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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.
*/
#include "modules/audio_processing/aec_dump/capture_stream_info.h"
namespace webrtc {
CaptureStreamInfo::CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task)
: task_(std::move(task)) {
RTC_DCHECK(task_);
task_->GetEvent()->set_type(audioproc::Event::STREAM);
}
CaptureStreamInfo::~CaptureStreamInfo() = default;
void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
RTC_DCHECK(task_);
auto* stream = task_->GetEvent()->mutable_stream();
for (size_t i = 0; i < src.num_channels(); ++i) {
const auto& channel_view = src.channel(i);
stream->add_input_channel(channel_view.begin(),
sizeof(float) * channel_view.size());
}
}
void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
RTC_DCHECK(task_);
auto* stream = task_->GetEvent()->mutable_stream();
for (size_t i = 0; i < src.num_channels(); ++i) {
const auto& channel_view = src.channel(i);
stream->add_output_channel(channel_view.begin(),
sizeof(float) * channel_view.size());
}
}
void CaptureStreamInfo::AddInput(const AudioFrame& frame) {
RTC_DCHECK(task_);
auto* stream = task_->GetEvent()->mutable_stream();
const size_t data_size =
sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
stream->set_input_data(frame.data(), data_size);
}
void CaptureStreamInfo::AddOutput(const AudioFrame& frame) {
RTC_DCHECK(task_);
auto* stream = task_->GetEvent()->mutable_stream();
const size_t data_size =
sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
stream->set_output_data(frame.data(), data_size);
}
void CaptureStreamInfo::AddAudioProcessingState(
const AecDump::AudioProcessingState& state) {
RTC_DCHECK(task_);
auto* stream = task_->GetEvent()->mutable_stream();
stream->set_delay(state.delay);
stream->set_drift(state.drift);
stream->set_level(state.level);
stream->set_keypress(state.keypress);
}
} // namespace webrtc
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