| /* |
| * 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 MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ |
| #define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/audio/audio_frame.h" |
| #include "modules/audio_processing/aec_dump/capture_stream_info.h" |
| #include "modules/audio_processing/aec_dump/write_to_file_task.h" |
| #include "modules/audio_processing/include/aec_dump.h" |
| #include "rtc_base/ignore_wundef.h" |
| #include "rtc_base/platform_file.h" |
| #include "rtc_base/race_checker.h" |
| #include "rtc_base/system/file_wrapper.h" |
| #include "rtc_base/task_queue.h" |
| #include "rtc_base/thread_annotations.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 "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: |
| // Does member variables initialization shared across all c-tors. |
| AecDumpImpl(FileWrapper debug_file, |
| int64_t max_log_size_bytes, |
| rtc::TaskQueue* worker_queue); |
| |
| ~AecDumpImpl() override; |
| |
| void WriteInitMessage(const ProcessingConfig& api_format, |
| int64_t time_now_ms) override; |
| void AddCaptureStreamInput(const AudioFrameView<const float>& src) override; |
| void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override; |
| void AddCaptureStreamInput(const AudioFrame& frame) override; |
| void AddCaptureStreamOutput(const AudioFrame& frame) override; |
| void AddAudioProcessingState(const AudioProcessingState& state) override; |
| void WriteCaptureStreamMessage() override; |
| |
| void WriteRenderStreamMessage(const AudioFrame& frame) override; |
| void WriteRenderStreamMessage( |
| const AudioFrameView<const float>& src) override; |
| |
| void WriteConfig(const InternalAPMConfig& config) override; |
| |
| void WriteRuntimeSetting( |
| const AudioProcessing::RuntimeSetting& runtime_setting) override; |
| |
| private: |
| std::unique_ptr<WriteToFileTask> CreateWriteToFileTask(); |
| |
| FileWrapper debug_file_; |
| int64_t num_bytes_left_for_log_ = 0; |
| rtc::RaceChecker race_checker_; |
| rtc::TaskQueue* worker_queue_; |
| CaptureStreamInfo capture_stream_info_; |
| }; |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_ |