| 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..ed06fd4d7d1efc623051d8f74230acb049921ac2
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
|
| @@ -0,0 +1,86 @@
|
| +/*
|
| + * 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_impl.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,
|
| + 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;
|
| +
|
| + CaptureStreamInfo* GetCaptureStreamInfo() 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_;
|
| + CaptureStreamInfoImpl capture_stream_info_;
|
| +};
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
|
|
|