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645f24cb86
This CL creates a new stream interface and uses it to replace most of the usage of AudioFrame in the non-test code. The CL changes some of the test code as well, as the other changes required that. The CL will be followed by 2 more related CLs. Bug: webrtc:5298 Change-Id: I5cfbe6079f30fc3fbf35b35fd077b6fb49c7def0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170040 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30799}
69 lines
2.1 KiB
C++
69 lines
2.1 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_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 <vector>
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#include "modules/audio_processing/aec_dump/write_to_file_task.h"
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#include "modules/audio_processing/include/aec_dump.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/ignore_wundef.h"
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#include "rtc_base/logging.h"
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// Files generated at build-time by the protobuf compiler.
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RTC_PUSH_IGNORING_WUNDEF()
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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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RTC_POP_IGNORING_WUNDEF()
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namespace webrtc {
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class CaptureStreamInfo {
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public:
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explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task);
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~CaptureStreamInfo();
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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<WriteToFileTask> GetTask() {
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RTC_DCHECK(task_);
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return std::move(task_);
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}
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void SetTask(std::unique_ptr<WriteToFileTask> task) {
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RTC_DCHECK(!task_);
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RTC_DCHECK(task);
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task_ = std::move(task);
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task_->GetEvent()->set_type(audioproc::Event::STREAM);
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}
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private:
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std::unique_ptr<WriteToFileTask> task_;
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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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