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Add latency to remote source api.

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Latency corresponds to base minimum delay on NetEq.

Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}
This commit is contained in:
Ruslan Burakov
2019-02-16 02:07:05 +01:00
committed by Commit Bot
parent 86f09741a7
commit 7ea460593c
13 changed files with 354 additions and 4 deletions
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6
api/media_stream_interface.h
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@ -201,6 +201,12 @@ class AudioSourceInterface : public MediaSourceInterface {
// be applied in the track in a way that does not affect clones of the track.
virtual void SetVolume(double volume) {}
// Sets the minimum latency of the remote source until audio playout. Actual
// observered latency may differ depending on the source. |latency| is in the
// range of [0.0, 10.0] seconds.
virtual void SetLatency(double latency) {}
virtual double GetLatency() const;
// Registers/unregisters observers to the audio source.
virtual void RegisterAudioObserver(AudioObserver* observer) {}
virtual void UnregisterAudioObserver(AudioObserver* observer) {}