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Use the sample rate as timescale for the audio encoder ring buffer offset - #489

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eerimoq:mainfrom
gabbelitoV2:use-the-sample-rate-as-timescale-for-the-audio-encoder-ring
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gabbelitoV2 wants to merge 1 commit into
eerimoq:mainfrom
gabbelitoV2:use-the-sample-rate-as-timescale-for-the-audio-encoder-ring

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The offset of an output buffer into the current input buffer is a sample count, but it was turned into a CMTime with the input buffer's presentation timescale (the host clock for capture buffers), i.e. nanoseconds. That is harmless when the output frame size divides the input buffer size (AAC with 1024 frame buffers) but gives a sawtooth in the timestamps for Opus (960 frames) and for sources with other buffer sizes.

Not built locally, relying on CI for the build.

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