1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_processing/aec_dump/capture_stream_info.h"
12
13 namespace webrtc {
14
AddInput(const AudioFrameView<const float> & src)15 void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
16 auto* stream = event_->mutable_stream();
17
18 for (int i = 0; i < src.num_channels(); ++i) {
19 const auto& channel_view = src.channel(i);
20 stream->add_input_channel(channel_view.begin(),
21 sizeof(float) * channel_view.size());
22 }
23 }
24
AddOutput(const AudioFrameView<const float> & src)25 void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
26 auto* stream = event_->mutable_stream();
27
28 for (int i = 0; i < src.num_channels(); ++i) {
29 const auto& channel_view = src.channel(i);
30 stream->add_output_channel(channel_view.begin(),
31 sizeof(float) * channel_view.size());
32 }
33 }
34
AddInput(const int16_t * const data,int num_channels,int samples_per_channel)35 void CaptureStreamInfo::AddInput(const int16_t* const data,
36 int num_channels,
37 int samples_per_channel) {
38 auto* stream = event_->mutable_stream();
39 const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
40 stream->set_input_data(data, data_size);
41 }
42
AddOutput(const int16_t * const data,int num_channels,int samples_per_channel)43 void CaptureStreamInfo::AddOutput(const int16_t* const data,
44 int num_channels,
45 int samples_per_channel) {
46 auto* stream = event_->mutable_stream();
47 const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
48 stream->set_output_data(data, data_size);
49 }
50
AddAudioProcessingState(const AecDump::AudioProcessingState & state)51 void CaptureStreamInfo::AddAudioProcessingState(
52 const AecDump::AudioProcessingState& state) {
53 auto* stream = event_->mutable_stream();
54 stream->set_delay(state.delay);
55 stream->set_drift(state.drift);
56 if (state.applied_input_volume.has_value()) {
57 stream->set_applied_input_volume(*state.applied_input_volume);
58 }
59 stream->set_keypress(state.keypress);
60 }
61 } // namespace webrtc
62