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/neteq_impl_unittest.cc

diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index b772dfa..0e087c8 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -92,7 +92,8 @@
 
     if (use_mock_delay_manager_) {
       std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
-          config_.max_packets_in_buffer, delay_peak_detector_, tick_timer_));
+          config_.max_packets_in_buffer, config_.min_delay_ms,
+          delay_peak_detector_, tick_timer_));
       mock_delay_manager_ = mock.get();
       EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
       deps.delay_manager = std::move(mock);