refactor:calculate encoder pts based on decoder pts #348
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src/AudioPipe.cpp
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| //Get reference to first buffer in queue | ||
| auto& frontAudioBuffer = queue.front(); | ||
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| if (frontAudioBuffer->GetTimestamp() > 2 * samples + prevPTS) |
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shouldn't be the timestamp threshold for detecting a discontinuity be a static constant and not depending on the samples?
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mathematically, i think it's more accurate to use num of samples for discontinuity detection instead of use constant, for the following reasons:
- the discontinuity may happen between rateBlockBuffer and frontAudioBuffer. If constant is used, we allow more tolerance for this case.
- Not sure if it's valid case, is it possible that connection exists all the time, and user switch codecs in between? I think use num of samples is more safe for this case as well.
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yes, I meant a timestamp threshold converted to the num samples based on the output clock rate.
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what I meant is that we are detecting the disconituity based on the leftovers of the rateBlockBuffer which is variable
src/AudioPipe.cpp
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| //Get reference to first buffer in queue | ||
| auto& frontAudioBuffer = queue.front(); | ||
|
|
||
| if (frontAudioBuffer->GetTimestamp() > 2 * samples + prevPTS) |
There was a problem hiding this comment.
what I meant is that we are detecting the disconituity based on the leftovers of the rateBlockBuffer which is variable
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