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vlc/modules/codec/avcodec/audio.c

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/*****************************************************************************
* audio.c: audio decoder using ffmpeg library
*****************************************************************************
* Copyright (C) 1999-2003 the VideoLAN team
* $Id$
*
* 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., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <vlc_common.h>
#include <vlc_aout.h>
#include <vlc_codec.h>
#include <vlc_avcodec.h>
/* ffmpeg header */
#ifdef HAVE_LIBAVCODEC_AVCODEC_H
# include <libavcodec/avcodec.h>
#elif defined(HAVE_FFMPEG_AVCODEC_H)
# include <ffmpeg/avcodec.h>
#else
# include <avcodec.h>
#endif
#include "avcodec.h"
/*****************************************************************************
* decoder_sys_t : decoder descriptor
*****************************************************************************/
struct decoder_sys_t
{
FFMPEG_COMMON_MEMBERS
/* Temporary buffer for libavcodec */
int i_output_max;
uint8_t *p_output;
/*
* Output properties
*/
audio_sample_format_t aout_format;
date_t end_date;
/*
*
*/
uint8_t *p_samples;
int i_samples;
/* */
int i_reject_count;
/* */
bool b_extract;
int pi_extraction[AOUT_CHAN_MAX];
int i_previous_channels;
int64_t i_previous_layout;
};
#define BLOCK_FLAG_PRIVATE_REALLOCATED (1 << BLOCK_FLAG_PRIVATE_SHIFT)
static void SetupOutputFormat( decoder_t *p_dec, bool b_trust );
/*****************************************************************************
* InitAudioDec: initialize audio decoder
*****************************************************************************
* The ffmpeg codec will be opened, some memory allocated.
*****************************************************************************/
int InitAudioDec( decoder_t *p_dec, AVCodecContext *p_context,
AVCodec *p_codec, int i_codec_id, const char *psz_namecodec )
{
decoder_sys_t *p_sys;
/* Allocate the memory needed to store the decoder's structure */
if( ( p_dec->p_sys = p_sys = malloc(sizeof(*p_sys)) ) == NULL )
{
return VLC_ENOMEM;
}
p_codec->type = CODEC_TYPE_AUDIO;
p_context->codec_type = CODEC_TYPE_AUDIO;
p_context->codec_id = i_codec_id;
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;
p_sys->b_delayed_open = false;
/* ***** 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_coded_sample = p_dec->fmt_in.audio.i_bitspersample;
if( p_dec->fmt_in.i_extra > 0 )
{
const uint8_t * const p_src = p_dec->fmt_in.p_extra;
int i_offset;
int i_size;
if( p_dec->fmt_in.i_codec == VLC_CODEC_FLAC )
{
i_offset = 8;
i_size = p_dec->fmt_in.i_extra - 8;
}
else if( p_dec->fmt_in.i_codec == VLC_CODEC_ALAC )
{
static const uint8_t p_pattern[] = { 0, 0, 0, 36, 'a', 'l', 'a', 'c' };
/* Find alac atom XXX it is a bit ugly */
for( i_offset = 0; i_offset < p_dec->fmt_in.i_extra - sizeof(p_pattern); i_offset++ )
{
if( !memcmp( &p_src[i_offset], p_pattern, sizeof(p_pattern) ) )
break;
}
i_size = __MIN( p_dec->fmt_in.i_extra - i_offset, 36 );
if( i_size < 36 )
i_size = 0;
}
else
{
i_offset = 0;
i_size = p_dec->fmt_in.i_extra;
}
if( i_size > 0 )
{
p_sys->p_context->extradata =
malloc( i_size + FF_INPUT_BUFFER_PADDING_SIZE );
if( p_sys->p_context->extradata )
{
uint8_t *p_dst = p_sys->p_context->extradata;
p_sys->p_context->extradata_size = i_size;
memcpy( &p_dst[0], &p_src[i_offset], i_size );
memset( &p_dst[i_size], 0, FF_INPUT_BUFFER_PADDING_SIZE );
}
}
}
else
{
p_sys->p_context->extradata_size = 0;
p_sys->p_context->extradata = NULL;
}
/* ***** Open the codec ***** */
int ret;
vlc_avcodec_lock();
ret = avcodec_open( p_sys->p_context, p_sys->p_codec );
vlc_avcodec_unlock();
if( ret < 0 )
{
msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec );
free( p_sys->p_context->extradata );
free( p_sys );
return VLC_EGENERIC;
}
msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec );
switch( i_codec_id )
{
case CODEC_ID_WAVPACK:
p_sys->i_output_max = 8 * sizeof(int32_t) * 131072;
break;
case CODEC_ID_TTA:
p_sys->i_output_max = p_sys->p_context->channels * sizeof(int32_t) * p_sys->p_context->sample_rate * 2;
break;
case CODEC_ID_FLAC:
p_sys->i_output_max = 8 * sizeof(int32_t) * 65535;
break;
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT( 52, 35, 0 )
case CODEC_ID_WMAPRO:
p_sys->i_output_max = 8 * sizeof(float) * 6144; /* (1 << 12) * 3/2 */
break;
#endif
default:
p_sys->i_output_max = 0;
break;
}
if( p_sys->i_output_max < AVCODEC_MAX_AUDIO_FRAME_SIZE )
p_sys->i_output_max = AVCODEC_MAX_AUDIO_FRAME_SIZE;
msg_Dbg( p_dec, "Using %d bytes output buffer", p_sys->i_output_max );
p_sys->p_output = av_malloc( p_sys->i_output_max );
p_sys->p_samples = NULL;
p_sys->i_samples = 0;
p_sys->i_reject_count = 0;
p_sys->b_extract = false;
p_sys->i_previous_channels = 0;
p_sys->i_previous_layout = 0;
/* */
p_dec->fmt_out.i_cat = AUDIO_ES;
/* Try to set as much information as possible but do not trust it */
SetupOutputFormat( p_dec, false );
date_Set( &p_sys->end_date, 0 );
if( p_dec->fmt_out.audio.i_rate )
date_Init( &p_sys->end_date, p_dec->fmt_out.audio.i_rate, 1 );
else if( p_dec->fmt_in.audio.i_rate )
date_Init( &p_sys->end_date, p_dec->fmt_in.audio.i_rate, 1 );
return VLC_SUCCESS;
}
/*****************************************************************************
* SplitBuffer: Needed because aout really doesn't like big audio chunk and
* wma produces easily > 30000 samples...
*****************************************************************************/
static 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 = decoder_NewAudioBuffer( p_dec, i_samples ) ) == NULL )
return NULL;
p_buffer->i_pts = date_Get( &p_sys->end_date );
p_buffer->i_length = date_Increment( &p_sys->end_date, i_samples )
- p_buffer->i_pts;
if( p_sys->b_extract )
aout_ChannelExtract( p_buffer->p_buffer, p_dec->fmt_out.audio.i_channels,
p_sys->p_samples, p_sys->p_context->channels, i_samples,
p_sys->pi_extraction, p_dec->fmt_out.audio.i_bitspersample );
else
memcpy( p_buffer->p_buffer, p_sys->p_samples, p_buffer->i_buffer );
p_sys->p_samples += i_samples * p_sys->p_context->channels * ( p_dec->fmt_out.audio.i_bitspersample / 8 );
p_sys->i_samples -= i_samples;
return p_buffer;
}
/*****************************************************************************
* DecodeAudio: Called to decode one frame
*****************************************************************************/
aout_buffer_t * 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_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
{
block_Release( p_block );
avcodec_flush_buffers( p_sys->p_context );
p_sys->i_samples = 0;
date_Set( &p_sys->end_date, 0 );
if( p_sys->i_codec_id == CODEC_ID_MP2 || p_sys->i_codec_id == CODEC_ID_MP3 )
p_sys->i_reject_count = 3;
return NULL;
}
if( p_sys->i_samples > 0 )
{
/* More data */
p_buffer = SplitBuffer( p_dec );
if( !p_buffer ) block_Release( p_block );
return p_buffer;
}
if( !date_Get( &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 )
{
block_Release( p_block );
return NULL;
}
if( (p_block->i_flags & BLOCK_FLAG_PRIVATE_REALLOCATED) == 0 )
{
*pp_block = p_block = block_Realloc( p_block, 0, p_block->i_buffer + FF_INPUT_BUFFER_PADDING_SIZE );
if( !p_block )
return NULL;
p_block->i_buffer -= FF_INPUT_BUFFER_PADDING_SIZE;
memset( &p_block->p_buffer[p_block->i_buffer], 0, FF_INPUT_BUFFER_PADDING_SIZE );
p_block->i_flags |= BLOCK_FLAG_PRIVATE_REALLOCATED;
}
do
{
i_output = __MAX( p_block->i_buffer, p_sys->i_output_max );
if( i_output > p_sys->i_output_max )
{
/* Grow output buffer if necessary (eg. for PCM data) */
p_sys->p_output = av_realloc( p_sys->p_output, i_output );
}
i_used = avcodec_decode_audio2( 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 (%zu bytes)",
p_block->i_buffer );
block_Release( p_block );
return NULL;
}
else if( (size_t)i_used > p_block->i_buffer )
{
i_used = p_block->i_buffer;
}
p_block->i_buffer -= i_used;
p_block->p_buffer += i_used;
} while( p_block->i_buffer > 0 && i_output <= 0 );
if( p_sys->p_context->channels <= 0 || p_sys->p_context->channels > 8 ||
p_sys->p_context->sample_rate <= 0 )
{
msg_Warn( p_dec, "invalid audio properties channels count %d, sample rate %d",
p_sys->p_context->channels, p_sys->p_context->sample_rate );
block_Release( p_block );
return NULL;
}
if( p_dec->fmt_out.audio.i_rate != (unsigned int)p_sys->p_context->sample_rate )
{
date_Init( &p_sys->end_date, p_sys->p_context->sample_rate, 1 );
date_Set( &p_sys->end_date, p_block->i_pts );
}
/* **** Set audio output parameters **** */
SetupOutputFormat( p_dec, true );
if( p_block->i_pts != 0 &&
p_block->i_pts != date_Get( &p_sys->end_date ) )
{
date_Set( &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 / (p_dec->fmt_out.audio.i_bitspersample / 8) / p_sys->p_context->channels;
p_sys->p_samples = p_sys->p_output;
/* Silent unwanted samples */
if( p_sys->i_reject_count > 0 )
{
memset( p_sys->p_output, 0, i_output );
p_sys->i_reject_count--;
}
p_buffer = SplitBuffer( p_dec );
if( !p_buffer ) block_Release( p_block );
return p_buffer;
}
/*****************************************************************************
* EndAudioDec: audio decoder destruction
*****************************************************************************/
void EndAudioDec( decoder_t *p_dec )
{
decoder_sys_t *p_sys = p_dec->p_sys;
av_free( p_sys->p_output );
}
/*****************************************************************************
*
*****************************************************************************/
void GetVlcAudioFormat( vlc_fourcc_t *pi_codec, unsigned *pi_bits, int i_sample_fmt )
{
switch( i_sample_fmt )
{
case SAMPLE_FMT_U8:
*pi_codec = VLC_CODEC_U8;
*pi_bits = 8;
break;
case SAMPLE_FMT_S32:
*pi_codec = VLC_CODEC_S32N;
*pi_bits = 32;
break;
case SAMPLE_FMT_FLT:
*pi_codec = VLC_CODEC_FL32;
*pi_bits = 32;
break;
case SAMPLE_FMT_DBL:
*pi_codec = VLC_CODEC_FL64;
*pi_bits = 64;
break;
case SAMPLE_FMT_S16:
default:
*pi_codec = VLC_CODEC_S16N;
*pi_bits = 16;
break;
}
}
static const uint64_t pi_channels_map[][2] =
{
{ CH_FRONT_LEFT, AOUT_CHAN_LEFT },
{ CH_FRONT_RIGHT, AOUT_CHAN_RIGHT },
{ CH_FRONT_CENTER, AOUT_CHAN_CENTER },
{ CH_LOW_FREQUENCY, AOUT_CHAN_LFE },
{ CH_BACK_LEFT, AOUT_CHAN_REARLEFT },
{ CH_BACK_RIGHT, AOUT_CHAN_REARRIGHT },
{ CH_FRONT_LEFT_OF_CENTER, 0 },
{ CH_FRONT_RIGHT_OF_CENTER, 0 },
{ CH_BACK_CENTER, AOUT_CHAN_REARCENTER },
{ CH_SIDE_LEFT, AOUT_CHAN_MIDDLELEFT },
{ CH_SIDE_RIGHT, AOUT_CHAN_MIDDLERIGHT },
{ CH_TOP_CENTER, 0 },
{ CH_TOP_FRONT_LEFT, 0 },
{ CH_TOP_FRONT_CENTER, 0 },
{ CH_TOP_FRONT_RIGHT, 0 },
{ CH_TOP_BACK_LEFT, 0 },
{ CH_TOP_BACK_CENTER, 0 },
{ CH_TOP_BACK_RIGHT, 0 },
{ CH_STEREO_LEFT, 0 },
{ CH_STEREO_RIGHT, 0 },
};
static void SetupOutputFormat( decoder_t *p_dec, bool b_trust )
{
decoder_sys_t *p_sys = p_dec->p_sys;
GetVlcAudioFormat( &p_dec->fmt_out.i_codec,
&p_dec->fmt_out.audio.i_bitspersample,
p_sys->p_context->sample_fmt );
p_dec->fmt_out.audio.i_rate = p_sys->p_context->sample_rate;
/* */
if( p_sys->i_previous_channels == p_sys->p_context->channels &&
p_sys->i_previous_layout == p_sys->p_context->channel_layout )
return;
if( b_trust )
{
p_sys->i_previous_channels = p_sys->p_context->channels;
p_sys->i_previous_layout = p_sys->p_context->channel_layout;
}
/* Specified order
* FIXME should we use fmt_in.audio.i_physical_channels or not ?
*/
const unsigned i_order_max = 8 * sizeof(p_sys->p_context->channel_layout);
uint32_t pi_order_src[i_order_max];
int i_channels_src = 0;
if( p_sys->p_context->channel_layout )
{
for( unsigned i = 0; i < sizeof(pi_channels_map)/sizeof(*pi_channels_map); i++ )
{
if( p_sys->p_context->channel_layout & pi_channels_map[i][0] )
pi_order_src[i_channels_src++] = pi_channels_map[i][1];
}
}
else
{
/* Create default order */
if( b_trust )
msg_Warn( p_dec, "Physical channel configuration not set : guessing" );
for( unsigned int i = 0; i < __MIN( i_order_max, (unsigned)p_sys->p_context->channels ); i++ )
{
if( i < sizeof(pi_channels_map)/sizeof(*pi_channels_map) )
pi_order_src[i_channels_src++] = pi_channels_map[i][1];
}
}
if( i_channels_src != p_sys->p_context->channels && b_trust )
msg_Err( p_dec, "Channel layout not understood" );
uint32_t i_layout_dst;
int i_channels_dst;
p_sys->b_extract = aout_CheckChannelExtraction( p_sys->pi_extraction,
&i_layout_dst, &i_channels_dst,
NULL, pi_order_src, i_channels_src );
if( i_channels_dst != i_channels_src && b_trust )
msg_Warn( p_dec, "%d channels are dropped", i_channels_src - i_channels_dst );
p_dec->fmt_out.audio.i_physical_channels =
p_dec->fmt_out.audio.i_original_channels = i_layout_dst;
p_dec->fmt_out.audio.i_channels = i_channels_dst;
}