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..70a2f400e874c69134d4ef56cb408a7a8920778a |
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+++ b/webrtc/call/rtp_transport_controller_receive.h |
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+/* |
+ * 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" |
+ |
+// For DeliveryStatus. |
+// TODO(nisse): This file ought to not depend on call. |
+#include "webrtc/call/call.h" |
+ |
+namespace webrtc { |
+ |
+class ReceiveSideCongestionController; |
+class RtpPacketReceived; |
+ |
+// This class represents a receiver of already parsed RTP packets. |
+class RtpPacketSinkInterface { |
+ public: |
+ virtual ~RtpPacketSinkInterface() {} |
+ virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0; |
+}; |
+ |
+// TODO(nisse): This additional interface may be a bit confusing, it's |
+// needed because RTP extensions are currently associated with each |
+// stream. We should be able to identify extensions earlier, since the |
+// mapping is the same for all streams on the same transport. Then we |
+// can delete this class, replacing it with RtpPacketSinkInterface. |
+// There's also the subtlety that the per-transport negotiation may |
+// differ from the extensions expected for a particular stream, and we |
+// may have code that breaks if we identify extension which is not |
+// consistent with the configuration of the stream. |
+ |
+// This class represents a receiver of RTP packets, which has the |
+// additional responsibility of identifying extension headers. I.e., |
+// the caller of OnRtpPacketReceive is expected to have called |
+// packet->Parse, but not packet->IdentifyExtensions. |
+class RtpPacketReceiverInterface { |
+ public: |
+ virtual ~RtpPacketReceiverInterface() {} |
+ virtual bool OnRtpPacketReceive(RtpPacketReceived* packet) = 0; |
+}; |
+ |
+// This class represents the RTP receive parsing and demuxing, for a |
+// single transport. 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 |
+// parsing and demuxing logic into separate classes. A demuxer should |
+// handle only a single transport. |
+class RtpTransportControllerReceiveInterface { |
+ public: |
+ // Configuration needed for media-independent processing of received |
+ // packets. |
+ struct Config { |
+ // Send side bandwidth estimation is defined in |
+ // draft-holmer-rmcat-transport-wide-cc-extensions. When invoking |
+ // the ReceiveSideCongestionController, we need to know if |
+ // send-side bwe is configured or not. |
+ bool use_send_side_bwe = false; |
+ }; |
+ |
+ // Creates the default implementation. |
+ static std::unique_ptr<RtpTransportControllerReceiveInterface> Create( |
+ ReceiveSideCongestionController* receive_side_cc, |
+ bool enable_receive_side_bwe); |
+ |
+ virtual ~RtpTransportControllerReceiveInterface() {} |
+ |
+ // Registers the receiver responsible for an ssrc. |
+ virtual void AddReceiver(uint32_t ssrc, |
+ const Config& config, |
+ RtpPacketReceiverInterface* receiver) = 0; |
+ // TODO(nisse): It's unclear what info is conveniently available at |
+ // remove time. For now, we take only the |receiver| pointer and |
+ // iterate over the mapping. |
+ virtual void RemoveReceiver(const RtpPacketReceiverInterface* receiver) = 0; |
+ |
+ // Used to represent auxillary sinks, currently used for FlexFec. |
+ // The responsible receiver must be registered first. |
+ virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0; |
+ // TODO(nisse): It's unclear what info is conveniently available at |
+ // remove time. For now, we take only the |receiver| pointer and |
+ // iterate over the mapping. |
+ virtual void RemoveSink(const RtpPacketSinkInterface* sink) = 0; |
+ |
+ // Process raw incoming RTP packets. |
+ // TODO(nisse): DeliveryStatus is needed for the current handling of |
+ // unsignalled ssrcs. Change return type to bool or void, once we do |
+ // that via AddPayload instead. |
+ virtual PacketReceiver::DeliveryStatus OnRtpPacket( |
+ int64_t arrival_time_ms, |
+ rtc::ArrayView<const uint8_t> packet) = 0; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |