/*****************************************************************************
* audio.c: audio decoder using ffmpeg library
*****************************************************************************
- * Copyright (C) 1999-2003 VideoLAN
- * $Id: audio.c,v 1.25 2003/11/22 23:39:14 fenrir Exp $
+ * Copyright (C) 1999-2003 the VideoLAN team
+ * $Id$
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
- * Gildas Bazin <gbazin@netcourrier.com>
+ * 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
*
* 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
*****************************************************************************/
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
#include <vlc/vlc.h>
-#include <vlc/decoder.h>
+#include <vlc_aout.h>
+#include <vlc_codec.h>
+#include <vlc_input.h>
/* ffmpeg header */
#ifdef HAVE_FFMPEG_AVCODEC_H
#include "ffmpeg.h"
-static unsigned int pi_channels_maps[6] =
+static unsigned int pi_channels_maps[7] =
{
0,
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_REARLEFT | AOUT_CHAN_REARRIGHT,
+ AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT | AOUT_CHAN_CENTER
+ | AOUT_CHAN_REARLEFT | AOUT_CHAN_REARRIGHT | AOUT_CHAN_LFE
};
/*****************************************************************************
*****************************************************************************/
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;
+ FFMPEG_COMMON_MEMBERS
/* Temporary buffer for libavcodec */
uint8_t *p_output;
*/
audio_sample_format_t aout_format;
audio_date_t end_date;
+
+ /*
+ *
+ */
+ uint8_t *p_samples;
+ int i_samples;
+
+ /* */
+ int i_reject_count;
+
+ int i_input_rate;
};
/*****************************************************************************
* 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 )
+ AVCodec *p_codec, int i_codec_id, const char *psz_namecodec )
{
decoder_sys_t *p_sys;
(decoder_sys_t *)malloc(sizeof(decoder_sys_t)) ) == NULL )
{
msg_Err( p_dec, "out of memory" );
- return VLC_EGENERIC;
+ return VLC_ENOMEM;
}
- p_dec->p_sys->p_context = p_context;
- p_dec->p_sys->p_codec = p_codec;
- p_dec->p_sys->i_codec_id = i_codec_id;
- p_dec->p_sys->psz_namecodec = psz_namecodec;
+ 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_dec->fmt_in.i_extra + FF_INPUT_BUFFER_PADDING_SIZE );
+ malloc( p_sys->p_context->extradata_size +
+ FF_INPUT_BUFFER_PADDING_SIZE );
memcpy( p_sys->p_context->extradata,
- p_dec->fmt_in.p_extra, p_dec->fmt_in.i_extra );
- memset( p_sys->p_context->extradata + p_dec->fmt_in.i_extra, 0,
+ (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 );
}
+ else
+ p_sys->p_context->extradata = NULL;
/* ***** Open the codec ***** */
+ vlc_mutex_t *lock = var_AcquireMutex( "avcodec" );
+ if( lock == NULL )
+ {
+ free( p_sys->p_context->extradata );
+ free( p_sys );
+ return VLC_ENOMEM;
+ }
+
if (avcodec_open( p_sys->p_context, p_sys->p_codec ) < 0)
{
+ vlc_mutex_unlock( lock );
msg_Err( p_dec, "cannot open codec (%s)", p_sys->psz_namecodec );
+ free( p_sys->p_context->extradata );
+ free( p_sys );
return VLC_EGENERIC;
}
- else
- {
- msg_Dbg( p_dec, "ffmpeg codec (%s) started", p_sys->psz_namecodec );
- }
+ vlc_mutex_unlock( lock );
+
+ 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_output = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
+ p_sys->p_samples = NULL;
+ p_sys->i_samples = 0;
+ p_sys->i_reject_count = 0;
+ p_sys->i_input_rate = INPUT_RATE_DEFAULT;
aout_DateSet( &p_sys->end_date, 0 );
+ if( p_dec->fmt_in.audio.i_rate )
+ aout_DateInit( &p_sys->end_date, p_dec->fmt_in.audio.i_rate );
/* 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...
+ *****************************************************************************/
+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 = 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 * p_sys->i_input_rate / INPUT_RATE_DEFAULT );
+
+ 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, i_samples;
- uint8_t *p_samples;
+ int i_used, i_output;
aout_buffer_t *p_buffer;
block_t *p_block;
p_block = *pp_block;
+ if( p_block->i_rate > 0 )
+ p_sys->i_input_rate = p_block->i_rate;
+
+ if( p_block->i_flags & (BLOCK_FLAG_DISCONTINUITY|BLOCK_FLAG_CORRUPTED) )
+ {
+ block_Release( p_block );
+ avcodec_flush_buffers( p_sys->p_context );
+
+ 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( !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 > AVCODEC_MAX_AUDIO_FRAME_SIZE )
+ {
+ /* Grow output buffer if necessary (eg. for PCM data) */
+ p_sys->p_output = realloc(p_sys->p_output, p_block->i_buffer);
+ }
+ *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 );
+
+#if LIBAVCODEC_VERSION_INT >= ((52<<16)+(0<<8)+0)
+ i_output = __MAX( AVCODEC_MAX_AUDIO_FRAME_SIZE, p_block->i_buffer );
+ 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 );
+#else
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 );
+#endif
- if( i_used < 0 )//|| i_output == 0 )
+ if( i_used < 0 || i_output < 0 )
{
if( i_used < 0 )
msg_Warn( p_dec, "cannot decode one frame (%d bytes)",
block_Release( p_block );
return NULL;
}
- else if( i_used > p_block->i_buffer )
+ else if( (size_t)i_used > p_block->i_buffer )
{
i_used = p_block->i_buffer;
}
return NULL;
}
- if( p_dec->fmt_out.audio.i_rate != p_sys->p_context->sample_rate )
+ if( p_dec->fmt_out.audio.i_rate != (unsigned int)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 );
p_block->i_pts = 0;
/* **** Now we can output these samples **** */
- i_samples = i_output / 2 / p_sys->p_context->channels;
- p_samples = p_sys->p_output;
+ p_sys->i_samples = i_output / sizeof(int16_t) / p_sys->p_context->channels;
+ p_sys->p_samples = p_sys->p_output;
- p_buffer = p_dec->pf_aout_buffer_new( p_dec, i_samples );
- if( !p_buffer )
+ /* Silent unwanted samples */
+ if( p_sys->i_reject_count > 0 )
{
- msg_Err( p_dec, "cannot get aout buffer" );
- block_Release( p_block );
- return NULL;
+ memset( p_sys->p_output, 0, i_output );
+ p_sys->i_reject_count--;
}
- 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_samples, p_buffer->i_nb_bytes );
-
+ p_buffer = SplitBuffer( p_dec );
+ if( !p_buffer ) block_Release( p_block );
return p_buffer;
}
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