/* * 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 #include #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& src); void AddOutput(const AudioFrameView& 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 FetchEvent() { std::unique_ptr result = std::move(event_); CreateNewEvent(); return result; } private: void CreateNewEvent() { event_ = std::make_unique(); event_->set_type(audioproc::Event::STREAM); } std::unique_ptr event_; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_