March 2008 - FFmpeg-soc - lists.ffmpeg.org

BFI
by Sisir Koppaka 16 Apr '08

16 Apr '08

>From avio.h: offset_t *url_fseek*(ByteIOContext *s, offset_t offset, int whence); What does the url_fseek() function do exactly? And whence has SEEK_XYZ values - what do they mean? Any pointers on where this info is located in the code? ----------------- Sisir Koppaka

7 113

[soc]: r2021 - in eac3: ac3dec.h ac3dec_data.c ac3dec_data.h eac3dec.c
by bwolowiec 09 Apr '08

09 Apr '08

Author: bwolowiec Date: Mon Mar 17 23:21:11 2008 New Revision: 2021 Log: added support for dependent stream channel map Modified: eac3/ac3dec.h eac3/ac3dec_data.c eac3/ac3dec_data.h eac3/eac3dec.c Modified: eac3/ac3dec.h ============================================================================== --- eac3/ac3dec.h (original) +++ eac3/ac3dec.h Mon Mar 17 23:21:11 2008 @@ -77,6 +77,7 @@ typedef struct AC3DecodeContext { int num_blocks; ///< Number of audio blocks int channel_mode; ///< Channel mode (acmod) int lfe_on; ///< Low frequency effect channel on (lfeon) + int channel_map; ///< Custom channel map int bitstream_id; ///< Bit stream identification (bsid) int center_mix_level; ///< Center mix level index int surround_mix_level; ///< Surround mix level index Modified: eac3/ac3dec_data.c ============================================================================== --- eac3/ac3dec_data.c (original) +++ eac3/ac3dec_data.c Mon Mar 17 23:21:11 2008 @@ -1105,6 +1105,23 @@ const uint8_t ff_eac3_frm_expstr[32][6] }; /** + * Default channel map for a dependent substream defined by acmod + */ +const uint16_t ff_eac3_default_chmap[8] = { + AC3_CHMAP_LEFT | AC3_CHMAP_RIGHT, // FIXME Ch1+Ch2 + AC3_CHMAP_CENTRE, + AC3_CHMAP_LEFT | AC3_CHMAP_RIGHT, + AC3_CHMAP_LEFT | AC3_CHMAP_CENTRE | AC3_CHMAP_RIGHT, + AC3_CHMAP_LEFT | AC3_CHMAP_RIGHT | AC3_CHMAP_CENTRE_SURROUND, + AC3_CHMAP_LEFT | AC3_CHMAP_CENTRE | AC3_CHMAP_RIGHT | + AC3_CHMAP_CENTRE_SURROUND, + AC3_CHMAP_LEFT | AC3_CHMAP_RIGHT | AC3_CHMAP_LEFT_SURROUND | + AC3_CHMAP_RIGHT_SURROUND, + AC3_CHMAP_LEFT | AC3_CHMAP_CENTRE | AC3_CHMAP_RIGHT | + AC3_CHMAP_LEFT_SURROUND | AC3_CHMAP_RIGHT_SURROUND +}; + +/** * Table E2.16 Default Coupling Banding Structure */ const uint8_t ff_eac3_default_cpl_band_struct[18] = Modified: eac3/ac3dec_data.h ============================================================================== --- eac3/ac3dec_data.h (original) +++ eac3/ac3dec_data.h Mon Mar 17 23:21:11 2008 @@ -37,4 +37,18 @@ extern const uint8_t ff_eac3_defecplbnds extern const uint8_t ff_ac3_rematrix_band_tab[5]; +extern const uint16_t ff_eac3_default_chmap[8]; + +/** Custom channel map locations bitmask + * Other channels described in documentation: Lc/Rc pair, Lrs/Rrs pair, Ts, Lsd/Rsd pair, Lw/Rw pair, Lvh/Rvh pair, Cvh, Reserved, LFE2 */ +enum { + AC3_CHMAP_LEFT = 1<<0, + AC3_CHMAP_CENTRE = 1<<1, + AC3_CHMAP_RIGHT = 1<<2, + AC3_CHMAP_LEFT_SURROUND = 1<<3, + AC3_CHMAP_RIGHT_SURROUND = 1<<4, + AC3_CHMAP_CENTRE_SURROUND = 1<<7, + AC3_CHMAP_LFE = 1<<15 +}; + #endif /* FFMPEG_AC3DEC_DATA_H */ Modified: eac3/eac3dec.c ============================================================================== --- eac3/eac3dec.c (original) +++ eac3/eac3dec.c Mon Mar 17 23:21:11 2008 @@ -331,9 +331,12 @@ static int parse_bsi(AC3DecodeContext *s /* dependent stream channel map */ if (s->stream_type == EAC3_STREAM_TYPE_DEPENDENT) { if (get_bits1(gbc)) { - skip_bits(gbc, 16); // skip custom channel map + s->channel_map = get_bits(gbc, 16); //custom channel map } else { - //TODO default channel map based on acmod and lfeon + //default channel map based on acmod and lfeon + s->channel_map = ff_eac3_default_chmap[s->channel_mode]; + if(s->lfe_on) + s->channel_map |= AC3_CHMAP_LFE; } } #endif

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Re: [FFmpeg-soc] [FFmpeg-devel] [Patch]GSoC 2008 qualification task TS Muxer
by zhentan feng 01 Apr '08

01 Apr '08

2008/3/25, Michael Niedermayer <michaelni(a)gmx.at>: > On Tue, Mar 25, 2008 at 11:57:16AM +0800, zhentan feng wrote: > > 2008/3/25, Michael Niedermayer <michaelni(a)gmx.at>: > > [...] > > > I modified the codes and attached 2 new patches. > > > > The patch names "TSMuxerPatch_svn_dev" is the 4 files against > > svn://svn.mplayerhq.hu/ffmpeg/trunk/libavformat > > I think if the patch works correctly, it would be our expect. > > > [...] > > Review done based on diffs relative to the correct ancestor files. > > > --- ffmpeg-svn/trunk/libavformat/mpegenc.c 2008-02-02 05:27:56.000000000 +0100 > > +++ mpegpes.h 2008-03-25 12:00:15.000000000 +0100 > [...] > > +/** > > + * PES packet description > > + */ > > typedef struct PacketDesc { > > int64_t pts; > > int64_t dts; > > @@ -39,10 +42,13 @@ > > struct PacketDesc *next; > > } PacketDesc; > > > > +/** > > + * PES stream structure > > + */ > > typedef struct { > > AVFifoBuffer fifo; > > uint8_t id; > > - int max_buffer_size; /* in bytes */ > > + int max_buffer_size; /**< in bytes */ > > int buffer_index; > > PacketDesc *predecode_packet; > > PacketDesc *premux_packet; > > These (and many other) changes are unrelated to factorizing the PES code out. Thanks. I have fixed it. > > > > --- ffmpeg-svn/trunk/libavformat/mpegenc.c 2008-02-02 05:27:56.000000000 +0100 > > +++ mpegpesenc.c 2008-03-25 12:00:15.000000000 +0100 > [...] > > switch(st->codec->codec_type) { > > case CODEC_TYPE_AUDIO: > > - if (st->codec->codec_id == CODEC_ID_AC3) { > > - stream->id = ac3_id++; > > - } else if (st->codec->codec_id == CODEC_ID_DTS) { > > - stream->id = dts_id++; > > - } else if (st->codec->codec_id == CODEC_ID_PCM_S16BE) { > > - stream->id = lpcm_id++; > > - for(j = 0; j < 4; j++) { > > - if (lpcm_freq_tab[j] == st->codec->sample_rate) > > - break; > > + if(ps_audio_bound != 0||ps_video_bound != 0){ > > + if (st->codec->codec_id == CODEC_ID_AC3) { > > + stream->id = ac3_id++; > > + } else if (st->codec->codec_id == CODEC_ID_DTS) { > > + stream->id = dts_id++; > > + } else if (st->codec->codec_id == CODEC_ID_PCM_S16BE) { > > + stream->id = lpcm_id++; > > + for(j = 0; j < 4; j++) { > > + if (lpcm_freq_tab[j] == st->codec->sample_rate) > > + break; > > All reindentions as well as other cosmetic changes must be in one or more > seperate patch(s) I see :) > > > [...] > > > @@ -1171,7 +480,6 @@ > > stream->next_packet= &pkt_desc->next; > > > > av_fifo_realloc(&stream->fifo, av_fifo_size(&stream->fifo) + size + 1); > > - > > if (s->is_dvd){ > > if (is_iframe && (s->packet_number == 0 || (pts - stream->vobu_start_pts >= 36000))) { // min VOBU length 0.4 seconds (mpucoder) > > stream->bytes_to_iframe = av_fifo_size(&stream->fifo); > > please remove all useless cosmetic changes fixed > > > > > > Index: mpegenc.c > > =================================================================== > > --- mpegenc.c (revision 12559) > > +++ mpegenc.c (working copy) > [...] > > @@ -181,7 +157,7 @@ > > int P_STD_max_mpeg_PS1 = 0; > > > > for(i=0;i<ctx->nb_streams;i++) { > > - StreamInfo *stream = ctx->streams[i]->priv_data; > > + PESStream *stream = ctx->streams[i]->priv_data; > > Why is the struct renamed? > If there is some sense in this, then it must be in a seperate patch. > If there is no sense in this then it should not be renamed at all. > > [...] > > @@ -314,82 +292,17 @@ > > > > s->audio_bound = 0; > > s->video_bound = 0; > > - mpa_id = AUDIO_ID; > > - ac3_id = AC3_ID; > > - dts_id = DTS_ID; > > - mpv_id = VIDEO_ID; > > - mps_id = SUB_ID; > > - lpcm_id = LPCM_ID; > > - for(i=0;i<ctx->nb_streams;i++) { > > - st = ctx->streams[i]; > > - stream = av_mallocz(sizeof(StreamInfo)); > > - if (!stream) > > - goto fail; > > - st->priv_data = stream; > > + > > trailing whitespace is forbidden in ffmpeg svn fixed > > > > + if(ff_pes_muxer_init(ctx,ps_audio_bound,ps_video_bound) != 0) > > + goto fail; > > > [...] > > @@ -621,26 +534,12 @@ > > put_byte(pb, 0xff); > > } > > > > -static int get_nb_frames(AVFormatContext *ctx, StreamInfo *stream, int len){ > > - int nb_frames=0; > > - PacketDesc *pkt_desc= stream->premux_packet; > > - > > - while(len>0){ > > - if(pkt_desc->size == pkt_desc->unwritten_size) > > - nb_frames++; > > - len -= pkt_desc->unwritten_size; > > - pkt_desc= pkt_desc->next; > > - } > > - > > - return nb_frames; > > -} > > - > > /* flush the packet on stream stream_index */ > > static int flush_packet(AVFormatContext *ctx, int stream_index, > > [...] > > > if (packet_size > 0) { > > + int ps_flag = 1; > > + ff_pes_cal_header(ps_flag,s,id,stream, > > + &packet_size,&header_len,&pts,&dts, > > + &payload_size,&startcode,&stuffing_size, > > + &trailer_size,&pad_packet_bytes); > > > > - /* packet header size */ > > - packet_size -= 6; > > + nb_frames= ff_pes_get_nb_frames(ctx, stream, payload_size - stuffing_size); > > > > - /* packet header */ > > - if (s->is_mpeg2) { > > - header_len = 3; > > - if (stream->packet_number==0) > > - header_len += 3; /* PES extension */ > [...] > > > > put_be16(ctx->pb, packet_size); > > @@ -987,105 +814,19 @@ > > } > > #endif > > > > -static int remove_decoded_packets(AVFormatContext *ctx, int64_t scr){ > > -// MpegMuxContext *s = ctx->priv_data; > > - int i; > > - > > - for(i=0; i<ctx->nb_streams; i++){ > > - AVStream *st = ctx->streams[i]; > > - StreamInfo *stream = st->priv_data; > > - PacketDesc *pkt_desc; > > - > > - while((pkt_desc= stream->predecode_packet) > > - && scr > pkt_desc->dts){ //FIXME > vs >= > > - if(stream->buffer_index < pkt_desc->size || > > - stream->predecode_packet == stream->premux_packet){ > > - av_log(ctx, AV_LOG_ERROR, > > - "buffer underflow i=%d bufi=%d size=%d\n", > > - i, stream->buffer_index, pkt_desc->size); > > - break; > > - } > > - stream->buffer_index -= pkt_desc->size; > > - > > - stream->predecode_packet= pkt_desc->next; > > - av_freep(&pkt_desc); > > - } > > - } > > - > > - return 0; > > -} > > - > > static int output_packet(AVFormatContext *ctx, int flush){ > > MpegMuxContext *s = ctx->priv_data; > > It would be very nice if you could split the different factorizations of > code into seperate patches. I see :) > > > [...] > > > +#define AUDIO_ID 0xc0 > > +#define VIDEO_ID 0xe0 > > +#define AC3_ID 0x80 > > +#define DTS_ID 0x8a > > +#define LPCM_ID 0xa0 > > +#define SUB_ID 0x20 > > code duplication > I will fix it. > > [...] > > > > > /* prepare packet header */ > > q = buf; > > *q++ = 0x47; > > val = (ts_st->pid >> 8); > > - if (is_start) > > + if (is_start) { > > val |= 0x40; > > + is_start = 0; > > + } > > *q++ = val; > > *q++ = ts_st->pid; > > *q++ = 0x10 | ts_st->cc | (write_pcr ? 0x20 : 0); > > ts_st->cc = (ts_st->cc + 1) & 0xf; > > if (write_pcr) { > > + /* add header and pcr bytes to pcr according to specs */ > > + pcr = ts->cur_pcr + (32+56) * 90000 / ts->mux_rate; > > *q++ = 7; /* AFC length */ > > *q++ = 0x10; /* flags: PCR present */ > > *q++ = pcr >> 25; > > @@ -524,59 +567,8 @@ > > *q++ = pcr >> 1; > > *q++ = (pcr & 1) << 7; > > *q++ = 0; > > + ts->last_pcr = pcr; > > } > > I assume these are bugfixes for the TS muxer? We certainly do want them but > they must be in a seperate patch. Not in the patch spliting the common PES > code out. yes, these are bugfixes.But it was modified by the student last year,not me. The situation is that I just download the codes from svn://svn.mplayerhq.hu/soc/dvbmuxer and I got 4 files mpegtsenc.c, mpegenc.c,mpegpesenc.c,mpegpes.h,which are already bug fixed for mpegtsenc.c against the svn-devl codes, and already extracted some common codes. According to baptiste.coudurie wrote in the early mail,I have just extracted more common codes and do not do any functional changes. As qualification task, I think I will do the next 2 things: 1.Test my codes , confirm it works correctly. 2.Create different patches for different situation: i) create patches against svn://svn.mplayerhq.hu/soc/dvbmuxer to show my changes from the last year's students. ii) create patches against svn-devel codes to show the bug fixed and common codes extracted from mpegtsenc.c and mpegenc.c to mpegpesenc.c. am I right? thank you very much. > > > [...] > > -- > Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB > > Frequently ignored awnser#1 FFmpeg bugs should be sent to our bugtracker. User > questions about the command line tools should be sent to the ffmpeg-user ML. > And questions about how to use libav* should be sent to the libav-user ML. > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.6 (GNU/Linux) > > iD8DBQFH6OSSYR7HhwQLD6sRAk+9AKCTQiebtxysgiwlCJI/MGRj8yeW8QCdHzmN > iPdnWr7Y+hhjjF3vz8K8sGU= > =1wM7 > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel(a)mplayerhq.hu > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel > -- Best wishes~

3 16

GSoC application deadline
by Benjamin Larsson 01 Apr '08

01 Apr '08

The application deadline is today (/5:00 PM PDT / 00:00 UTC April 1, 2008/), so make sure you submit your applications in time! MvH Benjamin Larsson

4 5

[soc]: r2062 - libavfilter/diffs/04_ffmpeg_filters.diff
by vitor 31 Mar '08

31 Mar '08

Author: vitor Date: Mon Mar 31 22:57:46 2008 New Revision: 2062 Log: Oops, I broke audio decoding Modified: libavfilter/diffs/04_ffmpeg_filters.diff Modified: libavfilter/diffs/04_ffmpeg_filters.diff ============================================================================== --- libavfilter/diffs/04_ffmpeg_filters.diff (original) +++ libavfilter/diffs/04_ffmpeg_filters.diff Mon Mar 31 22:57:46 2008 @@ -326,7 +326,15 @@ Index: ffmpeg.c if (ost->video_pad) { av_picture_pad((AVPicture*)final_picture, (AVPicture *)padding_src, -@@ -1165,6 +1434,17 @@ +@@ -1053,6 +1322,7 @@ + static short *samples= NULL; + AVSubtitle subtitle, *subtitle_to_free; + int got_subtitle; ++ int loop; + + if(ist->next_pts == AV_NOPTS_VALUE) + ist->next_pts= ist->pts; +@@ -1165,6 +1435,17 @@ &buffer_to_free); } @@ -344,19 +352,24 @@ Index: ffmpeg.c // preprocess audio (volume) if (ist->st->codec->codec_type == CODEC_TYPE_AUDIO) { if (audio_volume != 256) { -@@ -1205,6 +1485,11 @@ +@@ -1202,9 +1483,16 @@ + } + } + #endif ++ loop = ist->st->codec->codec_type != CODEC_TYPE_VIDEO || ++ avfilter_poll_frame(ist->out_video_filter->inputs[0]); /* if output time reached then transcode raw format, encode packets and output them */ if (start_time == 0 || ist->pts >= start_time) +#if ENABLE_AVFILTER -+ while(ist->st->codec->codec_type == CODEC_TYPE_VIDEO && -+ avfilter_poll_frame(ist->out_video_filter->inputs[0])) { -+ get_filtered_video_pic(ist->out_video_filter, &ist->picref, &picture, &ist->pts); ++ while(loop) { ++ if (ist->st->codec->codec_type == CODEC_TYPE_VIDEO) ++ get_filtered_video_pic(ist->out_video_filter, &ist->picref, &picture, &ist->pts); +#endif for(i=0;i<nb_ostreams;i++) { int frame_size; -@@ -1227,6 +1512,9 @@ +@@ -1227,6 +1515,9 @@ do_audio_out(os, ost, ist, data_buf, data_size); break; case CODEC_TYPE_VIDEO: @@ -366,9 +379,12 @@ Index: ffmpeg.c do_video_out(os, ost, ist, &picture, &frame_size); video_size += frame_size; if (vstats_filename && frame_size) -@@ -1286,6 +1574,12 @@ +@@ -1285,7 +1576,15 @@ + av_free_packet(&opkt); } } ++ loop = (ist->st->codec->codec_type == CODEC_TYPE_VIDEO) && ++ avfilter_poll_frame(ist->out_video_filter->inputs[0]); } + +#if ENABLE_AVFILTER @@ -379,7 +395,7 @@ Index: ffmpeg.c av_free(buffer_to_free); /* XXX: allocate the subtitles in the codec ? */ if (subtitle_to_free) { -@@ -1693,10 +1987,21 @@ +@@ -1693,10 +1992,21 @@ fprintf(stderr, "Cannot get resampling context\n"); exit(1); } @@ -401,7 +417,7 @@ Index: ffmpeg.c break; case CODEC_TYPE_SUBTITLE: ost->encoding_needed = 1; -@@ -3804,6 +4109,9 @@ +@@ -3804,6 +4114,9 @@ #ifdef CONFIG_VHOOK { "vhook", HAS_ARG | OPT_EXPERT | OPT_VIDEO, {(void*)add_frame_hooker}, "insert video processing module", "module" }, #endif

1 0

GSoC 2008 participation
by Bartlomiej Wolowiec 31 Mar '08

31 Mar '08

I would like to participate in GSoC this year and make some project for FFmpeg. As a qualification task I'm finishing last year's work - EAC3 decoder. This year I am keen to write some audio decoder. I'm interested in few ideas from the list. I will be pleased to hear any suggestions about them. G.723.1 Encoder: This task sounds good mainly because the documentations looks very good and there are decoder/encoder patterns. I've got one question: do you mean making fixed or floating-point encoder or both of them? ALAC Encoder: I would like to try to write the encoder base on alac.c and flac*.c, then try myself in efforts of developing existing decoder through reverse engineering. Nellymoser Encoder: I understand that here work should be based on the decoder code (I saw that the documentation is available, but it costs $6000), am I right? or possibly any other encoder is very necessary? (I'm keen to write it :) ) -- Bartlomiej Wolowiec

2 3

Proposal SoC
by Marcondes Ricarte 31 Mar '08

31 Mar '08

Hi, How is the submission of the activity to Google? There are templates for projects? The mentors will indicate the project for the candidates? As the proposal is written? Marcondes Ricarte

4 6

[soc]: r2061 - libavfilter/avfiltergraph.c
by vitor 30 Mar '08

30 Mar '08

Author: vitor Date: Sun Mar 30 17:46:38 2008 New Revision: 2061 Log: Replace consume_char() function by *(*buf)++ Modified: libavfilter/avfiltergraph.c Modified: libavfilter/avfiltergraph.c ============================================================================== --- libavfilter/avfiltergraph.c (original) +++ libavfilter/avfiltergraph.c Sun Mar 30 17:46:38 2008 @@ -216,22 +216,6 @@ static void consume_whitespace(const cha } /** - * get the next non-whitespace char - */ -static char consume_char(const char **buf) -{ - char out; - consume_whitespace(buf); - - out = **buf; - - if (out) - (*buf)++; - - return out; -} - -/** * Copy the first size bytes of input string to a null-terminated string, * removing any control character. Ex: "aaa'bb'c\'c\\" -> "aaabbc'c\" */ @@ -304,14 +288,14 @@ static char *consume_string(const char * */ static void parse_link_name(const char **buf, char **name) { - consume_char(buf); + (*buf)++; *name = consume_string(buf); if (!*name[0]) goto fail; - if (consume_char(buf) != ')') + if (*(*buf)++ != ')') goto fail; return; @@ -333,7 +317,7 @@ static int parse_filter(const char **buf name = consume_string(buf); if (**buf == '=') { - consume_char(buf); + (*buf)++; opts = consume_string(buf); } else { opts = NULL; @@ -393,8 +377,8 @@ static int parse_inouts(const char **buf */ int avfilter_graph_parse_chain(AVFilterGraph *graph, const char *filters, AVFilterContext *in, int inpad, AVFilterContext *out, int outpad) { - AVFilterInOut *inout=NULL; - AVFilterInOut *head=NULL; + AVFilterInOut *inout=NULL; + AVFilterInOut *head=NULL; int index = 0; char chr = 0; @@ -434,7 +418,7 @@ int avfilter_graph_parse_chain(AVFilterG } pad = parse_inouts(&filters, &inout, 0, LinkTypeOut, index); - chr = consume_char(&filters); + chr = *filters++; index++; } while (chr == ',' || chr == ';');

1 0

[soc]: r2060 - libavfilter/diffs/04_ffmpeg_filters.diff
by vitor 30 Mar '08

30 Mar '08

Author: vitor Date: Sun Mar 30 13:30:37 2008 New Revision: 2060 Log: Fix filtering when copying to two output files, for ex: ffmpeg -i file1.avi -vfilters vflip out1.avi out2.avi Patch by Mike Sullivan (mike <at> affinesystems.com) Modified: libavfilter/diffs/04_ffmpeg_filters.diff Modified: libavfilter/diffs/04_ffmpeg_filters.diff ============================================================================== --- libavfilter/diffs/04_ffmpeg_filters.diff (original) +++ libavfilter/diffs/04_ffmpeg_filters.diff Sun Mar 30 13:30:37 2008 @@ -344,31 +344,42 @@ Index: ffmpeg.c // preprocess audio (volume) if (ist->st->codec->codec_type == CODEC_TYPE_AUDIO) { if (audio_volume != 256) { -@@ -1227,10 +1507,23 @@ +@@ -1205,6 +1485,11 @@ + /* if output time reached then transcode raw format, + encode packets and output them */ + if (start_time == 0 || ist->pts >= start_time) ++#if ENABLE_AVFILTER ++ while(ist->st->codec->codec_type == CODEC_TYPE_VIDEO && ++ avfilter_poll_frame(ist->out_video_filter->inputs[0])) { ++ get_filtered_video_pic(ist->out_video_filter, &ist->picref, &picture, &ist->pts); ++#endif + for(i=0;i<nb_ostreams;i++) { + int frame_size; + +@@ -1227,6 +1512,9 @@ do_audio_out(os, ost, ist, data_buf, data_size); break; case CODEC_TYPE_VIDEO: +#if ENABLE_AVFILTER -+ while(avfilter_poll_frame(ist->out_video_filter->inputs[0])) { -+ get_filtered_video_pic(ist->out_video_filter, &ist->picref, &picture, &ist->pts); -+ ost->st->codec->sample_aspect_ratio = ist->picref->pixel_aspect; -+ do_video_out(os, ost, ist, &picture, &frame_size); -+ video_size += frame_size; -+ if (vstats_filename && frame_size) -+ do_video_stats(os, ost, frame_size); -+ if(ist->picref) -+ avfilter_unref_pic(ist->picref); -+ } -+#else ++ ost->st->codec->sample_aspect_ratio = ist->picref->pixel_aspect; ++#endif do_video_out(os, ost, ist, &picture, &frame_size); video_size += frame_size; if (vstats_filename && frame_size) - do_video_stats(os, ost, frame_size); +@@ -1286,6 +1574,12 @@ + } + } + } ++ ++#if ENABLE_AVFILTER ++ if(ist->picref) ++ avfilter_unref_pic(ist->picref); ++ } +#endif - break; - case CODEC_TYPE_SUBTITLE: - do_subtitle_out(os, ost, ist, &subtitle, -@@ -1693,10 +1986,21 @@ + av_free(buffer_to_free); + /* XXX: allocate the subtitles in the codec ? */ + if (subtitle_to_free) { +@@ -1693,10 +1987,21 @@ fprintf(stderr, "Cannot get resampling context\n"); exit(1); } @@ -390,7 +401,7 @@ Index: ffmpeg.c break; case CODEC_TYPE_SUBTITLE: ost->encoding_needed = 1; -@@ -3804,6 +4108,9 @@ +@@ -3804,6 +4109,9 @@ #ifdef CONFIG_VHOOK { "vhook", HAS_ARG | OPT_EXPERT | OPT_VIDEO, {(void*)add_frame_hooker}, "insert video processing module", "module" }, #endif

1 0

Status for QuickTime Edit List Support
by sathvik L 29 Mar '08

29 Mar '08

Hi , The link which has the qt implementation of xine demuxer was very helpful , thanks on that . My progress isn't going very well , as the deadline for my college projects are april 9 . I am putting all my effort to get this done :) Regards , Sathvik.

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