Index: webrtc/call/rtp_transport_controller_receive.h |
diff --git a/webrtc/call/rtp_transport_controller_receive.h b/webrtc/call/rtp_transport_controller_receive.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..e1f15ed8e5728d083bb2d5b96f8b79ba2883dad6 |
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+++ b/webrtc/call/rtp_transport_controller_receive.h |
@@ -0,0 +1,67 @@ |
+/* |
+ * 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. |
+ */ |
+#ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |
+ |
+#include <memory> |
+ |
+#include "webrtc/base/array_view.h" |
+#include "webrtc/base/criticalsection.h" |
+ |
+#include "webrtc/call/rtp_demuxer.h" |
+#include "webrtc/call/rtp_transport_controller_receive_interface.h" |
+ |
+namespace webrtc { |
+ |
+class ReceiveSideCongestionController; |
+class RtpPacketReceived; |
+ |
+// This class represents the RTP receive parsing and demuxing, for a |
danilchap
2017/05/23 12:53:09
What benefits this class has over RtpDemuxer? only
|
+// single RTP session. It isn't thread aware, leaving responsibility |
+// of multithreading issues to the user of this class. |
+// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP |
+// and not leave any RTCP processing to individual receive streams. |
+// TODO(nisse): Extract packet processing, including parsing and |
+// demuxing, into a separate RtpSessionReceiver classes. |
+class RtpTransportControllerReceive |
+ : public RtpTransportControllerReceiveInterface { |
+ public: |
+ // Implements RtpTransportControllerReceiveInterface. |
+ std::unique_ptr<RtpTransportReceiver> CreateReceiver( |
+ uint32_t ssrc, |
+ RtpPacketSinkInterface* sink) override; |
+ |
+ // Thread-safe wrappers for the corresponding RtpDemuxer methods. |
+ void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override; |
+ size_t RemoveSink(const RtpPacketSinkInterface* sink) override; |
+ |
+ // TODO(nisse): Not yet responsible for parsing. |
+ bool OnRtpPacket(const RtpPacketReceived& packet); |
+ |
+ private: |
+ class Receiver : public RtpTransportReceiver { |
+ public: |
+ Receiver(RtpTransportControllerReceive* transport, |
+ uint32_t ssrc, RtpPacketSinkInterface* sink); |
+ |
+ ~Receiver(); |
+ |
+ private: |
+ RtpTransportControllerReceive *transport_; |
+ RtpPacketSinkInterface* sink_; |
+ }; |
+ |
+ rtc::CriticalSection lock_; |
+ RtpDemuxer demuxer_ GUARDED_BY(&lock_); |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |