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/*
* 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 "modules/audio_processing/include/aec_dump.h"
#include "rtc_base/ignore_wundef.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:
CaptureStreamInfo() { CreateNewEvent(); }
CaptureStreamInfo(const CaptureStreamInfo&) = delete;
CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete;
~CaptureStreamInfo() = default;
void AddInput(const AudioFrameView<const float>& src);
void AddOutput(const AudioFrameView<const float>& src);
void AddInput(const int16_t* const data,
int num_channels,
int samples_per_channel);
void AddOutput(const int16_t* const data,
int num_channels,
int samples_per_channel);
void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
std::unique_ptr<audioproc::Event> FetchEvent() {
std::unique_ptr<audioproc::Event> result = std::move(event_);
CreateNewEvent();
return result;
}
private:
void CreateNewEvent() {
event_ = std::make_unique<audioproc::Event>();
event_->set_type(audioproc::Event::STREAM);
}
std::unique_ptr<audioproc::Event> event_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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