Index: webrtc/call/rtp_stream_receiver_controller.h |
diff --git a/webrtc/call/rtp_stream_receiver_controller.h b/webrtc/call/rtp_stream_receiver_controller.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..5c8ed6775e8739b59e3db79c967f3df7e7ab10cb |
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+++ b/webrtc/call/rtp_stream_receiver_controller.h |
@@ -0,0 +1,72 @@ |
+/* |
+ * 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_STREAM_RECEIVER_CONTROLLER_H_ |
+#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |
+ |
+#include <memory> |
+ |
+#include "webrtc/base/criticalsection.h" |
+#include "webrtc/call/rtp_demuxer.h" |
+#include "webrtc/call/rtp_stream_receiver_controller_interface.h" |
+ |
+namespace webrtc { |
+ |
+class RtpPacketReceived; |
+ |
+// This class represents the RTP receive parsing and demuxing, for a |
+// single RTP session. |
+// TODO(nisse): Add RTCP processing, we should aim to terminate RTCP |
+// and not leave any RTCP processing to individual receive streams. |
+// TODO(nisse): Extract per-packet processing, including parsing and |
+// demuxing, into a separate class. |
+class RtpStreamReceiverController |
+ : public RtpStreamReceiverControllerInterface { |
+ public: |
+ RtpStreamReceiverController(); |
+ ~RtpStreamReceiverController() override; |
+ |
+ // Implements RtpStreamReceiverControllerInterface. |
+ std::unique_ptr<RtpStreamReceiverInterface> 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 RtpStreamReceiverInterface { |
+ public: |
+ Receiver(RtpStreamReceiverController* controller, |
+ uint32_t ssrc, |
+ RtpPacketSinkInterface* sink); |
+ |
+ ~Receiver() override; |
+ |
+ private: |
+ RtpStreamReceiverController* const controller_; |
+ RtpPacketSinkInterface* const sink_; |
+ }; |
+ |
+ // TODO(nisse): Move to a TaskQueue for synchronization. When used |
+ // by Call, we expect construction and all methods but OnRtpPacket |
+ // to be called on the same thread, and OnRtpPacket to be called |
+ // by a single, but possibly distinct, thread. But applications not |
+ // using Call may have use threads differently. |
+ rtc::CriticalSection lock_; |
+ RtpDemuxer demuxer_ GUARDED_BY(&lock_); |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |