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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 "modules/audio_processing/aec_dump/capture_stream_info.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "rtc_base/ignore_wundef.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 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,
rtc::TaskQueue* 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_;
rtc::TaskQueue* worker_queue_;
CaptureStreamInfo capture_stream_info_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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