Repo created

TMessagesProj/jni/voip/webrtc/call/rtp_stream_receiver_controller.cc

@@ -0,0 +1,71 @@
+/*
+ *  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 "call/rtp_stream_receiver_controller.h"
+
+#include <memory>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+RtpStreamReceiverController::Receiver::Receiver(
+    RtpStreamReceiverController* controller,
+    uint32_t ssrc,
+    RtpPacketSinkInterface* sink)
+    : controller_(controller), sink_(sink) {
+  const bool sink_added = controller_->AddSink(ssrc, sink_);
+  if (!sink_added) {
+    RTC_LOG(LS_ERROR)
+        << "RtpStreamReceiverController::Receiver::Receiver: Sink "
+           "could not be added for SSRC="
+        << ssrc << ".";
+  }
+}
+
+RtpStreamReceiverController::Receiver::~Receiver() {
+  // This may fail, if corresponding AddSink in the constructor failed.
+  controller_->RemoveSink(sink_);
+}
+
+RtpStreamReceiverController::RtpStreamReceiverController() {}
+
+RtpStreamReceiverController::~RtpStreamReceiverController() = default;
+
+std::unique_ptr<RtpStreamReceiverInterface>
+RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
+                                            RtpPacketSinkInterface* sink) {
+  return std::make_unique<Receiver>(this, ssrc, sink);
+}
+
+bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
+  return demuxer_.OnRtpPacket(packet);
+}
+
+void RtpStreamReceiverController::OnRecoveredPacket(
+    const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
+  demuxer_.OnRtpPacket(packet);
+}
+
+bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
+                                          RtpPacketSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
+  return demuxer_.AddSink(ssrc, sink);
+}
+
+bool RtpStreamReceiverController::RemoveSink(
+    const RtpPacketSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&demuxer_sequence_);
+  return demuxer_.RemoveSink(sink);
+}
+
+}  // namespace webrtc