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Propagate base minimum delay to audio_receiver_stream

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Bug: webrtc:10287
Change-Id: Id7914976ef5b7eb708802119932b554d9ce4879e
Reviewed-on: https://webrtc-review.googlesource.com/c/121563
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26563}
This commit is contained in:
Ruslan Burakov
2019-02-06 09:45:56 +01:00
committed by Commit Bot
parent 9ce800d6d1
commit 3b50f9f9ce
9 changed files with 71 additions and 0 deletions
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9
modules/audio_coding/include/audio_coding_module.h
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@ -275,6 +275,15 @@ class AudioCodingModule {
//
virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
// Sets a base minimum for the playout delay. Base minimum delay sets lower
// bound minimum delay value which is set via SetMinimumPlayoutDelay.
//
// Returns true if value was successfully set, false overwise.
virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
// Returns current value of base minimum delay in milliseconds.
virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutTimestamp()
// The send timestamp of an RTP packet is associated with the decoded