February 2012 - FFmpeg-trac - lists.ffmpeg.org

#470(undetermined:new): libquicktime uncompressed formats
by FFmpeg 05 Feb '12

05 Feb '12

#470: libquicktime uncompressed formats -------------------------------------+------------------------------------- Reporter: cehoyos | Owner: Type: | Status: new enhancement | Component: Priority: wish | undetermined Version: git- | Keywords: v308 v408 master | v410 yuv4 mov Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- I will attach four uncompressed libquicktime samples (showing lena). They use the following video codecs: {{{ v308: 8 bit planar YUV 4:4:4 v408: 8 bit Planar YUVA 4:4:4:4 v410: 10 bit Packed YUV 4:4:4 yuv4: YUV 4:2:0 }}} Sources are in http://libquicktime.cvs.sourceforge.net/viewvc/libquicktime/libquicktime/pl… -- Ticket URL: <https://avcodec.org/trac/ffmpeg/ticket/470> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#452(FFmpeg:new): APE - Only 16-bit samples are supported
by FFmpeg 05 Feb '12

05 Feb '12

#452: APE - Only 16-bit samples are supported -------------------------------------+------------------------------------- Reporter: Flurby | Owner: michael Type: defect | Status: new Priority: normal | Component: FFmpeg Version: | Keywords: APE decoder unspecified | Blocking: Blocked By: | Analyzed by developer: 0 Reproduced by developer: 0 | -------------------------------------+------------------------------------- When I want to convert an .ape file to a .wav, ffmpeg don't want and says : "Only 16-bit samples are supported". bash$ffmpeg -i SideA.ape SideA.wav The result is : FFmpeg version 0.6.3-rpmfusion [...] [ape @ 0x263d400]Only 16-bit samples are supported Last message repeated 20 times Input #0, ape, from 'SideA.ape': Duration: 00:21:36.00, start: 0.000000, bitrate: 3148 kb/s Stream #0.0: Audio: ape, 96000 Hz, 2 channels File 'SideA.wav' already exists. Overwrite ? [y/N] y [ape @ 0x263d400]Only 16-bit samples are supported Output #0, wav, to 'SideA.wav': Stream #0.0: Audio: pcm_s16le, 96000 Hz, 2 channels, s16, 3072 kb/s Stream mapping: Stream #0.0 -> #0.0 Error while opening decoder for input stream #0.0 bash$ -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/452> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#438(avformat:new): broken frame after seek in ogg container
by FFmpeg 05 Feb '12

05 Feb '12

#438: broken frame after seek in ogg container ----------------------------------+--------------------------------------- Reporter: j | Type: defect Status: new | Priority: normal Component: avformat | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- ogg demuxer does not seek to keyframe: running ffmpeg -ss 23 -i test.ogv -s 640x480 -f mjpeg -an -vframes 1 test.jpg the output frame is not fully decoded. it did not seek to a keyframe and just dumps output. if insted one seeks on the decoded stream. the frame looks ok. ffmpeg -i test.ogv -ss 23 -s 640x480 -f mjpeg -an -vframes 1 test.jpg ffmpeg's ogg demuxer should be able to seek to the previous keyframe in the first case. and decode from there -- Ticket URL: <http://ffmpeg.org/trac/ffmpeg/ticket/438> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#941(avformat:new): Seek problems with ogg decoder
by FFmpeg 05 Feb '12

05 Feb '12

#941: Seek problems with ogg decoder ----------------------------------+--------------------------------------- Reporter: DonMoir | Type: defect Status: new | Priority: normal Component: avformat | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- ffmpeg-git-14d94a1 built on Jan 23 2012 17:40:00 with gcc 4.6.2 configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.101 / 51. 34.101 libavcodec 53. 57.105 / 53. 57.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.101 / 2. 59.101 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 There seems to be several problems with seeking in the ogg decoder (libavformat/oggdec.c) This file can be used to identify some of the key problems: http://sms.pangolin.com/temp/bad_seek_ogg_BuckBunny.zip (46 mb) There are no issues with playback of this file and only seeking is a problem. Fails to seek properly with SMPlayer, ffplay, and anything else using ffmpeg unless special handling is applied. In reference to the function ogg_read_seek in file oggdec.c. 1) I see it makes no attempt to create (if needbe) the AVIndexEntry table for the stream which is pretty much required to make seeking work. Most of the other formats do this when you call it's read_seek function. So I find that I need to create this table prior to calling avformat_seek_file for ogg files. This table can be created by calling av_read_packet until the timestamp of interest is reached. This is pretty quick but it should only be done when needbe. Other formats have other and quicker means to do this, but the read packet method works well enough if nothing else is available. For ogg files, once I know that the index_entries have been created as best they are going to be, I then call av_index_search_timestamp(pStream,timestamp,AVSEEK_FLAG_BACKWARD) and I use the timestamp associated with the returned index for my seek time. The AVSEEK_FLAG_BACKWARD flag makes sure the returned timestamp will be less than or equal to my requested timestamp. If -1 is returned form av_index_search_timestamp I deal with it but this should not happen if the index_entries have been built and all else is good. Now we can call avformat_seek_file with the modified timestamp and expect reasonable results. If you don't do this read_timestamp will fail in ff_seek_frame_binary which is called from ogg_read_seek. Note that av_index_search_timestamp is also pretty much worthless for ogg files if the index_entries have not been created. 2) ogg_read_seek does not do the proper cleanup after a seek. That is, there is stale information left that was in place prior to a seek. The ogg decoder uses private data that it places in AVFormatContext.priv_data. This is known to ogg as struct ogg*. Within this structure, ogg also has information for each stream. This is known to ogg as struct ogg_stream*. The stale information that I know about within the ogg_stream is lastpts and lastdts. These are not modified by a seek and whatever was in there before the seek will be used after a seek and these values will be used by the next av_read_packet. After a seek, I am having to call ogg_reset (also in oggdec.c) but this is clearly not the right thing to do. This ogg_reset is not available to higher level code but I have it hacked in for the present. The packet now will contain AV_NOPTS_VALUE for the pts and dts and this is not right either, but at least I can deal with it. Additional packet reads straighten this out. The lastdts and lastpts should of course reflect the next packet that is read. Not really sure how to clean it all up after a seek to make sure all is normal for ogg files. All the other decoders I have tested do this correctly. I think more needs to be cleaned up in addition to lastpts and lastdts but not sure. With the above changes, I can now seek into the buckbunny sample perfectly. Would be great if someone would revisit the seek code in the oggdec.c file and make sure it does the right thing. There are some other problems with ogg but the above 2 things goes a long way in fixing some of them. For the most part, most of my ogg files work fine after the above changes, but there are still some anoying problems with some files. It's like the audio packets are being used to create index_entries for a video stream for some files and not video packets and so this will throw the index_entries off. So while the seek works fine, it may be some time before the next video frame is available but audio is dead on. Not sure about this one and this is just for some ogg files. I can provide more information and code samples if required. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/941> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#959(undetermined:new): Media 100 codec support.
by FFmpeg 02 Feb '12

02 Feb '12

#959: Media 100 codec support. -------------------------------------+------------------------------------- Reporter: Galane | Type: Status: new | enhancement Component: | Priority: normal undetermined | Version: Keywords: | unspecified Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- I've put a couple of small MOV files, their resource forks, the iFinish transcoder for Windows (which was very hard to find) and a small MOV flattener utility here http://partsbyemc.com/pub/ Read the text file for more info. Hopefully with all this, Media 100 / iFinish support can be added to FFMPEG. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/959> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#960(avformat:new): Application provided invalid, non monotonically increasing dts to muxer
by FFmpeg 01 Feb '12

01 Feb '12

#960: Application provided invalid, non monotonically increasing dts to muxer ----------------------------------+-------------------------------------- Reporter: aurelien | Type: defect Status: new | Priority: important Component: avformat | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+-------------------------------------- I have a sample : H.264/AAC wrapped in MPEG-TS whith cannot be transcoded (or copy using -vcodec copy) using ffmpeg-git. jupiter@~/ffmpeg> ./ffmpeg -y -i /tmp/sample-non-monotonic-dts.ts -an -vcodec copy -f mpegts /dev/null ffmpeg version N-37401-g151ecc2 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 31 2012 12:54:08 with gcc 4.6.2 configuration: --prefix=/home/aurelien/local --enable-postproc --enable- nonfree --enable-gpl --enable-libopencore-amrnb --enable-libfaac --enable- libmp3lame --enable-shared --enable-libxvid --enable-libx264 --extra- cflags=-I/home/aurelien/local/include --extra- ldflags=-L/home/aurelien/local/lib --enable-version3 --disable- optimizations libavutil 51. 36.100 / 51. 36.100 libavcodec 54. 0.102 / 54. 0.102 libavformat 54. 0.100 / 54. 0.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 60.100 / 2. 60.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 52. 0.100 / 52. 0.100 [mpegts @ 0xda43a0] max_analyze_duration 5000000 reached at 5000000 [mpegts @ 0xda43a0] PES packet size mismatch Input #0, mpegts, from '/tmp/sample-non-monotonic-dts.ts': Duration: 00:01:08.67, start: 0.500011, bitrate: 2189 kb/s Program 1 Stream #0:0[0x3e8]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 16:11 DAR 20:11], 50 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x1e2](fra): Audio: mp2 ([3][0][0][0] / 0x0003), 48000 Hz, stereo, s16, 192 kb/s (clean effects) [mpegts @ 0xdab3e0] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to '/dev/null': Metadata: encoder : Lavf54.0.100 Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 720x576 [SAR 16:11 DAR 20:11], q=2-31, 50 fps, 90k tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [mpegts @ 0xdab3e0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 23399 >= 23399 av_interleaved_write_frame(): Invalid argument It does not work with ffmpeg version shipped with Debian Squeeze either. jupiter@~> ffmpeg -y -i /tmp/sample-non-monotonic-dts.ts -an -vcodec copy -f mpegts /dev/null FFmpeg version SVN-r0.5.6-4:0.5.6-3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.6-3 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable- libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable- zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable- libfaad --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Dec 31 2011 16:14:46, gcc: 4.4.5 Input #0, mpegts, from '/tmp/sample-non-monotonic-dts.ts': Duration: 00:01:08.67, start: 0.500022, bitrate: 2189 kb/s Program 1 Stream #0.0[0x3e8]: Video: h264, yuv420p, 720x576 [PAR 16:11 DAR 20:11], 50 tbr, 90k tbn, 50 tbc Stream #0.1[0x1e2](fra): Audio: mp2, 48000 Hz, stereo, s16, 192 kb/s Output #0, mpegts, to '/dev/null': Stream #0.0: Video: 0x0000, yuv420p, 720x576 [PAR 16:11 DAR 20:11], q=2-31, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding [mpegts @ 0x170b870]dts < pcr, TS is invalid Last message repeated 220 times [NULL @ 0x170a710]error, non monotone timestamps 23398 >= 23398 av_interleaved_write_frame(): Error while opening file Sample name : sample-non-monotonic-dts.ts (uploaded on upload.ffmpeg.org) Here are two frames taken from the sample : NEW DATA PID : 1000 Size : 5280 PTS : 81000 DTS : 68400 NEW DATA PID : 1000 Size : 1072 PTS : 68400 DTS : no value I guess the problem is the second frame has pts=dts=68400 which is also the dts of the previous frame though I don't know if the sample is valid regarding ISO 13818-1. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/960> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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