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