Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 16 matching lines @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/ptr_util.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream_impl.h"
-#include "webrtc/call/rtp_demuxer.h"
+#include "webrtc/call/rtp_transport_controller_receive.h"
 #include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
@@ skipping 158 matching lines @@
   // Audio, Video, and FlexFEC receive streams are owned by the client that
   // creates them.
   std::set<AudioReceiveStream*> audio_receive_streams_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
 
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
-  // TODO(nisse): Should eventually be part of injected
-  // RtpTransportControllerReceive, with a single demuxer in the bundled case.
-  RtpDemuxer audio_rtp_demuxer_ GUARDED_BY(receive_crit_);
-  RtpDemuxer video_rtp_demuxer_ GUARDED_BY(receive_crit_);
+  // TODO(nisse): Should eventually be injected at creation,
+  // with a single object in the bundled case.
+  RtpTransportControllerReceive audio_transport_receive_;
+  RtpTransportControllerReceive video_transport_receive_;
 
   // This extra map is used for receive processing which is
   // independent of media type.
 
   // TODO(nisse): In the RTP transport refactoring, we should have a
   // single mapping from ssrc to a more abstract receive stream, with
   // accessor methods for all configuration we need at this level.
   struct ReceiveRtpConfig {
     ReceiveRtpConfig() = default;  // Needed by std::map
     ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
@@ skipping 323 matching lines @@
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream =
-      new AudioReceiveStream(transport_send_->packet_router(), config,
-                             config_.audio_state, event_log_);
+      new AudioReceiveStream(
+          &audio_transport_receive_, transport_send_->packet_router(), config,
+          config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
-    audio_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     receive_rtp_config_[config.rtp.remote_ssrc] =
         ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
     audio_receive_streams_.insert(receive_stream);
 
     ConfigureSync(config.sync_group);
   }
   {
     ReadLockScoped read_lock(*send_crit_);
     auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
     if (it != audio_send_ssrcs_.end()) {
@@ skipping 11 matching lines @@
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   webrtc::internal::AudioReceiveStream* audio_receive_stream =
       static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
   {
     WriteLockScoped write_lock(*receive_crit_);
     const AudioReceiveStream::Config& config = audio_receive_stream->config();
     uint32_t ssrc = config.rtp.remote_ssrc;
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
-    size_t num_deleted = audio_rtp_demuxer_.RemoveSink(audio_receive_stream);
-    RTC_DCHECK(num_deleted == 1);
     audio_receive_streams_.erase(audio_receive_stream);
     const std::string& sync_group = audio_receive_stream->config().sync_group;
     const auto it = sync_stream_mapping_.find(sync_group);
     if (it != sync_stream_mapping_.end() &&
         it->second == audio_receive_stream) {
       sync_stream_mapping_.erase(it);
       ConfigureSync(sync_group);
     }
     receive_rtp_config_.erase(ssrc);
   }
@@ skipping 68 matching lines @@
   UpdateAggregateNetworkState();
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     webrtc::VideoReceiveStream::Config configuration) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   VideoReceiveStream* receive_stream =
-      new VideoReceiveStream(num_cpu_cores_, transport_send_->packet_router(),
-                             std::move(configuration),
-                             module_process_thread_.get(), call_stats_.get());
+      new VideoReceiveStream(
+          &video_transport_receive_, num_cpu_cores_,
+          transport_send_->packet_router(), std::move(configuration),
+          module_process_thread_.get(), call_stats_.get());
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
   ReceiveRtpConfig receive_config(config.rtp.extensions,
                                   UseSendSideBwe(config));
   {
     WriteLockScoped write_lock(*receive_crit_);
-    video_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     if (config.rtp.rtx_ssrc) {
-      video_rtp_demuxer_.AddSink(config.rtp.rtx_ssrc, receive_stream);
       // We record identical config for the rtx stream as for the main
       // stream. Since the transport_send_cc negotiation is per payload
       // type, we may get an incorrect value for the rtx stream, but
       // that is unlikely to matter in practice.
       receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
     }
     receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
     video_receive_streams_.insert(receive_stream);
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
   event_log_->LogVideoReceiveStreamConfig(config);
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl =
       static_cast<VideoReceiveStream*>(receive_stream);
   const VideoReceiveStream::Config& config = receive_stream_impl->config();
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
     // separate SSRC there can be either one or two.
-    size_t num_deleted = video_rtp_demuxer_.RemoveSink(receive_stream_impl);
-    RTC_DCHECK_GE(num_deleted, 1);
     receive_rtp_config_.erase(config.rtp.remote_ssrc);
     if (config.rtp.rtx_ssrc) {
       receive_rtp_config_.erase(config.rtp.rtx_ssrc);
     }
     video_receive_streams_.erase(receive_stream_impl);
     ConfigureSync(config.sync_group);
   }
 
   receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
       ->RemoveStream(config.rtp.remote_ssrc);
 
   UpdateAggregateNetworkState();
   delete receive_stream_impl;
 }
 
 FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
     const FlexfecReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   RecoveredPacketReceiver* recovered_packet_receiver = this;
+
   FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
-      config, recovered_packet_receiver, call_stats_->rtcp_rtt_stats(),
-      module_process_thread_.get());
-
+      &video_transport_receive_, config, recovered_packet_receiver,
+      call_stats_->rtcp_rtt_stats(), module_process_thread_.get());
   {
     WriteLockScoped write_lock(*receive_crit_);
-    video_rtp_demuxer_.AddSink(config.remote_ssrc, receive_stream);
-
-    for (auto ssrc : config.protected_media_ssrcs)
-      video_rtp_demuxer_.AddSink(ssrc, receive_stream);
-
     RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
                receive_rtp_config_.end());
     receive_rtp_config_[config.remote_ssrc] =
         ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
   }
 
   // TODO(brandtr): Store config in RtcEventLog here.
 
   return receive_stream;
 }
@@ skipping 10 matching lines @@
   {
     WriteLockScoped write_lock(*receive_crit_);
 
     const FlexfecReceiveStream::Config& config =
         receive_stream_impl->GetConfig();
     uint32_t ssrc = config.remote_ssrc;
     receive_rtp_config_.erase(ssrc);
 
     // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
     // destroyed.
-    video_rtp_demuxer_.RemoveSink(receive_stream_impl);
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
   }
 
   delete receive_stream_impl;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
@@ skipping 351 matching lines @@
   // on parsed_packet to the receive streams.
   rtc::Optional<RtpPacketReceived> parsed_packet =
       ParseRtpPacket(packet, length, &packet_time);
 
   if (!parsed_packet)
     return DELIVERY_PACKET_ERROR;
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   if (media_type == MediaType::AUDIO) {
-    if (audio_rtp_demuxer_.OnRtpPacket(*parsed_packet)) {
+    if (audio_transport_receive_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
       event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return DELIVERY_OK;
     }
   } else if (media_type == MediaType::VIDEO) {
-    if (video_rtp_demuxer_.OnRtpPacket(*parsed_packet)) {
+    if (video_transport_receive_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
       event_log_->LogRtpHeader(kIncomingPacket, media_type, packet, length);
       return DELIVERY_OK;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
@@ skipping 15 matching lines @@
 // audio packets with FlexFEC.
 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
   ReadLockScoped read_lock(*receive_crit_);
   rtc::Optional<RtpPacketReceived> parsed_packet =
       ParseRtpPacket(packet, length, nullptr);
   if (!parsed_packet)
     return;
 
   parsed_packet->set_recovered(true);
 
-  video_rtp_demuxer_.OnRtpPacket(*parsed_packet);
+  video_transport_receive_.OnRtpPacket(*parsed_packet);
 }
 
 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
                                      MediaType media_type) {
   auto it = receive_rtp_config_.find(packet.Ssrc());
   bool use_send_side_bwe =
       (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
 
   RTPHeader header;
   packet.GetHeader(&header);
@@ skipping 13 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc