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Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine

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The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.

This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.

Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
This commit is contained in:
henrik.lundin
2016-08-22 15:39:53 -07:00
committed by Commit bot
parent e131ea50b4
commit b3f1c5d2fe
8 changed files with 53 additions and 5 deletions
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9
webrtc/modules/audio_coding/include/audio_coding_module.h
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@ -678,6 +678,15 @@ class AudioCodingModule {
//
virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
///////////////////////////////////////////////////////////////////////////
// int FilteredCurrentDelayMs()
// Returns the current total delay from NetEq (packet buffer and sync buffer)
// in ms, with smoothing applied to even out short-time fluctuations due to
// jitter. The packet buffer part of the delay is not updated during DTX/CNG
// periods.
//
virtual int FilteredCurrentDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutData10Ms(
// Get 10 milliseconds of raw audio data for playout, at the given sampling