Delete class SSRCDatabase, and its global ssrc registry.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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.
  */
 
@@ skipping 82 matching lines @@
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
                      configuration.intra_frame_callback,
                      configuration.transport_feedback_callback,
                      configuration.bitrate_allocation_observer,
                      this),
       clock_(configuration.clock),
       audio_(configuration.audio),
-      collision_detected_(false),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
       packet_overhead_(28),  // IPV4 UDP.
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
@@ skipping 235 matching lines @@
   return state;
 }
 
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false
     if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
       LOG(LS_WARNING) << "Failed to send RTCP BYE";
     }
 
-    collision_detected_ = false;
-
     // Generate a new SSRC for the next "call" if false

[nisse-webrtc, 2017/01/31 14:23:42]

This is no longer true, the SetSendingStatus is a nop. 

Is this feature used for anything? If we need to select a new ssrc, where should
that allocation move?

[nisse-webrtc, 2017/02/01 10:52:50]

I just deleted this function and the below logic. It was used only for automatic
ssrc selection.

     rtp_sender_.SetSendingStatus(sending);
 
     // Make sure that RTCP objects are aware of our SSRC (it could have changed
     // Due to collision)
     uint32_t SSRC = rtp_sender_.SSRC();
     rtcp_sender_.SetSSRC(SSRC);
     SetRtcpReceiverSsrcs(SSRC);
 
     return 0;
   }
@@ skipping 406 matching lines @@
 bool ModuleRtpRtcpImpl::SetFecParameters(
     const FecProtectionParams& delta_params,
     const FecProtectionParams& key_params) {
   return rtp_sender_.SetFecParameters(delta_params, key_params);
 }
 
 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
   // Inform about the incoming SSRC.
   rtcp_sender_.SetRemoteSSRC(ssrc);
   rtcp_receiver_.SetRemoteSSRC(ssrc);
-
-  // Check for a SSRC collision.
-  if (rtp_sender_.SSRC() == ssrc && !collision_detected_) {
-    // If we detect a collision change the SSRC but only once.
-    collision_detected_ = true;
-    uint32_t new_ssrc = rtp_sender_.GenerateNewSSRC();
-    if (new_ssrc == 0) {
-      // Configured via API ignore.
-      return;
-    }
-    if (RtcpMode::kOff != rtcp_sender_.Status()) {
-      // Send RTCP bye on the current SSRC.
-      SendRTCP(kRtcpBye);
-    }
-    // Change local SSRC and inform all objects about the new SSRC.
-    rtcp_sender_.SetSSRC(new_ssrc);
-    SetRtcpReceiverSsrcs(new_ssrc);
-  }
 }
 
 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
                                     uint32_t* video_rate,
                                     uint32_t* fec_rate,
                                     uint32_t* nack_rate) const {
   *total_rate = rtp_sender_.BitrateSent();
   *video_rate = rtp_sender_.VideoBitrateSent();
   *fec_rate = rtp_sender_.FecOverheadRate();
   *nack_rate = rtp_sender_.NackOverheadRate();
@@ skipping 95 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc