webrtc/modules/audio_coding/acm2
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
..
acm_receive_test.cc Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_receive_test.h Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
acm_receiver.cc Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_receiver.h Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_receiver_unittest.cc Update ACM to use RTPHeader instead of WebRtcRTPHeader 2019-02-18 08:01:31 +00:00
acm_resampler.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
acm_resampler.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
acm_send_test.cc Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
acm_send_test.h (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_module.cc Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
audio_coding_module_unittest.cc Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
call_statistics.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
call_statistics.h Move NetworkStatistics and AudioDecodingCallStats from common_types.h 2018-11-19 11:55:34 +00:00
call_statistics_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00