Rename ViEReceiver and move ownership to VideoReceiveStream.

webrtc/video/rtp_stream_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 "webrtc/video/rtp_stream_receiver.h"
 
 #include <vector>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/pacing/packet_router.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/rtp_cvo.h"
@@ skipping 38 matching lines @@
   rtp_rtcp->SetSendingStatus(false);
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
 }
 
 
 static const int kPacketLogIntervalMs = 10000;
 
-ViEReceiver::ViEReceiver(vcm::VideoReceiver* video_receiver,
-                         RemoteBitrateEstimator* remote_bitrate_estimator,
-                         RtpFeedback* rtp_feedback,
-                         Transport* transport,
-                         RtcpRttStats* rtt_stats,
-                         PacedSender* paced_sender,
-                         PacketRouter* packet_router)
+RtpStreamReceiver::RtpStreamReceiver(
+    vcm::VideoReceiver* video_receiver,
+    RemoteBitrateEstimator* remote_bitrate_estimator,
+    Transport* transport,
+    RtcpRttStats* rtt_stats,
+    PacedSender* paced_sender,
+    PacketRouter* packet_router)
     : clock_(Clock::GetRealTimeClock()),
       video_receiver_(video_receiver),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       packet_router_(packet_router),
       ntp_estimator_(clock_),
       rtp_payload_registry_(RTPPayloadStrategy::CreateStrategy(false)),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
                                                      this,
-                                                     rtp_feedback,
+                                                     this,
                                                      &rtp_payload_registry_)),
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       fec_receiver_(FecReceiver::Create(this)),
       receiving_(false),
       restored_packet_in_use_(false),
       last_packet_log_ms_(-1),
       rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
                                     transport,
                                     rtt_stats,
                                     &rtcp_packet_type_counter_observer_,
                                     remote_bitrate_estimator_,
                                     paced_sender,
                                     packet_router)) {
   packet_router_->AddRtpModule(rtp_rtcp_.get());
   rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp);
 }
 
-ViEReceiver::~ViEReceiver() {
+RtpStreamReceiver::~RtpStreamReceiver() {
   packet_router_->RemoveRtpModule(rtp_rtcp_.get());
   UpdateHistograms();
 }
 
-void ViEReceiver::UpdateHistograms() {
+void RtpStreamReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
     RTC_LOGGED_HISTOGRAM_PERCENTAGE(
         "WebRTC.Video.ReceivedFecPacketsInPercent",
         static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
     RTC_LOGGED_HISTOGRAM_PERCENTAGE(
         "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
         static_cast<int>(counter.num_recovered_packets * 100 /
                          counter.num_fec_packets));
   }
 }
 
-bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
+bool RtpStreamReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
   int8_t old_pltype = -1;
   if (rtp_payload_registry_.ReceivePayloadType(
           video_codec.plName, kVideoPayloadTypeFrequency, 0,
           video_codec.maxBitrate, &old_pltype) != -1) {
     rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
   }
 
   return rtp_receiver_->RegisterReceivePayload(
              video_codec.plName, video_codec.plType, kVideoPayloadTypeFrequency,
              0, 0) == 0;
 }
 
-void ViEReceiver::SetNackStatus(bool enable,
-                                int max_nack_reordering_threshold) {
+void RtpStreamReceiver::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) {
+void RtpStreamReceiver::SetRtxPayloadType(int payload_type,
+                                          int associated_payload_type) {
   rtp_payload_registry_.SetRtxPayloadType(payload_type,
                                           associated_payload_type);
 }
 
-void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
+void RtpStreamReceiver::SetUseRtxPayloadMappingOnRestore(bool val) {
   rtp_payload_registry_.set_use_rtx_payload_mapping_on_restore(val);
 }
 
-void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
+void RtpStreamReceiver::SetRtxSsrc(uint32_t ssrc) {
   rtp_payload_registry_.SetRtxSsrc(ssrc);
 }
 
-bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
+bool RtpStreamReceiver::GetRtxSsrc(uint32_t* ssrc) const {
   return rtp_payload_registry_.GetRtxSsrc(ssrc);
 }
 
-bool ViEReceiver::IsFecEnabled() const {
+bool RtpStreamReceiver::IsFecEnabled() const {
   return rtp_payload_registry_.ulpfec_payload_type() > -1;
 }
 
-uint32_t ViEReceiver::GetRemoteSsrc() const {
+uint32_t RtpStreamReceiver::GetRemoteSsrc() const {
   return rtp_receiver_->SSRC();
 }
 
-int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
+int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
   return rtp_receiver_->CSRCs(csrcs);
 }
 
-RtpReceiver* ViEReceiver::GetRtpReceiver() const {
+RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const {
   return rtp_receiver_.get();
 }
 
-void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
-                                                  int id) {
+void RtpStreamReceiver::EnableReceiveRtpHeaderExtension(
+    const std::string& extension, int id) {
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
 
-void ViEReceiver::RegisterRtcpPacketTypeCounterObserver(
+void RtpStreamReceiver::RegisterRtcpPacketTypeCounterObserver(
     RtcpPacketTypeCounterObserver* observer) {
   rtcp_packet_type_counter_observer_.Set(observer);
 }
 
 
-int32_t ViEReceiver::OnReceivedPayloadData(const uint8_t* payload_data,
-                                           const size_t payload_size,
-                                           const WebRtcRTPHeader* rtp_header) {
+int32_t RtpStreamReceiver::OnReceivedPayloadData(
+    const uint8_t* payload_data,
+    const size_t payload_size,
+    const WebRtcRTPHeader* rtp_header) {
   RTC_DCHECK(video_receiver_);
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
       ntp_estimator_.Estimate(rtp_header->header.timestamp);
   if (video_receiver_->IncomingPacket(payload_data, payload_size,
                                       rtp_header_with_ntp) != 0) {
     // Check this...
     return -1;
   }
   return 0;
 }
 
-bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
-                                    size_t rtp_packet_length) {
+bool RtpStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
+                                          size_t rtp_packet_length) {
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) {
     return false;
   }
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
   bool in_order = IsPacketInOrder(header);
   return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
 }
 
-bool ViEReceiver::DeliverRtp(const uint8_t* rtp_packet,
-                             size_t rtp_packet_length,
-                             const PacketTime& packet_time) {
+// TODO(pbos): Remove as soon as audio can handle a changing payload type
+// without this callback.
+int32_t RtpStreamReceiver::OnInitializeDecoder(
+    const int8_t payload_type,
+    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+    const int frequency,
+    const size_t channels,
+    const uint32_t rate) {
+  RTC_NOTREACHED();
+  return 0;
+}
+
+void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) {
+  rtp_rtcp_->SetRemoteSSRC(ssrc);
+}
+
+bool RtpStreamReceiver::DeliverRtp(const uint8_t* rtp_packet,
+                                   size_t rtp_packet_length,
+                                   const PacketTime& packet_time) {
   RTC_DCHECK(remote_bitrate_estimator_);
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
   }
 
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
@@ skipping 34 matching lines @@
   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) {
+bool RtpStreamReceiver::ReceivePacket(const uint8_t* packet,
+                                      size_t packet_length,
+                                      const RTPHeader& header,
+                                      bool in_order) {
   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,
                                                  &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) {
+bool RtpStreamReceiver::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 (packet[header.headerLength] == ulpfec_pt) {
       rtp_receive_statistics_->FecPacketReceived(header, packet_length);
       // Notify video_receiver about received FEC packets to avoid NACKing these
       // packets.
       NotifyReceiverOfFecPacket(header);
     }
     if (fec_receiver_->AddReceivedRedPacket(
             header, packet, packet_length, ulpfec_pt) != 0) {
@@ skipping 25 matching lines @@
       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) {
+void RtpStreamReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) {
   int8_t last_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,
                                                  &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(nullptr, 0, &rtp_header);
 }
 
-bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
-                              size_t rtcp_packet_length) {
+bool RtpStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
+                                    size_t rtcp_packet_length) {
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
   }
 
   rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
 
   int64_t rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
   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 (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
                            &rtp_timestamp) != 0) {
     // Waiting for RTCP.
     return true;
   }
   ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
 
   return true;
 }
 
-void ViEReceiver::StartReceive() {
+void RtpStreamReceiver::StartReceive() {
   rtc::CritScope lock(&receive_cs_);
   receiving_ = true;
 }
 
-void ViEReceiver::StopReceive() {
+void RtpStreamReceiver::StopReceive() {
   rtc::CritScope lock(&receive_cs_);
   receiving_ = false;
 }
 
-ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const {
+ReceiveStatistics* RtpStreamReceiver::GetReceiveStatistics() const {
   return rtp_receive_statistics_.get();
 }
 
-bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const {
+bool RtpStreamReceiver::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 {
+bool RtpStreamReceiver::IsPacketRetransmitted(const RTPHeader& header,
+                                              bool in_order) const {
   // Retransmissions are handled separately if RTX is enabled.
   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(), nullptr, nullptr, &min_rtt, nullptr);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc