1 files changed, 60 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_transceiver_interface.cc b/third_party/libwebrtc/api/rtp_transceiver_interface.cc
new file mode 100644
index 0000000000..7267b286be
--- /dev/null
+++ b/third_party/libwebrtc/api/rtp_transceiver_interface.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2018 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 "api/rtp_transceiver_interface.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RtpTransceiverInit::RtpTransceiverInit() = default;
+
+RtpTransceiverInit::RtpTransceiverInit(const RtpTransceiverInit& rhs) = default;
+
+RtpTransceiverInit::~RtpTransceiverInit() = default;
+
+absl::optional<RtpTransceiverDirection>
+RtpTransceiverInterface::fired_direction() const {
+  return absl::nullopt;
+}
+
+bool RtpTransceiverInterface::stopping() const {
+  return false;
+}
+
+void RtpTransceiverInterface::Stop() {
+  StopInternal();
+}
+
+RTCError RtpTransceiverInterface::StopStandard() {
+  RTC_DCHECK_NOTREACHED()
+      << "DEBUG: RtpTransceiverInterface::StopStandard called";
+  return RTCError::OK();
+}
+
+void RtpTransceiverInterface::StopInternal() {
+  RTC_DCHECK_NOTREACHED()
+      << "DEBUG: RtpTransceiverInterface::StopInternal called";
+}
+
+// TODO(bugs.webrtc.org/11839) Remove default implementations when clients
+// are updated.
+void RtpTransceiverInterface::SetDirection(
+    RtpTransceiverDirection new_direction) {
+  SetDirectionWithError(new_direction);
+}
+
+RTCError RtpTransceiverInterface::SetDirectionWithError(
+    RtpTransceiverDirection new_direction) {
+  RTC_DCHECK_NOTREACHED() << "Default implementation called";
+  return RTCError::OK();
+}
+
+}  // namespace webrtc