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Unified Diff: webrtc/modules/audio_processing/aec_dump/capture_stream_info_impl.cc

Issue 2838133003: Implementation of new AecDump interface. (Closed)
Patch Set: Complete and tested AecDump implementation. Created 3 years, 8 months ago
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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

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