[vlc-commits] avcodec: use context->channels in audio encoding
Ilkka Ollakka
git at videolan.org
Sun Sep 22 16:08:55 CEST 2013
vlc/vlc-2.1 | branch: master | Ilkka Ollakka <ileoo at videolan.org> | Sun Sep 22 16:52:19 2013 +0300| [eb8e5c4dc130cc04abc28c7b1c9225fac708950f] | committer: Ilkka Ollakka
avcodec: use context->channels in audio encoding
Input and output channel count should be anyway the same and it's
hopefully littlebit clearer than fmt_in and fmt_out mixups.
(cherry picked from commit 4f11edd7bcacea1fa0b44161b76091e98aeed7a2)
Signed-off-by: Ilkka Ollakka <ileoo at videolan.org>
> http://git.videolan.org/gitweb.cgi/vlc/vlc-2.1.git/?a=commit;h=eb8e5c4dc130cc04abc28c7b1c9225fac708950f
---
modules/codec/avcodec/encoder.c | 30 +++++++++++++++---------------
1 file changed, 15 insertions(+), 15 deletions(-)
diff --git a/modules/codec/avcodec/encoder.c b/modules/codec/avcodec/encoder.c
index 8193d07..a82d235 100644
--- a/modules/codec/avcodec/encoder.c
+++ b/modules/codec/avcodec/encoder.c
@@ -605,9 +605,8 @@ int OpenEncoder( vlc_object_t *p_this )
else if( p_enc->fmt_in.i_cat == AUDIO_ES )
{
/* work around bug in libmp3lame encoding */
- if( i_codec_id == AV_CODEC_ID_MP3 && p_enc->fmt_in.audio.i_channels > 2 )
- p_enc->fmt_in.audio.i_channels = 2;
-
+ if( i_codec_id == AV_CODEC_ID_MP3 && p_enc->fmt_out.audio.i_channels > 2 )
+ p_enc->fmt_out.audio.i_channels = 2;
p_context->codec_type = AVMEDIA_TYPE_AUDIO;
p_context->sample_fmt = p_codec->sample_fmts ?
p_codec->sample_fmts[0] :
@@ -653,7 +652,7 @@ int OpenEncoder( vlc_object_t *p_this )
date_Set( &p_sys->buffer_date, 0 );
p_context->time_base.num = 1;
p_context->time_base.den = p_context->sample_rate;
- p_context->channels = p_enc->fmt_out.audio.i_channels;
+ p_context->channels = p_enc->fmt_out.audio.i_channels;
#if LIBAVUTIL_VERSION_CHECK( 52, 2, 6, 0, 0)
p_context->channel_layout = av_get_default_channel_layout( p_context->channels );
#endif
@@ -870,7 +869,8 @@ int OpenEncoder( vlc_object_t *p_this )
p_sys->i_frame_size = p_context->frame_size > 1 ?
p_context->frame_size :
FF_MIN_BUFFER_SIZE;
- p_sys->p_buffer = malloc( p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels);
+ p_sys->i_buffer_out = p_sys->i_frame_size * p_sys->i_sample_bytes * p_sys->p_context->channels;
+ p_sys->p_buffer = malloc( p_sys->i_buffer_out );
if ( unlikely( p_sys->p_buffer == NULL ) )
{
goto error;
@@ -880,7 +880,6 @@ int OpenEncoder( vlc_object_t *p_this )
//b_variable tells if we can feed any size frames to encoder
p_sys->b_variable = p_context->frame_size ? false : true;
- p_sys->i_buffer_out = p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels;
if( p_sys->b_planar )
{
@@ -1119,9 +1118,10 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
int got_packet,i_out;
size_t buffer_delay = 0, i_samples_left = 0;
+
//i_bytes_left is amount of bytes we get
i_samples_left = p_aout_buf ? p_aout_buf->i_nb_samples : 0;
- buffer_delay = p_sys->i_samples_delay * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels;
+ buffer_delay = p_sys->i_samples_delay * p_sys->i_sample_bytes * p_sys->p_context->channels;
//p_sys->i_buffer_out = p_sys->i_frame_size * chan * p_sys->i_sample_bytes
//Calculate how many bytes we would need from current buffer to fill frame
@@ -1138,7 +1138,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
)
{
//How much we need to copy from new packet
- const int leftover = leftover_samples * p_enc->fmt_in.audio.i_channels * p_sys->i_sample_bytes;
+ const int leftover = leftover_samples * p_sys->p_context->channels * p_sys->i_sample_bytes;
#if LIBAVUTIL_VERSION_CHECK( 51,27,2,46,100 )
const int align = 0;
@@ -1161,7 +1161,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
// We need to deinterleave from p_aout_buf to p_buffer the leftover bytes
if( p_sys->b_planar )
aout_Deinterleave( p_sys->p_interleave_buf, p_sys->p_buffer,
- p_sys->i_frame_size, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec );
+ p_sys->i_frame_size, p_sys->p_context->channels, p_enc->fmt_in.i_codec );
else
memcpy( p_sys->p_buffer + buffer_delay, p_aout_buf->p_buffer, leftover);
@@ -1178,7 +1178,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
memset( p_sys->p_buffer + (leftover+buffer_delay), 0, padding_size );
buffer_delay += padding_size;
}
- if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
+ if( avcodec_fill_audio_frame( p_sys->frame, p_sys->p_context->channels,
p_sys->p_context->sample_fmt, p_sys->b_planar ? p_sys->p_interleave_buf : p_sys->p_buffer,
p_sys->i_buffer_out,
align) < 0 )
@@ -1269,11 +1269,11 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
if( p_sys->b_planar )
{
aout_Deinterleave( p_sys->p_buffer, p_aout_buf->p_buffer,
- p_sys->frame->nb_samples, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec );
+ p_sys->frame->nb_samples, p_sys->p_context->channels, p_enc->fmt_in.i_codec );
}
- if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
+ if( avcodec_fill_audio_frame( p_sys->frame, p_sys->p_context->channels,
p_sys->p_context->sample_fmt,
p_sys->b_planar ? p_sys->p_buffer : p_aout_buf->p_buffer,
__MIN(p_sys->i_buffer_out, p_aout_buf->i_buffer),
@@ -1283,8 +1283,8 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
p_sys->frame->nb_samples = 0;
}
- p_aout_buf->p_buffer += (p_sys->frame->nb_samples * p_enc->fmt_in.audio.i_channels * p_sys->i_sample_bytes);
- p_aout_buf->i_buffer -= (p_sys->frame->nb_samples * p_enc->fmt_in.audio.i_channels * p_sys->i_sample_bytes);
+ p_aout_buf->p_buffer += (p_sys->frame->nb_samples * p_sys->p_context->channels * p_sys->i_sample_bytes);
+ p_aout_buf->i_buffer -= (p_sys->frame->nb_samples * p_sys->p_context->channels * p_sys->i_sample_bytes);
p_aout_buf->i_nb_samples -= p_sys->frame->nb_samples;
date_Increment( &p_sys->buffer_date, p_sys->frame->nb_samples );
@@ -1324,7 +1324,7 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
if( p_aout_buf->i_nb_samples > 0 )
{
memcpy( p_sys->p_buffer + buffer_delay, p_aout_buf->p_buffer,
- p_aout_buf->i_nb_samples * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels);
+ p_aout_buf->i_nb_samples * p_sys->i_sample_bytes * p_sys->p_context->channels);
p_sys->i_samples_delay += p_aout_buf->i_nb_samples;
}
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