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Commit 4f11edd7 authored by Ilkka Ollakka's avatar Ilkka Ollakka
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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.
parent c2e4f7c6
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Showing with 15 additions and 15 deletions
modules/codec/avcodec/encoder.c
View file @ 4f11edd7
......@@ -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
......@@ -878,7 +877,8 @@ errmsg:
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;
......@@ -888,7 +888,6 @@ errmsg:
//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 )
{
......@@ -1127,9 +1126,10 @@ 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
......@@ -1146,7 +1146,7 @@ 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;
......@@ -1169,7 +1169,7 @@ 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);
......@@ -1186,7 +1186,7 @@ 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,
leftover + buffer_delay,
align) < 0 )
......@@ -1277,11 +1277,11 @@ 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),
......@@ -1291,8 +1291,8 @@ 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 );
......@@ -1332,7 +1332,7 @@ 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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