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}

audio/channel_receive.h

diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 02d0e4b..9027623 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -135,6 +135,7 @@
     uint32_t remote_ssrc,
     size_t jitter_buffer_max_packets,
     bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms,
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
     absl::optional<AudioCodecPairId> codec_pair_id,
     rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,