Builder ffmpeg-solaris10-i386 Build #14004
Results:
Failed shell_2 shell_3 shell_4 shell_5
SourceStamp:
| Project | ffmpeg |
| Repository | https://git.ffmpeg.org/ffmpeg.git |
| Branch | master |
| Revision | 44d7755f7ec894dd52076cc37fca78c82556bb73 |
| Got Revision | 44d7755f7ec894dd52076cc37fca78c82556bb73 |
| Changes | 10 changes |
BuildSlave:
unstable10xReason:
The SingleBranchScheduler scheduler named 'schedule-ffmpeg-solaris10-i386' triggered this build
Steps and Logfiles:
-
git update ( 9 secs )
-
shell 'gsed -i ...' ( 0 secs )
-
shell_1 'gsed -i ...' ( 0 secs )
-
shell_2 'gsed -i ...' failed ( 0 secs )
-
shell_3 './configure --samples="../../../ffmpeg/fate-suite" ...' failed ( 8 secs )
-
shell_4 'gmake fate-rsync' failed ( 0 secs )
-
shell_5 '../../../ffmpeg/fate.sh ../../../ffmpeg/fate_config.sh' failed ( 0 secs )
Build Properties:
| Name | Value | Source |
|---|---|---|
| branch | master | Build |
| builddir | /export/home/buildbot/slave/ffmpeg-solaris10-i386 | slave |
| buildername | ffmpeg-solaris10-i386 | Builder |
| buildnumber | 14004 | Build |
| codebase | Build | |
| got_revision | 44d7755f7ec894dd52076cc37fca78c82556bb73 | Git |
| project | ffmpeg | Build |
| repository | https://git.ffmpeg.org/ffmpeg.git | Build |
| revision | 44d7755f7ec894dd52076cc37fca78c82556bb73 | Build |
| scheduler | schedule-ffmpeg-solaris10-i386 | Scheduler |
| slavename | unstable10x | BuildSlave |
| workdir | /export/home/buildbot/slave/ffmpeg-solaris10-i386 | slave (deprecated) |
Forced Build Properties:
| Name | Label | Value |
|---|
Responsible Users:
- James Almerjamrial@gmail.com
Timing:
| Start | Tue Jun 16 14:29:30 2026 |
| End | Tue Jun 16 14:29:49 2026 |
| Elapsed | 18 secs |
All Changes:
:
Change #271306
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:23 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 0c29837dac4f67c9a5ca6f96cb0cbb5572b692af Comments
avcodec/mpeg4audio: add a frame_length field to MPEG4AudioConfig Will be useful to get fixed frame sizes outside decoders. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavcodec/mpeg4audio.c
- libavcodec/mpeg4audio.h
Change #271307
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:23 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 47b4be8865cf9b295de8de54e3a0d23fafbc190f Comments
avformat/isom: export codecpar frame_size Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavformat/isom.c
Change #271308
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:23 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 68b53a8dbfdfbcce72d5b676edfcddb2f5d7a070 Comments
avformat/demux: discard trimming samples in codecs with fixed frame size When a demuxer reports the last packet with a duration smaller than the real coded duration, this information is not relayed to the decoder, which will happily output all the trimming samples anyway. Fix that by ensuring we export a discard padding information as side data. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavformat/demux.c
Change #271309
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:23 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 2153c6795cf93bc374357a4408f425213d872398 Comments
avcodec/decode: don't discard the existing skip_samples value if a new side data doesn't report any Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavcodec/decode.c
Change #271310
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 51f5f60443eec9944c9d0cf24e0766edf403d7ef Comments
avformat/movenc: use a common denominator across all tracks as movie timescale The default of 1000 may result in off by 1 errors when rescaling certain durations, as is the case of fate-gaplessenc-itunes-to-ipod-aac, so lets try to prevent that by using a global timescale every track can agree with whenever possible. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavformat/movenc.c
- libavformat/movenc.h
- tests/fate/mov.mak
- tests/ref/acodec/alac
- tests/ref/acodec/pcm-s16be
- tests/ref/acodec/pcm-s24be
- tests/ref/acodec/pcm-s32be
- tests/ref/acodec/pcm-s8
- tests/ref/fate/adtstoasc_ticket3715
- tests/ref/fate/autorotate
- tests/ref/fate/binsub-movtextenc
- tests/ref/fate/copy-psp
- tests/ref/fate/copy-trac236
- tests/ref/fate/copy-trac3074
- tests/ref/fate/filter-meta-4560-rotate0
- tests/ref/fate/gaplessenc-itunes-to-ipod-aac
- tests/ref/fate/generic-tags-remux-mov
- tests/ref/fate/h264-bsf-dts2pts
- tests/ref/fate/hevc-bsf-dts2pts-cra
- tests/ref/fate/hevc-bsf-dts2pts-idr
- tests/ref/fate/hevc-bsf-dts2pts-idr-cra
- tests/ref/fate/media100
- tests/ref/fate/mov-channel-description
- tests/ref/fate/mov-cover-image
- tests/ref/fate/mov-dovi-hvce-mp4-to-mp4
- tests/ref/fate/mov-mp4-chapters
- tests/ref/fate/mov-mp4-disposition-mpegts-remux
- tests/ref/fate/mov-mp4-fragmented-ttml-dfxp
- tests/ref/fate/mov-mp4-fragmented-ttml-stpp
- tests/ref/fate/mov-mp4-iamf-5_1_4
- tests/ref/fate/mov-mp4-iamf-7_1_4-video-last
- tests/ref/fate/mov-mp4-iamf-ambisonic_1
- tests/ref/fate/mov-mp4-iamf-stereo
- tests/ref/fate/mov-mp4-pcm
- tests/ref/fate/mov-mp4-pcm-float
- tests/ref/fate/mov-write-amve
- tests/ref/fate/movenc
- tests/ref/fate/prores-metadata
- tests/ref/lavf-fate/evc.mp4
- tests/ref/lavf-fate/h264.mp4
- tests/ref/lavf-fate/hevc.mp4
- tests/ref/lavf-fate/qtrle_mace6.mov
- tests/ref/lavf-fate/vvc.mp4
- tests/ref/lavf/ismv
- tests/ref/lavf/mov
- tests/ref/lavf/mov_hybrid_frag
- tests/ref/lavf/mov_rtphint
- tests/ref/lavf/mp4
Change #271311
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 298e5f810cc56ce5622522416676de89b3ff8b91 Comments
tests/audiomatch: don't attempt to print floating point values Instead check that the result is sane Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- tests/audiomatch.c
- tests/ref/fate/audiomatch-afconvert-16000-mono-he-adts
- tests/ref/fate/audiomatch-afconvert-16000-mono-he-m4a
- tests/ref/fate/audiomatch-afconvert-16000-mono-lc-adts
- tests/ref/fate/audiomatch-afconvert-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he-adts
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he2-adts
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he2-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-lc-adts
- tests/ref/fate/audiomatch-afconvert-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-afconvert-44100-mono-he-adts
- tests/ref/fate/audiomatch-afconvert-44100-mono-he-m4a
- tests/ref/fate/audiomatch-afconvert-44100-mono-lc-adts
- tests/ref/fate/audiomatch-afconvert-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he-adts
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he2-adts
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he2-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-lc-adts
- tests/ref/fate/audiomatch-afconvert-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-dolby-44100-mono-he-mp4
- tests/ref/fate/audiomatch-dolby-44100-mono-lc-mp4
- tests/ref/fate/audiomatch-dolby-44100-stereo-he-mp4
- tests/ref/fate/audiomatch-dolby-44100-stereo-he2-mp4
- tests/ref/fate/audiomatch-dolby-44100-stereo-lc-mp4
- tests/ref/fate/audiomatch-faac-16000-mono-lc-adts
- tests/ref/fate/audiomatch-faac-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-faac-16000-stereo-lc-adts
- tests/ref/fate/audiomatch-faac-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-faac-44100-mono-lc-adts
- tests/ref/fate/audiomatch-faac-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-faac-44100-stereo-lc-adts
- tests/ref/fate/audiomatch-faac-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-nero-16000-mono-he-m4a
- tests/ref/fate/audiomatch-nero-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-he-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-he2-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-nero-44100-mono-he-m4a
- tests/ref/fate/audiomatch-nero-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-he-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-he2-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-quicktime7-44100-stereo-lc-mp4
- tests/ref/fate/audiomatch-quicktimeX-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-square-aac
- tests/ref/fate/audiomatch-square-mp3
Change #271312
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision da04251772a39213aab009972c6dec89dfa56a94 Comments
avformat/mov: export information about the last actual sample in a stream This way the generic demux code can calculate how many trimming samples should be discarded. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavformat/mov.c
- tests/fate/aac.mak
- tests/ref/fate/audiomatch-afconvert-16000-mono-he-m4a
- tests/ref/fate/audiomatch-afconvert-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-he2-m4a
- tests/ref/fate/audiomatch-afconvert-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-afconvert-44100-mono-he-m4a
- tests/ref/fate/audiomatch-afconvert-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-he2-m4a
- tests/ref/fate/audiomatch-afconvert-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-faac-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-faac-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-faac-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-faac-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-nero-16000-mono-he-m4a
- tests/ref/fate/audiomatch-nero-16000-mono-lc-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-he-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-he2-m4a
- tests/ref/fate/audiomatch-nero-16000-stereo-lc-m4a
- tests/ref/fate/audiomatch-nero-44100-mono-he-m4a
- tests/ref/fate/audiomatch-nero-44100-mono-lc-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-he-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-he2-m4a
- tests/ref/fate/audiomatch-nero-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-quicktime7-44100-stereo-lc-mp4
- tests/ref/fate/audiomatch-quicktimeX-44100-stereo-lc-m4a
- tests/ref/fate/audiomatch-square-aac
- tests/ref/fate/autorotate
- tests/ref/fate/copy-psp
- tests/ref/fate/gaplessenc-itunes-to-ipod-aac
- tests/ref/fate/gaplessenc-pcm-to-mov-aac
- tests/ref/fate/gaplessinfo-itunes1
- tests/ref/fate/gaplessinfo-itunes2
- tests/ref/fate/generic-tags-remux-mov
- tests/ref/fate/matroska-dovi-write-config8
- tests/ref/fate/mov-440hz-10ms
- tests/ref/fate/mov-aac-2048-priming
- tests/ref/fate/segment-adts-to-mkv-header-002
- tests/ref/fate/segment-adts-to-mkv-header-all
Change #271313
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 60d50e2f2949e633c27a174f5dedd9495b02e0ba Comments
tests/fate/gapless: print packet side data Only a hinting "|" delimiter character was being printed. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- tests/fate-run.sh
- tests/ref/fate/gaplessenc-itunes-to-ipod-aac
- tests/ref/fate/gaplessenc-pcm-to-mov-aac
- tests/ref/fate/gaplessinfo-itunes1
- tests/ref/fate/gaplessinfo-itunes2
Change #271314
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision f18bc95323f06f4fce073142a939141b64906639 Comments
tests/fate/aac: use mp4 instead of adts for encoding tests It properly signals primming and padding samples, which lets us remove all the comparison offsets. But leave one test using adts, to not reduce coverage. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- tests/fate/aac.mak
Change #271315
Category ffmpeg Changed by James Almer <jamrial@gmail.com> Changed at Tue 16 Jun 2026 14:18:24 Repository https://git.ffmpeg.org/ffmpeg.git Project ffmpeg Branch master Revision 44d7755f7ec894dd52076cc37fca78c82556bb73 Comments
avformat/mov: export initial padding Some muxers, like Matroska, use it to write priming samples. fate-segment-adts-to-mkv no longer uses the ref file from fate-segment-adts-to-mkv-header-all as it's demuxed through the hls demuxer and this commit exposed a bug where initial padding is not being propagated. Signed-off-by: James Almer <jamrial@gmail.com>
Changed files
- libavformat/isom.h
- libavformat/mov.c
- tests/fate/segment.mak
- tests/ref/fate/segment-adts-to-mkv
- tests/ref/fate/segment-adts-to-mkv-header-000
- tests/ref/fate/segment-adts-to-mkv-header-001
- tests/ref/fate/segment-adts-to-mkv-header-002
- tests/ref/fate/segment-adts-to-mkv-header-all