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/*****************************************************************************
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* audio.c: audio decoder using ffmpeg library
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*****************************************************************************
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* Copyright (C) 1999-2003 the VideoLAN team
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* Authors: Laurent Aimar <fenrir@via.ecp.fr>
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* Gildas Bazin <gbazin@videolan.org>
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
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*****************************************************************************/
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/*****************************************************************************
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*****************************************************************************/
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#include <vlc_common.h>
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#include <vlc_codec.h>
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#include <vlc_input.h>
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#ifdef HAVE_LIBAVCODEC_AVCODEC_H
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# include <libavcodec/avcodec.h>
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#elif defined(HAVE_FFMPEG_AVCODEC_H)
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# include <ffmpeg/avcodec.h>
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static const unsigned int pi_channels_maps[7] =
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AOUT_CHAN_CENTER, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT,
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AOUT_CHAN_CENTER | AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARLEFT
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| AOUT_CHAN_REARRIGHT,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
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| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT,
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AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
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| AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT | AOUT_CHAN_LFE
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/*****************************************************************************
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* decoder_sys_t : decoder descriptor
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*****************************************************************************/
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/* Temporary buffer for libavcodec */
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audio_sample_format_t aout_format;
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audio_date_t end_date;
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/*****************************************************************************
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* InitAudioDec: initialize audio decoder
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*****************************************************************************
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* The ffmpeg codec will be opened, some memory allocated.
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*****************************************************************************/
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int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
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AVCodec *p_codec, int i_codec_id, const char *psz_namecodec )
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/* Allocate the memory needed to store the decoder's structure */
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if( ( p_dec->p_sys = p_sys =
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(decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL )
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p_sys->p_context = p_context;
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p_sys->p_codec = p_codec;
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p_sys->i_codec_id = i_codec_id;
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p_sys->psz_namecodec = psz_namecodec;
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/* ***** Fill p_context with init values ***** */
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p_sys->p_context->sample_rate = p_dec->fmt_in.audio.i_rate;
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p_sys->p_context->channels = p_dec->fmt_in.audio.i_channels;
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if( !p_dec->fmt_in.audio.i_physical_channels )
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msg_Warn( p_dec, "Physical channel configuration not set : guessing" );
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p_dec->fmt_in.audio.i_original_channels =
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p_dec->fmt_in.audio.i_physical_channels =
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pi_channels_maps[p_sys->p_context->channels];
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p_dec->fmt_out.audio.i_physical_channels =
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p_dec->fmt_out.audio.i_original_channels =
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p_dec->fmt_in.audio.i_physical_channels;
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p_sys->p_context->block_align = p_dec->fmt_in.audio.i_blockalign;
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p_sys->p_context->bit_rate = p_dec->fmt_in.i_bitrate;
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#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
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p_sys->p_context->bits_per_sample = p_dec->fmt_in.audio.i_bitspersample;
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p_sys->p_context->bits_per_coded_sample = p_dec->fmt_in.audio.i_bitspersample;
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if( p_dec->fmt_in.i_extra > 0 )
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const uint8_t * const p_src = p_dec->fmt_in.p_extra;
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if( p_dec->fmt_in.i_codec == VLC_FOURCC( 'f', 'l', 'a', 'c' ) )
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i_size = p_dec->fmt_in.i_extra - 8;
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else if( p_dec->fmt_in.i_codec == VLC_FOURCC( 'a', 'l', 'a', 'c' ) )
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static const uint8_t p_pattern[] = { 0, 0, 0, 36, 'a', 'l', 'a', 'c' };
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/* Find alac atom XXX it is a bit ugly */
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for( i_offset = 0; i_offset < p_dec->fmt_in.i_extra - sizeof(p_pattern); i_offset++ )
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if( !memcmp( &p_src[i_offset], p_pattern, sizeof(p_pattern) ) )
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i_size = __MIN( p_dec->fmt_in.i_extra - i_offset, 36 );
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i_size = p_dec->fmt_in.i_extra;
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p_sys->p_context->extradata =
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malloc( i_size + FF_INPUT_BUFFER_PADDING_SIZE );
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if( p_sys->p_context->extradata )
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uint8_t *p_dst = p_sys->p_context->extradata;
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p_sys->p_context->extradata_size = i_size;
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memcpy( &p_dst[0], &p_src[i_offset], i_size );
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memset( &p_dst[i_size], 0, FF_INPUT_BUFFER_PADDING_SIZE );
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p_sys->p_context->extradata_size = 0;
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p_sys->p_context->extradata = NULL;
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/* ***** Open the codec ***** */
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vlc_mutex_t *lock = var_AcquireMutex( "avcodec" );
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free( p_sys->p_context->extradata );
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if (avcodec_open( p_sys->p_context, p_sys->p_codec ) < 0)
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vlc_mutex_unlock( lock );
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msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec );
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free( p_sys->p_context->extradata );
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vlc_mutex_unlock( lock );
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msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec );
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p_sys->p_output = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
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p_sys->p_samples = NULL;
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p_sys->i_samples = 0;
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p_sys->i_reject_count = 0;
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p_sys->i_input_rate = INPUT_RATE_DEFAULT;
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aout_DateSet( &p_sys->end_date, 0 );
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if( p_dec->fmt_in.audio.i_rate )
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aout_DateInit( &p_sys->end_date, p_dec->fmt_in.audio.i_rate );
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/* Set output properties */
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p_dec->fmt_out.i_cat = AUDIO_ES;
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p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE;
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p_dec->fmt_out.audio.i_bitspersample = 16;
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/*****************************************************************************
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* SplitBuffer: Needed because aout really doesn't like big audio chunk and
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* wma produces easily > 30000 samples...
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*****************************************************************************/
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static aout_buffer_t *SplitBuffer( decoder_t *p_dec )
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decoder_sys_t *p_sys = p_dec->p_sys;
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int i_samples = __MIN( p_sys->i_samples, 4096 );
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aout_buffer_t *p_buffer;
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if( i_samples == 0 ) return NULL;
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if( ( p_buffer = p_dec->pf_aout_buffer_new( p_dec, i_samples ) ) == NULL )
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msg_Err( p_dec, "cannot get aout buffer" );
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p_buffer->start_date = aout_DateGet( &p_sys->end_date );
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p_buffer->end_date = aout_DateIncrement( &p_sys->end_date,
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i_samples * p_sys->i_input_rate / INPUT_RATE_DEFAULT );
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memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_nb_bytes );
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p_sys->p_samples += p_buffer->i_nb_bytes;
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p_sys->i_samples -= i_samples;
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/*****************************************************************************
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* DecodeAudio: Called to decode one frame
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*****************************************************************************/
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aout_buffer_t * DecodeAudio ( decoder_t *p_dec, block_t **pp_block )
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decoder_sys_t *p_sys = p_dec->p_sys;
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int i_used, i_output;
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aout_buffer_t *p_buffer;
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if( !pp_block || !*pp_block ) return NULL;
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if( p_block->i_rate > 0 )
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p_sys->i_input_rate = p_block->i_rate;
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if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
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block_Release( p_block );
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avcodec_flush_buffers( p_sys->p_context );
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if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 )
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p_sys->i_reject_count = 3;
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if( p_sys->i_samples > 0 )
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p_buffer = SplitBuffer( p_dec );
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if( !p_buffer ) block_Release( p_block );
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if( !aout_DateGet( &p_sys->end_date ) && !p_block->i_pts )
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/* We've just started the stream, wait for the first PTS. */
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block_Release( p_block );
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if( p_block->i_buffer <= 0 )
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block_Release( p_block );
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if( p_block->i_buffer > AVCODEC_MAX_AUDIO_FRAME_SIZE )
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/* Grow output buffer if necessary (eg. for PCM data) */
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p_sys->p_output = realloc(p_sys->p_output, p_block->i_buffer);
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*pp_block = p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE );
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p_block->i_buffer -= FF_INPUT_BUFFER_PADDING_SIZE;
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memset( &p_block->p_buffer[p_block->i_buffer], 0, FF_INPUT_BUFFER_PADDING_SIZE );
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#if LIBAVCODEC_VERSION_INT >= ((52<<16)+(0<<8)+0)
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i_output = __MAX( AVCODEC_MAX_AUDIO_FRAME_SIZE, p_block->i_buffer );
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i_used = avcodec_decode_audio2( p_sys->p_context,
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(int16_t*)p_sys->p_output, &i_output,
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p_block->p_buffer, p_block->i_buffer );
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i_used = avcodec_decode_audio( p_sys->p_context,
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(int16_t*)p_sys->p_output, &i_output,
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p_block->p_buffer, p_block->i_buffer );
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if( i_used < 0 || i_output < 0 )
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msg_Warn( p_dec, "cannot decode one frame (%zu bytes)",
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block_Release( p_block );
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else if( (size_t)i_used > p_block->i_buffer )
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i_used = p_block->i_buffer;
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p_block->i_buffer -= i_used;
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p_block->p_buffer += i_used;
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if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 6 ||
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p_sys->p_context->sample_rate <= 0 )
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msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
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p_sys->p_context->channels, p_sys->p_context->sample_rate );
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block_Release( p_block );
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if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate )
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aout_DateInit( &p_sys->end_date, p_sys->p_context->sample_rate );
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aout_DateSet( &p_sys->end_date, p_block->i_pts );
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/* **** Set audio output parameters **** */
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p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate;
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p_dec->fmt_out.audio.i_channels = p_sys->p_context->channels;
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p_dec->fmt_out.audio.i_original_channels =
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p_dec->fmt_out.audio.i_physical_channels =
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pi_channels_maps[p_sys->p_context->channels];
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if( p_block->i_pts != 0 &&
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p_block->i_pts != aout_DateGet( &p_sys->end_date ) )
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aout_DateSet( &p_sys->end_date, p_block->i_pts );
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/* **** Now we can output these samples **** */
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p_sys->i_samples = i_output / sizeof(int16_t) / p_sys->p_context->channels;
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p_sys->p_samples = p_sys->p_output;
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/* Silent unwanted samples */
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if( p_sys->i_reject_count > 0 )
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memset( p_sys->p_output, 0, i_output );
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p_sys->i_reject_count--;
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p_buffer = SplitBuffer( p_dec );
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if( !p_buffer ) block_Release( p_block );
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/*****************************************************************************
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* EndAudioDec: audio decoder destruction
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*****************************************************************************/
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void EndAudioDec( decoder_t *p_dec )
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decoder_sys_t *p_sys = p_dec->p_sys;
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free( p_sys->p_output );