webrtc/modules/audio_coding
Chen Xing e08648dc70 Add AbsoluteCaptureTime to RtpPacketInfo.
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.

Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
2019-08-07 10:12:56 +00:00
..
acm2 Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_ 2019-08-07 09:53:22 +00:00
audio_network_adaptor Format almost everything. 2019-07-08 13:45:15 +00:00
codecs Remove empty OWNERS file. 2019-07-22 11:55:23 +00:00
include Delete obsolete method AudioCodingModule::SetBitRate 2019-08-07 08:37:25 +00:00
neteq Add AbsoluteCaptureTime to RtpPacketInfo. 2019-08-07 10:12:56 +00:00
test Format almost everything. 2019-07-08 13:45:15 +00:00
audio_coding.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
BUILD.gn Switch neteq_rtpplay into an executable. 2019-07-25 08:45:21 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00