| Commit message (Collapse) | Author | Age | Files | Lines |
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options if global_verbosity_level >= 3
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If dither is "auto", only enable dither if source depth > dest depth
Lossy codecs do not have a fixed bit depth, so we treat them as >= 24
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Newest versions appear to have a "common.h" somewhere that is
interfering with libhb/common.h.
move headers into "handbrake" subdirectory
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Rename hb.h to handbrake.h
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Fixes https://github.com/HandBrake/HandBrake/issues/2124
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Use 5.1 back for AAC because 5.1 side uses a not-so-universally
supported feature to signal the non-standard layout
Fixes https://github.com/HandBrake/HandBrake/issues/1744
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option stereo_mode" message in the log. This option is changed in ffmpeg.
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Eliminates deprecation warnings for avresample
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avcodec_close(..)
Hence rename hb_avcodec_close -> hb_avcodec_free_context and pass the required ptr-ptr.
avcodec_free_context(..) ensures releasing of all resources attached to the context.
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Found via `codespell -q 3 --skip="./gtk/po`
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Fixes https://github.com/HandBrake/HandBrake/issues/815
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On windows builds, there is an upstream bug in the lame
encoder that causes an extra output packet that has the same
timestamp as the second to last packet. This causes an error
during muxing. Drop the packet with the duplicate timestamp.
Fixes https://github.com/HandBrake/HandBrake/issues/726
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Setting the time_base to 90khz reduces rounding errors when converting
back and forth from the default which was sample_rate.
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When exactly input_samples were left in the input buffer, we were
dropping them when they could be encoded.
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This prevents libav from adding an mp4 edit list entry that causes a
properly functioning player to drop the first couple of audio frames.
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The series of commits starting with Tim's
"libav: remove our custom QSV wrapper."
Closes #46, closes #276, and closes #416.
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avcodec_encode_audio2 is deprecated, use avcodec_send_frame and
avcodec_receive_packet
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Remove:
hb_audio_samplerate_get_best()
Add:
hb_audio_samplerate_is_supported()
hb_audio_samplerate_find_closest()
hb_audio_samplerate_get_sr_shift()
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Fixes https://forum.handbrake.fr/viewtopic.php?f=12&t=33345
Some players expect each packet to start on an mp3 frame header. Our
mp3lame encoder did not ensure this and resulted in failure to play
audio on these players.
libav already has the necessary code to parse headers and
accumulate a full frame of data, so use it.
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This brings together several independent implementations of a simple
buffer list manager.
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@7157 b64f7644-9d1e-0410-96f1-a4d463321fa5
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... instead of a 0 length buffer.
This fixes this issue:
https://forum.handbrake.fr/viewtopic.php?f=12&t=31959
Theora can create 0 length output. These 0 length frames indicate
duplicate frames. So we can't use 0 length buffers to indicate the end
of the stream.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@7143 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Dolby Digital Plus (E-AC-3), Dolby TrueHD and FLAC can now be passed through without re-encoding. They aren't covered by Auto Passthru yet, however.
In addition, encoding to E-AC-3 is now possible.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6908 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6852 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@6042 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Filters were leaking buffers when a job is cancelled.
decavcodec could leak when job cancelled.
decavcodec leaked audio extradata in BSInfo
encavcodec and encavcodecaudio leaked AVCodecContext
sync leaked subtitle_sanitizer_t data
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5853 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5620 b64f7644-9d1e-0410-96f1-a4d463321fa5
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(currently, 16 KiB).
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5335 b64f7644-9d1e-0410-96f1-a4d463321fa5
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A minimum output buffer size of 8KB is needed by libav
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5329 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5318 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5292 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5290 b64f7644-9d1e-0410-96f1-a4d463321fa5
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configure --enable-fdk-aac to enable this encoder
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5287 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Works with encoders that accept 16-bit signed integers as input (currently, only ffflac).
When supported, the default method is standard triangular.
CLI users can request a specific dither algorithm via the --adither option.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5241 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Works via a dedicated encoder: "FLAC (24-bit)", "ffflac24".
Users who prefer 16-bit can use the previous FLAC encoder: "FLAC (ffmpeg)", "ffflac".
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5134 b64f7644-9d1e-0410-96f1-a4d463321fa5
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Miscellaneous bugfixes and improvements.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5124 b64f7644-9d1e-0410-96f1-a4d463321fa5
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- encavcodecaudio: use libavresample directly (instead of via the hb_audio_resample wrapper), and add support for planar output
- hb_audio_resample: add support for planar input
- hb_audio_remap: add support for planar input
- deca52: output planar float samples (no re-interleaving)
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5073 b64f7644-9d1e-0410-96f1-a4d463321fa5
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git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5042 b64f7644-9d1e-0410-96f1-a4d463321fa5
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- allow setting a sample_fmt other than AV_SAMPLE_FMT_FLT
- fall back on planar/packed variant of the requested format if available
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5019 b64f7644-9d1e-0410-96f1-a4d463321fa5
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