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The difference to the original is new bitexactness strings. The
reason for reland is breaking downstream projects.
Original CL description:
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.
TBR=ossu@webrtc.org
Bug: webrtc:8649
Change-Id: I6261b18c69fd666d43ab34ed8f1bc9d5cc82b21f
Reviewed-on: https://webrtc-review.googlesource.com/c/123882
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26809}
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|---|---|---|
| .. | ||
| acm_receive_test.cc | ||
| acm_receive_test.h | ||
| acm_receiver.cc | ||
| acm_receiver.h | ||
| acm_receiver_unittest.cc | ||
| acm_resampler.cc | ||
| acm_resampler.h | ||
| acm_send_test.cc | ||
| acm_send_test.h | ||
| audio_coding_module.cc | ||
| audio_coding_module_unittest.cc | ||
| call_statistics.cc | ||
| call_statistics.h | ||
| call_statistics_unittest.cc | ||