webrtc/modules/audio_coding
Alex Loiko 9c31ac2323 Tests for multi-stream Opus.
This CL (mainly) adds bit-exactness tests for multi-stream Opus. The
tests are in audio_coding_unittest.cc. Some refactoring of
AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it
possible. A few checks for "channels \in {1, 2}" are replaced with
"channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few
other changes are made to be able to write and read multi-channel WAV
files.

The SDP changes are NOT included; as of this CL there is no way to set
up a multi-stream opus en/de-coder from SDP strings.

Bug: webrtc:8649
Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c
Reviewed-on: https://webrtc-review.googlesource.com/c/114883
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26742}
2019-02-18 17:09:59 +00:00
..
acm2 Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
audio_network_adaptor Remove webrtc::ProtoString. 2019-02-16 11:11:45 +00:00
codecs Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
include Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
neteq Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
test Update remaining audio test code to not use WebRtcRTPHeader. 2019-02-18 13:29:35 +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 webrtc::ProtoString. 2019-02-16 11:11:45 +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