📄 audio.c
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/***************************************************************************** * audio.c: audio decoder using ffmpeg library ***************************************************************************** * Copyright (C) 1999-2003 VideoLAN * $Id: audio.c 10191 2005-03-07 20:13:56Z robux4 $ * * Authors: Laurent Aimar <fenrir@via.ecp.fr> * Gildas Bazin <gbazin@videolan.org> * * This program is free software; you can redistribute it and/or modify * 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 * GNU General Public License for more details. * * 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. *****************************************************************************//***************************************************************************** * Preamble *****************************************************************************/#include <vlc/vlc.h>#include <vlc/decoder.h>/* ffmpeg header */#ifdef HAVE_FFMPEG_AVCODEC_H# include <ffmpeg/avcodec.h>#else# include <avcodec.h>#endif#include "ffmpeg.h"static unsigned int pi_channels_maps[7] ={ 0, AOUT_CHAN_CENTER, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT, AOUT_CHAN_CENTER | AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT, AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT | AOUT_CHAN_LFE};/***************************************************************************** * decoder_sys_t : decoder descriptor *****************************************************************************/struct decoder_sys_t{ /* Common part between video and audio decoder */ int i_cat; int i_codec_id; char *psz_namecodec; AVCodecContext *p_context; AVCodec *p_codec; /* Temporary buffer for libavcodec */ uint8_t *p_output; /* * Output properties */ audio_sample_format_t aout_format; audio_date_t end_date; /* * */ uint8_t *p_samples; int i_samples;};/***************************************************************************** * InitAudioDec: initialize audio decoder ***************************************************************************** * The ffmpeg codec will be opened, some memory allocated. *****************************************************************************/int E_(InitAudioDec)( decoder_t *p_dec, AVCodecContext *p_context, AVCodec *p_codec, int i_codec_id, char *psz_namecodec ){ decoder_sys_t *p_sys; vlc_value_t lockval; var_Get( p_dec->p_libvlc, "avcodec", &lockval ); /* Allocate the memory needed to store the decoder's structure */ if( ( p_dec->p_sys = p_sys = (decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL ) { msg_Err( p_dec, "out of memory" ); return VLC_EGENERIC; } p_sys->p_context = p_context; p_sys->p_codec = p_codec; p_sys->i_codec_id = i_codec_id; p_sys->psz_namecodec = psz_namecodec; /* ***** Fill p_context with init values ***** */ p_sys->p_context->sample_rate = p_dec->fmt_in.audio.i_rate; p_sys->p_context->channels = p_dec->fmt_in.audio.i_channels; p_sys->p_context->block_align = p_dec->fmt_in.audio.i_blockalign; p_sys->p_context->bit_rate = p_dec->fmt_in.i_bitrate; p_sys->p_context->bits_per_sample = p_dec->fmt_in.audio.i_bitspersample; if( ( p_sys->p_context->extradata_size = p_dec->fmt_in.i_extra ) > 0 ) { int i_offset = 0; if( p_dec->fmt_in.i_codec == VLC_FOURCC( 'f', 'l', 'a', 'c' ) ) i_offset = 8; p_sys->p_context->extradata_size -= i_offset; p_sys->p_context->extradata = malloc( p_sys->p_context->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE ); memcpy( p_sys->p_context->extradata, (char*)p_dec->fmt_in.p_extra + i_offset, p_sys->p_context->extradata_size ); memset( (char*)p_sys->p_context->extradata + p_sys->p_context->extradata_size, 0, FF_INPUT_BUFFER_PADDING_SIZE ); } /* ***** Open the codec ***** */ vlc_mutex_lock( lockval.p_address ); if (avcodec_open( p_sys->p_context, p_sys->p_codec ) < 0) { vlc_mutex_unlock( lockval.p_address ); msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec ); free( p_sys ); return VLC_EGENERIC; } vlc_mutex_unlock( lockval.p_address ); msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec ); p_sys->p_output = malloc( 3 * AVCODEC_MAX_AUDIO_FRAME_SIZE ); p_sys->p_samples = NULL; p_sys->i_samples = 0; if( p_dec->fmt_in.audio.i_rate ) { aout_DateInit( &p_sys->end_date, p_dec->fmt_in.audio.i_rate ); aout_DateSet( &p_sys->end_date, 0 ); } /* Set output properties */ p_dec->fmt_out.i_cat = AUDIO_ES; p_dec->fmt_out.i_codec = AOUT_FMT_S16_NE; p_dec->fmt_out.audio.i_bitspersample = 16; return VLC_SUCCESS;}/***************************************************************************** * SplitBuffer: Needed because aout really doesn't like big audio chunk and * wma produces easily > 30000 samples... *****************************************************************************/aout_buffer_t *SplitBuffer( decoder_t *p_dec ){ decoder_sys_t *p_sys = p_dec->p_sys; int i_samples = __MIN( p_sys->i_samples, 4096 ); aout_buffer_t *p_buffer; if( i_samples == 0 ) return NULL; if( ( p_buffer = p_dec->pf_aout_buffer_new( p_dec, i_samples ) ) == NULL ) { msg_Err( p_dec, "cannot get aout buffer" ); return NULL; } p_buffer->start_date = aout_DateGet( &p_sys->end_date ); p_buffer->end_date = aout_DateIncrement( &p_sys->end_date, i_samples ); memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_nb_bytes ); p_sys->p_samples += p_buffer->i_nb_bytes; p_sys->i_samples -= i_samples; return p_buffer;}/***************************************************************************** * DecodeAudio: Called to decode one frame *****************************************************************************/aout_buffer_t *E_( DecodeAudio )( decoder_t *p_dec, block_t **pp_block ){ decoder_sys_t *p_sys = p_dec->p_sys; int i_used, i_output; aout_buffer_t *p_buffer; block_t *p_block; if( !pp_block || !*pp_block ) return NULL; p_block = *pp_block; if( p_block->i_buffer <= 0 && p_sys->i_samples > 0 ) { /* More data */ p_buffer = SplitBuffer( p_dec ); if( !p_buffer ) block_Release( p_block ); return p_buffer; } if( !aout_DateGet( &p_sys->end_date ) && !p_block->i_pts ) { /* We've just started the stream, wait for the first PTS. */ block_Release( p_block ); return NULL; } if( p_block->i_buffer <= 0 || ( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) ) ) { block_Release( p_block ); return NULL; } i_used = avcodec_decode_audio( p_sys->p_context, (int16_t*)p_sys->p_output, &i_output, p_block->p_buffer, p_block->i_buffer ); if( i_used < 0 || i_output < 0 ) { if( i_used < 0 ) msg_Warn( p_dec, "cannot decode one frame (%d bytes)", p_block->i_buffer ); block_Release( p_block ); return NULL; } else if( i_used > p_block->i_buffer ) { i_used = p_block->i_buffer; } p_block->i_buffer -= i_used; p_block->p_buffer += i_used; if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 6 ) { msg_Warn( p_dec, "invalid channels count %d", p_sys->p_context->channels ); block_Release( p_block ); return NULL; } if( p_dec->fmt_out.audio.i_rate != p_sys->p_context->sample_rate ) { aout_DateInit( &p_sys->end_date, p_sys->p_context->sample_rate ); aout_DateSet( &p_sys->end_date, p_block->i_pts ); } /* **** Set audio output parameters **** */ p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate; p_dec->fmt_out.audio.i_channels = p_sys->p_context->channels; p_dec->fmt_out.audio.i_original_channels = p_dec->fmt_out.audio.i_physical_channels = pi_channels_maps[p_sys->p_context->channels]; if( p_block->i_pts != 0 && p_block->i_pts != aout_DateGet( &p_sys->end_date ) ) { aout_DateSet( &p_sys->end_date, p_block->i_pts ); } p_block->i_pts = 0; /* **** Now we can output these samples **** */ p_sys->i_samples = i_output / 2 / p_sys->p_context->channels; p_sys->p_samples = p_sys->p_output; p_buffer = SplitBuffer( p_dec ); if( !p_buffer ) block_Release( p_block ); return p_buffer;}/***************************************************************************** * EndAudioDec: audio decoder destruction *****************************************************************************/void E_(EndAudioDec)( decoder_t *p_dec ){ decoder_sys_t *p_sys = p_dec->p_sys; if( p_sys->p_output ) free( p_sys->p_output );}
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