#include "hb.h"
#include "hbffmpeg.h"
-
-//#include "libavcodec/audioconvert.h"
-#include "../contrib/ffmpeg/libavcodec/audioconvert.h"
+#include "libavcodec/audioconvert.h"
static int decavcodecInit( hb_work_object_t *, hb_job_t * );
static int decavcodecWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
}
if ( pv->buffer )
{
- free( pv->buffer );
+ av_free( pv->buffer );
pv->buffer = NULL;
}
free( pv );
hb_work_private_t * pv = w->private_data;
hb_buffer_t * in = *buf_in, * buf, * last = NULL;
int pos, len, out_size, i, uncompressed_len;
- short buffer[AVCODEC_MAX_AUDIO_FRAME_SIZE];
+ short* bufaligned;
uint64_t cur;
unsigned char *parser_output_buffer;
int parser_output_buffer_len;
cur = ( in->start < 0 )? pv->pts_next : in->start;
+ bufaligned = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
pos = 0;
while( pos < in->size )
{
- len = av_parser_parse( pv->parser, pv->context,
- &parser_output_buffer, &parser_output_buffer_len,
- in->data + pos, in->size - pos, cur, cur );
+ len = av_parser_parse2( pv->parser, pv->context,
+ &parser_output_buffer, &parser_output_buffer_len,
+ in->data + pos, in->size - pos, cur, cur, AV_NOPTS_VALUE );
out_size = 0;
uncompressed_len = 0;
if (parser_output_buffer_len)
{
- out_size = sizeof(buffer);
- uncompressed_len = avcodec_decode_audio2( pv->context, buffer,
- &out_size,
- parser_output_buffer,
- parser_output_buffer_len );
+ AVPacket avp;
+ av_init_packet( &avp );
+ avp.data = parser_output_buffer;
+ avp.size = parser_output_buffer_len;
+
+ out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ uncompressed_len = avcodec_decode_audio3( pv->context, bufaligned, &out_size, &avp );
}
if( out_size )
{
pv->context->sample_rate;
cur = buf->stop;
- s16 = buffer;
+ s16 = bufaligned;
fl32 = (float *) buf->data;
for( i = 0; i < out_size / 2; i++ )
{
pv->pts_next = cur;
+ av_free( bufaligned );
return HB_WORK_OK;
}
AVCodecParserContext *parser = av_parser_init( codec->id );
AVCodecContext *context = avcodec_alloc_context();
hb_avcodec_open( context, codec );
-#ifdef SYS_CYGWIN
- uint8_t *buffer = memalign(16, AVCODEC_MAX_AUDIO_FRAME_SIZE);
-#else
- uint8_t *buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
-#endif
+ uint8_t *buffer = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
unsigned char *pbuffer;
int pos = 0, pbuffer_size;
while ( pos < buf->size )
{
- int len = av_parser_parse( parser, context, &pbuffer, &pbuffer_size,
- buf->data + pos, buf->size - pos,
- buf->start, buf->start );
+ int len = av_parser_parse2( parser, context, &pbuffer, &pbuffer_size,
+ buf->data + pos, buf->size - pos,
+ buf->start, buf->start, AV_NOPTS_VALUE );
pos += len;
if ( pbuffer_size > 0 )
{
- len = avcodec_decode_audio2( context, (int16_t*)buffer, &out_size,
- pbuffer, pbuffer_size );
+ AVPacket avp;
+ av_init_packet( &avp );
+ avp.data = pbuffer;
+ avp.size = pbuffer_size;
+
+ len = avcodec_decode_audio3( context, (int16_t*)buffer, &out_size, &avp );
if ( len > 0 && context->sample_rate > 0 )
{
info->bitrate = context->bit_rate;
}
}
}
- free( buffer );
+ av_free( buffer );
av_parser_close( parser );
hb_avcodec_close( context );
return ret;
hb_chapter_t *c = hb_list_item( pv->job->title->list_chapter, chap_num - 1 );
if ( c && c->title )
{
- hb_log( "%s: \"%s\" (%d) at frame %u time %lld",
+ hb_log( "%s: \"%s\" (%d) at frame %u time %"PRId64,
pv->context->codec->name, c->title, chap_num, pv->nframes, pts );
}
else
{
- hb_log( "%s: Chapter %d at frame %u time %lld",
+ hb_log( "%s: Chapter %d at frame %u time %"PRId64,
pv->context->codec->name, chap_num, pv->nframes, pts );
}
}
static int decodeFrame( hb_work_private_t *pv, uint8_t *data, int size )
{
- int got_picture;
+ int got_picture, oldlevel = 0;
AVFrame frame;
+ AVPacket avp;
- if ( avcodec_decode_video( pv->context, &frame, &got_picture, data, size ) < 0 )
+ if ( global_verbosity_level <= 1 )
+ {
+ oldlevel = av_log_get_level();
+ av_log_set_level( AV_LOG_QUIET );
+ }
+
+ av_init_packet( &avp );
+ avp.data = data;
+ avp.size = size;
+ if ( avcodec_decode_video2( pv->context, &frame, &got_picture, &avp ) < 0 )
{
++pv->decode_errors;
}
+ if ( global_verbosity_level <= 1 )
+ {
+ av_log_set_level( oldlevel );
+ }
if( got_picture )
{
// ffmpeg makes it hard to attach a pts to a frame. if the MPEG ES
{
frame_dur += frame.repeat_pict * frame_dur * 0.5;
}
+ // XXX Unlike every other video decoder, the Raw decoder doesn't
+ // use the standard buffer allocation routines so we never
+ // get to put a PTS in the frame. Do it now.
+ if ( pv->context->codec_id == CODEC_ID_RAWVIDEO )
+ {
+ frame.pts = pv->pts;
+ pv->pts = -1;
+ }
// If there was no pts for this frame, assume constant frame rate
// video & estimate the next frame time from the last & duration.
double pts = frame.pts;
do {
uint8_t *pout;
int pout_len;
- int len = av_parser_parse( pv->parser, pv->context, &pout, &pout_len,
- data + pos, size - pos, pts, dts );
+ int len = av_parser_parse2( pv->parser, pv->context, &pout, &pout_len,
+ data + pos, size - pos, pts, dts, AV_NOPTS_VALUE );
pos += len;
if ( pout_len > 0 )
info->rate = 27000000;
info->rate_base = (int64_t)context->time_base.num * 27000000LL /
context->time_base.den;
+ if ( context->ticks_per_frame > 1 )
+ {
+ // for ffmpeg 0.5 & later, the H.264 & MPEG-2 time base is
+ // field rate rather than frame rate so convert back to frames.
+ info->rate_base *= context->ticks_per_frame;
+ }
- /* Sometimes there's no pixel aspect set in the source. In that case,
- assume a 1:1 PAR. Otherwise, preserve the source PAR. */
- info->pixel_aspect_width = context->sample_aspect_ratio.num ?
- context->sample_aspect_ratio.num : 1;
- info->pixel_aspect_height = context->sample_aspect_ratio.den ?
- context->sample_aspect_ratio.den : 1;
-
+ info->pixel_aspect_width = context->sample_aspect_ratio.num;
+ info->pixel_aspect_height = context->sample_aspect_ratio.den;
+
+ /* Sometimes there's no pixel aspect set in the source ffmpeg context
+ * which appears to come from the video stream. In that case,
+ * try the pixel aspect in AVStream (which appears to come from
+ * the container). Else assume a 1:1 PAR. */
+ if ( info->pixel_aspect_width == 0 ||
+ info->pixel_aspect_height == 0 )
+ {
+ AVStream *st = hb_ffmpeg_avstream( w->codec_param );
+ info->pixel_aspect_width = st->sample_aspect_ratio.num ?
+ st->sample_aspect_ratio.num : 1;
+ info->pixel_aspect_height = st->sample_aspect_ratio.den ?
+ st->sample_aspect_ratio.den : 1;
+ }
/* ffmpeg returns the Pixel Aspect Ratio (PAR). Handbrake wants the
* Display Aspect Ratio so we convert by scaling by the Storage
* Aspect Ratio (w/h). We do the calc in floating point to get the
int16_t *buffer = pv->buffer;
if ( buffer == NULL )
{
- // XXX ffmpeg bug workaround
- // malloc a buffer for the audio decode. On an x86, ffmpeg
- // uses mmx/sse instructions on this buffer without checking
- // that it's 16 byte aligned and this will cause an abort if
- // the buffer is allocated on our stack. Rather than doing
- // complicated, machine dependent alignment here we use the
- // fact that malloc returns an aligned pointer on most architectures.
-
- #ifdef SYS_CYGWIN
- // Cygwin's malloc doesn't appear to return 16-byte aligned memory so use memalign instead.
- pv->buffer = memalign(16, AVCODEC_MAX_AUDIO_FRAME_SIZE);
- #else
- pv->buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
- #endif
-
+ pv->buffer = av_malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
buffer = pv->buffer;
}
+
+ AVPacket avp;
+ av_init_packet( &avp );
+ avp.data = data + pos;
+ avp.size = size - pos;
+
int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
- int len = avcodec_decode_audio2( context, buffer, &out_size,
- data + pos, size - pos );
+ int len = avcodec_decode_audio3( context, buffer, &out_size, &avp );
if ( len <= 0 )
{
return;
NULL, 0 );
// get output buffer size (in 2-byte samples) then malloc a buffer
out_size = ( out_size * 2 ) / isamp;
- buffer = malloc( out_size );
+ buffer = av_malloc( out_size );
// we're doing straight sample format conversion which behaves as if
// there were only one channel.
// if we allocated a buffer for sample format conversion, free it
if ( buffer != pv->buffer )
{
- free( buffer );
+ av_free( buffer );
}
}
}
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