Artem Titov 3ecec176a8 Extract third party part of g711 codec into separate target ...

Bug: webrtc:8366
Change-Id: I34c7ea707213e0c1a50826896da01f70c072eae5
Reviewed-on: https://webrtc-review.googlesource.com/84741
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23724}

2018-06-25 11:26:59 +00:00

..

acm2

Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"

2018-06-21 12:36:44 +00:00

audio_network_adaptor

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

codecs

Extract third party part of g711 codec into separate target

2018-06-25 11:26:59 +00:00

include

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

neteq

Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.

2018-06-21 14:23:53 +00:00

test

Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"

2018-06-21 12:36:44 +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

Extract third party part of g711 codec into separate target

2018-06-25 11:26:59 +00:00

DEPS

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

OWNERS

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