X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=src%2Faudio_output%2Ffilters.c;h=14aca008c7f3b7a2bbff01d8a7719cb9f6fd3fba;hb=924bda6286af1120d86a3b70f98aa06e1c7106ae;hp=382ca6a2f89ab17d671e7a5be7cd3e9540572162;hpb=54929735f8c480fcbdecb870e97405670acc6ce8;p=vlc diff --git a/src/audio_output/filters.c b/src/audio_output/filters.c index 382ca6a2f8..14aca008c7 100644 --- a/src/audio_output/filters.c +++ b/src/audio_output/filters.c @@ -1,8 +1,8 @@ /***************************************************************************** * filters.c : audio output filters management ***************************************************************************** - * Copyright (C) 2002 VideoLAN - * $Id: filters.c,v 1.12 2002/10/20 12:23:48 massiot Exp $ + * Copyright (C) 2002-2007 the VideoLAN team + * $Id$ * * Authors: Christophe Massiot * @@ -10,7 +10,7 @@ * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. - * + * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the @@ -18,23 +18,26 @@ * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA. + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ -#include /* calloc(), malloc(), free() */ -#include +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif -#include +#include +#include #ifdef HAVE_ALLOCA_H -# include +# include #endif -#include "audio_output.h" +#include #include "aout_internal.h" +#include /***************************************************************************** * FindFilter: find an audio filter for a specific transformation @@ -43,8 +46,11 @@ 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 ) { - aout_filter_t * p_filter = vlc_object_create( p_aout, - sizeof(aout_filter_t) ); + static const char typename[] = "audio output"; + aout_filter_t * p_filter; + + p_filter = vlc_custom_create( p_aout, sizeof(*p_filter), + VLC_OBJECT_GENERIC, typename ); if ( p_filter == NULL ) return NULL; vlc_object_attach( p_filter, p_aout ); @@ -52,19 +58,21 @@ static aout_filter_t * FindFilter( aout_instance_t * p_aout, memcpy( &p_filter->input, p_input_format, sizeof(audio_sample_format_t) ); memcpy( &p_filter->output, p_output_format, sizeof(audio_sample_format_t) ); - p_filter->p_module = module_Need( p_filter, "audio filter", NULL ); + p_filter->p_module = module_Need( p_filter, "audio filter", NULL, 0 ); if ( p_filter->p_module == NULL ) { vlc_object_detach( p_filter ); - vlc_object_destroy( p_filter ); + vlc_object_release( p_filter ); return NULL; } + p_filter->b_continuity = false; + return p_filter; } /***************************************************************************** - * SplitConversion: split a conversion in two parts + * SplitConversion: split a conversion in two parts ***************************************************************************** * Returns the number of conversions required by the first part - 0 if only * one conversion was asked. @@ -72,16 +80,18 @@ static aout_filter_t * FindFilter( aout_instance_t * p_aout, * developer passed SplitConversion( toto, titi, titi, ... ). That is legal. * SplitConversion( toto, titi, toto, ... ) isn't. *****************************************************************************/ -static int SplitConversion( aout_instance_t * p_aout, - const audio_sample_format_t * p_input_format, - const audio_sample_format_t * p_output_format, - audio_sample_format_t * p_middle_format ) +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 ) { - vlc_bool_t b_format = + bool 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_channels != p_output_format->i_channels); + bool b_rate = (p_input_format->i_rate != p_output_format->i_rate); + bool 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; @@ -93,56 +103,78 @@ static int SplitConversion( aout_instance_t * p_aout, 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_channels = p_input_format->i_channels; + 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; } /* 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_FiltersCreatePipeline: create a filters pipeline to transform a sample * format to another ***************************************************************************** - * TODO : allow the user to add/remove specific filters + * pi_nb_filters must be initialized before calling this function *****************************************************************************/ int aout_FiltersCreatePipeline( aout_instance_t * p_aout, - aout_filter_t ** pp_filters, + 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 ) { + aout_filter_t** pp_filters = pp_filters_start + *pi_nb_filters; audio_sample_format_t temp_format; int i_nb_conversions; if ( AOUT_FMTS_IDENTICAL( p_input_format, p_output_format ) ) { msg_Dbg( p_aout, "no need for any filter" ); - *pi_nb_filters = 0; return 0; } aout_FormatsPrint( p_aout, "filter(s)", p_input_format, p_output_format ); + if( *pi_nb_filters + 1 > AOUT_MAX_FILTERS ) + { + msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS ); + intf_UserFatal( p_aout, 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 = 1; + ++*pi_nb_filters; return 0; } /* 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_aout, p_input_format, + i_nb_conversions = SplitConversion( p_input_format, p_output_format, &temp_format ); if ( !i_nb_conversions ) { @@ -155,10 +187,8 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout, if ( pp_filters[0] == NULL && i_nb_conversions == 2 ) { /* Try with only one conversion. */ - SplitConversion( p_aout, p_input_format, &temp_format, - &temp_format ); - pp_filters[0] = FindFilter( p_aout, p_input_format, - &temp_format ); + SplitConversion( p_input_format, &temp_format, &temp_format ); + pp_filters[0] = FindFilter( p_aout, p_input_format, &temp_format ); } if ( pp_filters[0] == NULL ) { @@ -169,20 +199,37 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout, /* 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, 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( p_aout, - &pp_filters[0]->output, - p_output_format, &temp_format ); + i_nb_conversions = SplitConversion( &pp_filters[0]->output, + p_output_format, &temp_format ); if ( !i_nb_conversions ) { - vlc_object_detach( pp_filters[0] ); - vlc_object_destroy( pp_filters[0] ); + ReleaseFilter( pp_filters[0] ); msg_Err( p_aout, "couldn't find a filter for the second part of the conversion" ); + return -1; + } + if( *pi_nb_filters + 3 > AOUT_MAX_FILTERS ) + { + ReleaseFilter( pp_filters[0] ); + msg_Err( p_aout, "max filter reached (%d)", AOUT_MAX_FILTERS ); + intf_UserFatal( p_aout, 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, &temp_format ); @@ -191,31 +238,28 @@ int aout_FiltersCreatePipeline( aout_instance_t * p_aout, if ( pp_filters[1] == NULL || pp_filters[2] == NULL ) { - vlc_object_detach( pp_filters[0] ); - vlc_object_destroy( pp_filters[0] ); + ReleaseFilter( pp_filters[0] ); if ( pp_filters[1] != NULL ) { - vlc_object_detach( pp_filters[1] ); - vlc_object_destroy( pp_filters[1] ); + ReleaseFilter( pp_filters[1] ); } if ( pp_filters[2] != NULL ) { - vlc_object_detach( pp_filters[2] ); - vlc_object_destroy( pp_filters[2] ); + ReleaseFilter( pp_filters[2] ); } msg_Err( p_aout, "couldn't find filters for the second part of the conversion" ); + return -1; } - *pi_nb_filters = 3; + *pi_nb_filters += 3; + msg_Dbg( p_aout, "found 3 filters for the whole conversion" ); } else { - *pi_nb_filters = 2; + *pi_nb_filters += 2; + msg_Dbg( p_aout, "found 2 filters for the whole conversion" ); } - /* We have enough filters. */ - msg_Dbg( p_aout, "found %d filters for the whole conversion", - *pi_nb_filters ); return 0; } @@ -227,12 +271,13 @@ void aout_FiltersDestroyPipeline( aout_instance_t * p_aout, 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_destroy( pp_filters[i] ); + vlc_object_release( pp_filters[i] ); } } @@ -246,15 +291,17 @@ void aout_FiltersHintBuffers( aout_instance_t * p_aout, { int i; + (void)p_aout; /* unused */ + for ( i = i_nb_filters - 1; i >= 0; 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 + * 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 + * 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, @@ -292,15 +339,15 @@ void aout_FiltersPlay( aout_instance_t * p_aout, aout_filter_t * p_filter = pp_filters[i]; aout_buffer_t * p_output_buffer; + /* 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 * 1000000 - / p_filter->input.i_rate, *pp_input_buffer, - p_output_buffer ); + ((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 * shall be set by the filter plug-in. */ @@ -312,6 +359,9 @@ void aout_FiltersPlay( aout_instance_t * p_aout, aout_BufferFree( *pp_input_buffer ); *pp_input_buffer = p_output_buffer; } + + if( p_output_buffer->i_nb_samples <= 0 ) + break; } }