diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc | 61 |
1 files changed, 61 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc new file mode 100644 index 0000000000..7d82a39729 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.cc @@ -0,0 +1,61 @@ +/* + * 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 { + +void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) { + auto* stream = event_->mutable_stream(); + + for (int 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) { + auto* stream = event_->mutable_stream(); + + for (int 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 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 |