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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}
58 lines
1.6 KiB
C++
58 lines
1.6 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 MODULES_AUDIO_PROCESSING_AEC_DUMP_WRITE_TO_FILE_TASK_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_WRITE_TO_FILE_TASK_H_
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#include <memory>
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#include <string>
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#include <utility>
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#include "rtc_base/checks.h"
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#include "rtc_base/event.h"
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#include "rtc_base/ignore_wundef.h"
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#include "rtc_base/platform_file.h"
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#include "rtc_base/task_queue.h"
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#include "system_wrappers/include/file_wrapper.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 "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 WriteToFileTask : public rtc::QueuedTask {
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public:
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WriteToFileTask(webrtc::FileWrapper* debug_file,
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int64_t* num_bytes_left_for_log);
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~WriteToFileTask() override;
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audioproc::Event* GetEvent();
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private:
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bool IsRoomForNextEvent(size_t event_byte_size) const;
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void UpdateBytesLeft(size_t event_byte_size);
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bool Run() override;
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webrtc::FileWrapper* debug_file_;
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audioproc::Event event_;
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int64_t* num_bytes_left_for_log_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_WRITE_TO_FILE_TASK_H_
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