Chromium Code Reviews| Index: webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h | 
| diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..c4d2a683a63da9e4fefb89efb1849cf8fc8082d5 | 
| --- /dev/null | 
| +++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h | 
| @@ -0,0 +1,91 @@ | 
| +/* | 
| + * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ | 
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ | 
| + | 
| +#include <memory> | 
| +#include <string> | 
| +#include <vector> | 
| + | 
| +#include "webrtc/base/ignore_wundef.h" | 
| +#include "webrtc/base/platform_file.h" | 
| +#include "webrtc/base/protobuf_utils.h" | 
| +#include "webrtc/base/task_queue.h" | 
| +#include "webrtc/base/thread_checker.h" | 
| +#include "webrtc/modules/audio_processing/aec_dump/capture_stream_info.h" | 
| +#include "webrtc/modules/audio_processing/aec_dump/write_to_file_task.h" | 
| +#include "webrtc/modules/audio_processing/include/aec_dump.h" | 
| +#include "webrtc/modules/include/module_common_types.h" | 
| +#include "webrtc/system_wrappers/include/file_wrapper.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 "webrtc/modules/audio_processing/debug.pb.h" | 
| +#endif | 
| +RTC_POP_IGNORING_WUNDEF() | 
| + | 
| +namespace rtc { | 
| +class TaskQueue; | 
| +} // namespace rtc | 
| + | 
| +namespace webrtc { | 
| + | 
| +// Task-queue based implementation of AecDump. It is thread safe by | 
| +// relying on locks in TaskQueue. | 
| +class AecDumpImpl : public AecDump { | 
| + public: | 
| + AecDumpImpl(rtc::PlatformFile file, | 
| 
 
peah-webrtc
2017/05/16 06:30:38
Do these constructors need to be public? If we ins
 
aleloi
2017/05/16 20:10:17
It would expose some implementation in the interfa
 
peah-webrtc
2017/05/17 05:36:32
Good point! Lets keep the factories :-)
 
 | 
| + int64_t max_log_size_bytes, | 
| + rtc::TaskQueue* worker_queue); | 
| + AecDumpImpl(std::string file_name, | 
| + int64_t max_log_size_bytes, | 
| + rtc::TaskQueue* worker_queue); | 
| + AecDumpImpl(FILE* handle, | 
| + int64_t max_log_size_bytes, | 
| + rtc::TaskQueue* worker_queue); | 
| + ~AecDumpImpl() override; | 
| + | 
| + void AddCaptureStreamInput(const FloatAudioFrame& src) override; | 
| + void AddCaptureStreamOutput(const FloatAudioFrame& src) override; | 
| + | 
| + void AddCaptureStreamInput(const AudioFrame& frame) override; | 
| + void AddCaptureStreamOutput(const AudioFrame& frame) override; | 
| + void AddAudioProcessingState(const AudioProcessingState& state) override; | 
| + | 
| + void WriteInitMessage(const InternalAPMStreamsConfig& api_format) override; | 
| + void WriteRenderStreamMessage(const AudioFrame& frame) override; | 
| + void WriteRenderStreamMessage(const FloatAudioFrame& src) override; | 
| + void WriteCaptureStreamMessage() override; | 
| + void WriteConfig(const InternalAPMConfig& config, bool forced) override; | 
| + | 
| + private: | 
| + // Does member variables initialization shared across all c-tors. | 
| + AecDumpImpl(int64_t max_log_size_bytes, rtc::TaskQueue* worker_queue); | 
| + std::unique_ptr<WriteToFileTask> CreateWriteToFileTask(); | 
| + | 
| + // Implementation detail of WriteConfig: If not |forced|, only | 
| + // writes the current config if it is different from the last saved | 
| + // one; if |forced|, writes the config regardless of the last saved. | 
| + ProtoString last_serialized_capture_config_ GUARDED_BY(config_string_lock_) = | 
| + ""; | 
| + std::unique_ptr<FileWrapper> debug_file_; | 
| + int64_t num_bytes_left_for_log_ = 0; | 
| + | 
| + rtc::TaskQueue* worker_queue_; | 
| + rtc::CriticalSection config_string_lock_; | 
| 
 
peah-webrtc
2017/05/16 06:30:38
I'm not convinced that this lock is really needed.
 
aleloi
2017/05/16 20:10:17
I also can't see the need for it. It was definitel
 
 | 
| + CaptureStreamInfo capture_stream_info_; | 
| +}; | 
| +} // namespace webrtc | 
| + | 
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ |