Test submission of complete AEC-dump refactoring.

webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h

+/*
+ *  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/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/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(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;
+
+  std::unique_ptr<CaptureStreamInfo> GetCaptureStreamInfo() override;
+
+  void WriteInitMessage(const ProcessingConfig& api_format) override;
+  void WriteRenderStreamMessage(const AudioFrame& frame) override;
+  void WriteRenderStreamMessage(
+      std::vector<rtc::ArrayView<const float>> src) override;
+  void WriteCaptureStreamMessage(
+      std::unique_ptr<CaptureStreamInfo> capture_stream_info) override;
+  void WriteConfig(const InternalAPMConfig& config, bool forced) override;
+
+ private:
+  void PostTask(std::unique_ptr<audioproc::Event> event);
+
+  // 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.
+  std::string 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_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_