8b3db59b6e
This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa. Reason for revert: Order of initialization of global static strings. Original change's description: > Reland of https://webrtc-review.googlesource.com/c/src/+/114883 > > The difference to the original is new bitexactness strings AND > global static file string constants. 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. > > Bug: webrtc:8649 > Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2 > Reviewed-on: https://webrtc-review.googlesource.com/c/123387 > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26774} TBR=aleloi@webrtc.org,ossu@webrtc.org Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8649 Reviewed-on: https://webrtc-review.googlesource.com/c/123580 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26777}
44 lines
1.3 KiB
C++
44 lines
1.3 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
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#include <string>
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#include "common_audio/wav_file.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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namespace test {
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class OutputWavFile : public AudioSink {
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public:
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// Creates an OutputWavFile, opening a file named |file_name| for writing.
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// The output file is a PCM encoded wav file.
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OutputWavFile(const std::string& file_name, int sample_rate_hz)
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: wav_writer_(file_name, sample_rate_hz, 1) {}
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bool WriteArray(const int16_t* audio, size_t num_samples) override {
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wav_writer_.WriteSamples(audio, num_samples);
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return true;
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}
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private:
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WavWriter wav_writer_;
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RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
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