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Add PeerConnection option to configure minimum audio jitter buffer delay.

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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
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4
modules/audio_coding/neteq/delay_manager_unittest.cc
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@ -27,6 +27,7 @@ using ::testing::_;
class DelayManagerTest : public ::testing::Test {
protected:
static const int kMaxNumberOfPackets = 240;
static const int kMinDelayMs = 0;
static const int kTimeStepMs = 10;
static const int kFs = 8000;
static const int kFrameSizeMs = 20;
@ -56,7 +57,8 @@ void DelayManagerTest::SetUp() {
void DelayManagerTest::RecreateDelayManager() {
EXPECT_CALL(detector_, Reset()).Times(1);
dm_.reset(new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_));
dm_.reset(new DelayManager(kMaxNumberOfPackets, kMinDelayMs, &detector_,
&tick_timer_));
}
void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {