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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"
#include <cstddef>
#include <cstdint>
#include "api/audio/audio_view.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
for (int i = 0; i < src.num_channels(); ++i) {
AddInputChannel(src.channel(i));
}
}
void CaptureStreamInfo::AddInputChannel(MonoView<const float> channel) {
auto* stream = event_->mutable_stream();
stream->add_input_channel(channel.begin(), sizeof(float) * channel.size());
}
void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
for (int i = 0; i < src.num_channels(); ++i) {
AddOutputChannel(src.channel(i));
}
}
void CaptureStreamInfo::AddOutputChannel(MonoView<const float> channel) {
auto* stream = event_->mutable_stream();
stream->add_output_channel(channel.begin(), sizeof(float) * channel.size());
}
void CaptureStreamInfo::AddInput(const int16_t* const data,
int num_channels,
int samples_per_channel) {
auto* stream = event_->mutable_stream();
const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
stream->set_input_data(data, data_size);
}
void CaptureStreamInfo::AddOutput(const int16_t* const data,
int num_channels,
int samples_per_channel) {
auto* stream = event_->mutable_stream();
const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
stream->set_output_data(data, data_size);
}
void CaptureStreamInfo::AddAudioProcessingState(
const AecDump::AudioProcessingState& state) {
auto* stream = event_->mutable_stream();
stream->set_delay(state.delay);
stream->set_drift(state.drift);
if (state.applied_input_volume.has_value()) {
stream->set_applied_input_volume(*state.applied_input_volume);
}
stream->set_keypress(state.keypress);
}
} // namespace webrtc
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