webrtc_m130/test/testsupport/copy_to_file_audio_capturer.cc
Artem Titov 66a29b9953 Introduce CopyToFileAudioCapturer.
It will be used to dump generated audio from TestAudioDeviceModule into
user defuned file in peer connection level test framework.

Bug: webrtc:10138
Change-Id: I6e3db36aaf1303ab148e8812937c4f9cd1b49315
Reviewed-on: https://webrtc-review.googlesource.com/c/117220
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26267}
2019-01-15 15:06:55 +00:00

47 lines
1.5 KiB
C++

/*
* Copyright (c) 2019 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.
*/
#include "test/testsupport/copy_to_file_audio_capturer.h"
#include <utility>
#include "absl/memory/memory.h"
namespace webrtc {
namespace test {
CopyToFileAudioCapturer::CopyToFileAudioCapturer(
std::unique_ptr<TestAudioDeviceModule::Capturer> delegate,
std::string stream_dump_file_name)
: delegate_(std::move(delegate)),
wav_writer_(absl::make_unique<WavWriter>(std::move(stream_dump_file_name),
delegate_->SamplingFrequency(),
delegate_->NumChannels())) {}
CopyToFileAudioCapturer::~CopyToFileAudioCapturer() = default;
int CopyToFileAudioCapturer::SamplingFrequency() const {
return delegate_->SamplingFrequency();
}
int CopyToFileAudioCapturer::NumChannels() const {
return delegate_->NumChannels();
}
bool CopyToFileAudioCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
bool result = delegate_->Capture(buffer);
if (result) {
wav_writer_->WriteSamples(buffer->data(), buffer->size());
}
return result;
}
} // namespace test
} // namespace webrtc