Jared Siskin f2ed401679 Fix unscaled timestamps passed to nack_tracker ... If timestamp_scaler_ is used, then rtp_header.timestamp, passed to UpdateLastDecodedPacket, will advance at a different rate than the scaled timestamp packet->timestamp, passed to UpdateLastDecodedPacket. NackTracker::EstimateTimestamp uses timestamp_last_received_rtp_, and NackTracker::TimeToPlay uses timestamp_last_decoded_rtp_. This difference causes TimeToPlay to continuously increase to huge values, so that every missing packet will be returned from GetNackList, even if RTT > real time to play. Change-Id: Ie6ca347972edea98a202c9cdd26c6ab3f45a73c4 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222841 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34361}		2021-06-23 08:41:50 +00:00
..
acm2	Avoid using legacy rtp parser in neteq test::Packet	2021-06-17 08:38:14 +00:00
audio_network_adaptor	Remove check for WebRTC-SendSideBwe-WithOverhead in bitrate controller.	2020-11-09 23:11:26 +00:00
codecs	Not dropping the refresh DTX packets but substituting them by 1 byte packets.	2021-05-11 19:47:34 +00:00
g3doc	Fix documentation owners formating	2021-04-23 11:07:28 +00:00
include	Replace RTC_DEPRECATED with ABSL_DEPRECATED	2021-02-22 12:53:23 +00:00
neteq	Fix unscaled timestamps passed to nack_tracker	2021-06-23 08:41:50 +00:00
test	Revert "opus: take SILK vad result into account for voice detection"	2020-11-04 07:29:48 +00:00
audio_coding.gni	build: remove WEBRTC_CODEC_RED	2020-05-26 11:01:26 +00:00
BUILD.gn	Avoid using legacy rtp parser in neteq test::Packet	2021-06-17 08:38:14 +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