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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
66 lines
2 KiB
C++
66 lines
2 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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#include <memory>
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#include <utility>
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#include <vector>
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#include "webrtc/modules/audio_processing/aec_dump/write_to_file_task.h"
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#include "webrtc/modules/audio_processing/include/aec_dump.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/ignore_wundef.h"
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#include "webrtc/rtc_base/logging.h"
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// Files generated at build-time by the protobuf compiler.
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RTC_PUSH_IGNORING_WUNDEF()
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
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#else
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#include "webrtc/modules/audio_processing/debug.pb.h"
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#endif
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RTC_POP_IGNORING_WUNDEF()
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namespace webrtc {
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class CaptureStreamInfo {
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public:
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explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task);
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~CaptureStreamInfo();
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void AddInput(const FloatAudioFrame& src);
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void AddOutput(const FloatAudioFrame& src);
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void AddInput(const AudioFrame& frame);
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void AddOutput(const AudioFrame& frame);
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void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
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std::unique_ptr<WriteToFileTask> GetTask() {
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RTC_DCHECK(task_);
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return std::move(task_);
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}
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void SetTask(std::unique_ptr<WriteToFileTask> task) {
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RTC_DCHECK(!task_);
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RTC_DCHECK(task);
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task_ = std::move(task);
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task_->GetEvent()->set_type(audioproc::Event::STREAM);
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}
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private:
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std::unique_ptr<WriteToFileTask> task_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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