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efeeba0864
Bug: webrtc:15623 Change-Id: Ia184993769f74d51e68a5a536d5fdde26890bcfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325481 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41058}
82 lines
3.1 KiB
C++
82 lines
3.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "modules/audio_processing/aec_dump/capture_stream_info.h"
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#include "modules/audio_processing/include/aec_dump.h"
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#include "rtc_base/race_checker.h"
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#include "rtc_base/system/file_wrapper.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/thread_annotations.h"
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// Files generated at build-time by the protobuf compiler.
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
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#else
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#include "modules/audio_processing/debug.pb.h"
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#endif
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namespace webrtc {
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// Task-queue based implementation of AecDump. It is thread safe by
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// relying on locks in TaskQueue.
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class AecDumpImpl : public AecDump {
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public:
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// `max_log_size_bytes` - maximum number of bytes to write to the debug file,
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// `max_log_size_bytes == -1` means the log size will be unlimited.
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AecDumpImpl(FileWrapper debug_file,
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int64_t max_log_size_bytes,
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rtc::TaskQueue* worker_queue);
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AecDumpImpl(const AecDumpImpl&) = delete;
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AecDumpImpl& operator=(const AecDumpImpl&) = delete;
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~AecDumpImpl() override;
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void WriteInitMessage(const ProcessingConfig& api_format,
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int64_t time_now_ms) override;
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void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
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void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
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void AddCaptureStreamInput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) override;
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void AddCaptureStreamOutput(const int16_t* const data,
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int num_channels,
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int samples_per_channel) override;
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void AddAudioProcessingState(const AudioProcessingState& state) override;
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void WriteCaptureStreamMessage() override;
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void WriteRenderStreamMessage(const int16_t* const data,
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int num_channels,
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int samples_per_channel) override;
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void WriteRenderStreamMessage(
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const AudioFrameView<const float>& src) override;
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void WriteConfig(const InternalAPMConfig& config) override;
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void WriteRuntimeSetting(
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const AudioProcessing::RuntimeSetting& runtime_setting) override;
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private:
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void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event);
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FileWrapper debug_file_;
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int64_t num_bytes_left_for_log_ = 0;
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rtc::RaceChecker race_checker_;
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rtc::TaskQueue* worker_queue_;
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CaptureStreamInfo capture_stream_info_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
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