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<title>jellyfin/MediaBrowser.Controller/MediaEncoding/EncodingHelper.cs, branch v12.0-rc5</title>
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</subtitle>
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<updated>2026-08-04T17:52:29Z</updated>
<entry>
<title>Fix PCM audio transcoding to wav returning HTTP 500 and headerless output</title>
<updated>2026-08-04T17:52:29Z</updated>
<author>
<name>vdatanet</name>
<email>joan@vdata.net</email>
</author>
<published>2026-08-04T17:52:29Z</published>
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<id>urn:sha1:3578e9a332c64d1aeef6be5e4f2c1a8d271169a8</id>
<content type='text'>
`GetProgressiveAudioFullCommandLine` forced the raw PCM muxer and a bogus
sample rate whenever the audio encoder was `pcm_*`, regardless of the
container the client asked for. Two separate failures came out of it:

- `-ar ` + `state.BaseRequest.AudioBitRate` used a *bitrate* as a *sample
  rate*, and `AudioBitRate` is optional. When it is absent the argument
  degrades to a bare `-ar`, ffmpeg aborts with `Expected number for ar but
  found: -ar` / `Error opening output files: Invalid argument` (exit 234)
  and the request fails with HTTP 500. Every `GET /Audio/{id}/stream.wav`
  that does not carry an explicit `AudioBitRate` hits this.
  The sample rate was already being set correctly a few lines below from
  `OutputAudioSampleRate`, so the line is dropped rather than repaired.

- `-f s16le` overrode the muxer even for a real container. A request that
  did supply a bitrate (`/Audio/{id}/universal` passes
  `MaxStreamingBitrate`) survived the first bug but produced raw headerless
  samples served under an `audio/wav` content type, so clients saw a body
  with no RIFF header. The raw muxer is now only forced when the requested
  container is actually raw PCM, which keeps the I2S/MCU route from #10321
  working.

Also drop the `containerInternal = ".pcm"` assignment in
`StreamingHelpers.GetStreamingState`: it is written after
`state.OutputContainer` has already been read from the same variable and is
never read again, so it has no effect and only obscures where the output
container comes from.

Verified against ffmpeg 8.1.2 with a 96 kHz FLAC source: before, the wav
command line exits 234; after, it produces a valid `RIFF/WAVE` 48 kHz stereo
`pcm_s16le` file, while the raw `.pcm` route still yields exactly
2 s x 48000 x 2ch x 2 B = 384000 bytes of headerless samples.
</content>
</entry>
<entry>
<title>Fix format negotiation in hybrid SW decode and CUDA tonemap pipeline</title>
<updated>2026-07-15T05:54:45Z</updated>
<author>
<name>nyanmisaka</name>
<email>nst799610810@gmail.com</email>
</author>
<published>2026-07-15T05:47:48Z</published>
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<id>urn:sha1:4503ad295ca69de46c38fbdbebd2a61cc9961aa9</id>
<content type='text'>
The CUDA hwcontext in FFmpeg 8.1 has added support for 10bit
fully-planar formats, but few CUDA filters support them.

Signed-off-by: nyanmisaka &lt;nst799610810@gmail.com&gt;
</content>
</entry>
<entry>
<title>Fix audio sample rate forced to 48 kHz for non-Opus codecs</title>
<updated>2026-06-21T07:43:31Z</updated>
<author>
<name>danne</name>
<email>danne.e85@gmail.com</email>
</author>
<published>2026-06-13T20:44:44Z</published>
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<id>urn:sha1:e4383493a96da86c99516bfdf69ecd609c2dfec2</id>
<content type='text'>
GetProgressiveAudioFullCommandLine applied the libopus-only sample rate
quantization to every codec except Opus, inverting the intended guard.
A requested rate such as 44100 Hz was therefore snapped to 48000 Hz for
AAC/MP3/FLAC, while Opus (which actually requires the quantization) was
skipped entirely.

Apply the quantization only when the output codec is Opus, and pass the
requested sample rate through unchanged for all other codecs.

Fixes #17026

Co-Authored-By: Claude Opus 4.8 &lt;noreply@anthropic.com&gt;
</content>
</entry>
<entry>
<title>Extend TranscodingReason reporting</title>
<updated>2026-06-15T07:29:24Z</updated>
<author>
<name>Shadowghost</name>
<email>Ghost_of_Stone@web.de</email>
</author>
<published>2026-06-15T07:29:24Z</published>
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<id>urn:sha1:1dd5a850807bc4269fce66d2d4191e67027285c8</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Fix duplicate trickplay images for interlaced videos</title>
<updated>2026-06-09T12:18:17Z</updated>
<author>
<name>nyanmisaka</name>
<email>nst799610810@gmail.com</email>
</author>
<published>2026-06-09T12:18:17Z</published>
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<id>urn:sha1:c693047467e2f454b4ceae7269ceb97df5e71bba</id>
<content type='text'>
Signed-off-by: nyanmisaka &lt;nst799610810@gmail.com&gt;
</content>
</entry>
<entry>
<title>Fix A/V desync when resuming HLS with video transcode + audio copy (#16580)</title>
<updated>2026-05-28T17:26:28Z</updated>
<author>
<name>Piotr Niełacny</name>
<email>piotr.nielacny@gmail.com</email>
</author>
<published>2026-05-28T17:26:28Z</published>
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<id>urn:sha1:8d544e48424d9ddbb1f97d354ed6e6a3f749cbfb</id>
<content type='text'>
Fix A/V desync when resuming HLS with video transcode + audio copy</content>
</entry>
<entry>
<title>Fix inconsistent extradata generated by hevc_vaapi on AMD driver</title>
<updated>2026-05-23T19:34:54Z</updated>
<author>
<name>nyanmisaka</name>
<email>nst799610810@gmail.com</email>
</author>
<published>2026-05-23T19:31:56Z</published>
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<id>urn:sha1:69b0b63a950a7a076a682fb81d3a0ff32239c1a0</id>
<content type='text'>
This change is required for upstream ffmpeg 8+, because its
mp4 muxer will drop in-band PS when using codec tag hvc1.

Signed-off-by: nyanmisaka &lt;nst799610810@gmail.com&gt;
</content>
</entry>
<entry>
<title>Fix StyleCop pattern matching whitespace false-positive</title>
<updated>2026-05-21T18:48:43Z</updated>
<author>
<name>jakobkukla</name>
<email>jakob.kukla@gmail.com</email>
</author>
<published>2025-10-27T15:51:34Z</published>
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<id>urn:sha1:eb2cef1b7eaac440f3cc1d3ee2f36a32f72ae37a</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Run tree-wide dotnet format</title>
<updated>2026-05-21T18:48:41Z</updated>
<author>
<name>jakobkukla</name>
<email>jakob.kukla@gmail.com</email>
</author>
<published>2026-04-16T09:38:01Z</published>
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<id>urn:sha1:37350282cc18a977162dcf14fe9b6068d29d6edf</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Normalize VobSub .sub to .idx for embedding, add EncodingHelper tests</title>
<updated>2026-05-19T11:03:07Z</updated>
<author>
<name>Piotr Niełacny</name>
<email>piotr.nielacny@gmail.com</email>
</author>
<published>2026-03-24T13:26:25Z</published>
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<id>urn:sha1:405d987557b2638afc89edf3dff20360e39cb09a</id>
<content type='text'>
Move the .sub to .idx path normalization outside the burn-in check so
it applies to subtitle embedding as well. ffmpeg requires the .idx file
to read VobSub subtitles.

Add unit tests for GetMapArgs and GetInputArgument covering internal
subs, external SRT, multi-file SRT, multi-stream MKS containers, and
VobSub .sub/.idx path normalization.
</content>
</entry>
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