mirror of
https://github.com/mollyim/webrtc.git
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5a6ae02e90
This is a reland of 80b95de765
Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
>
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}
Bug: webrtc:6463
Change-Id: I12154ef65744c1b7811974a1d871e05ed3fbbc27
Reviewed-on: https://webrtc-review.googlesource.com/c/118660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26337}
84 lines
2.9 KiB
C++
84 lines
2.9 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_AEC_DUMP_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/aec_dump/capture_stream_info.h"
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#include "modules/audio_processing/aec_dump/write_to_file_task.h"
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#include "modules/audio_processing/include/aec_dump.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/race_checker.h"
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#include "rtc_base/system/file_wrapper.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread_annotations.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 rtc {
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class TaskQueue;
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} // namespace rtc
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namespace webrtc {
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// Task-queue based implementation of AecDump. It is thread safe by
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// relying on locks in TaskQueue.
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class AecDumpImpl : public AecDump {
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public:
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// Does member variables initialization shared across all c-tors.
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AecDumpImpl(FileWrapper debug_file,
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int64_t max_log_size_bytes,
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rtc::TaskQueue* worker_queue);
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~AecDumpImpl() override;
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void WriteInitMessage(const ProcessingConfig& api_format,
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int64_t time_now_ms) override;
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void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
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void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
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void AddCaptureStreamInput(const AudioFrame& frame) override;
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void AddCaptureStreamOutput(const AudioFrame& frame) override;
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void AddAudioProcessingState(const AudioProcessingState& state) override;
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void WriteCaptureStreamMessage() override;
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void WriteRenderStreamMessage(const AudioFrame& frame) override;
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void WriteRenderStreamMessage(
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const AudioFrameView<const float>& src) override;
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void WriteConfig(const InternalAPMConfig& config) override;
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void WriteRuntimeSetting(
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const AudioProcessing::RuntimeSetting& runtime_setting) override;
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private:
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std::unique_ptr<WriteToFileTask> CreateWriteToFileTask();
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FileWrapper debug_file_;
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int64_t num_bytes_left_for_log_ = 0;
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rtc::RaceChecker race_checker_;
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rtc::TaskQueue* worker_queue_;
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CaptureStreamInfo capture_stream_info_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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