Delete class SSRCDatabase, and its global ssrc registry.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 9ceebd99e02257027ebf137c7dfd59b8054e89ad..802e3e0133a3f67b06d08c7a62f160a4fbd4bd4b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -111,10 +111,8 @@ RTPSender::RTPSender(
       send_packet_observer_(send_packet_observer),
       bitrate_callback_(bitrate_callback),
       // RTP variables
-      ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
       remote_ssrc_(0),
       sequence_number_forced_(false),
-      ssrc_forced_(false),
       last_rtp_timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
@@ -128,11 +126,6 @@ RTPSender::RTPSender(
       send_side_bwe_with_overhead_(
           webrtc::field_trial::FindFullName(
               "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
-  ssrc_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_ != 0);
-  ssrc_rtx_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_rtx_ != 0);
-
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
   // Random start, 16 bits. Can't be 0.
@@ -157,12 +150,6 @@ RTPSender::~RTPSender() {
   // variables but we grab them in all other methods. (what's the design?)
   // Start documenting what thread we're on in what method so that it's easier
   // to understand performance attributes and possibly remove locks.
-  if (remote_ssrc_ != 0) {
-    ssrc_db_->ReturnSSRC(remote_ssrc_);
-  }
-  ssrc_db_->ReturnSSRC(ssrc_);
-
-  SSRCDatabase::ReturnSSRCDatabase();
   while (!payload_type_map_.empty()) {
     std::map<int8_t, RtpUtility::Payload*>::iterator it =
         payload_type_map_.begin();
@@ -334,12 +321,13 @@ int RTPSender::RtxStatus() const {
 
 void RTPSender::SetRtxSsrc(uint32_t ssrc) {
   rtc::CritScope lock(&send_critsect_);
-  ssrc_rtx_ = ssrc;
+  ssrc_rtx_.emplace(ssrc);
 }
 
 uint32_t RTPSender::RtxSsrc() const {
   rtc::CritScope lock(&send_critsect_);
-  return ssrc_rtx_;
+  RTC_DCHECK(ssrc_rtx_);
+  return ssrc_rtx_.value_or(0);

[danilchap, 2017/02/14 13:23:05]

there is a DCHECK above, so no need to handle empty optional case:
return *ssrc_rtx_;

[nisse-webrtc, 2017/02/14 13:44:20]

Done. (Then the DCHECK is a bit redundant too, we'll always crash one way or the
other if the member is unset. But let's keep the DCHECK for clarity.)

 }
 
 void RTPSender::SetRtxPayloadType(int payload_type,
@@ -401,7 +389,12 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
   {
     // Drop this packet if we're not sending media packets.
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_) {
+      LOG(LS_ERROR)  << "SSRC unset";
+      return false;
+    }
+
+    ssrc = *ssrc_;
     sequence_number = sequence_number_;
     rtp_timestamp = timestamp_offset_ + capture_timestamp;
     if (transport_frame_id_out)
@@ -521,7 +514,13 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
         if (!audio_configured_ && !last_packet_marker_bit_) {
           break;
         }
-        ssrc = ssrc_;
+        if (!ssrc_) {
+          LOG(LS_ERROR)  << "SSRC unset";
+          return 0;
+        }
+
+        ssrc = *ssrc_;
+
         sequence_number = sequence_number_;
         ++sequence_number_;
         payload_type = payload_type_;
@@ -545,7 +544,11 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
               (now_ms - last_timestamp_time_ms_) * kTimestampTicksPerMs;
           capture_time_ms += (now_ms - last_timestamp_time_ms_);
         }
-        ssrc = ssrc_rtx_;
+        if (!ssrc_rtx_) {
+          LOG(LS_ERROR)  << "RTX SSRC unset";
+          return 0;
+        }
+        ssrc = *ssrc_rtx_;
         sequence_number = sequence_number_rtx_;
         ++sequence_number_rtx_;
         payload_type = rtx_payload_type_map_.begin()->second;
@@ -919,7 +922,9 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
   int max_delay_ms = 0;
   {
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_)
+      return;
+    ssrc = *ssrc_;
   }
   {
     rtc::CritScope cs(&statistics_crit_);
@@ -959,7 +964,9 @@ void RTPSender::ProcessBitrate() {
   uint32_t ssrc;
   {
     rtc::CritScope lock(&send_critsect_);
-    ssrc = ssrc_;
+    if (!ssrc_)
+      return;
+    ssrc = *ssrc_;
   }
 
   rtc::CritScope lock(&statistics_crit_);
@@ -993,7 +1000,8 @@ std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
   rtc::CritScope lock(&send_critsect_);
   std::unique_ptr<RtpPacketToSend> packet(
       new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
-  packet->SetSsrc(ssrc_);
+  RTC_DCHECK(ssrc_);
+  packet->SetSsrc(*ssrc_);
   packet->SetCsrcs(csrcs_);
   // Reserve extensions, if registered, RtpSender set in SendToNetwork.
   packet->ReserveExtension<AbsoluteSendTime>();
@@ -1010,7 +1018,7 @@ bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
   rtc::CritScope lock(&send_critsect_);
   if (!sending_media_)
     return false;
-  RTC_DCHECK_EQ(packet->Ssrc(), ssrc_);
+  RTC_DCHECK(packet->Ssrc() == ssrc_);
   packet->SetSequenceNumber(sequence_number_++);
 
   // Remember marker bit to determine if padding can be inserted with
@@ -1042,23 +1050,6 @@ bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
   return true;
 }
 
-void RTPSender::SetSendingStatus(bool enabled) {
-  if (!enabled) {
-    rtc::CritScope lock(&send_critsect_);
-    if (!ssrc_forced_) {
-      // Generate a new SSRC.
-      ssrc_db_->ReturnSSRC(ssrc_);
-      ssrc_ = ssrc_db_->CreateSSRC();
-      RTC_DCHECK(ssrc_ != 0);
-    }
-    // Don't initialize seq number if SSRC passed externally.
-    if (!sequence_number_forced_ && !ssrc_forced_) {
-      // Generate a new sequence number.
-      sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
-    }
-  }
-}
-
 void RTPSender::SetSendingMediaStatus(bool enabled) {
   rtc::CritScope lock(&send_critsect_);
   sending_media_ = enabled;
@@ -1079,29 +1070,14 @@ uint32_t RTPSender::TimestampOffset() const {
   return timestamp_offset_;
 }
 
-uint32_t RTPSender::GenerateNewSSRC() {
-  // If configured via API, return 0.
-  rtc::CritScope lock(&send_critsect_);
-
-  if (ssrc_forced_) {
-    return 0;
-  }
-  ssrc_ = ssrc_db_->CreateSSRC();
-  RTC_DCHECK(ssrc_ != 0);
-  return ssrc_;
-}
-
 void RTPSender::SetSSRC(uint32_t ssrc) {
   // This is configured via the API.
   rtc::CritScope lock(&send_critsect_);
 
-  if (ssrc_ == ssrc && ssrc_forced_) {
+  if (ssrc_ == ssrc) {
     return;  // Since it's same ssrc, don't reset anything.
   }
-  ssrc_forced_ = true;
-  ssrc_db_->ReturnSSRC(ssrc_);
-  ssrc_db_->RegisterSSRC(ssrc);
-  ssrc_ = ssrc;
+  ssrc_.emplace(ssrc);
   if (!sequence_number_forced_) {
     sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
   }
@@ -1109,7 +1085,7 @@ void RTPSender::SetSSRC(uint32_t ssrc) {
 
 uint32_t RTPSender::SSRC() const {
   rtc::CritScope lock(&send_critsect_);
-  return ssrc_;
+  return *ssrc_;
 }
 
 rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
@@ -1189,6 +1165,8 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
     if (!sending_media_)
       return nullptr;
 
+    RTC_DCHECK(ssrc_rtx_);
+
     // Replace payload type.
     auto kv = rtx_payload_type_map_.find(packet.PayloadType());
     if (kv == rtx_payload_type_map_.end())
@@ -1199,7 +1177,7 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
     rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
 
     // Replace SSRC.
-    rtx_packet->SetSsrc(ssrc_rtx_);
+    rtx_packet->SetSsrc(*ssrc_rtx_);
   }
 
   uint8_t* rtx_payload =