March 2014 - FFmpeg-trac - lists.ffmpeg.org

Re: [FFmpeg-trac] #3367(avcodec:closed): vlc2.1.2, crash when using hw accel vdpau
by FFmpeg 03 Mar '14

03 Mar '14

#3367: vlc2.1.2, crash when using hw accel vdpau ------------------------------------+----------------------------------- Reporter: promeneur | Owner: Type: defect | Status: closed Priority: important | Component: avcodec Version: git-master | Resolution: fixed Keywords: videolan | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ------------------------------------+----------------------------------- Changes (by cehoyos): * keywords: => videolan * resolution: => fixed * status: new => closed * component: undetermined => avcodec * version: unspecified => git-master Comment: I suspect this issue is fixed since a22907e9 -- Ticket URL: <https://trac.ffmpeg.org/ticket/3367#comment:3> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #1582(swscale:open): Banding/discoloration in darker gradients when converting to rgb24
by FFmpeg 03 Mar '14

03 Mar '14

#1582: Banding/discoloration in darker gradients when converting to rgb24 -------------------------------------+------------------------------------- Reporter: y3kcjd5 | Owner: Type: defect | Status: open Priority: normal | Component: swscale Version: git-master | Resolution: Keywords: swscale, | Blocked By: bounty | Reproduced by developer: 1 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Comment (by y3kcjd5): New findings: With "+full_chroma_int" I was able to fix the discoloration, but I was still getting some mild luma banding until I finally figured out to use "-sws_flags full_chroma_int+accurate_rnd", at which point output was finally for all intents and purposes visually indiscernible from input. +1 lobby for this being default behavior. -- Ticket URL: <https://trac.ffmpeg.org/ticket/1582#comment:14> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3277(undetermined:new): Crash in libavcodec with vlc (was: Segfault with sample.)
by FFmpeg 02 Mar '14

02 Mar '14

#3277: Crash in libavcodec with vlc -------------------------------------+------------------------------------- Reporter: | Owner: SevereOverfl0w | Status: new Type: defect | Component: Priority: important | undetermined Version: unspecified | Resolution: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- -- Ticket URL: <https://trac.ffmpeg.org/ticket/3277#comment:10> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3229(avcodec:open): Wrong field order for mjpeg (was: ffplay shows bad images)
by FFmpeg 02 Mar '14

02 Mar '14

#3229: Wrong field order for mjpeg ------------------------------------+----------------------------------- Reporter: hxuanyu | Owner: Type: defect | Status: open Priority: normal | Component: avcodec Version: git-master | Resolution: Keywords: mjpeg | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 0 | ------------------------------------+----------------------------------- -- Ticket URL: <https://trac.ffmpeg.org/ticket/3229#comment:6> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3391(undetermined:new): Gaps in input timestamps cause invalid Ogg Opus output
by FFmpeg 02 Mar '14

02 Mar '14

#3391: Gaps in input timestamps cause invalid Ogg Opus output -------------------------------------+------------------------------------- Reporter: MarkZV | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- When transcoding to Ogg Opus from a format with per-packet timestamps, such as mp4 audio, if the input contains a gap in packet timestamps (e.g. due to a lost/corrupted packet), FFmpeg writes invalid Ogg Opus output. Specifically, the Ogg granule position written by FFmpeg increments by more than the number of samples contained in packets that complete on the page. This violates [https://tools.ietf.org/html/draft-ietf-codec- oggopus-03#section-4 section 4 of the specification] and may cause synchronization problems. Because the Ogg container does not have explicit per-frame or per-packet timestamps, Ogg Opus does not allow gaps in the audio. Instead, an equal duration of silence or predicted samples may be substituted for the missing samples when encoding. If pre-encoded Opus is being copied to Ogg without encoding, any missing Opus packets may be replaced with Opus PLC (packet loss concealment) packets as described in [https://tools.ietf.org/html/draft-ietf-codec-oggopus-03#section-4.1 section 4.1 of the specification]. {{{ $ ffmpeg -v 9 -loglevel 99 -i lostpacket.m4a out.opus ffmpeg version N-60696-g38a08e0 Copyright (c) 2000-2014 the FFmpeg developers built on Feb 17 2014 01:24:12 with gcc 4.8.2 (MacPots gcc48 4.8.2_0) configuration: --enable-debug --assert-level=2 --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable- libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-fontconfig --enable-libfreetype --mandir=/opt/local/share/man --enable-pthreads --cc=/opt/local/bin/gcc-mp-4.8 --arch=x86_64 --enable- yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-libfdk-aac --enable-libfaac libavutil 52. 64.100 / 52. 64.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 1.102 / 4. 1.102 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'. Reading option '-i' ... matched as input file with argument 'lostpacket.m4a'. Reading option 'out.opus' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument 9. Successfully parsed a group of options. Parsing a group of options: input file lostpacket.m4a. Successfully parsed a group of options. Opening an input file: lostpacket.m4a. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] ISO: File Type Major Brand: M4A [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] Before avformat_find_stream_info() pos: 28861 bytes read:28861 seeks:0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] After avformat_find_stream_info() pos: 48 bytes read:28861 seeks:0 frames:1 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'lostpacket.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.32.101 Duration: 00:00:03.54, start: 0.042667, bitrate: 65 kb/s Stream #0:0(und), 1, 1/48000: Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 61 kb/s (default) Metadata: handler_name : SoundHandler Successfully opened the file. Parsing a group of options: output file out.opus. Successfully parsed a group of options. Opening an output file: out.opus. Successfully opened the file. detected 4 logical cores [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'time_base' to value '1/48000' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'sample_rate' to value '48000' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'sample_fmt' to value 'fltp' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'channel_layout' to value '0x3' [graph 0 input from stream 0:0 @ 0x102a12fc0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3 [audio format for output stream 0:0 @ 0x102a138a0] Setting 'sample_fmts' to value 's16|flt' [audio format for output stream 0:0 @ 0x102a138a0] Setting 'sample_rates' to value '48000|24000|16000|12000|8000' [audio format for output stream 0:0 @ 0x102a138a0] Setting 'channel_layouts' to value '0x4|0x3|0x7|0x33|0x37|0x3f|0x70f|0x63f' [audio format for output stream 0:0 @ 0x102a138a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0' [AVFilterGraph @ 0x102a11be0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x102a14080] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:flt r:48000Hz [libopus @ 0x10300f800] No bit rate set. Defaulting to 96000 bps. Output #0, opus, to 'out.opus': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.33.100 Stream #0:0(und), 0, 1/48000: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s (default) Metadata: handler_name : SoundHandler major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.33.100 Stream mapping: Stream #0:0 -> #0:0 (aac -> libopus) Press [q] to stop, [?] for help [libopus @ 0x10300f800] Trying to remove 584 more samples than there are in the queue [output stream 0:0 @ 0x102a13300] EOF on sink link output stream 0:0:default. No more output streams to write to, finishing. size= 49kB time=00:00:03.57 bitrate= 112.8kbits/s video:0kB audio:49kB subtitle:0 data:0 global headers:0kB muxing overhead 1.290400% 166 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x102a12c80] Statistics: 0 seeks, 6 writeouts [AVIOContext @ 0x102a11dc0] Statistics: 28861 bytes read, 0 seeks $ }}} The incorrect Ogg Opus file produces warnings from the opusinfo tool (part of [http://www.opus-codec.org/downloads/ opus-tools]); "Sample count behind granule": {{{ $ opusinfo out.opus Processing file "out.opus"... New logical stream (#1, serial: f039efde): type opus Encoded with Lavf55.33.100 User comments section follows... language=und handler_name=SoundHandler major_brand=M4A minor_version=512 compatible_brands=isomiso2 encoder=Lavf55.33.100 WARNING: Sample count behind granule (95040>96064) in stream 1 WARNING: Sample count behind granule (143040>144064) in stream 1 WARNING: Sample count behind granule (170880>171320) in stream 1 Opus stream 1: Pre-skip: 312 Playback gain: 0 dB Channels: 2 Original sample rate: 48000Hz Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min) Page duration: 1000.0ms (max), 890.0ms (avg), 580.0ms (min) Total data length: 50394 bytes (overhead: 1.31%) Playback length: 0m:03.562s Average bitrate: 113.2 kb/s, w/o overhead: 111.7 kb/s Logical stream 1 ended $ }}} It is expected that FFmpeg would produce a valid Ogg Opus file, in this case by inserting silence or predicted samples in place of the missing samples before encoding. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3391> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3335(undetermined:closed): Audio Rematrix error going from 16ch
by FFmpeg 01 Mar '14

01 Mar '14

#3335: Audio Rematrix error going from 16ch -------------------------------------+------------------------------------- Reporter: | Owner: zcybercomputing | Status: closed Type: defect | Component: Priority: normal | undetermined Version: unspecified | Resolution: invalid Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Changes (by cehoyos): * priority: important => normal * status: reopened => closed * resolution: => invalid Comment: Did you try the link in comment:5? -- Ticket URL: <https://trac.ffmpeg.org/ticket/3335#comment:11> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3335(undetermined:reopened): Audio Rematrix error going from 16ch
by FFmpeg 01 Mar '14

01 Mar '14

#3335: Audio Rematrix error going from 16ch -------------------------------------+------------------------------------- Reporter: | Owner: zcybercomputing | Status: reopened Type: defect | Component: Priority: important | undetermined Version: unspecified | Resolution: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Comment (by zcybercomputing): What I want to do is use the first two channels and ignore everything else. If I was doing 5.1, then they would be the first 6 channels and the rest would be discarded. This is the standard way of mapping channels when working with SDI signals (which is what produces the 16ch stream in the first place). I am sorry, I have not been able to locate part of the documentation that sufficiently explains how to structure the syntax. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3335#comment:10> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3335(undetermined:reopened): Audio Rematrix error going from 16ch
by FFmpeg 01 Mar '14

01 Mar '14

#3335: Audio Rematrix error going from 16ch -------------------------------------+------------------------------------- Reporter: | Owner: zcybercomputing | Status: reopened Type: defect | Component: Priority: important | undetermined Version: unspecified | Resolution: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Changes (by Cigaes): * priority: normal => important Comment: Replying to [comment:7 zcybercomputing]: > C:\Users\3ABN>"C:\Users\3ABN\Another Gui\ffmpeg-20140115-git- 785dc14-win64-static\bin\ffmpeg.exe" -i "F:\test.mov" -y -vcodec prores -profile:v 1 -acodec pcm_s16le -ar 48000 -ac 2 "F:\Capture0000test.mov" Your file has 16 channels, you ask to make that two. How? Take the first two and drop the others? Mix the 0-7 together and 8-15 together? Mix the even ones together and the odd ones together? None of the above? FFmpeg does not guess that kind of thing, this is not a magic trick: > [auto-inserted resampler 0 @ 000000000525ac60] [SWR @ 0000000002940900] Rematrix is needed between 16 channels and stereo but there is not enough information to do it -- Ticket URL: <https://trac.ffmpeg.org/ticket/3335#comment:9> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3335(undetermined:reopened): Audio Rematrix error going from 16ch
by FFmpeg 01 Mar '14

01 Mar '14

#3335: Audio Rematrix error going from 16ch -------------------------------------+------------------------------------- Reporter: | Owner: zcybercomputing | Status: reopened Type: defect | Component: Priority: normal | undetermined Version: unspecified | Resolution: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Changes (by cehoyos): * priority: important => normal Comment: Your command line is simply invalid, conversion from "16 channels" to "stereo" is not defined (and cannot be defined). Please test the pan filter and report back. Generally, please don't change ticket priorities. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3335#comment:8> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3335(undetermined:reopened): Audio Rematrix error going from 16ch
by FFmpeg 01 Mar '14

01 Mar '14

#3335: Audio Rematrix error going from 16ch -------------------------------------+------------------------------------- Reporter: | Owner: zcybercomputing | Status: reopened Type: defect | Component: Priority: important | undetermined Version: unspecified | Resolution: Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Changes (by zcybercomputing): * priority: normal => important * status: closed => reopened * resolution: needs_more_info => Comment: I figured out what the problem is. The file contains a single stream that contains 16 channels of audio. For some reason ffmpeg refuses to manipulate this type of audio stream. This type of setup is very common in broadcast video recorders, and an inability to work with it is problematic to workflow. reference to another user with this problem. http://ffmpeg.org/pipermail/ffmpeg-user/2013-October/018067.html Example file: https://drive.google.com/file/d/0B5GSKNa8Y3Dga05Uc09WRmtwVDg/edit?usp=shari… Output: Microsoft Windows [Version 6.1.7601] Copyright (c) 2009 Microsoft Corporation. All rights reserved. C:\Users\3ABN>"C:\Users\3ABN\Another Gui\ffmpeg-20140115-git- 785dc14-win64-stati c\bin\ffmpeg.exe" -i "F:\test.mov" -y -vcodec prores -profile:v 1 -acodec pcm_s1 6le -ar 48000 -ac 2 "F:\Capture0000test.mov" ffmpeg version N-59852-g785dc14 Copyright (c) 2000-2014 the FFmpeg developers built on Jan 14 2014 22:07:30 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable- libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable- libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable- libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 62.100 / 52. 62.100 libavcodec 55. 48.101 / 55. 48.101 libavformat 55. 23.103 / 55. 23.103 libavdevice 55. 5.102 / 55. 5.102 libavfilter 4. 1.100 / 4. 1.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0000000002947800] Stream #1: not enough frames to est imate rate; consider increasing probesize Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'F:\test.mov': Metadata: creation_time : 2014-02-21 02:25:06 Duration: 00:00:02.74, start: 0.000000, bitrate: 81448 kb/s Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 720x486, 6 2874 kb/s, SAR 10:11 DAR 400:297, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc (defaul t) Metadata: creation_time : 2014-02-21 02:25:06 handler_name : Apple Alias Data Handler Stream #0:1(eng): Audio: pcm_s24le (lpcm / 0x6D63706C), 48000 Hz, 16 channel s, s32, 18432 kb/s (default) Metadata: creation_time : 2014-02-21 02:25:06 handler_name : Apple Alias Data Handler [auto-inserted resampler 0 @ 000000000525ac60] [SWR @ 0000000002940900] Rematrix is needed between 16 channels and stereo but there is not enough information to do it [auto-inserted resampler 0 @ 000000000525ac60] Failed to configure output pad on auto-inserted resampler 0 Error opening filters! C:\Users\3ABN> -- Ticket URL: <https://trac.ffmpeg.org/ticket/3335#comment:7> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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