mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
2 KiB
C++
66 lines
2 KiB
C++
/*
|
|
* 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 "modules/audio_processing/aec_dump/write_to_file_task.h"
|
|
#include "modules/audio_processing/include/aec_dump.h"
|
|
#include "modules/include/module_common_types.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 FloatAudioFrame& src);
|
|
void AddOutput(const FloatAudioFrame& 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_
|