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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h | 66 |
1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h new file mode 100644 index 0000000000..0819bbcb23 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h @@ -0,0 +1,66 @@ +/* + * 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_CAPTURE_STREAM_INFO_H_ +#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ + +#include <memory> +#include <utility> + +#include "modules/audio_processing/include/aec_dump.h" +#include "rtc_base/ignore_wundef.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 { + +class CaptureStreamInfo { + public: + CaptureStreamInfo() { CreateNewEvent(); } + CaptureStreamInfo(const CaptureStreamInfo&) = delete; + CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete; + ~CaptureStreamInfo() = default; + + void AddInput(const AudioFrameView<const float>& src); + void AddOutput(const AudioFrameView<const float>& src); + + void AddInput(const int16_t* const data, + int num_channels, + int samples_per_channel); + void AddOutput(const int16_t* const data, + int num_channels, + int samples_per_channel); + + void AddAudioProcessingState(const AecDump::AudioProcessingState& state); + + std::unique_ptr<audioproc::Event> FetchEvent() { + std::unique_ptr<audioproc::Event> result = std::move(event_); + CreateNewEvent(); + return result; + } + + private: + void CreateNewEvent() { + event_ = std::make_unique<audioproc::Event>(); + event_->set_type(audioproc::Event::STREAM); + } + std::unique_ptr<audioproc::Event> event_; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_ |