X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fdecavcodec.c;h=8b09afcf77dac79aaf135d80368e1b44661037b2;hb=c593146bf3fab6290c71cbbb974e0a756e43f5e0;hp=4d74fc68f5269d7fdeaa99c51dee0ccd2b976c13;hpb=68e3a3c68e1af09f9547b2deca6bc4716e7ee10d;p=handbrake-jp%2Fhandbrake-jp-git.git
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c
index 4d74fc68..8b09afcf 100644
--- a/libhb/decavcodec.c
+++ b/libhb/decavcodec.c
@@ -1,16 +1,73 @@
/* $Id: decavcodec.c,v 1.6 2005/03/06 04:08:54 titer Exp $
This file is part of the HandBrake source code.
- Homepage: .
+ Homepage: .
It may be used under the terms of the GNU General Public License. */
-#include "hb.h"
+/* This module is Handbrake's interface to the ffmpeg decoder library
+ (libavcodec & small parts of libavformat). It contains four Handbrake
+ "work objects":
+
+ decavcodec connects HB to an ffmpeg audio decoder
+ decavcodecv connects HB to an ffmpeg video decoder
+
+ (Two different routines are needed because the ffmpeg library
+ has different decoder calling conventions for audio & video.
+ The audio decoder should have had its name changed to "decavcodeca"
+ but I got lazy.) These work objects are self-contained & follow all
+ of HB's conventions for a decoder module. They can be used like
+ any other HB decoder (deca52, decmpeg2, etc.).
+
+ decavcodecai "internal" (incestuous?) version of decavcodec
+ decavcodecvi "internal" (incestuous?) version of decavcodecv
+
+ These routine are functionally equivalent to the routines above but
+ can only be used by the ffmpeg-based stream reader in libhb/stream.c.
+ The reason they exist is because the ffmpeg library leaves some of
+ the information needed by the decoder in the AVStream (the data
+ structure used by the stream reader) and we need to retrieve it
+ to successfully decode frames. But in HB the reader and decoder
+ modules are in completely separate threads and nothing goes between
+ them but hb_buffers containing frames to be decoded. I.e., there's
+ no easy way for the ffmpeg stream reader to pass a pointer to its
+ AVStream over to the ffmpeg video or audio decoder. So the *i work
+ objects use a private back door to the stream reader to get access
+ to the AVStream (routines hb_ffmpeg_avstream and hb_ffmpeg_context)
+ and the codec_param passed to these work objects is the key to this
+ back door (it's basically an index that allows the correct AVStream
+ to be retrieved).
+
+ The normal & *i objects share a lot of code (the basic frame decoding
+ and bitstream info code is factored out into subroutines that can be
+ called by either) but the top level routines of the *i objects
+ (decavcodecviWork, decavcodecviInfo, etc.) are different because:
+ 1) they *have* to use the AVCodecContext that's contained in the
+ reader's AVStream rather than just allocating & using their own,
+ 2) the Info routines have access to stuff kept in the AVStream in addition
+ to stuff kept in the AVCodecContext. This shouldn't be necessary but
+ crucial information like video frame rate that should be in the
+ AVCodecContext is either missing or wrong in the version of ffmpeg
+ we're currently using.
-#include "ffmpeg/avcodec.h"
+ A consequence of the above is that the non-i work objects *can't* use
+ information from the AVStream because there isn't one - they get their
+ data from either the dvd reader or the mpeg reader, not the ffmpeg stream
+ reader. That means that they have to make up for deficiencies in the
+ AVCodecContext info by using stuff kept in the HB "title" struct. It
+ also means that ffmpeg codecs that randomly scatter state needed by
+ the decoder across both the AVCodecContext & the AVStream (e.g., the
+ VC1 decoder) can't easily be used by the HB mpeg stream reader.
+ */
-int decavcodecInit( hb_work_object_t *, hb_job_t * );
-int decavcodecWork( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
-void decavcodecClose( hb_work_object_t * );
+#include "hb.h"
+#include "hbffmpeg.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 ** );
+static void decavcodecClose( hb_work_object_t * );
+static int decavcodecInfo( hb_work_object_t *, hb_work_info_t * );
+static int decavcodecBSInfo( hb_work_object_t *, const hb_buffer_t *, hb_work_info_t * );
hb_work_object_t hb_decavcodec =
{
@@ -18,35 +75,125 @@ hb_work_object_t hb_decavcodec =
"MPGA decoder (libavcodec)",
decavcodecInit,
decavcodecWork,
- decavcodecClose
+ decavcodecClose,
+ decavcodecInfo,
+ decavcodecBSInfo
};
+#define HEAP_SIZE 8
+typedef struct {
+ // there are nheap items on the heap indexed 1..nheap (i.e., top of
+ // heap is 1). The 0th slot is unused - a marker is put there to check
+ // for overwrite errs.
+ int64_t h[HEAP_SIZE+1];
+ int nheap;
+} pts_heap_t;
+
struct hb_work_private_s
{
- hb_job_t * job;
-
- AVCodecContext * context;
- int64_t pts_last;
+ hb_job_t *job;
+ AVCodecContext *context;
+ AVCodecParserContext *parser;
+ hb_list_t *list;
+ double duration; // frame duration (for video)
+ double pts_next; // next pts we expect to generate
+ int64_t pts; // (video) pts passing from parser to decoder
+ int64_t chap_time; // time of next chap mark (if new_chap != 0)
+ int new_chap; // output chapter mark pending
+ uint32_t nframes;
+ uint32_t ndrops;
+ uint32_t decode_errors;
+ int brokenByMicrosoft; // video stream may contain packed b-frames
+ hb_buffer_t* delayq[HEAP_SIZE];
+ pts_heap_t pts_heap;
+ void* buffer;
+ struct SwsContext *sws_context; // if we have to rescale or convert color space
};
+static int64_t heap_pop( pts_heap_t *heap )
+{
+ int64_t result;
+
+ if ( heap->nheap <= 0 )
+ {
+ return -1;
+ }
+
+ // return the top of the heap then put the bottom element on top,
+ // decrease the heap size by one & rebalence the heap.
+ result = heap->h[1];
+
+ int64_t v = heap->h[heap->nheap--];
+ int parent = 1;
+ int child = parent << 1;
+ while ( child <= heap->nheap )
+ {
+ // find the smallest of the two children of parent
+ if (child < heap->nheap && heap->h[child] > heap->h[child+1] )
+ ++child;
+
+ if (v <= heap->h[child])
+ // new item is smaller than either child so it's the new parent.
+ break;
+
+ // smallest child is smaller than new item so move it up then
+ // check its children.
+ int64_t hp = heap->h[child];
+ heap->h[parent] = hp;
+ parent = child;
+ child = parent << 1;
+ }
+ heap->h[parent] = v;
+ return result;
+}
+
+static void heap_push( pts_heap_t *heap, int64_t v )
+{
+ if ( heap->nheap < HEAP_SIZE )
+ {
+ ++heap->nheap;
+ }
+
+ // stick the new value on the bottom of the heap then bubble it
+ // up to its correct spot.
+ int child = heap->nheap;
+ while (child > 1) {
+ int parent = child >> 1;
+ if (heap->h[parent] <= v)
+ break;
+ // move parent down
+ int64_t hp = heap->h[parent];
+ heap->h[child] = hp;
+ child = parent;
+ }
+ heap->h[child] = v;
+}
+
/***********************************************************************
* hb_work_decavcodec_init
***********************************************************************
*
**********************************************************************/
-int decavcodecInit( hb_work_object_t * w, hb_job_t * job )
+static int decavcodecInit( hb_work_object_t * w, hb_job_t * job )
{
AVCodec * codec;
+
hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
w->private_data = pv;
pv->job = job;
- codec = avcodec_find_decoder( CODEC_ID_MP2 );
+ int codec_id = w->codec_param;
+ /*XXX*/
+ if ( codec_id == 0 )
+ codec_id = CODEC_ID_MP2;
+
+ codec = avcodec_find_decoder( codec_id );
+ pv->parser = av_parser_init( codec_id );
+
pv->context = avcodec_alloc_context();
- avcodec_open( pv->context, codec );
- pv->pts_last = -1;
+ hb_avcodec_open( pv->context, codec );
return 0;
}
@@ -56,10 +203,42 @@ int decavcodecInit( hb_work_object_t * w, hb_job_t * job )
***********************************************************************
*
**********************************************************************/
-void decavcodecClose( hb_work_object_t * w )
+static void decavcodecClose( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
- avcodec_close( pv->context );
+
+ if ( pv )
+ {
+ if ( pv->job && pv->context && pv->context->codec )
+ {
+ hb_log( "%s-decoder done: %u frames, %u decoder errors, %u drops",
+ pv->context->codec->name, pv->nframes, pv->decode_errors,
+ pv->ndrops );
+ }
+ if ( pv->sws_context )
+ {
+ sws_freeContext( pv->sws_context );
+ }
+ if ( pv->parser )
+ {
+ av_parser_close(pv->parser);
+ }
+ if ( pv->context && pv->context->codec )
+ {
+ hb_avcodec_close( pv->context );
+ }
+ if ( pv->list )
+ {
+ hb_list_close( &pv->list );
+ }
+ if ( pv->buffer )
+ {
+ free( pv->buffer );
+ pv->buffer = NULL;
+ }
+ free( pv );
+ w->private_data = NULL;
+ }
}
/***********************************************************************
@@ -67,34 +246,51 @@ void decavcodecClose( hb_work_object_t * w )
***********************************************************************
*
**********************************************************************/
-int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
+static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * in = *buf_in, * buf, * last = NULL;
- int pos, len, out_size, i;
+ int pos, len, out_size, i, uncompressed_len;
short buffer[AVCODEC_MAX_AUDIO_FRAME_SIZE];
uint64_t cur;
+ unsigned char *parser_output_buffer;
+ int parser_output_buffer_len;
- *buf_out = NULL;
-
- if( in->start < 0 ||
- ( pv->pts_last > 0 &&
- in->start > pv->pts_last &&
- in->start - pv->pts_last < 5000 ) ) /* Hacky */
+ if ( (*buf_in)->size <= 0 )
{
- cur = pv->pts_last;
+ /* EOF on input stream - send it downstream & say that we're done */
+ *buf_out = *buf_in;
+ *buf_in = NULL;
+ return HB_WORK_DONE;
}
- else
+
+ *buf_out = NULL;
+
+ if ( in->start < -1 && pv->pts_next <= 0 )
{
- cur = in->start;
+ // discard buffers that start before video time 0
+ return HB_WORK_OK;
}
+ cur = ( in->start < 0 )? pv->pts_next : in->start;
+
pos = 0;
while( pos < in->size )
{
- len = avcodec_decode_audio( pv->context, buffer, &out_size,
- in->data + pos, in->size - pos );
+ len = av_parser_parse( pv->parser, pv->context,
+ &parser_output_buffer, &parser_output_buffer_len,
+ in->data + pos, in->size - pos, cur, cur );
+ 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 );
+ }
if( out_size )
{
short * s16;
@@ -102,8 +298,21 @@ int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
buf = hb_buffer_init( 2 * out_size );
+ int sample_size_in_bytes = 2; // Default to 2 bytes
+ switch (pv->context->sample_fmt)
+ {
+ case SAMPLE_FMT_S16:
+ sample_size_in_bytes = 2;
+ break;
+ /* We should handle other formats here - but that needs additional format conversion work below */
+ /* For now we'll just report the error and try to carry on */
+ default:
+ hb_log("decavcodecWork - Unknown Sample Format from avcodec_decode_audio (%d) !", pv->context->sample_fmt);
+ break;
+ }
+
buf->start = cur;
- buf->stop = cur + 90000 * ( out_size / 4 ) /
+ buf->stop = cur + 90000 * ( out_size / (sample_size_in_bytes * pv->context->channels) ) /
pv->context->sample_rate;
cur = buf->stop;
@@ -127,8 +336,943 @@ int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
pos += len;
}
- pv->pts_last = cur;
+ pv->pts_next = cur;
return HB_WORK_OK;
}
+static int decavcodecInfo( hb_work_object_t *w, hb_work_info_t *info )
+{
+ hb_work_private_t *pv = w->private_data;
+
+ memset( info, 0, sizeof(*info) );
+
+ if ( pv && pv->context )
+ {
+ AVCodecContext *context = pv->context;
+ info->bitrate = context->bit_rate;
+ info->rate = context->time_base.num;
+ info->rate_base = context->time_base.den;
+ info->profile = context->profile;
+ info->level = context->level;
+ return 1;
+ }
+ return 0;
+}
+
+static const int chan2layout[] = {
+ HB_INPUT_CH_LAYOUT_MONO, // We should allow no audio really.
+ HB_INPUT_CH_LAYOUT_MONO,
+ HB_INPUT_CH_LAYOUT_STEREO,
+ HB_INPUT_CH_LAYOUT_2F1R,
+ HB_INPUT_CH_LAYOUT_2F2R,
+ HB_INPUT_CH_LAYOUT_3F2R,
+ HB_INPUT_CH_LAYOUT_4F2R,
+ HB_INPUT_CH_LAYOUT_STEREO,
+ HB_INPUT_CH_LAYOUT_STEREO,
+};
+
+static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf,
+ hb_work_info_t *info )
+{
+ hb_work_private_t *pv = w->private_data;
+ int ret = 0;
+
+ memset( info, 0, sizeof(*info) );
+
+ if ( pv && pv->context )
+ {
+ return decavcodecInfo( w, info );
+ }
+ // XXX
+ // We should parse the bitstream to find its parameters but for right
+ // now we just return dummy values if there's a codec that will handle it.
+ AVCodec *codec = avcodec_find_decoder( w->codec_param? w->codec_param :
+ CODEC_ID_MP2 );
+ if ( ! codec )
+ {
+ // there's no ffmpeg codec for this audio type - give up
+ return -1;
+ }
+
+ static char codec_name[64];
+ info->name = strncpy( codec_name, codec->name, sizeof(codec_name)-1 );
+
+ AVCodecParserContext *parser = av_parser_init( codec->id );
+ AVCodecContext *context = avcodec_alloc_context();
+ hb_avcodec_open( context, codec );
+#if defined( SYS_CYGWIN )
+ uint8_t *buffer = memalign(16, AVCODEC_MAX_AUDIO_FRAME_SIZE);
+#else
+ uint8_t *buffer = malloc( AVCODEC_MAX_AUDIO_FRAME_SIZE );
+#endif
+ 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 );
+ pos += len;
+ if ( pbuffer_size > 0 )
+ {
+ len = avcodec_decode_audio2( context, (int16_t*)buffer, &out_size,
+ pbuffer, pbuffer_size );
+ if ( len > 0 && context->sample_rate > 0 )
+ {
+ info->bitrate = context->bit_rate;
+ info->rate = context->sample_rate;
+ info->rate_base = 1;
+ info->channel_layout = chan2layout[context->channels & 7];
+ ret = 1;
+ break;
+ }
+ }
+ }
+ free( buffer );
+ av_parser_close( parser );
+ hb_avcodec_close( context );
+ return ret;
+}
+
+/* -------------------------------------------------------------
+ * General purpose video decoder using libavcodec
+ */
+
+static uint8_t *copy_plane( uint8_t *dst, uint8_t* src, int dstride, int sstride,
+ int h )
+{
+ if ( dstride == sstride )
+ {
+ memcpy( dst, src, dstride * h );
+ return dst + dstride * h;
+ }
+ int lbytes = dstride <= sstride? dstride : sstride;
+ while ( --h >= 0 )
+ {
+ memcpy( dst, src, lbytes );
+ src += sstride;
+ dst += dstride;
+ }
+ return dst;
+}
+
+// copy one video frame into an HB buf. If the frame isn't in our color space
+// or at least one of its dimensions is odd, use sws_scale to convert/rescale it.
+// Otherwise just copy the bits.
+static hb_buffer_t *copy_frame( hb_work_private_t *pv, AVFrame *frame )
+{
+ AVCodecContext *context = pv->context;
+ int w, h;
+ if ( ! pv->job )
+ {
+ // if the dimensions are odd, drop the lsb since h264 requires that
+ // both width and height be even.
+ w = ( context->width >> 1 ) << 1;
+ h = ( context->height >> 1 ) << 1;
+ }
+ else
+ {
+ w = pv->job->title->width;
+ h = pv->job->title->height;
+ }
+ hb_buffer_t *buf = hb_video_buffer_init( w, h );
+ uint8_t *dst = buf->data;
+
+ if ( context->pix_fmt != PIX_FMT_YUV420P || w != context->width ||
+ h != context->height )
+ {
+ // have to convert to our internal color space and/or rescale
+ AVPicture dstpic;
+ avpicture_fill( &dstpic, dst, PIX_FMT_YUV420P, w, h );
+
+ if ( ! pv->sws_context )
+ {
+ pv->sws_context = sws_getContext( context->width, context->height, context->pix_fmt,
+ w, h, PIX_FMT_YUV420P,
+ SWS_LANCZOS|SWS_ACCURATE_RND,
+ NULL, NULL, NULL );
+ }
+ sws_scale( pv->sws_context, frame->data, frame->linesize, 0, h,
+ dstpic.data, dstpic.linesize );
+ }
+ else
+ {
+ dst = copy_plane( dst, frame->data[0], w, frame->linesize[0], h );
+ w = (w + 1) >> 1; h = (h + 1) >> 1;
+ dst = copy_plane( dst, frame->data[1], w, frame->linesize[1], h );
+ dst = copy_plane( dst, frame->data[2], w, frame->linesize[2], h );
+ }
+ return buf;
+}
+
+static int get_frame_buf( AVCodecContext *context, AVFrame *frame )
+{
+ hb_work_private_t *pv = context->opaque;
+ frame->pts = pv->pts;
+ pv->pts = -1;
+ return avcodec_default_get_buffer( context, frame );
+}
+
+static void log_chapter( hb_work_private_t *pv, int chap_num, int64_t pts )
+{
+ 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",
+ pv->context->codec->name, c->title, chap_num, pv->nframes, pts );
+ }
+ else
+ {
+ hb_log( "%s: Chapter %d at frame %u time %lld",
+ pv->context->codec->name, chap_num, pv->nframes, pts );
+ }
+}
+
+static void flushDelayQueue( hb_work_private_t *pv )
+{
+ hb_buffer_t *buf;
+ int slot = pv->nframes & (HEAP_SIZE-1);
+
+ // flush all the video packets left on our timestamp-reordering delay q
+ while ( ( buf = pv->delayq[slot] ) != NULL )
+ {
+ buf->start = heap_pop( &pv->pts_heap );
+ hb_list_add( pv->list, buf );
+ pv->delayq[slot] = NULL;
+ slot = ( slot + 1 ) & (HEAP_SIZE-1);
+ }
+}
+
+static int decodeFrame( hb_work_private_t *pv, uint8_t *data, int size )
+{
+ int got_picture, oldlevel = 0;
+ AVFrame frame;
+
+ if ( global_verbosity_level <= 1 )
+ {
+ oldlevel = av_log_get_level();
+ av_log_set_level( AV_LOG_QUIET );
+ }
+ if ( avcodec_decode_video( pv->context, &frame, &got_picture, data, size ) < 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
+ // packet had a pts we handed it to av_parser_parse (if the packet had
+ // no pts we set it to -1 but before the parse we can't distinguish between
+ // the start of a video frame with no pts & an intermediate packet of
+ // some frame which never has a pts). we hope that when parse returns
+ // the frame to us the pts we originally handed it will be in parser->pts.
+ // we put this pts into pv->pts so that when a avcodec_decode_video
+ // finally gets around to allocating an AVFrame to hold the decoded
+ // frame we can stuff that pts into the frame. if all of these relays
+ // worked at this point frame.pts should hold the frame's pts from the
+ // original data stream or -1 if it didn't have one. in the latter case
+ // we generate the next pts in sequence for it.
+ double frame_dur = pv->duration;
+ if ( frame_dur <= 0 )
+ {
+ frame_dur = 90000. * (double)pv->context->time_base.num /
+ (double)pv->context->time_base.den;
+ pv->duration = frame_dur;
+ }
+ if ( frame.repeat_pict )
+ {
+ frame_dur += frame.repeat_pict * frame_dur * 0.5;
+ }
+ // 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;
+ if ( pts < 0 )
+ {
+ pts = pv->pts_next;
+ }
+ pv->pts_next = pts + frame_dur;
+
+ hb_buffer_t *buf;
+
+ // if we're doing a scan or this content couldn't have been broken
+ // by Microsoft we don't worry about timestamp reordering
+ if ( ! pv->job || ! pv->brokenByMicrosoft )
+ {
+ buf = copy_frame( pv, &frame );
+ buf->start = pts;
+ hb_list_add( pv->list, buf );
+ ++pv->nframes;
+ return got_picture;
+ }
+
+ // XXX This following probably addresses a libavcodec bug but I don't
+ // see an easy fix so we workaround it here.
+ //
+ // The M$ 'packed B-frames' atrocity results in decoded frames with
+ // the wrong timestamp. E.g., if there are 2 b-frames the timestamps
+ // we see here will be "2 3 1 5 6 4 ..." instead of "1 2 3 4 5 6".
+ // The frames are actually delivered in the right order but with
+ // the wrong timestamp. To get the correct timestamp attached to
+ // each frame we have a delay queue (longer than the max number of
+ // b-frames) & a sorting heap for the timestamps. As each frame
+ // comes out of the decoder the oldest frame in the queue is removed
+ // and associated with the smallest timestamp. Then the new frame is
+ // added to the queue & its timestamp is pushed on the heap.
+ // This does nothing if the timestamps are correct (i.e., the video
+ // uses a codec that Micro$oft hasn't broken yet) but the frames
+ // get timestamped correctly even when M$ has munged them.
+
+ // remove the oldest picture from the frame queue (if any) &
+ // give it the smallest timestamp from our heap. The queue size
+ // is a power of two so we get the slot of the oldest by masking
+ // the frame count & this will become the slot of the newest
+ // once we've removed & processed the oldest.
+ int slot = pv->nframes & (HEAP_SIZE-1);
+ if ( ( buf = pv->delayq[slot] ) != NULL )
+ {
+ buf->start = heap_pop( &pv->pts_heap );
+
+ if ( pv->new_chap && buf->start >= pv->chap_time )
+ {
+ buf->new_chap = pv->new_chap;
+ pv->new_chap = 0;
+ pv->chap_time = 0;
+ log_chapter( pv, buf->new_chap, buf->start );
+ }
+ else if ( pv->nframes == 0 )
+ {
+ log_chapter( pv, pv->job->chapter_start, buf->start );
+ }
+ hb_list_add( pv->list, buf );
+ }
+
+ // add the new frame to the delayq & push its timestamp on the heap
+ pv->delayq[slot] = copy_frame( pv, &frame );
+ heap_push( &pv->pts_heap, pts );
+
+ ++pv->nframes;
+ }
+
+ return got_picture;
+}
+
+static void decodeVideo( hb_work_private_t *pv, uint8_t *data, int size,
+ int64_t pts, int64_t dts )
+{
+ /*
+ * The following loop is a do..while because we need to handle both
+ * data & the flush at the end (signaled by size=0). At the end there's
+ * generally a frame in the parser & one or more frames in the decoder
+ * (depending on the bframes setting).
+ */
+ int pos = 0;
+ 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 );
+ pos += len;
+
+ if ( pout_len > 0 )
+ {
+ pv->pts = pv->parser->pts;
+ decodeFrame( pv, pout, pout_len );
+ }
+ } while ( pos < size );
+
+ /* the stuff above flushed the parser, now flush the decoder */
+ if ( size <= 0 )
+ {
+ while ( decodeFrame( pv, NULL, 0 ) )
+ {
+ }
+ flushDelayQueue( pv );
+ }
+}
+
+static hb_buffer_t *link_buf_list( hb_work_private_t *pv )
+{
+ hb_buffer_t *head = hb_list_item( pv->list, 0 );
+
+ if ( head )
+ {
+ hb_list_rem( pv->list, head );
+
+ hb_buffer_t *last = head, *buf;
+
+ while ( ( buf = hb_list_item( pv->list, 0 ) ) != NULL )
+ {
+ hb_list_rem( pv->list, buf );
+ last->next = buf;
+ last = buf;
+ }
+ }
+ return head;
+}
+
+
+static int decavcodecvInit( hb_work_object_t * w, hb_job_t * job )
+{
+
+ hb_work_private_t *pv = calloc( 1, sizeof( hb_work_private_t ) );
+ w->private_data = pv;
+ pv->job = job;
+ pv->list = hb_list_init();
+
+ int codec_id = w->codec_param;
+ pv->parser = av_parser_init( codec_id );
+ pv->context = avcodec_alloc_context2( CODEC_TYPE_VIDEO );
+
+ /* we have to wrap ffmpeg's get_buffer to be able to set the pts (?!) */
+ pv->context->opaque = pv;
+ pv->context->get_buffer = get_frame_buf;
+
+ return 0;
+}
+
+static int next_hdr( hb_buffer_t *in, int offset )
+{
+ uint8_t *dat = in->data;
+ uint16_t last2 = 0xffff;
+ for ( ; in->size - offset > 1; ++offset )
+ {
+ if ( last2 == 0 && dat[offset] == 0x01 )
+ // found an mpeg start code
+ return offset - 2;
+
+ last2 = ( last2 << 8 ) | dat[offset];
+ }
+
+ return -1;
+}
+
+static int find_hdr( hb_buffer_t *in, int offset, uint8_t hdr_type )
+{
+ if ( in->size - offset < 4 )
+ // not enough room for an mpeg start code
+ return -1;
+
+ for ( ; ( offset = next_hdr( in, offset ) ) >= 0; ++offset )
+ {
+ if ( in->data[offset+3] == hdr_type )
+ // found it
+ break;
+ }
+ return offset;
+}
+
+static int setup_extradata( hb_work_object_t *w, hb_buffer_t *in )
+{
+ hb_work_private_t *pv = w->private_data;
+
+ // we can't call the avstream funcs but the read_header func in the
+ // AVInputFormat may set up some state in the AVContext. In particular
+ // vc1t_read_header allocates 'extradata' to deal with header issues
+ // related to Microsoft's bizarre engineering notions. We alloc a chunk
+ // of space to make vc1 work then associate the codec with the context.
+ if ( w->codec_param != CODEC_ID_VC1 )
+ {
+ // we haven't been inflicted with M$ - allocate a little space as
+ // a marker and return success.
+ pv->context->extradata_size = 16;
+ pv->context->extradata = av_malloc(pv->context->extradata_size);
+ return 0;
+ }
+
+ // find the start and and of the sequence header
+ int shdr, shdr_end;
+ if ( ( shdr = find_hdr( in, 0, 0x0f ) ) < 0 )
+ {
+ // didn't find start of seq hdr
+ return 1;
+ }
+ if ( ( shdr_end = next_hdr( in, shdr + 4 ) ) < 0 )
+ {
+ shdr_end = in->size;
+ }
+ shdr_end -= shdr;
+
+ // find the start and and of the entry point header
+ int ehdr, ehdr_end;
+ if ( ( ehdr = find_hdr( in, 0, 0x0e ) ) < 0 )
+ {
+ // didn't find start of entry point hdr
+ return 1;
+ }
+ if ( ( ehdr_end = next_hdr( in, ehdr + 4 ) ) < 0 )
+ {
+ ehdr_end = in->size;
+ }
+ ehdr_end -= ehdr;
+
+ // found both headers - allocate an extradata big enough to hold both
+ // then copy them into it.
+ pv->context->extradata_size = shdr_end + ehdr_end;
+ pv->context->extradata = av_malloc(pv->context->extradata_size + 8);
+ memcpy( pv->context->extradata, in->data + shdr, shdr_end );
+ memcpy( pv->context->extradata + shdr_end, in->data + ehdr, ehdr_end );
+ memset( pv->context->extradata + shdr_end + ehdr_end, 0, 8);
+ return 0;
+}
+
+static int decavcodecvWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
+ hb_buffer_t ** buf_out )
+{
+ hb_work_private_t *pv = w->private_data;
+ hb_buffer_t *in = *buf_in;
+ int64_t pts = AV_NOPTS_VALUE;
+ int64_t dts = pts;
+
+ *buf_in = NULL;
+
+ /* if we got an empty buffer signaling end-of-stream send it downstream */
+ if ( in->size == 0 )
+ {
+ decodeVideo( pv, in->data, in->size, pts, dts );
+ hb_list_add( pv->list, in );
+ *buf_out = link_buf_list( pv );
+ return HB_WORK_DONE;
+ }
+
+ // if this is the first frame open the codec (we have to wait for the
+ // first frame because of M$ VC1 braindamage).
+ if ( pv->context->extradata_size == 0 )
+ {
+ if ( setup_extradata( w, in ) )
+ {
+ // we didn't find the headers needed to set up extradata.
+ // the codec will abort if we open it so just free the buf
+ // and hope we eventually get the info we need.
+ hb_buffer_close( &in );
+ return HB_WORK_OK;
+ }
+ AVCodec *codec = avcodec_find_decoder( w->codec_param );
+ // There's a mis-feature in ffmpeg that causes the context to be
+ // incorrectly initialized the 1st time avcodec_open is called.
+ // If you close it and open a 2nd time, it finishes the job.
+ hb_avcodec_open( pv->context, codec );
+ hb_avcodec_close( pv->context );
+ hb_avcodec_open( pv->context, codec );
+ }
+
+ if( in->start >= 0 )
+ {
+ pts = in->start;
+ dts = in->renderOffset;
+ }
+ if ( in->new_chap )
+ {
+ pv->new_chap = in->new_chap;
+ pv->chap_time = pts >= 0? pts : pv->pts_next;
+ }
+ decodeVideo( pv, in->data, in->size, pts, dts );
+ hb_buffer_close( &in );
+ *buf_out = link_buf_list( pv );
+ return HB_WORK_OK;
+}
+
+static int decavcodecvInfo( hb_work_object_t *w, hb_work_info_t *info )
+{
+ hb_work_private_t *pv = w->private_data;
+
+ memset( info, 0, sizeof(*info) );
+
+ if ( pv && pv->context )
+ {
+ AVCodecContext *context = pv->context;
+ info->bitrate = context->bit_rate;
+ info->width = context->width;
+ info->height = context->height;
+
+ /* ffmpeg gives the frame rate in frames per second while HB wants
+ * it in units of the 27MHz MPEG clock. */
+ 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;
+
+ /* 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
+ * rounding right. */
+ info->aspect = (double)info->pixel_aspect_width *
+ (double)context->width /
+ (double)info->pixel_aspect_height /
+ (double)context->height;
+
+ info->profile = context->profile;
+ info->level = context->level;
+ info->name = context->codec->name;
+ return 1;
+ }
+ return 0;
+}
+
+static int decavcodecvBSInfo( hb_work_object_t *w, const hb_buffer_t *buf,
+ hb_work_info_t *info )
+{
+ return 0;
+}
+
+hb_work_object_t hb_decavcodecv =
+{
+ WORK_DECAVCODECV,
+ "Video decoder (libavcodec)",
+ decavcodecvInit,
+ decavcodecvWork,
+ decavcodecClose,
+ decavcodecvInfo,
+ decavcodecvBSInfo
+};
+
+
+// This is a special decoder for ffmpeg streams. The ffmpeg stream reader
+// includes a parser and passes information from the parser to the decoder
+// via a codec context kept in the AVStream of the reader's AVFormatContext.
+// We *have* to use that codec context to decode the stream or we'll get
+// garbage. ffmpeg_title_scan put a cookie that can be used to get to that
+// codec context in our codec_param.
+
+// this routine gets the appropriate context pointer from the ffmpeg
+// stream reader. it can't be called until we get the first buffer because
+// we can't guarantee that reader will be called before the our init
+// routine and if our init is called first we'll get a pointer to the
+// old scan stream (which has already been closed).
+static void init_ffmpeg_context( hb_work_object_t *w )
+{
+ hb_work_private_t *pv = w->private_data;
+ pv->context = hb_ffmpeg_context( w->codec_param );
+
+ // during scan the decoder gets closed & reopened which will
+ // close the codec so reopen it if it's not there
+ if ( ! pv->context->codec )
+ {
+ AVCodec *codec = avcodec_find_decoder( pv->context->codec_id );
+ hb_avcodec_open( pv->context, codec );
+ }
+ // set up our best guess at the frame duration.
+ // the frame rate in the codec is usually bogus but it's sometimes
+ // ok in the stream.
+ AVStream *st = hb_ffmpeg_avstream( w->codec_param );
+
+ if ( st->nb_frames && st->duration )
+ {
+ // compute the average frame duration from the total number
+ // of frames & the total duration.
+ pv->duration = ( (double)st->duration * (double)st->time_base.num ) /
+ ( (double)st->nb_frames * (double)st->time_base.den );
+ }
+ else
+ {
+ // XXX We don't have a frame count or duration so try to use the
+ // far less reliable time base info in the stream.
+ // Because the time bases are so screwed up, we only take values
+ // in the range 8fps - 64fps.
+ AVRational tb;
+ if ( st->time_base.num * 64 > st->time_base.den &&
+ st->time_base.den > st->time_base.num * 8 )
+ {
+ tb = st->time_base;
+ }
+ else if ( st->r_frame_rate.den * 64 > st->r_frame_rate.num &&
+ st->r_frame_rate.num > st->r_frame_rate.den * 8 )
+ {
+ tb.num = st->r_frame_rate.den;
+ tb.den = st->r_frame_rate.num;
+ }
+ else
+ {
+ tb.num = 1001; /*XXX*/
+ tb.den = 24000; /*XXX*/
+ }
+ pv->duration = (double)tb.num / (double)tb.den;
+ }
+ pv->duration *= 90000.;
+
+ // we have to wrap ffmpeg's get_buffer to be able to set the pts (?!)
+ pv->context->opaque = pv;
+ pv->context->get_buffer = get_frame_buf;
+
+ // avi, mkv and possibly mp4 containers can contain the M$ VFW packed
+ // b-frames abortion that messes up frame ordering and timestamps.
+ // XXX ffmpeg knows which streams are broken but doesn't expose the
+ // info externally. We should patch ffmpeg to add a flag to the
+ // codec context for this but until then we mark all ffmpeg streams
+ // as suspicious.
+ pv->brokenByMicrosoft = 1;
+}
+
+static void prepare_ffmpeg_buffer( hb_buffer_t * in )
+{
+ // ffmpeg requires an extra 8 bytes of zero at the end of the buffer and
+ // will seg fault in odd, data dependent ways if it's not there. (my guess
+ // is this is a case of a local performance optimization creating a global
+ // performance degradation since all the time wasted by extraneous data
+ // copies & memory zeroing has to be huge compared to the minor reduction
+ // in inner-loop instructions this affords - modern cpus bottleneck on
+ // memory bandwidth not instruction bandwidth).
+ if ( in->size + FF_INPUT_BUFFER_PADDING_SIZE > in->alloc )
+ {
+ // have to realloc to add the padding
+ hb_buffer_realloc( in, in->size + FF_INPUT_BUFFER_PADDING_SIZE );
+ }
+ memset( in->data + in->size, 0, FF_INPUT_BUFFER_PADDING_SIZE );
+}
+
+static int decavcodecviInit( hb_work_object_t * w, hb_job_t * job )
+{
+
+ hb_work_private_t *pv = calloc( 1, sizeof( hb_work_private_t ) );
+ w->private_data = pv;
+ pv->job = job;
+ pv->list = hb_list_init();
+ pv->pts_next = -1;
+ pv->pts = -1;
+ return 0;
+}
+
+static int decavcodecviWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
+ hb_buffer_t ** buf_out )
+{
+ hb_work_private_t *pv = w->private_data;
+ if ( ! pv->context )
+ {
+ init_ffmpeg_context( w );
+ }
+ hb_buffer_t *in = *buf_in;
+ *buf_in = NULL;
+
+ /* if we got an empty buffer signaling end-of-stream send it downstream */
+ if ( in->size == 0 )
+ {
+ /* flush any frames left in the decoder */
+ while ( decodeFrame( pv, NULL, 0 ) )
+ {
+ }
+ flushDelayQueue( pv );
+ hb_list_add( pv->list, in );
+ *buf_out = link_buf_list( pv );
+ return HB_WORK_DONE;
+ }
+
+ int64_t pts = in->start;
+ if( pts >= 0 )
+ {
+ // use the first timestamp as our 'next expected' pts
+ if ( pv->pts_next < 0 )
+ {
+ pv->pts_next = pts;
+ }
+ pv->pts = pts;
+ }
+
+ if ( in->new_chap )
+ {
+ pv->new_chap = in->new_chap;
+ pv->chap_time = pts >= 0? pts : pv->pts_next;
+ }
+ prepare_ffmpeg_buffer( in );
+ decodeFrame( pv, in->data, in->size );
+ hb_buffer_close( &in );
+ *buf_out = link_buf_list( pv );
+ return HB_WORK_OK;
+}
+
+static int decavcodecviInfo( hb_work_object_t *w, hb_work_info_t *info )
+{
+ if ( decavcodecvInfo( w, info ) )
+ {
+ hb_work_private_t *pv = w->private_data;
+ if ( ! pv->context )
+ {
+ init_ffmpeg_context( w );
+ }
+ // we have the frame duration in units of the 90KHz pts clock but
+ // need it in units of the 27MHz MPEG clock. */
+ info->rate = 27000000;
+ info->rate_base = pv->duration * 300.;
+ return 1;
+ }
+ return 0;
+}
+
+static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size )
+{
+ AVCodecContext *context = pv->context;
+ int pos = 0;
+
+ while ( pos < size )
+ {
+ 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.
+
+ #if defined( 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
+
+ buffer = pv->buffer;
+ }
+ int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ int len = avcodec_decode_audio2( context, buffer, &out_size,
+ data + pos, size - pos );
+ if ( len <= 0 )
+ {
+ return;
+ }
+ pos += len;
+ if( out_size > 0 )
+ {
+ // We require signed 16-bit ints for the output format. If
+ // we got something different convert it.
+ if ( context->sample_fmt != SAMPLE_FMT_S16 )
+ {
+ // Note: av_audio_convert seems to be a work-in-progress but
+ // looks like it will eventually handle general audio
+ // mixdowns which would allow us much more flexibility
+ // in handling multichannel audio in HB. If we were doing
+ // anything more complicated than a one-for-one format
+ // conversion we'd probably want to cache the converter
+ // context in the pv.
+ int isamp = av_get_bits_per_sample_format( context->sample_fmt ) / 8;
+ AVAudioConvert *ctx = av_audio_convert_alloc( SAMPLE_FMT_S16, 1,
+ context->sample_fmt, 1,
+ NULL, 0 );
+ // get output buffer size (in 2-byte samples) then malloc a buffer
+ out_size = ( out_size * 2 ) / isamp;
+ buffer = malloc( out_size );
+
+ // we're doing straight sample format conversion which behaves as if
+ // there were only one channel.
+ const void * const ibuf[6] = { pv->buffer };
+ void * const obuf[6] = { buffer };
+ const int istride[6] = { isamp };
+ const int ostride[6] = { 2 };
+
+ av_audio_convert( ctx, obuf, ostride, ibuf, istride, out_size >> 1 );
+ av_audio_convert_free( ctx );
+ }
+ hb_buffer_t *buf = hb_buffer_init( 2 * out_size );
+
+ // convert from bytes to total samples
+ out_size >>= 1;
+
+ double pts = pv->pts_next;
+ buf->start = pts;
+ pts += out_size * pv->duration;
+ buf->stop = pts;
+ pv->pts_next = pts;
+
+ float *fl32 = (float *)buf->data;
+ int i;
+ for( i = 0; i < out_size; ++i )
+ {
+ fl32[i] = buffer[i];
+ }
+ hb_list_add( pv->list, buf );
+
+ // if we allocated a buffer for sample format conversion, free it
+ if ( buffer != pv->buffer )
+ {
+ free( buffer );
+ }
+ }
+ }
+}
+
+static int decavcodecaiWork( hb_work_object_t *w, hb_buffer_t **buf_in,
+ hb_buffer_t **buf_out )
+{
+ if ( (*buf_in)->size <= 0 )
+ {
+ /* EOF on input stream - send it downstream & say that we're done */
+ *buf_out = *buf_in;
+ *buf_in = NULL;
+ return HB_WORK_DONE;
+ }
+
+ hb_work_private_t *pv = w->private_data;
+
+ if ( (*buf_in)->start < -1 && pv->pts_next <= 0 )
+ {
+ // discard buffers that start before video time 0
+ *buf_out = NULL;
+ return HB_WORK_OK;
+ }
+
+ if ( ! pv->context )
+ {
+ init_ffmpeg_context( w );
+ // duration is a scaling factor to go from #bytes in the decoded
+ // frame to frame time (in 90KHz mpeg ticks). 'channels' converts
+ // total samples to per-channel samples. 'sample_rate' converts
+ // per-channel samples to seconds per sample and the 90000
+ // is mpeg ticks per second.
+ pv->duration = 90000. /
+ (double)( pv->context->sample_rate * pv->context->channels );
+ }
+ hb_buffer_t *in = *buf_in;
+
+ // if the packet has a timestamp use it if we don't have a timestamp yet
+ // or if there's been a timing discontinuity of more than 100ms.
+ if ( in->start >= 0 &&
+ ( pv->pts_next < 0 || ( in->start - pv->pts_next ) > 90*100 ) )
+ {
+ pv->pts_next = in->start;
+ }
+ prepare_ffmpeg_buffer( in );
+ decodeAudio( pv, in->data, in->size );
+ *buf_out = link_buf_list( pv );
+
+ return HB_WORK_OK;
+}
+
+hb_work_object_t hb_decavcodecvi =
+{
+ WORK_DECAVCODECVI,
+ "Video decoder (ffmpeg streams)",
+ decavcodecviInit,
+ decavcodecviWork,
+ decavcodecClose,
+ decavcodecviInfo,
+ decavcodecvBSInfo
+};
+
+hb_work_object_t hb_decavcodecai =
+{
+ WORK_DECAVCODECAI,
+ "Audio decoder (ffmpeg streams)",
+ decavcodecviInit,
+ decavcodecaiWork,
+ decavcodecClose,
+ decavcodecInfo,
+ decavcodecBSInfo
+};