_filter ); return p_filter; } -/***************************************************************************** - * SplitConversion: split a conversion in two parts - ***************************************************************************** - * Returns the number of conversions required by the first part - 0 if only - * one conversion was asked. - * Beware : p_output_format can be modified during this function if the - * developer passed SplitConversion( toto, titi, titi, ... ). That is legal. - * SplitConversion( toto, titi, toto, ... ) isn't. - *****************************************************************************/ -static int SplitConversion( const audio_sample_format_t * p_input_format, - const audio_sample_format_t * p_output_format, - audio_sample_format_t * p_middle_format ) +/** + * Splits audio format conversion in two simpler conversions + * @return 0 on successful split, -1 if the input and output formats are too + * similar to split the conversion. + */ +static int SplitConversion( const audio_sample_format_t *restrict infmt, + const audio_sample_format_t *restrict outfmt, + audio_sample_format_t *midfmt ) { - vlc_bool_t b_format = - (p_input_format->i_format != p_output_format->i_format); - vlc_bool_t b_rate = (p_input_format->i_rate != p_output_format->i_rate); - vlc_bool_t b_channels = - (p_input_format->i_physical_channels - != p_output_format->i_physical_channels) - || (p_input_format->i_original_channels - != p_output_format->i_original_channels); - int i_nb_conversions = b_format + b_rate + b_channels; - - if ( i_nb_conversions <= 1 ) return 0; - - memcpy( p_middle_format, p_output_format, sizeof(audio_sample_format_t) ); + *midfmt = *outfmt; - if ( i_nb_conversions == 2 ) + if( infmt->i_rate != outfmt->i_rate ) + midfmt->i_rate = infmt->i_rate; + else + if( infmt->i_physical_channels != outfmt->i_physical_channels + || infmt->i_original_channels != outfmt->i_original_channels ) { - if ( !b_format || !b_channels ) - { - p_middle_format->i_rate = p_input_format->i_rate; - aout_FormatPrepare( p_middle_format ); - return 1; - } - - /* !b_rate */ - p_middle_format->i_physical_channels - = p_input_format->i_physical_channels; - p_middle_format->i_original_channels - = p_input_format->i_original_channels; - aout_FormatPrepare( p_middle_format ); - return 1; + midfmt->i_physical_channels = infmt->i_physical_channels; + midfmt->i_original_channels = infmt->i_original_channels; + } + else + { + assert( infmt->i_format != outfmt->i_format ); + if( AOUT_FMT_LINEAR( infmt ) ) + /* NOTE: Use S16N as intermediate. We have all conversions to S16N, + * and all useful conversions from S16N. TODO: FL32 if HAVE_FPU. */ + midfmt->i_format = VLC_CODEC_S16N; + else + if( AOUT_FMT_LINEAR( outfmt ) ) + /* NOTE: our non-linear -> linear filters always output 32-bits */ + midfmt->i_format = HAVE_FPU ? VLC_CODEC_FL32 : VLC_CODEC_FI32; + else + return -1; /* no indirect non-linear -> non-linear */ } - /* i_nb_conversion == 3 */ - p_middle_format->i_rate = p_input_format->i_rate; - aout_FormatPrepare( p_middle_format ); - return 2; -} - -static void ReleaseFilter( aout_filter_t * p_filter ) -{ - module_Unneed( p_filter, p_filter->p_module ); - vlc_object_detach( p_filter ); - vlc_object_release( p_filter ); + aout_FormatPrepare( midfmt ); + return AOUT_FMTS_IDENTICAL( infmt, midfmt ) ? -1 : 0; } -/***************************************************************************** - * aout_FiltersCreatePipeline: create a filters pipeline to transform a sample - * format to another - ***************************************************************************** - * pi_nb_filters must be initialized before calling this function - *****************************************************************************/ -int aout_FiltersCreatePipeline( aout_instance_t * p_aout, - aout_filter_t ** pp_filters_start, - int * pi_nb_filters, - const audio_sample_format_t * p_input_format, - const audio_sample_format_t * p_output_format ) +#undef aout_FiltersCreatePipeline +/** + * Allocates audio format conversion filters + * @param obj parent VLC object for new filters + * @param filters table of filters [IN/OUT] + * @param nb_filters pointer to the number of filters in the table [IN/OUT] + * @param infmt input audio format + * @param outfmt output audio format + * @return 0 on success, -1 on failure + */ +int aout_FiltersCreatePipeline( vlc_object_t *obj, + filter_t **filters, + int *nb_filters, + const audio_sample_format_t *restrict infmt, + const audio_sample_format_t *restrict outfmt ) { - aout_filter_t** pp_filters = pp_filters_start + *pi_nb_filters; - audio_sample_format_t temp_format; - int i_nb_conversions; + audio_sample_format_t curfmt = *outfmt; + unsigned i = 0, max = *nb_filters - AOUT_MAX_FILTERS; - if ( AOUT_FMTS_IDENTICAL( p_input_format, p_output_format ) ) - { - msg_Dbg( p_aout, "no need for any filter" ); - return 0; - } - - aout_FormatsPrint( p_aout, "filter(s)", p_input_format, p_output_format ); + filters += *nb_filters; + aout_FormatsPrint( obj, "filter(s)", infmt, outfmt ); - if( *pi_nb_filters + 1 > AOUT_MAX_FILTERS ) + while( !AOUT_FMTS_IDENTICAL( infmt, &curfmt ) ) { - msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS ); - intf_UserFatal( p_aout, VLC_FALSE, _("Audio filtering failed"), - _("The maximum number of filters (%d) was reached."), - AOUT_MAX_FILTERS ); - return -1; - } - - /* Try to find a filter to do the whole conversion. */ - pp_filters[0] = FindFilter( p_aout, p_input_format, p_output_format ); - if ( pp_filters[0] != NULL ) - { - msg_Dbg( p_aout, "found a filter for the whole conversion" ); - ++*pi_nb_filters; - return 0; - } + if( i >= max ) + { + msg_Err( obj, "max (%u) filters reached", AOUT_MAX_FILTERS ); + dialog_Fatal( obj, _("Audio filtering failed"), + _("The maximum number of filters (%u) was reached."), + AOUT_MAX_FILTERS ); + goto rollback; + } - /* We'll have to split the conversion. We always do the downmixing - * before the resampling, because the audio decoder can probably do it - * for us. */ - i_nb_conversions = SplitConversion( p_input_format, - p_output_format, &temp_format ); - if ( !i_nb_conversions ) - { - /* There was only one conversion to do, and we already failed. */ - msg_Err( p_aout, "couldn't find a filter for the conversion" ); - return -1; - } + /* Make room and prepend a filter */ + memmove( filters + 1, filters, i * sizeof( *filters ) ); - pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format ); - if ( pp_filters[0] == NULL && i_nb_conversions == 2 ) - { - /* Try with only one conversion. */ - SplitConversion( p_input_format, &temp_format, &temp_format ); - pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format ); - } - if ( pp_filters[0] == NULL ) - { - msg_Err( p_aout, - "couldn't find a filter for the first part of the conversion" ); - return -1; - } - - /* We have the first stage of the conversion. Find a filter for - * the rest. */ - if( *pi_nb_filters + 2 > AOUT_MAX_FILTERS ) - { - ReleaseFilter( pp_filters[0] ); - msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS ); - intf_UserFatal( p_aout, VLC_FALSE, _("Audio filtering failed"), - _("The maximum number of filters (%d) was reached."), - AOUT_MAX_FILTERS ); - return -1; - } - pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->output, - p_output_format ); - if ( pp_filters[1] == NULL ) - { - /* Try to split the conversion. */ - i_nb_conversions = SplitConversion( &pp_filters[0]->output, - p_output_format, &temp_format ); - if ( !i_nb_conversions ) + *filters = FindFilter( obj, infmt, &curfmt ); + if( *filters != NULL ) { - ReleaseFilter( pp_filters[0] ); - msg_Err( p_aout, - "couldn't find a filter for the second part of the conversion" ); - return -1; + i++; + break; /* done! */ } - if( *pi_nb_filters + 3 > AOUT_MAX_FILTERS ) + + audio_sample_format_t midfmt; + /* Split the conversion */ + if( SplitConversion( infmt, &curfmt, &midfmt ) ) { - ReleaseFilter( pp_filters[0] ); - msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS ); - intf_UserFatal( p_aout, VLC_FALSE, _("Audio filtering failed"), - _("The maximum number of filters (%d) was reached."), - AOUT_MAX_FILTERS ); - return -1; + msg_Err( obj, "conversion pipeline failed: %4.4s -> %4.4s", + (const char *)&infmt->i_format, + (const char *)&outfmt->i_format ); + goto rollback; } - pp_filters[1] = FindFilter( p_aout, &pp_filters[0]->output, - &temp_format ); - pp_filters[2] = FindFilter( p_aout, &temp_format, - p_output_format ); - if ( pp_filters[1] == NULL || pp_filters[2] == NULL ) + *filters = FindFilter( obj, &midfmt, &curfmt ); + if( *filters == NULL ) { - ReleaseFilter( pp_filters[0] ); - if ( pp_filters[1] != NULL ) - { - ReleaseFilter( pp_filters[1] ); - } - if ( pp_filters[2] != NULL ) - { - ReleaseFilter( pp_filters[2] ); - } - msg_Err( p_aout, - "couldn't find filters for the second part of the conversion" ); - return -1; + msg_Err( obj, "cannot find filter for simple conversion" ); + goto rollback; } - *pi_nb_filters += 3; - msg_Dbg( p_aout, "found 3 filters for the whole conversion" ); - } - else - { - *pi_nb_filters += 2; - msg_Dbg( p_aout, "found 2 filters for the whole conversion" ); + curfmt = midfmt; + i++; } + msg_Dbg( obj, "conversion pipeline completed" ); + *nb_filters += i; return 0; -} -/***************************************************************************** - * aout_FiltersDestroyPipeline: deallocate a filters pipeline - *****************************************************************************/ -void aout_FiltersDestroyPipeline( aout_instance_t * p_aout, - aout_filter_t ** pp_filters, - int i_nb_filters ) -{ - int i; - (void)p_aout; - - for ( i = 0; i < i_nb_filters; i++ ) - { - module_Unneed( pp_filters[i], pp_filters[i]->p_module ); - vlc_object_detach( pp_filters[i] ); - vlc_object_release( pp_filters[i] ); - } +rollback: + aout_FiltersDestroyPipeline( filters, i ); + return -1; } -/***************************************************************************** - * aout_FiltersHintBuffers: fill in aout_alloc_t structures to optimize - * buffer allocations - *****************************************************************************/ -void aout_FiltersHintBuffers( aout_instance_t * p_aout, - aout_filter_t ** pp_filters, - int i_nb_filters, aout_alloc_t * p_first_alloc ) +/** + * Destroys a chain of audio filters. + */ +void aout_FiltersDestroyPipeline( filter_t *const *filters, unsigned n ) { - int i; - - (void)p_aout; /* unused */ - - for ( i = i_nb_filters - 1; i >= 0; i-- ) + for( unsigned i = 0; i < n; i++ ) { - aout_filter_t * p_filter = pp_filters[i]; - - int i_output_size = p_filter->output.i_bytes_per_frame - * p_filter->output.i_rate * AOUT_MAX_INPUT_RATE - / p_filter->output.i_frame_length; - int i_input_size = p_filter->input.i_bytes_per_frame - * p_filter->input.i_rate * AOUT_MAX_INPUT_RATE - / p_filter->input.i_frame_length; - - p_first_alloc->i_bytes_per_sec = __MAX( p_first_alloc->i_bytes_per_sec, - i_output_size ); + filter_t *p_filter = filters[i]; - if ( p_filter->b_in_place ) - { - p_first_alloc->i_bytes_per_sec = __MAX( - p_first_alloc->i_bytes_per_sec, - i_input_size ); - p_filter->output_alloc.i_alloc_type = AOUT_ALLOC_NONE; - } - else - { - /* We're gonna need a buffer allocation. */ - memcpy( &p_filter->output_alloc, p_first_alloc, - sizeof(aout_alloc_t) ); - p_first_alloc->i_alloc_type = AOUT_ALLOC_STACK; - p_first_alloc->i_bytes_per_sec = i_input_size; - } + module_unneed( p_filter, p_filter->p_module ); + free( p_filter->p_owner ); + vlc_object_release( p_filter ); } } -/***************************************************************************** - * aout_FiltersPlay: play a buffer - *****************************************************************************/ -void aout_FiltersPlay( aout_instance_t * p_aout, - aout_filter_t ** pp_filters, - int i_nb_filters, aout_buffer_t ** pp_input_buffer ) +/** + * Filters an audio buffer through a chain of filters. + */ +void aout_FiltersPlay( filter_t *const *pp_filters, + unsigned i_nb_filters, block_t ** pp_block ) { - int i; + block_t *p_block = *pp_block; - for ( i = 0; i < i_nb_filters; i++ ) + /* TODO: use filter chain */ + for( unsigned i = 0; i < i_nb_filters; i++ ) { - aout_filter_t * p_filter = pp_filters[i]; - aout_buffer_t * p_output_buffer; + filter_t * p_filter = pp_filters[i]; - /* Resamplers can produce slightly more samples than (i_in_nb * - * p_filter->output.i_rate / p_filter->input.i_rate) so we need - * slightly bigger buffers. */ - aout_BufferAlloc( &p_filter->output_alloc, - ((mtime_t)(*pp_input_buffer)->i_nb_samples + 2) - * 1000000 / p_filter->input.i_rate, - *pp_input_buffer, p_output_buffer ); - if ( p_output_buffer == NULL ) - { - msg_Err( p_aout, "out of memory" ); - return; - } - /* Please note that p_output_buffer->i_nb_samples & i_nb_bytes + /* Please note that p_block->i_nb_samples & i_buffer * shall be set by the filter plug-in. */ - - p_filter->pf_do_work( p_aout, p_filter, *pp_input_buffer, - p_output_buffer ); - - if ( !p_filter->b_in_place ) - { - aout_BufferFree( *pp_input_buffer ); - *pp_input_buffer = p_output_buffer; - } - - if( p_output_buffer->i_nb_samples <= 0 ) - break; + p_block = p_filter->pf_audio_filter( p_filter, p_block ); } + *pp_block = p_block; } -