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

..

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