Henrik Lundin 4e268edb53 Add two new RTP header extensions to neteq_rtpplay ... This change adds flags and default values for two more RTP header extensions: VideoContentType and VideoTiming. This will silence a number of annoying warnings when running with application logs. Bug: none Change-Id: I9bb01ea2519813d3c47553ecff384141fbede23e Reviewed-on: https://webrtc-review.googlesource.com/75300 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23178}		2018-05-08 16:05:12 +00:00
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acm2	ACM: Properly initialize last_audio_buffer_ array	2018-05-08 11:40:04 +00:00
audio_network_adaptor	Replacing rtc::TimeDelta with webrtc::TimeDelta.	2018-05-08 13:22:53 +00:00
codecs	Replacing rtc::TimeDelta with webrtc::TimeDelta.	2018-05-08 13:22:53 +00:00
include	Remove dependencies on modules:module_api from AudioProcessing.	2018-04-12 22:05:27 +00:00
neteq	Add two new RTP header extensions to neteq_rtpplay	2018-05-08 16:05:12 +00:00
test	Remove incompatiblities with absl::optional in audio_coding	2018-04-17 12:05:13 +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	Break out the part of the iSAC codec that's used for Voice Activity Detection	2018-05-04 08:53:34 +00:00
DEPS	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
OWNERS	Moving src/webrtc into src/.	2017-09-15 04:25:06 +00:00