2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883

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}

modules/audio_coding/neteq/tools/input_audio_file.cc

@@ -18,9 +18,11 @@ namespace test {
 InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end)
     : loop_at_end_(loop_at_end) {
   fp_ = fopen(file_name.c_str(), "rb");
+  RTC_DCHECK(fp_) << file_name << " could not be opened.";
 }
 
 InputAudioFile::~InputAudioFile() {
+  RTC_DCHECK(fp_);
   fclose(fp_);
 }