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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}
63 lines
2 KiB
C++
63 lines
2 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_CAPTURE_STREAM_INFO_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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#include <memory>
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#include <utility>
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#include "modules/audio_processing/include/aec_dump.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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class CaptureStreamInfo {
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public:
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CaptureStreamInfo() { CreateNewEvent(); }
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CaptureStreamInfo(const CaptureStreamInfo&) = delete;
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CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete;
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~CaptureStreamInfo() = default;
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void AddInput(const AudioFrameView<const float>& src);
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void AddOutput(const AudioFrameView<const float>& src);
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void AddInput(const int16_t* const data,
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int num_channels,
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int samples_per_channel);
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void AddOutput(const int16_t* const data,
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int num_channels,
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int samples_per_channel);
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void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
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std::unique_ptr<audioproc::Event> FetchEvent() {
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std::unique_ptr<audioproc::Event> result = std::move(event_);
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CreateNewEvent();
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return result;
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}
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private:
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void CreateNewEvent() {
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event_ = std::make_unique<audioproc::Event>();
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event_->set_type(audioproc::Event::STREAM);
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}
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std::unique_ptr<audioproc::Event> event_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
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