commit 10403ae87c73e99c5d2b6be74709263bc4e1f32d
author Jakob Ivarsson <jakobi@webrtc.org> Tue Nov 27 14:45:20 2018
committer Commit Bot <commit-bot@chromium.org> Tue Nov 27 19:49:48 2018
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}

api/audio_options.h

diff --git a/api/audio_options.h b/api/audio_options.h
index 8ae8319..c2d1f44 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -54,6 +54,8 @@
   absl::optional<int> audio_jitter_buffer_max_packets;
   // Audio receiver jitter buffer (NetEq) fast accelerate mode.
   absl::optional<bool> audio_jitter_buffer_fast_accelerate;
+  // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
+  absl::optional<int> audio_jitter_buffer_min_delay_ms;
   // Audio processing to detect typing.
   absl::optional<bool> typing_detection;
   absl::optional<bool> experimental_agc;