aec_dump_impl.h - mozsearch

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/*

 *  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/task_queue/task_queue_base.h"

#include "modules/audio_processing/aec_dump/capture_stream_info.h"

#include "modules/audio_processing/include/aec_dump.h"

#include "rtc_base/race_checker.h"

#include "rtc_base/system/file_wrapper.h"

#include "rtc_base/thread_annotations.h"

// Files generated at build-time by the protobuf compiler.

#ifdef WEBRTC_ANDROID_PLATFORM_BUILD

#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"

#else

#include "modules/audio_processing/debug.pb.h"

#endif

namespace webrtc {

// Task-queue based implementation of AecDump. It is thread safe by

// relying on locks in TaskQueue.

class AecDumpImpl : public AecDump {

 public:

  // `max_log_size_bytes` - maximum number of bytes to write to the debug file,

  // `max_log_size_bytes == -1` means the log size will be unlimited.

  AecDumpImpl(FileWrapper debug_file,

              int64_t max_log_size_bytes,

              absl::Nonnull<TaskQueueBase*> worker_queue);

  AecDumpImpl(const AecDumpImpl&) = delete;

  AecDumpImpl& operator=(const AecDumpImpl&) = delete;

  ~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 int16_t* const data,

                             int num_channels,

                             int samples_per_channel) override;

  void AddCaptureStreamOutput(const int16_t* const data,

                              int num_channels,

                              int samples_per_channel) override;

  void AddAudioProcessingState(const AudioProcessingState& state) override;

  void WriteCaptureStreamMessage() override;

  void WriteRenderStreamMessage(const int16_t* const data,

                                int num_channels,

                                int samples_per_channel) override;

  void WriteRenderStreamMessage(

      const AudioFrameView<const float>& src) override;

  void WriteConfig(const InternalAPMConfig& config) override;

  void WriteRuntimeSetting(

      const AudioProcessing::RuntimeSetting& runtime_setting) override;

 private:

  void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event);

  FileWrapper debug_file_;

  int64_t num_bytes_left_for_log_ = 0;

  rtc::RaceChecker race_checker_;

  absl::Nonnull<TaskQueueBase*> worker_queue_;

  CaptureStreamInfo capture_stream_info_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_