Delete class SSRCDatabase, and its global ssrc registry.

webrtc/voice_engine/channel.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 565 matching lines @@
       rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
   if (header.payload_type_frequency < 0)
     return false;
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
 MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
     int32_t id,
     AudioFrame* audioFrame) {
   unsigned int ssrc;
-  RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+  RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
   event_log_proxy_->LogAudioPlayout(ssrc);
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   bool muted;
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
                                      &muted) == -1) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
     // In all likelihood, the audio in this frame is garbage. We return an
     // error so that the audio mixer module doesn't add it to the mix. As
     // a result, it won't be played out and the actions skipped here are
@@ skipping 2422 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc