blob: da8fb58895c3d90e6b5287406a51888af94175d0 [file] [log] [blame]
/*
* Copyright (c) 2017 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_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
#include <memory>
#include <utility>
#include <vector>
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/aec_dump/write_to_file_task.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "rtc_base/checks.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/logging.h"
// Files generated at build-time by the protobuf compiler.
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
#else
#include "modules/audio_processing/debug.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
namespace webrtc {
class CaptureStreamInfo {
public:
explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task);
~CaptureStreamInfo();
void AddInput(const AudioFrameView<const float>& src);
void AddOutput(const AudioFrameView<const float>& src);
void AddInput(const AudioFrame& frame);
void AddOutput(const AudioFrame& frame);
void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
std::unique_ptr<WriteToFileTask> GetTask() {
RTC_DCHECK(task_);
return std::move(task_);
}
void SetTask(std::unique_ptr<WriteToFileTask> task) {
RTC_DCHECK(!task_);
RTC_DCHECK(task);
task_ = std::move(task);
task_->GetEvent()->set_type(audioproc::Event::STREAM);
}
private:
std::unique_ptr<WriteToFileTask> task_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_