commit 10403ae87c73e99c5d2b6be74709263bc4e1f32d
author Jakob Ivarsson <jakobi@webrtc.org> Tue Nov 27 15:45:20 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Tue Nov 27 19:49:48 2018 +0000
tree e21a547bd770eb8c71d3289c2c15761778d12c63
parent c7f1a0af92e10a2f3b0d65b4b7c73001329ee1b5

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}

modules/audio_coding/neteq/include/neteq.h

diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index e1d166c..2820fd8 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -113,6 +113,7 @@
     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;