Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/call/rtp_transport_controller_receive.cc

+/*
+ *  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 <map>
+#include <utility>
+#include <vector>
+
+#include "webrtc/call/rtp_transport_controller_receive.h"
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+class RtpTransportControllerReceive
+    : public RtpTransportControllerReceiveInterface {
+ public:
+  RtpTransportControllerReceive(RtpPacketObserverInterface* observer);
+
+  // ImplementRtpTransportControllerReceiveInterface
+  bool AddReceiver(uint32_t ssrc,
+                   MediaType media_type,
+                   const Config& config,
+                   RtpPacketReceiverInterface* receiver) override;
+  bool RemoveReceiver(const RtpPacketReceiverInterface* receiver) override;
+
+  bool AddSink(uint32_t ssrc,
+               MediaType media_type,
+               RtpPacketSinkInterface* sink) override;
+  bool RemoveSink(const RtpPacketSinkInterface* sink) override;
+
+#if 0
+  bool RegisterPayload(uint8_t payload_type, MediaType media_type,
+                       RtpPacketReceiverInterface *receiver) override;
+#endif
+  PacketReceiver::DeliveryStatus OnRtpPacket(
+      MediaType media_type,
+      int64_t arrival_time_ms,
+      rtc::ArrayView<const uint8_t> packet) override;
+
+  ~RtpTransportControllerReceive() override;
+
+ private:
+  struct Stream {
+    MediaType media_type = MediaType::ANY;
+    Config config;
+    RtpPacketReceiverInterface* receiver = nullptr;
+    std::vector<RtpPacketSinkInterface*> auxillary_sinks;
+
+    Stream(MediaType media_type,
+           Config config,
+           RtpPacketReceiverInterface* receiver)
+        : media_type(media_type), config(config), receiver(receiver) {}
+  };
+
+  Stream* LookupStream(uint32_t ssrc, MediaType media_type);
+
+  // Indexed by ssrc. We could use a map indexed by pairs (media_type, ssrc),
+  // except that we want special handling of MediaType:ANY specially.
+  std::multimap<uint32_t, Stream> streams_;
+  RtpPacketObserverInterface* observer_;
+};
+
+RtpTransportControllerReceive::RtpTransportControllerReceive(
+    RtpPacketObserverInterface* observer)
+    : observer_(observer) {}
+
+RtpTransportControllerReceive::~RtpTransportControllerReceive() {
+  RTC_DCHECK(streams_.empty());
+}
+
+RtpTransportControllerReceive::Stream*
+RtpTransportControllerReceive::LookupStream(uint32_t ssrc,
+                                            MediaType media_type) {
+  auto range = streams_.equal_range(ssrc);
+  for (auto it = range.first; it != range.second; ++it) {
+    if (media_type == MediaType::ANY || media_type == it->second.media_type)
+      return &it->second;
+  }
+  return nullptr;
+}
+
+bool RtpTransportControllerReceive::AddReceiver(
+    uint32_t ssrc,
+    MediaType media_type,
+    const Config& config,
+    RtpPacketReceiverInterface* receiver) {
+  if (LookupStream(ssrc, media_type)) {
+    return false;
+  }
+  streams_.insert(
+      std::pair<uint32_t, Stream>(ssrc, Stream(media_type, config, receiver)));
+  return true;
+}
+
+bool RtpTransportControllerReceive::RemoveReceiver(
+    const RtpPacketReceiverInterface* receiver) {
+  for (auto it = streams_.begin(); it != streams_.end(); ++it) {
+    if (it->second.receiver == receiver) {
+      streams_.erase(it);
+      return true;
+    }
+  }
+  return false;
+}
+
+bool RtpTransportControllerReceive::AddSink(uint32_t ssrc,
+                                            MediaType media_type,
+                                            RtpPacketSinkInterface* sink) {
+  Stream* stream = LookupStream(ssrc, media_type);
+  if (stream) {
+    stream->auxillary_sinks.push_back(sink);
+    return true;
+  }
+  return false;
+}
+
+bool RtpTransportControllerReceive::RemoveSink(
+    const RtpPacketSinkInterface* sink) {
+  bool found = false;
+  for (auto it : streams_) {
+    auto sinks_end = it.second.auxillary_sinks.end();
+    auto sinks_it =
+        std::remove(it.second.auxillary_sinks.begin(), sinks_end, sink);
+    if (sinks_it != sinks_end) {
+      it.second.auxillary_sinks.erase(sinks_it, sinks_end);
+      found = true;
+    }
+  }
+  return found;
+}
+
+PacketReceiver::DeliveryStatus RtpTransportControllerReceive::OnRtpPacket(
+    MediaType media_type,
+    int64_t arrival_time_ms,
+    rtc::ArrayView<const uint8_t> raw_packet) {
+  RtpPacketReceived parsed_packet;
+  if (!parsed_packet.Parse(raw_packet.data(), raw_packet.size()))
+    return PacketReceiver::DELIVERY_PACKET_ERROR;
+  parsed_packet.set_arrival_time_ms(arrival_time_ms);
+
+  Stream* stream = LookupStream(parsed_packet.Ssrc(), media_type);
+  if (!stream) {
+    // TODO(nisse): Lookup payload, for unsignalled streams.
+    return PacketReceiver::DELIVERY_UNKNOWN_SSRC;
+  }
+  if (!stream->receiver->OnRtpPacketReceive(&parsed_packet))
+    return PacketReceiver::DELIVERY_PACKET_ERROR;
+  for (auto it : stream->auxillary_sinks) {
+    it->OnRtpPacket(parsed_packet);
+  }
+  if (observer_)
+    observer_->OnRtpPacket(media_type, stream->config, parsed_packet);
+  return PacketReceiver::DELIVERY_OK;
+}
+
+//  static
+std::unique_ptr<RtpTransportControllerReceiveInterface>
+RtpTransportControllerReceiveInterface::Create(
+    RtpPacketObserverInterface* observer) {
+  return std::unique_ptr<RtpTransportControllerReceiveInterface>(
+      new RtpTransportControllerReceive(observer));
+}
+
+}  // namespace webrtc