Ruslan Burakov 8af8896596 Expose jitter buffer flushes metric in new getStats api. ...

Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}

2018-11-23 11:41:43 +00:00

..

acm2

Expose jitter buffer flushes metric in new getStats api.

2018-11-23 11:41:43 +00:00

audio_network_adaptor

Removing ANA enabling field trials.

2018-11-22 22:26:28 +00:00

codecs

Adds trial to use link capacity estimate in Opus encoder.

2018-11-22 09:21:12 +00:00

include

Expose jitter buffer flushes metric in new getStats api.

2018-11-23 11:41:43 +00:00

neteq

Delete unused method NetEq::PacketBufferStatistics

2018-11-23 09:39:32 +00:00

test

Hide the AudioEncoderCng class behind a create function

2018-11-02 13:00:05 +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

Delete unused NetEq Rtcp stats.

2018-11-22 08:00:54 +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