Index: webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.cc |
diff --git a/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.cc b/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..8fe308a89174d58eff39817a8faf25b46ca196c6 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.cc |
@@ -0,0 +1,69 @@ |
+/* |
+ * 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 "webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.h" |
+ |
+namespace webrtc { |
+CaptureStreamInfoImpl::CaptureStreamInfoImpl( |
+ std::unique_ptr<audioproc::Event> event) |
+ : event_(std::move(event)) { |
+ RTC_DCHECK(event_); |
+ event_->set_type(audioproc::Event::STREAM); |
+} |
+ |
+CaptureStreamInfoImpl::~CaptureStreamInfoImpl() = default; |
+ |
+void CaptureStreamInfoImpl::AddInput(FloatAudioFrame src) { |
+ auto* stream = event_->mutable_stream(); |
+ |
+ for (size_t 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 CaptureStreamInfoImpl::AddOutput(FloatAudioFrame src) { |
+ auto* stream = event_->mutable_stream(); |
+ |
+ for (size_t 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 CaptureStreamInfoImpl::AddInput(const AudioFrame& frame) { |
+ audioproc::Stream* stream = event_->mutable_stream(); |
+ const size_t data_size = |
+ sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; |
+ stream->set_input_data(frame.data_, data_size); |
+} |
+ |
+void CaptureStreamInfoImpl::AddOutput(const AudioFrame& frame) { |
+ audioproc::Stream* stream = event_->mutable_stream(); |
+ const size_t data_size = |
+ sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; |
+ stream->set_output_data(frame.data_, data_size); |
+} |
+ |
+void CaptureStreamInfoImpl::set_delay(int delay) { |
+ event_->mutable_stream()->set_delay(delay); |
+} |
+void CaptureStreamInfoImpl::set_drift(int drift) { |
+ event_->mutable_stream()->set_drift(drift); |
+} |
+void CaptureStreamInfoImpl::set_level(int level) { |
+ event_->mutable_stream()->set_level(level); |
+} |
+void CaptureStreamInfoImpl::set_keypress(bool keypress) { |
+ event_->mutable_stream()->set_keypress(keypress); |
+} |
+} // namespace webrtc |