[vlc-commits] commit: Fixed avcodec audio encoder wrapper. (Laurent Aimar )
git at videolan.org
git at videolan.org
Sun May 2 00:27:11 CEST 2010
vlc | branch: master | Laurent Aimar <fenrir at videolan.org> | Sun May 2 00:09:48 2010 +0200| [95f32b1bf51a5bb03488a59f18feac42b80479d4] | committer: Laurent Aimar
Fixed avcodec audio encoder wrapper.
It closes #3538 and #3496.
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=95f32b1bf51a5bb03488a59f18feac42b80479d4
---
modules/codec/avcodec/encoder.c | 61 ++++++++++++++++++++++++++-------------
1 files changed, 41 insertions(+), 20 deletions(-)
diff --git a/modules/codec/avcodec/encoder.c b/modules/codec/avcodec/encoder.c
index ecbbe91..a39abcc 100644
--- a/modules/codec/avcodec/encoder.c
+++ b/modules/codec/avcodec/encoder.c
@@ -58,6 +58,8 @@
#define MAX_FRAME_DELAY (FF_MAX_B_FRAMES + 2)
+#define RAW_AUDIO_FRAME_SIZE (2048)
+
/*****************************************************************************
* Local prototypes
*****************************************************************************/
@@ -115,6 +117,7 @@ struct encoder_sys_t
/*
* Audio properties
*/
+ int i_sample_bytes;
int i_frame_size;
int i_samples_delay;
mtime_t i_pts;
@@ -551,7 +554,10 @@ int OpenEncoder( vlc_object_t *p_this )
if( i_codec_id == CODEC_ID_MP3 && p_enc->fmt_in.audio.i_channels > 2 )
p_enc->fmt_in.audio.i_channels = 2;
- p_context->codec_type = CODEC_TYPE_AUDIO;
+ p_context->codec_type = CODEC_TYPE_AUDIO;
+ p_context->sample_fmt = p_codec->sample_fmts ?
+ p_codec->sample_fmts[0] :
+ SAMPLE_FMT_S16;
p_enc->fmt_in.i_codec = VLC_CODEC_S16N;
p_context->sample_rate = p_enc->fmt_out.audio.i_rate;
p_context->channels = p_enc->fmt_out.audio.i_channels;
@@ -697,11 +703,23 @@ int OpenEncoder( vlc_object_t *p_this )
if( p_enc->fmt_in.i_cat == AUDIO_ES )
{
- p_sys->i_buffer_out = 2 * AVCODEC_MAX_AUDIO_FRAME_SIZE;
- p_sys->p_buffer_out = malloc( p_sys->i_buffer_out );
- p_sys->i_frame_size = p_context->frame_size * 2 * p_context->channels;
- p_sys->p_buffer = malloc( p_sys->i_frame_size );
+ GetVlcAudioFormat( &p_enc->fmt_in.i_codec,
+ &p_enc->fmt_in.audio.i_bitspersample,
+ p_sys->p_context->sample_fmt );
+ p_sys->i_sample_bytes = (p_enc->fmt_in.audio.i_bitspersample / 8) *
+ p_context->channels;
+ p_sys->i_frame_size = p_context->frame_size > 1 ?
+ p_context->frame_size :
+ RAW_AUDIO_FRAME_SIZE;
+ p_sys->p_buffer = malloc( p_sys->i_frame_size * p_sys->i_sample_bytes );
p_enc->fmt_out.audio.i_blockalign = p_context->block_align;
+ p_enc->fmt_out.audio.i_bitspersample = aout_BitsPerSample( vlc_fourcc_GetCodec( AUDIO_ES, p_enc->fmt_out.i_codec ) );
+
+ if( p_context->frame_size > 1 )
+ p_sys->i_buffer_out = 8 * AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ else
+ p_sys->i_buffer_out = p_sys->i_frame_size * p_sys->i_sample_bytes;
+ p_sys->p_buffer_out = malloc( p_sys->i_buffer_out );
}
msg_Dbg( p_enc, "found encoder %s", psz_namecodec );
@@ -906,6 +924,7 @@ static block_t *EncodeVideo( encoder_t *p_enc, picture_t *p_pict )
static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
{
encoder_sys_t *p_sys = p_enc->p_sys;
+
block_t *p_block, *p_chain = NULL;
uint8_t *p_buffer = p_aout_buf->p_buffer;
@@ -918,27 +937,29 @@ static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
p_sys->i_samples_delay += i_samples;
- while( p_sys->i_samples_delay >= p_sys->p_context->frame_size )
+ while( p_sys->i_samples_delay >= p_sys->i_frame_size )
{
- int16_t *p_samples;
+ void *p_samples;
int i_out;
if( i_samples_delay )
{
/* Take care of the left-over from last time */
- int i_delay_size = i_samples_delay * 2 *
- p_sys->p_context->channels;
- int i_size = p_sys->i_frame_size - i_delay_size;
+ int i_delay_size = i_samples_delay;
+ int i_size = (p_sys->i_frame_size - i_delay_size) *
+ p_sys->i_sample_bytes;
- p_samples = (int16_t *)p_sys->p_buffer;
- memcpy( p_sys->p_buffer + i_delay_size, p_buffer, i_size );
- p_buffer -= i_delay_size;
+ memcpy( p_sys->p_buffer + i_delay_size * p_sys->i_sample_bytes,
+ p_buffer, i_size );
+ p_buffer -= i_delay_size * p_sys->i_sample_bytes;
i_samples += i_samples_delay;
i_samples_delay = 0;
+
+ p_samples = p_sys->p_buffer;
}
else
{
- p_samples = (int16_t *)p_buffer;
+ p_samples = p_buffer;
}
i_out = avcodec_encode_audio( p_sys->p_context, p_sys->p_buffer_out,
@@ -947,9 +968,9 @@ static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
#if 0
msg_Warn( p_enc, "avcodec_encode_audio: %d", i_out );
#endif
- p_buffer += p_sys->i_frame_size;
- p_sys->i_samples_delay -= p_sys->p_context->frame_size;
- i_samples -= p_sys->p_context->frame_size;
+ p_buffer += p_sys->i_frame_size * p_sys->i_sample_bytes;
+ p_sys->i_samples_delay -= p_sys->i_frame_size;
+ i_samples -= p_sys->i_frame_size;
if( i_out <= 0 )
continue;
@@ -958,7 +979,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
memcpy( p_block->p_buffer, p_sys->p_buffer_out, i_out );
p_block->i_length = (mtime_t)1000000 *
- (mtime_t)p_sys->p_context->frame_size /
+ (mtime_t)p_sys->i_frame_size /
(mtime_t)p_sys->p_context->sample_rate;
p_block->i_dts = p_block->i_pts = p_sys->i_pts;
@@ -971,9 +992,9 @@ static block_t *EncodeAudio( encoder_t *p_enc, aout_buffer_t *p_aout_buf )
/* Backup the remaining raw samples */
if( i_samples )
{
- memcpy( &p_sys->p_buffer[i_samples_delay * 2 * p_sys->p_context->channels],
+ memcpy( &p_sys->p_buffer[i_samples_delay * p_sys->i_sample_bytes],
p_buffer,
- i_samples * 2 * p_sys->p_context->channels );
+ i_samples * p_sys->i_sample_bytes );
}
return p_chain;
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