Add PeerConnection option to configure minimum audio jitter buffer delay.

Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
This commit is contained in:
Jakob Ivarsson
2018-11-27 15:45:20 +01:00
committed by Commit Bot
parent c7f1a0af92
commit 10403ae87c
18 changed files with 66 additions and 13 deletions

View File

@ -113,6 +113,7 @@ class NetEq {
bool enable_post_decode_vad = false;
size_t max_packets_in_buffer = 50;
int max_delay_ms = 2000;
int min_delay_ms = 0;
bool enable_fast_accelerate = false;
bool enable_muted_state = false;
absl::optional<AudioCodecPairId> codec_pair_id;