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webrtc/modules/audio_coding
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Evan Shrubsole c3891e3a4e [Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper 
Bug: webrtc:13982
Change-Id: I35c08921c8c1be31f0de4bd81f918250bee25313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288961
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39052}
2023-01-10 09:53:17 +00:00
..
acm2 Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_network_adaptor Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
codecs Stop setting OPUS_SIGNAL_VOICE when DTX is enabled. 2022-12-20 11:06:48 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq [Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper 2023-01-10 09:53:17 +00:00
test Stop setting OPUS_SIGNAL_VOICE when DTX is enabled. 2022-12-20 11:06:48 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn [Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper 2023-01-10 09:53:17 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00