Remove add/removal of ViEReceiver RTP modules.

webrtc/video/vie_receiver.cc

 /*
  *  Copyright (c) 2012 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 "webrtc/video/vie_receiver.h"
 
 #include <vector>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/timestamp_extrapolator.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
 static const int kPacketLogIntervalMs = 10000;
 
 ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm,
                          RemoteBitrateEstimator* remote_bitrate_estimator,
                          RtpFeedback* rtp_feedback)
     : clock_(Clock::GetRealTimeClock()),
+      vcm_(module_vcm),
+      remote_bitrate_estimator_(remote_bitrate_estimator),
+      ntp_estimator_(clock_),
+      rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
       rtp_header_parser_(RtpHeaderParser::Create()),
-      rtp_payload_registry_(
-          new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
-      rtp_receiver_(
-          RtpReceiver::CreateVideoReceiver(clock_,
-                                           this,
-                                           rtp_feedback,
-                                           rtp_payload_registry_.get())),
+      rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
+                                                     this,
+                                                     rtp_feedback,
+                                                     &rtp_payload_registry_)),
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       fec_receiver_(FecReceiver::Create(this)),
-      rtp_rtcp_(NULL),
-      vcm_(module_vcm),
-      remote_bitrate_estimator_(remote_bitrate_estimator),
-      ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
       receiving_(false),
       restored_packet_in_use_(false),
       last_packet_log_ms_(-1) {}
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE(
         "WebRTC.Video.ReceivedFecPacketsInPercent",
         static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
                              static_cast<int>(counter.num_recovered_packets *
                                               100 / counter.num_fec_packets));
   }
 }
 
 bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
   int8_t old_pltype = -1;
-  if (rtp_payload_registry_->ReceivePayloadType(video_codec.plName,
+  if (rtp_payload_registry_.ReceivePayloadType(video_codec.plName,
                                                 kVideoPayloadTypeFrequency,
                                                 0,
                                                 video_codec.maxBitrate,
                                                 &old_pltype) != -1) {
-    rtp_payload_registry_->DeRegisterReceivePayload(old_pltype);
+    rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
   }
 
-  return RegisterPayload(video_codec);
-}
-
-bool ViEReceiver::RegisterPayload(const VideoCodec& video_codec) {
-  return rtp_receiver_->RegisterReceivePayload(video_codec.plName,
-                                               video_codec.plType,
-                                               kVideoPayloadTypeFrequency,
-                                               0,
-                                               video_codec.maxBitrate) == 0;
+  return rtp_receiver_->RegisterReceivePayload(
+             video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
+             0, 0) == 0;
 }
 
 void ViEReceiver::SetNackStatus(bool enable,
                                 int max_nack_reordering_threshold) {
   if (!enable) {
     // Reset the threshold back to the lower default threshold when NACK is
     // disabled since we no longer will be receiving retransmissions.
     max_nack_reordering_threshold = kDefaultMaxReorderingThreshold;
   }
   rtp_receive_statistics_->SetMaxReorderingThreshold(
       max_nack_reordering_threshold);
   rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff);
 }
 
 void ViEReceiver::SetRtxPayloadType(int payload_type,
                                     int associated_payload_type) {
-  rtp_payload_registry_->SetRtxPayloadType(payload_type,
+  rtp_payload_registry_.SetRtxPayloadType(payload_type,
                                            associated_payload_type);
 }
 
 void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
-  rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(val);
+  rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(val);
 }
 
 void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
-  rtp_payload_registry_->SetRtxSsrc(ssrc);
+  rtp_payload_registry_.SetRtxSsrc(ssrc);
 }
 
 bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
-  return rtp_payload_registry_->GetRtxSsrc(ssrc);
+  return rtp_payload_registry_.GetRtxSsrc(ssrc);
 }
 
 bool ViEReceiver::IsFecEnabled() const {
-  return rtp_payload_registry_->ulpfec_payload_type() > -1;
+  return rtp_payload_registry_.ulpfec_payload_type() > -1;
 }
 
 uint32_t ViEReceiver::GetRemoteSsrc() const {
   return rtp_receiver_->SSRC();
 }
 
 int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
   return rtp_receiver_->CSRCs(csrcs);
 }
 
-void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) {
-  rtp_rtcp_ = module;
+void ViEReceiver::Init(const std::vector<RtpRtcp*>& modules) {
+  rtp_rtcp_ = modules;
 }
 
 RtpReceiver* ViEReceiver::GetRtpReceiver() const {
   return rtp_receiver_.get();
 }
 
-void ViEReceiver::RegisterRtpRtcpModules(
-    const std::vector<RtpRtcp*>& rtp_modules) {
-  rtc::CritScope lock(&receive_cs_);
-  // Only change the "simulcast" modules, the base module can be accessed
-  // without a lock whereas the simulcast modules require locking as they can be
-  // changed in runtime.
-  rtp_rtcp_simulcast_ =
-      std::vector<RtpRtcp*>(rtp_modules.begin() + 1, rtp_modules.end());
-}
-
 void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
                                                   int id) {
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
 
 int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
                                            const size_t payload_size,
                                            const WebRtcRTPHeader* rtp_header) {
   RTC_DCHECK(vcm_);
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
-      ntp_estimator_->Estimate(rtp_header->header.timestamp);
+      ntp_estimator_.Estimate(rtp_header->header.timestamp);
   if (vcm_->IncomingPacket(payload_data,
                            payload_size,
                            rtp_header_with_ntp) != 0) {
     // Check this...
     return -1;
   }
   return 0;
 }
 
 bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
@@ skipping 47 matching lines @@
       LOG(LS_INFO) << ss.str();
       last_packet_log_ms_ = now_ms;
     }
   }
 
   remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
                                             header, true);
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
-  rtp_payload_registry_->SetIncomingPayloadType(header);
+  rtp_payload_registry_.SetIncomingPayloadType(header);
   bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
   // Update receive statistics after ReceivePacket.
   // Receive statistics will be reset if the payload type changes (make sure
   // that the first packet is included in the stats).
   rtp_receive_statistics_->IncomingPacket(
       header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
   return ret;
 }
 
 bool ViEReceiver::ReceivePacket(const uint8_t* packet,
                                 size_t packet_length,
                                 const RTPHeader& header,
                                 bool in_order) {
-  if (rtp_payload_registry_->IsEncapsulated(header)) {
+  if (rtp_payload_registry_.IsEncapsulated(header)) {
     return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
   }
   const uint8_t* payload = packet + header.headerLength;
   assert(packet_length >= header.headerLength);
   size_t payload_length = packet_length - header.headerLength;
   PayloadUnion payload_specific;
-  if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
+  if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
     return false;
   }
   return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
                                           payload_specific, in_order);
 }
 
 bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
                                                     size_t packet_length,
                                                     const RTPHeader& header) {
-  if (rtp_payload_registry_->IsRed(header)) {
-    int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type();
+  if (rtp_payload_registry_.IsRed(header)) {
+    int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
     if (packet[header.headerLength] == ulpfec_pt) {
       rtp_receive_statistics_->FecPacketReceived(header, packet_length);
       // Notify vcm about received FEC packets to avoid NACKing these packets.
       NotifyReceiverOfFecPacket(header);
     }
     if (fec_receiver_->AddReceivedRedPacket(
             header, packet, packet_length, ulpfec_pt) != 0) {
       return false;
     }
     return fec_receiver_->ProcessReceivedFec() == 0;
-  } else if (rtp_payload_registry_->IsRtx(header)) {
+  } else if (rtp_payload_registry_.IsRtx(header)) {
     if (header.headerLength + header.paddingLength == packet_length) {
       // This is an empty packet and should be silently dropped before trying to
       // parse the RTX header.
       return true;
     }
     // Remove the RTX header and parse the original RTP header.
     if (packet_length < header.headerLength)
       return false;
     if (packet_length > sizeof(restored_packet_))
       return false;
     rtc::CritScope lock(&receive_cs_);
     if (restored_packet_in_use_) {
       LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
       return false;
     }
-    if (!rtp_payload_registry_->RestoreOriginalPacket(
+    if (!rtp_payload_registry_.RestoreOriginalPacket(
             restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
             header)) {
       LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
                       << header.ssrc << " payload type: "
                       << static_cast<int>(header.payloadType);
       return false;
     }
     restored_packet_in_use_ = true;
     bool ret = OnRecoveredPacket(restored_packet_, packet_length);
     restored_packet_in_use_ = false;
     return ret;
   }
   return false;
 }
 
 void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
   int8_t last_media_payload_type =
-      rtp_payload_registry_->last_received_media_payload_type();
+      rtp_payload_registry_.last_received_media_payload_type();
   if (last_media_payload_type < 0) {
     LOG(LS_WARNING) << "Failed to get last media payload type.";
     return;
   }
   // Fake an empty media packet.
   WebRtcRTPHeader rtp_header = {};
   rtp_header.header = header;
   rtp_header.header.payloadType = last_media_payload_type;
   rtp_header.header.paddingLength = 0;
   PayloadUnion payload_specific;
-  if (!rtp_payload_registry_->GetPayloadSpecifics(last_media_payload_type,
+  if (!rtp_payload_registry_.GetPayloadSpecifics(last_media_payload_type,
                                                   &payload_specific)) {
     LOG(LS_WARNING) << "Failed to get payload specifics.";
     return;
   }
   rtp_header.type.Video.codec = payload_specific.Video.videoCodecType;
   rtp_header.type.Video.rotation = kVideoRotation_0;
   if (header.extension.hasVideoRotation) {
     rtp_header.type.Video.rotation =
         ConvertCVOByteToVideoRotation(header.extension.videoRotation);
   }
   OnReceivedPayloadData(NULL, 0, &rtp_header);
 }
 
 bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
                               size_t rtcp_packet_length) {

[stefan-webrtc, 2016/02/29 09:15:04]

DCHECK or CHECK

[pbos-webrtc, 2016/02/29 09:58:57]

Done.

+  assert(!rtp_rtcp_.empty());  // Should be set by owner at construction time.
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
-
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_)
-      rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
-  }
-  assert(rtp_rtcp_);  // Should be set by owner at construction time.
-  int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
-  if (ret != 0) {
-    return false;
   }
 
+  // Deliver to simulcast modules first, then base one.

[stefan-webrtc, 2016/02/29 09:15:04]

Do you know why? Seems like it would be cleaner to just loop over the modules
and deliver rtcp to all of them.

[pbos-webrtc, 2016/02/29 09:58:57]

Done.

+  for (size_t i = 1; i < rtp_rtcp_.size(); ++i)
+    rtp_rtcp_[i]->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+  // TODO(pbos): Is it important to check the return code here?
+  if (rtp_rtcp_[0]->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length) != 0)
+    return false;
+
   int64_t rtt = 0;
-  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
+  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
   if (rtt == 0) {
     // Waiting for valid rtt.
     return true;
   }
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;
-  if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
-                                &rtp_timestamp)) {
+  if (rtp_rtcp_[0]->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
+                              &rtp_timestamp) != 0) {
     // Waiting for RTCP.
     return true;
   }
-  ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+  ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
 
   return true;
 }
 
 void ViEReceiver::StartReceive() {
   rtc::CritScope lock(&receive_cs_);
   receiving_ = true;
 }
 
 void ViEReceiver::StopReceive() {
   rtc::CritScope lock(&receive_cs_);
   receiving_ = false;
 }
 
 ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
   return rtp_receive_statistics_.get();
 }
 
 bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const {
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   return statistician->IsPacketInOrder(header.sequenceNumber);
 }
 
 bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header,
                                         bool in_order) const {
   // Retransmissions are handled separately if RTX is enabled.
-  if (rtp_payload_registry_->RtxEnabled())
+  if (rtp_payload_registry_.RtxEnabled())
     return false;
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
-  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
+  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc