webrtc/modules/audio_coding/codecs
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
..
cng (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
g711 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
g722 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
ilbc Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00
isac Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00
opus Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
pcm16b (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
red (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
tools (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_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00
legacy_encoded_audio_frame.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
legacy_encoded_audio_frame.h [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
legacy_encoded_audio_frame_unittest.cc Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00