webrtc/api/audio_codecs/opus
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
..
audio_decoder_opus.cc [clang-tidy] Apply performance-move-const-arg fixes. 2019-02-01 15:02:36 +00:00
audio_decoder_opus.h Audio codecs API: Remove some weasel words in the docs 2018-10-22 08:52:15 +00:00
audio_encoder_opus.cc Replace rtc::Optional with absl::optional in api 2018-06-21 12:50:03 +00:00
audio_encoder_opus.h Audio codecs API: Remove some weasel words in the docs 2018-10-22 08:52:15 +00:00
audio_encoder_opus_config.cc Tests for multi-stream Opus. 2019-02-18 17:09:59 +00:00
audio_encoder_opus_config.h Audio codecs API: Remove some weasel words in the docs 2018-10-22 08:52:15 +00:00
BUILD.gn Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. 2018-10-23 09:24:15 +00:00