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Commit 8fc2c5f5 authored by Ilkka Ollakka's avatar Ilkka Ollakka
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avcodec: cleanup encodeAudio 

Handle better cases when we don't have input frame size
same as encoder input frame size. Also gather input samples
untill we get full frame fillet and not just next block.

Set length based on used samples and not frame_size

Also try to make memory calculations cleaner so they don't
overshoot and write stuff all over.

Refs #8265
parent 64d1ed4e
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Inline Side-by-side
Showing with 96 additions and 78 deletions
modules/codec/avcodec/encoder.c
View file @ 8fc2c5f5
...@@ -114,10 +114,12 @@ struct encoder_sys_t ...@@ -114,10 +114,12 @@ struct encoder_sys_t
/* /*
* Audio properties * Audio properties
*/ */
int i_sample_bytes; size_t i_sample_bytes;
int i_frame_size; size_t i_frame_size;
int i_samples_delay; size_t i_samples_delay; //How much samples in delay buffer
bool b_planar;
mtime_t i_pts; mtime_t i_pts;
date_t buffer_date;
/* Encoding settings */ /* Encoding settings */
int i_key_int; int i_key_int;
...@@ -304,6 +306,7 @@ int OpenEncoder( vlc_object_t *p_this ) ...@@ -304,6 +306,7 @@ int OpenEncoder( vlc_object_t *p_this )
p_enc->p_sys = p_sys; p_enc->p_sys = p_sys;
p_sys->i_samples_delay = 0; p_sys->i_samples_delay = 0;
p_sys->p_codec = p_codec; p_sys->p_codec = p_codec;
p_sys->b_planar = false;
p_sys->p_buffer = NULL; p_sys->p_buffer = NULL;
p_sys->i_buffer_out = 0; p_sys->i_buffer_out = 0;
...@@ -617,9 +620,10 @@ int OpenEncoder( vlc_object_t *p_this ) ...@@ -617,9 +620,10 @@ int OpenEncoder( vlc_object_t *p_this )
} }
} }
} }
p_sys->b_planar = av_sample_fmt_is_planar( p_context->sample_fmt );
// Try if we can use interleaved format for codec input as VLC doesn't really do planar audio yet // Try if we can use interleaved format for codec input as VLC doesn't really do planar audio yet
// FIXME: Remove when planar/interleaved audio in vlc is equally supported // FIXME: Remove when planar/interleaved audio in vlc is equally supported
if( av_sample_fmt_is_planar( p_context->sample_fmt ) ) if( p_sys->b_planar )
{ {
msg_Dbg( p_enc, "Trying to find packet sample format instead of planar %s", av_get_sample_fmt_name( p_context->sample_fmt ) ); msg_Dbg( p_enc, "Trying to find packet sample format instead of planar %s", av_get_sample_fmt_name( p_context->sample_fmt ) );
for( unsigned int i=0; p_codec->sample_fmts[i] != -1; i++ ) for( unsigned int i=0; p_codec->sample_fmts[i] != -1; i++ )
...@@ -634,8 +638,11 @@ int OpenEncoder( vlc_object_t *p_this ) ...@@ -634,8 +638,11 @@ int OpenEncoder( vlc_object_t *p_this )
} }
msg_Dbg( p_enc, "Ended up using %s as sample format", av_get_sample_fmt_name( p_context->sample_fmt ) ); msg_Dbg( p_enc, "Ended up using %s as sample format", av_get_sample_fmt_name( p_context->sample_fmt ) );
p_enc->fmt_in.i_codec = GetVlcAudioFormat( p_context->sample_fmt ); p_enc->fmt_in.i_codec = GetVlcAudioFormat( p_context->sample_fmt );
p_sys->b_planar = av_sample_fmt_is_planar( p_context->sample_fmt );
p_context->sample_rate = p_enc->fmt_out.audio.i_rate; p_context->sample_rate = p_enc->fmt_out.audio.i_rate;
date_Init( &p_sys->buffer_date, p_enc->fmt_out.audio.i_rate, 1 );
date_Set( &p_sys->buffer_date, 0 );
p_context->time_base.num = 1; p_context->time_base.num = 1;
p_context->time_base.den = p_context->sample_rate; 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;
...@@ -844,12 +851,11 @@ int OpenEncoder( vlc_object_t *p_this ) ...@@ -844,12 +851,11 @@ int OpenEncoder( vlc_object_t *p_this )
p_enc->fmt_in.i_codec = GetVlcAudioFormat( p_sys->p_context->sample_fmt ); p_enc->fmt_in.i_codec = GetVlcAudioFormat( p_sys->p_context->sample_fmt );
p_enc->fmt_in.audio.i_bitspersample = aout_BitsPerSample( p_enc->fmt_in.i_codec ); p_enc->fmt_in.audio.i_bitspersample = aout_BitsPerSample( p_enc->fmt_in.i_codec );
p_sys->i_sample_bytes = (p_enc->fmt_in.audio.i_bitspersample / 8) * 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_sys->i_frame_size = p_context->frame_size > 1 ?
p_context->frame_size : p_context->frame_size :
FF_MIN_BUFFER_SIZE; FF_MIN_BUFFER_SIZE;
p_sys->p_buffer = malloc( p_sys->i_frame_size * p_sys->i_sample_bytes ); p_sys->p_buffer = malloc( p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels);
if ( unlikely( p_sys->p_buffer == NULL ) ) if ( unlikely( p_sys->p_buffer == NULL ) )
{ {
goto error; goto error;
...@@ -857,7 +863,7 @@ int OpenEncoder( vlc_object_t *p_this ) ...@@ -857,7 +863,7 @@ int OpenEncoder( vlc_object_t *p_this )
p_enc->fmt_out.audio.i_blockalign = p_context->block_align; p_enc->fmt_out.audio.i_blockalign = p_context->block_align;
p_enc->fmt_out.audio.i_bitspersample = aout_BitsPerSample( p_enc->fmt_out.i_codec ); p_enc->fmt_out.audio.i_bitspersample = aout_BitsPerSample( p_enc->fmt_out.i_codec );
p_sys->i_buffer_out = p_sys->i_frame_size * p_sys->i_sample_bytes; p_sys->i_buffer_out = p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels;
} }
p_sys->frame = avcodec_alloc_frame(); p_sys->frame = avcodec_alloc_frame();
...@@ -1091,49 +1097,60 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf ) ...@@ -1091,49 +1097,60 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
encoder_sys_t *p_sys = p_enc->p_sys; encoder_sys_t *p_sys = p_enc->p_sys;
block_t *p_block, *p_chain = NULL; block_t *p_block, *p_chain = NULL;
int got_packet,i_out,i_samples_left=0,i_data_offset = 0; int got_packet,i_out;
size_t buffer_delay = 0, i_samples_left = 0;
//i_samples_left is amount of samples we get //i_bytes_left is amount of bytes we get
i_samples_left = p_aout_buf ? p_aout_buf->i_nb_samples : 0; 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;
if( p_sys->i_samples_delay > 0 )
//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
size_t leftover_samples = __MAX(0,__MIN((ssize_t)i_samples_left, (ssize_t)(p_sys->i_frame_size - p_sys->i_samples_delay)));
// Check if we have enough samples in delay_buffer and current p_aout_buf to fill frame
// Or if we are cleaning up
if( ( buffer_delay > 0 ) &&
( ( p_aout_buf && ( leftover_samples <= p_aout_buf->i_nb_samples ) &&
( (leftover_samples + p_sys->i_samples_delay ) >= p_sys->i_frame_size )
) ||
( !p_aout_buf )
)
)
{ {
AVPacket packet;
//How much we need to copy from new packet //How much we need to copy from new packet
const int leftover = __MAX(0,__MIN(i_samples_left, (p_sys->i_frame_size - p_sys->i_samples_delay))); const int leftover = leftover_samples * p_enc->fmt_in.audio.i_channels * p_sys->i_sample_bytes;
AVPacket packet = {0};
avcodec_get_frame_defaults( p_sys->frame ); avcodec_get_frame_defaults( p_sys->frame );
p_sys->frame->nb_samples = p_sys->i_samples_delay + leftover;
p_sys->frame->format = p_sys->p_context->sample_fmt; p_sys->frame->format = p_sys->p_context->sample_fmt;
p_sys->frame->pts = date_Get( &p_sys->buffer_date );
p_sys->frame->nb_samples = leftover_samples + p_sys->i_samples_delay;
date_Increment( &p_sys->buffer_date, p_sys->i_frame_size );
//Copy samples from new packet to buffer to get frame size if( likely( p_aout_buf ) )
if( likely( leftover ) )
{ {
if( av_sample_fmt_is_planar( p_sys->p_context->sample_fmt ) ) p_aout_buf->i_nb_samples -= leftover_samples;
aout_Deinterleave( p_sys->p_buffer+(p_sys->i_samples_delay*p_sys->i_sample_bytes*p_enc->fmt_in.audio.i_channels ),
p_aout_buf->p_buffer, leftover, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec ); // We need to deinterleave from p_aout_buf to p_buffer the leftover bytes
if( p_sys->b_planar )
aout_Deinterleave( p_sys->p_buffer+buffer_delay, p_aout_buf->p_buffer,
leftover_samples, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec );
else else
memcpy( p_sys->p_buffer+(p_sys->i_samples_delay*p_sys->i_sample_bytes), memcpy( p_sys->p_buffer + buffer_delay, p_aout_buf->p_buffer, leftover);
p_aout_buf->p_buffer,
leftover * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels
);
}
p_aout_buf->p_buffer += leftover;
p_aout_buf->i_buffer -= leftover;
p_aout_buf->i_pts = date_Get( &p_sys->buffer_date );
}
if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels, if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
p_sys->p_context->sample_fmt, p_sys->p_context->sample_fmt, p_sys->p_buffer,
p_sys->p_buffer, leftover + buffer_delay,
(p_sys->i_samples_delay + leftover) * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels, 0) < 0 )
0) < 0 ) msg_Err( p_enc, "filling error on fillup" );
msg_Err( p_enc, "Filling on leftovers error i_leftover %d i_samples_left %d samples_delay %d frame size %d", leftover, i_samples_left, p_sys->i_samples_delay, p_sys->i_frame_size );
if( likely( p_aout_buf ) )
p_sys->frame->pts = p_aout_buf->i_pts -
(mtime_t)1000000 * (mtime_t)p_sys->i_samples_delay /
(mtime_t)p_enc->fmt_in.audio.i_rate;
buffer_delay = 0;
p_sys->i_samples_delay = 0; p_sys->i_samples_delay = 0;
i_data_offset += leftover * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels;
i_samples_left -= leftover;
p_block = block_Alloc( p_sys->i_buffer_out ); p_block = block_Alloc( p_sys->i_buffer_out );
av_init_packet( &packet ); av_init_packet( &packet );
...@@ -1141,34 +1158,33 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf ) ...@@ -1141,34 +1158,33 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
packet.size = p_block->i_buffer; packet.size = p_block->i_buffer;
i_out = avcodec_encode_audio2( p_sys->p_context, &packet, p_sys->frame, &got_packet ); i_out = avcodec_encode_audio2( p_sys->p_context, &packet, p_sys->frame, &got_packet );
p_block->i_buffer = packet.size;
if( unlikely( !got_packet || ( i_out < 0 ) || !packet.size ) )
if( unlikely( !got_packet || ( i_out < 0 ) ) )
{ {
if( i_out < 0 ) if( i_out < 0 )
{ {
msg_Err( p_enc,"Encoding problem..."); msg_Err( p_enc,"Encoding problem..");
return p_chain; return p_chain;
} }
} else { block_Release( p_block );
p_block->i_buffer = packet.size; return NULL;
}
p_block->i_length = (mtime_t)1000000 * p_block->i_buffer = packet.size;
(mtime_t)p_sys->i_frame_size / p_block->i_length = (mtime_t)1000000 *
(mtime_t)p_sys->p_context->sample_rate; (mtime_t)p_sys->frame->nb_samples /
(mtime_t)p_sys->p_context->sample_rate;
p_block->i_dts = p_block->i_pts = packet.pts; p_block->i_dts = p_block->i_pts = packet.pts;
block_ChainAppend( &p_chain, p_block ); block_ChainAppend( &p_chain, p_block );
}
} }
if( unlikely( !p_aout_buf ) ) if( unlikely( !p_aout_buf ) )
{ {
msg_Dbg(p_enc,"Flushing.."); msg_Dbg(p_enc,"Flushing..");
do { do {
AVPacket packet; AVPacket packet = {0};
p_block = block_Alloc( p_sys->i_buffer_out ); p_block = block_Alloc( p_sys->i_buffer_out );
av_init_packet( &packet ); av_init_packet( &packet );
packet.data = p_block->p_buffer; packet.data = p_block->p_buffer;
...@@ -1183,9 +1199,9 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf ) ...@@ -1183,9 +1199,9 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
p_block->i_dts = p_block->i_pts = packet.pts; p_block->i_dts = p_block->i_pts = packet.pts;
if( !i_out && got_packet ) if( i_out >= 0 && got_packet )
block_ChainAppend( &p_chain, p_block ); block_ChainAppend( &p_chain, p_block );
} while( got_packet && !i_out ); } while( got_packet && (i_out>=0) );
return p_chain; return p_chain;
} }
...@@ -1193,32 +1209,34 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf ) ...@@ -1193,32 +1209,34 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
{ {
AVPacket packet = {0}; AVPacket packet = {0};
if( unlikely( p_aout_buf->i_pts > VLC_TS_INVALID &&
p_aout_buf->i_pts != date_Get( &p_sys->buffer_date ) ) )
date_Set( &p_sys->buffer_date, p_aout_buf->i_pts );
avcodec_get_frame_defaults( p_sys->frame );
p_sys->frame->nb_samples = p_sys->i_frame_size; p_sys->frame->nb_samples = p_sys->i_frame_size;
p_sys->frame->format = p_sys->p_context->sample_fmt; p_sys->frame->format = p_sys->p_context->sample_fmt;
p_sys->frame->pts = date_Get( &p_sys->buffer_date );
if( av_sample_fmt_is_planar( p_sys->p_context->sample_fmt ) ) if( p_sys->b_planar )
{ {
aout_Deinterleave( p_sys->p_buffer, p_aout_buf->p_buffer+i_data_offset, aout_Deinterleave( p_sys->p_buffer, p_aout_buf->p_buffer,
p_sys->i_frame_size, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec ); p_sys->frame->nb_samples, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec );
if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
p_sys->p_context->sample_fmt, }
p_sys->p_buffer,
p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels, if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
0) < 0 )
msg_Err( p_enc, "filling error on encode" );
} else {
if( avcodec_fill_audio_frame( p_sys->frame, p_enc->fmt_in.audio.i_channels,
p_sys->p_context->sample_fmt, p_sys->p_context->sample_fmt,
p_aout_buf->p_buffer+i_data_offset, p_sys->b_planar ? p_sys->p_buffer : p_aout_buf->p_buffer,
p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels, __MIN(p_sys->i_buffer_out, p_aout_buf->i_buffer),
0) < 0 ) 0) < 0 )
msg_Err( p_enc, "filling error on encode" ); msg_Err( p_enc, "filling error on encode" );
}
i_samples_left -= p_sys->i_frame_size; p_aout_buf->p_buffer += (p_sys->frame->nb_samples * p_enc->fmt_in.audio.i_channels * p_sys->i_sample_bytes);
i_data_offset += p_sys->i_frame_size * p_sys->i_sample_bytes * p_enc->fmt_in.audio.i_channels; 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->i_nb_samples -= p_sys->frame->nb_samples;
date_Increment( &p_sys->buffer_date, p_sys->frame->nb_samples );
p_sys->frame->pts = p_aout_buf->i_pts;
p_block = block_Alloc( p_sys->i_buffer_out ); p_block = block_Alloc( p_sys->i_buffer_out );
av_init_packet( &packet ); av_init_packet( &packet );
...@@ -1242,25 +1260,25 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf ) ...@@ -1242,25 +1260,25 @@ static block_t *EncodeAudio( encoder_t *p_enc, block_t *p_aout_buf )
p_block->i_buffer = packet.size; p_block->i_buffer = packet.size;
p_block->i_length = (mtime_t)1000000 * p_block->i_length = (mtime_t)1000000 *
(mtime_t)p_sys->i_frame_size / (mtime_t)p_sys->frame->nb_samples /
(mtime_t)p_sys->p_context->sample_rate; (mtime_t)p_sys->p_context->sample_rate;
p_block->i_dts = p_block->i_pts = packet.pts; p_block->i_dts = p_block->i_pts = packet.pts;
block_ChainAppend( &p_chain, p_block ); block_ChainAppend( &p_chain, p_block );
} }
if( i_samples_left < 0 )
msg_Err( p_enc, "I_data_left overflow");
// We have leftover samples that don't fill frame_size, and libavcodec doesn't seem to like // We have leftover samples that don't fill frame_size, and libavcodec doesn't seem to like
// that frame has more data than p_sys->i_frame_size most of the cases currently. // that frame has more data than p_sys->i_frame_size most of the cases currently.
if( i_samples_left > 0 ) if( p_aout_buf->i_nb_samples > 0 )
{ {
if( av_sample_fmt_is_planar( p_sys->p_context->sample_fmt ) ) if( p_sys->b_planar )
aout_Deinterleave( p_sys->p_buffer, p_aout_buf->p_buffer+i_data_offset, i_samples_left, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec ); aout_Deinterleave( p_sys->p_buffer+buffer_delay, p_aout_buf->p_buffer,
else p_aout_buf->i_nb_samples, p_enc->fmt_in.audio.i_channels, p_enc->fmt_in.i_codec );
memcpy( p_sys->p_buffer, p_aout_buf->p_buffer+i_data_offset , i_samples_left*p_sys->i_sample_bytes*p_enc->fmt_in.audio.i_channels); else
p_sys->i_samples_delay = i_samples_left; 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_sys->i_samples_delay += p_aout_buf->i_nb_samples;
} }
return p_chain; return p_chain;
......
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