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`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.
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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 <noreply@anthropic.com>
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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.
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Compute the in-file stream index for external subtitles instead of
hardcoding -map 1:0. For single-stream files (SRT/ASS/VTT) the index
is always 0, preserving existing behavior. For multi-stream containers
like MKS, the correct track is selected by counting sibling streams
that share the same Path.
Add unit tests for GetMapArgs covering internal subs, external SRT,
multiple external files, and multi-stream MKS containers.
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