|  | /* | 
|  | *  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_ |