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_ |