webrtc/modules/audio_coding
Ruslan Burakov edbea46295 Allow to change base minimum delay on NetEq.
This is first step to allow to set latency
from client code in Chromium.
Existing minimum latency hasn't been used because it can clash
with video syncronization code.

Bug: webrtc:10287
Change-Id: Ia38906506069a1abfa01698dc62df283fc15cfbc
Reviewed-on: https://webrtc-review.googlesource.com/c/121423
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#26536}
2019-02-04 17:19:55 +00:00
..
acm2 Removes all const Clock*. 2019-01-30 13:03:37 +00:00
audio_network_adaptor [clang-tidy] Apply performance-for-range-copy fixes. 2019-01-28 09:53:50 +00:00
codecs [clang-tidy] Apply performance-move-const-arg fixes. 2019-02-01 15:02:36 +00:00
include Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats(). 2019-01-16 11:44:10 +00:00
neteq Allow to change base minimum delay on NetEq. 2019-02-04 17:19:55 +00:00
test (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +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 Remove rtc_base/scoped_ref_ptr.h. 2019-01-25 20:29:58 +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