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 |
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index 0000000000000000000000000000000000000000..0d9534c081ba55d15d83a2a2c8d605f5cae06a9a |
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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_ |
+ |
the sun
2017/03/15 14:03:04
I find it confusing that we have 3 interfaces for
nisse-webrtc
2017/03/15 14:55:00
I added a TODO comment explaining how to get rid o
|
+#include "webrtc/base/array_view.h" |
+ |
+// For MediaType and DeliveryStatus. |
+// TODO(nisse): This file ought to not depend on call. We could move |
+// MediaType definition here, and perhaps rename it to transport_id in |
+// the process, since its main purpose is to disambiguate ssrc |
+// collisions between transports. |
+#include "webrtc/call/call.h" |
+ |
+namespace webrtc { |
+ |
+class RtpPacketReceived; |
+ |
+// This class represents a receiver of already parsed RTP packets. |
+class RtpPacketSinkInterface { |
+ public: |
+ virtual void OnRtpPacket(const RtpPacketReceived& packet) = 0; |
+ virtual ~RtpPacketSinkInterface() {} |
+}; |
+ |
+// 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. |
the sun
2017/03/15 14:03:04
Why? That seems like a hard-to-enforce contract.
nisse-webrtc
2017/03/15 14:55:00
It's kind-of enforced via const; an implementation
|
+class RtpPacketReceiverInterface { |
+ public: |
+ virtual bool OnRtpPacketReceive(RtpPacketReceived* packet) = 0; |
+ virtual ~RtpPacketReceiverInterface() {} |
+}; |
+ |
+class RtpPacketObserverInterface; |
+ |
+// This class represents the RTP receive demuxing. 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. |
+class RtpTransportControllerReceiveInterface { |
+ public: |
+ // Configuration needed for media-independent processing of received |
+ // packets, in particular, feeding information to the congestion |
+ // controller. |
+ |
+ // TODO(nisse): It turns out this data is only used for the |
+ // RtpPacketObserverInterface callback. Since we have no interest in |
+ // the details, make it a void* pointer or use a template parameter |
+ // for the type? |
+ struct Config { |
+ // See draft-holmer-rmcat-transport-wide-cc-extensions for details. |
+ bool use_send_side_bwe = false; |
+ }; |
+ |
+ // The Register* functions return true if registration succeeds, |
+ // false if the ssrc is already taken. |
+ |
+ // Registers the receiver responsible for an ssrc. The |media_type| |
+ // identifies the incoming transport, since we may use separate |
+ // transport for audio and video, with independent ssrc spaces. |
+ virtual bool AddReceiver(uint32_t ssrc, |
+ MediaType media_type, |
+ 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 bool RemoveReceiver(const RtpPacketReceiverInterface* receiver); |
+ |
+ // Used to represent auxillary sinks, currently used for FlexFec. |
+ // The responsible receiver must be registered first. |
+ virtual bool AddSink(uint32_t ssrc, |
the sun
2017/03/15 14:03:04
If the "RtpTransportControllerReceiveInterface"s p
nisse-webrtc
2017/03/15 14:55:00
Agree it's a bit confusing, but they have differen
|
+ MediaType media_type, |
+ 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 bool RemoveSink(const RtpPacketSinkInterface* sink); |
the sun
2017/03/15 14:03:04
= 0
Note: https://google.github.io/styleguide/cpp
nisse-webrtc
2017/03/15 14:55:00
Done.
|
+ |
+#if 0 |
+ // TODO(nisse): Not yet implemented. |
+ // Incoming packets with unknown ssrcs represent unsignalled |
+ // streams. We dispatch on payload type and media type. The receiver |
+ // will typically create a new RtpReceiver to pass the packet to. A |
+ // true return value from the callback means that we should retry |
+ // lookup. |
+ virtual bool AddPayload(uint8_t payload_type, MediaType media_type, |
+ RtpPacketReceiverInterface *receiver) = 0; |
+#endif |
+ // 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( |
+ MediaType media_type, |
+ rtc::ArrayView<const uint8_t> packet) = 0; |
+ |
+ virtual ~RtpTransportControllerReceiveInterface(); |
+ |
+ // Creates the default implementation. |
+ static RtpTransportControllerReceiveInterface* Create( |
+ RtpPacketObserverInterface* observer); |
+}; |
+ |
+// Callback invoked for all processed RTP packets, used for feeding |
+// the congestion controller (which is more tightly coupled to the |
+// send side). |
+class RtpPacketObserverInterface { |
+ public: |
+ virtual void OnRtpPacket( |
+ // TODO(nisse): Add media type / transport id to the RtpParsedPacket. |
+ MediaType media_type, |
+ const RtpTransportControllerReceiveInterface::Config, |
+ const RtpPacketReceived& packet) = 0; |
+ virtual ~RtpPacketObserverInterface(); |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_H_ |