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Builder ffmpegsos-solaris10-sparc Build #13956

Results:

Failed shell_2 shell_3 shell_4 shell_5

SourceStamp:

Projectffmpeg
Repositoryhttps://git.ffmpeg.org/ffmpeg.git
Branchmaster
Revisionc6e28adabac09745f3406ffbf3833b8ac7f3f46a
Got Revisionc6e28adabac09745f3406ffbf3833b8ac7f3f46a
Changes15 changes

BuildSlave:

unstable10s

Reason:

The SingleBranchScheduler scheduler named 'schedule-ffmpegsos-solaris10-sparc' triggered this build

Steps and Logfiles:

  1. git update ( 15 secs )
    1. stdio
  2. shell 'gsed -i ...' ( 0 secs )
    1. stdio
  3. shell_1 'gsed -i ...' ( 0 secs )
    1. stdio
  4. shell_2 'gsed -i ...' failed ( 0 secs )
    1. stdio
  5. shell_3 './configure --samples="../../../ffmpeg/fate-suite" ...' failed ( 8 secs )
    1. stdio
    2. config.log
  6. shell_4 'gmake fate-rsync' failed ( 0 secs )
    1. stdio
  7. shell_5 '../../../ffmpeg/fate.sh ../../../ffmpeg/fate_config_sos.sh' failed ( 0 secs )
    1. stdio
    2. configure.log
    3. compile.log
    4. test.log

Build Properties:

NameValueSource
branch master Build
builddir /export/home/buildbot-unstable10s/slave/ffmpegsos-solaris10-sparc slave
buildername ffmpegsos-solaris10-sparc Builder
buildnumber 13956 Build
codebase Build
got_revision c6e28adabac09745f3406ffbf3833b8ac7f3f46a Git
project ffmpeg Build
repository https://git.ffmpeg.org/ffmpeg.git Build
revision c6e28adabac09745f3406ffbf3833b8ac7f3f46a Build
scheduler schedule-ffmpegsos-solaris10-sparc Scheduler
slavename unstable10s BuildSlave
workdir /export/home/buildbot-unstable10s/slave/ffmpegsos-solaris10-sparc slave (deprecated)

Forced Build Properties:

NameLabelValue

Responsible Users:

  1. Kacper Michajłow

Timing:

StartSat Sep 5 23:15:15 2026
EndSat Sep 5 23:15:41 2026
Elapsed25 secs

All Changes:

:

  1. Change #280898

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:24
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 2bdbf5c67e86fd482c2979e5df17154a68a81392

    Comments

    avcodec/dstdec: fix decoding of uncompressed frames
The DSD payload of an uncompressed frame was copied packed into the
output buffer, but the in-place DSD to PCM conversion expects the DSD
bytes in every 4th byte, in the place of the float sample they produce.

This was always broken, but I guess, the uncompressed DST is something
that exists only on paper.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dstdec.c

  2. Change #280899

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:24
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 655dc1af76bc8a2270710994ba6c95126675e9ae

    Comments

    avutil/samplefmt: add AV_SAMPLE_FMT_DSD
DSD (Direct Stream Digital) is one-bit sigma-delta modulated audio,
found on Super Audio CDs and in DSF/DSDIFF files. libavcodec has
several decoders producing it internally (dsd_*, dst, wavpack DSD),
each converting to float PCM as part of decoding, which makes the
bitstream itself inaccessible to API users.

Add a dedicated sample format so raw DSD can be carried through the
pipeline, e.g. for bit-perfect pass-through to DSD capable DACs.

The format has no planar variant, altform references itself.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • doc/APIchanges
    • libavutil/samplefmt.c
    • libavutil/samplefmt.h
    • libavutil/version.h
    • tests/ref/fate/samplefmt

  3. Change #280900

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:24
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 90a20fd15b6aa9336bab7dec8e2d18b70703786d

    Comments

    avformat: set the sample format of DSD streams
The DSF, DSDIFF and WavPack demuxers know that a stream carries DSD, so
report AV_SAMPLE_FMT_DSD as its sample format.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavformat/dsfdec.c
    • libavformat/iff.c
    • libavformat/wvdec.c

  4. Change #280901

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:24
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 72b81d5b64c13e36ea9a87f8e096616a20944098

    Comments

    swresample: support AV_SAMPLE_FMT_DSD input
Convert DSD to PCM with the same 96-tap symmetric lowpass filter the
libavcodec DSD decoders use. The implementation is copied from
libavcodec/dsd.c, minus the LSBF table variants, since AV_SAMPLE_FMT_DSD
is defined as MSBF.

The libavcodec copy will be removed in future commits as decoders are
transitioned to SWR.

The conversion is stateful (per-channel filter history), so the audio
conversion functions gain a per-channel state argument.

DSD silence is the 0x69 bit pattern instead of the 0x80 used for PCM.

Only a direct DSD to float conversion function is provided. For format
pairs without a direct conversion function (e.g. DSD to s16 with equal
rates), swr_init() now falls back from the full_convert shortcut to
the generic two-stage path instead of failing, and reports unsupported
conversions with a proper error message instead of ENOMEM.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libswresample/Makefile
    • libswresample/audioconvert.c
    • libswresample/audioconvert.h
    • libswresample/dsd2pcm.c
    • libswresample/dsd2pcm.h
    • libswresample/reverse.c
    • libswresample/swresample.c
    • libswresample/version.h
    • tests/ref/fate/source

  5. Change #280902

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:24
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 6acb801dcd9c203032af077713d07721898aa3c0

    Comments

    avcodec/dsddec: support raw DSD output
Output the raw bitstream as AV_SAMPLE_FMT_DSD when requested. The
default float PCM output is unchanged.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dsddec.c
    • libavcodec/version.h

  6. Change #280903

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 1853c91628442ccca2374c3a690056dbfb379061

    Comments

    avcodec/dstdec: support raw DSD output
Output the losslessly decompressed bitstream as AV_SAMPLE_FMT_DSD. The
default float output is unchanged.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dstdec.c
    • libavcodec/version.h

  7. Change #280904

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 078224ae98dfba868bccb0e8726a9ed4ef278573

    Comments

    avcodec/wavpack: support raw DSD output
Output DSD frames as AV_SAMPLE_FMT_DSD. The default float output is
unchanged.

Raw frames carry no state between each other, so this mode does not
allocate the shared DSD context.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/version.h
    • libavcodec/wavpack.c

  8. Change #280905

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 5bd86141b64981bc049f8244863c777b27490961

    Comments

    avcodec/dsd: add helper for DSD to PCM conversion via libswresample
libswresample now supports AV_SAMPLE_FMT_DSD input using the same
conversion as the local dsd2pcm implementation. This will be used in
following commits and replace dsd2pcm impl.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • configure
    • libavcodec/dsd.c
    • libavcodec/dsd.h

  9. Change #280906

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 8d07c6698062997736b51e0daf2fc497921dd345

    Comments

    avcodec/dsddec: use libswresample for the PCM output
Repack the input to raw DSD unconditionally and convert it to the
default AV_SAMPLE_FMT_FLTP output with libswresample instead of the
local dsd2pcm copy.

If libswresample is disabled, the PCM output is not offered and the
decoder outputs only AV_SAMPLE_FMT_DSD.

The per-channel slice threading of the float conversion is gone. The
conversion now runs in a single swr_convert() call.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dsddec.c

  10. Change #280907

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 2c7e3ed47ac26f3b6becb14ffddfa9ccf90d82ff

    Comments

    avcodec/dstdec: use libswresample for the PCM output
Decode into packed DSD bytes unconditionally and convert to the default
AV_SAMPLE_FMT_FLT output with libswresample instead of the local
dsd2pcm copy.

If libswresample is disabled, the PCM output is not offered and the
decoder outputs only AV_SAMPLE_FMT_DSD.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dstdec.c
    • tests/fate/audio.mak

  11. Change #280908

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:25
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision b130b306f4d07acd5efabe9a9d92f6b471df019e

    Comments

    avcodec/wavpack: use libswresample for the PCM output
Unpack DSD frames into an interleaved scratch buffer and convert to
the default AV_SAMPLE_FMT_FLTP output with libswresample instead of
the in-place local dsd2pcm conversion.

The SwrContext carries the DSD to PCM filter state across frames and
replaces the DSDContext array as the RefStruct-shared state for frame
threading.

If libswresample is disabled, the PCM output is not offered and the
decoder outputs only AV_SAMPLE_FMT_DSD. The per-channel slice
threading of the float conversion is gone, the conversion now runs in
a single swr_convert() call.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/wavpack.c
    • tests/fate/wavpack.mak

  12. Change #280909

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:26
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision 62156c59203873b985bf666ebc25bac74ab24d29

    Comments

    avcodec/dsd: remove unused dsd2pcm implementation
All decoders convert their DSD output to PCM with libswresample now.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dsd.c
    • libavcodec/dsd.h

  13. Change #280910

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:26
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision baa01b0e748b8e770f070b493c98cdffcf832b7c

    Comments

    avcodec: add a raw DSD encoder
AV_SAMPLE_FMT_DSD frames are already the bitstream, so the dsd_msbf
"encoder" is a plain copy of the samples into packets.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • Changelog
    • doc/general_contents.texi
    • libavcodec/Makefile
    • libavcodec/allcodecs.c
    • libavcodec/dsdenc.c
    • libavcodec/version.h

  14. Change #280911

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:26
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision dd48d68d51d5db19105ce946f2049a74897ef221

    Comments

    tests/fate: cover raw DSD decoder output
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • tests/fate/audio.mak
    • tests/fate/wavpack.mak
    • tests/ref/fate/dsf-dst-dsd
    • tests/ref/fate/wavpack-dsd
    • tests/ref/fate/wavpack-dsd-pcm

  15. Change #280912

    Category ffmpeg
    Changed by Kacper Michajłow <kasper93ohnoyoudont@gmail.com>
    Changed at Sun 06 Sep 2026 00:48:26
    Repository https://git.ffmpeg.org/ffmpeg.git
    Project ffmpeg
    Branch master
    Revision c6e28adabac09745f3406ffbf3833b8ac7f3f46a

    Comments

    avcodec: deprecate the PCM output of the DSD decoders
Now that AV_SAMPLE_FMT_DSD exists, the decoders will prefer this output
and PCM conversion is deprecated and will be removed. Users can use
libswresample directly.

Signed-off-by: Kacper Michajłow <kasper93@gmail.com>

    Changed files

    • libavcodec/dsd.c
    • libavcodec/dsd.h
    • libavcodec/dsddec.c
    • libavcodec/dstdec.c
    • libavcodec/version.h
    • libavcodec/version_major.h
    • libavcodec/wavpack.c
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