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https://github.com/mollyim/webrtc.git
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3153b363cd
Making it clear that the field is used to store the applied input volume and not the recommended input volume. Bug: webrtc:7494, b/241923537 Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38051}
59 lines
2.2 KiB
C++
59 lines
2.2 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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#include "modules/audio_processing/aec_dump/capture_stream_info.h"
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namespace webrtc {
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void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
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auto* stream = event_->mutable_stream();
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for (int i = 0; i < src.num_channels(); ++i) {
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const auto& channel_view = src.channel(i);
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stream->add_input_channel(channel_view.begin(),
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sizeof(float) * channel_view.size());
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}
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}
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void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
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auto* stream = event_->mutable_stream();
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for (int i = 0; i < src.num_channels(); ++i) {
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const auto& channel_view = src.channel(i);
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stream->add_output_channel(channel_view.begin(),
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sizeof(float) * channel_view.size());
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}
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}
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void CaptureStreamInfo::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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auto* stream = event_->mutable_stream();
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const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
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stream->set_input_data(data, data_size);
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}
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void CaptureStreamInfo::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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auto* stream = event_->mutable_stream();
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const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
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stream->set_output_data(data, data_size);
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}
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void CaptureStreamInfo::AddAudioProcessingState(
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const AecDump::AudioProcessingState& state) {
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auto* stream = event_->mutable_stream();
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stream->set_delay(state.delay);
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stream->set_drift(state.drift);
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stream->set_applied_input_volume(state.applied_input_volume);
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stream->set_keypress(state.keypress);
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}
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} // namespace webrtc
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