1 files changed, 100 insertions, 0 deletions
diff --git a/chromium/third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame.cc b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame.cc
new file mode 100644
index 00000000000..6dbaa7fedf7
--- /dev/null
+++ b/chromium/third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame.cc
@@ -0,0 +1,100 @@
+// Copyright 2020 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame.h"
+
+#include <utility>
+
+#include "third_party/blink/renderer/core/typed_arrays/dom_array_buffer.h"
+#include "third_party/blink/renderer/modules/peerconnection/rtc_encoded_audio_frame_delegate.h"
+#include "third_party/blink/renderer/platform/wtf/text/string_builder.h"
+#include "third_party/webrtc/api/frame_transformer_interface.h"
+
+namespace blink {
+
+RTCEncodedAudioFrame::RTCEncodedAudioFrame(
+    std::unique_ptr<webrtc::TransformableFrameInterface> webrtc_frame)
+    : delegate_(base::MakeRefCounted<RTCEncodedAudioFrameDelegate>(
+          std::move(webrtc_frame),
+          Vector<uint32_t>())) {}
+
+RTCEncodedAudioFrame::RTCEncodedAudioFrame(
+    std::unique_ptr<webrtc::TransformableAudioFrameInterface>
+        webrtc_audio_frame) {
+  Vector<uint32_t> contributing_sources;
+  if (webrtc_audio_frame) {
+    wtf_size_t num_csrcs = webrtc_audio_frame->GetHeader().numCSRCs;
+    contributing_sources.ReserveInitialCapacity(num_csrcs);
+    for (wtf_size_t i = 0; i < num_csrcs; i++) {
+      contributing_sources.push_back(
+          webrtc_audio_frame->GetHeader().arrOfCSRCs[i]);
+    }
+  }
+  delegate_ = base::MakeRefCounted<RTCEncodedAudioFrameDelegate>(
+      std::move(webrtc_audio_frame), std::move(contributing_sources));
+}
+
+RTCEncodedAudioFrame::RTCEncodedAudioFrame(
+    scoped_refptr<RTCEncodedAudioFrameDelegate> delegate)
+    : delegate_(std::move(delegate)) {}
+
+uint64_t RTCEncodedAudioFrame::timestamp() const {
+  return delegate_->Timestamp();
+}
+
+DOMArrayBuffer* RTCEncodedAudioFrame::data() const {
+  if (!frame_data_) {
+    frame_data_ = delegate_->CreateDataBuffer();
+  }
+  return frame_data_;
+}
+
+DOMArrayBuffer* RTCEncodedAudioFrame::additionalData() const {
+  return nullptr;
+}
+
+void RTCEncodedAudioFrame::setData(DOMArrayBuffer* data) {
+  frame_data_ = data;
+}
+
+uint32_t RTCEncodedAudioFrame::synchronizationSource() const {
+  return delegate_->Ssrc();
+}
+
+Vector<uint32_t> RTCEncodedAudioFrame::contributingSources() const {
+  return delegate_->ContributingSources();
+}
+
+String RTCEncodedAudioFrame::toString() const {
+  StringBuilder sb;
+  sb.Append("RTCEncodedAudioFrame{timestamp: ");
+  sb.AppendNumber(timestamp());
+  sb.Append("us, size: ");
+  sb.AppendNumber(data() ? data()->ByteLengthAsSizeT() : 0);
+  sb.Append("}");
+  return sb.ToString();
+}
+
+void RTCEncodedAudioFrame::SyncDelegate() const {
+  delegate_->SetData(frame_data_);
+}
+
+scoped_refptr<RTCEncodedAudioFrameDelegate> RTCEncodedAudioFrame::Delegate()
+    const {
+  SyncDelegate();
+  return delegate_;
+}
+
+std::unique_ptr<webrtc::TransformableFrameInterface>
+RTCEncodedAudioFrame::PassWebRtcFrame() {
+  SyncDelegate();
+  return delegate_->PassWebRtcFrame();
+}
+
+void RTCEncodedAudioFrame::Trace(Visitor* visitor) {
+  ScriptWrappable::Trace(visitor);
+  visitor->Trace(frame_data_);
+}
+
+}  // namespace blink