mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-18 08:07:56 +01:00
65438812ba
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}
45 lines
1.4 KiB
C++
45 lines
1.4 KiB
C++
/*
|
|
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
|
|
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
|
|
|
|
#include <string>
|
|
|
|
#include "common_audio/wav_file.h"
|
|
#include "modules/audio_coding/neteq/tools/audio_sink.h"
|
|
#include "rtc_base/constructor_magic.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
class OutputWavFile : public AudioSink {
|
|
public:
|
|
// Creates an OutputWavFile, opening a file named |file_name| for writing.
|
|
// The output file is a PCM encoded wav file.
|
|
OutputWavFile(const std::string& file_name,
|
|
int sample_rate_hz,
|
|
int num_channels = 1)
|
|
: wav_writer_(file_name, sample_rate_hz, num_channels) {}
|
|
|
|
bool WriteArray(const int16_t* audio, size_t num_samples) override {
|
|
wav_writer_.WriteSamples(audio, num_samples);
|
|
return true;
|
|
}
|
|
|
|
private:
|
|
WavWriter wav_writer_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(OutputWavFile);
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
|