Delete class SSRCDatabase, and its global ssrc registry.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index a8345435f90e7d3c65ccb2e5bae2a8ad5bfa1c99..be921624d33682fb232d53eab61016689d23787e 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -575,9 +575,8 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
 MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
     int32_t id,
     AudioFrame* audioFrame) {
-  unsigned int ssrc;
-  RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
-  event_log_proxy_->LogAudioPlayout(ssrc);
+
+  event_log_proxy_->LogAudioPlayout(0);
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   bool muted;
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
@@ -896,6 +895,7 @@ Channel::Channel(int32_t channelId,
       _includeAudioLevelIndication(false),
       transport_overhead_per_packet_(0),
       rtp_overhead_per_packet_(0),
+      random_(rtc::TimeNanos()),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
@@ -935,6 +935,8 @@ Channel::Channel(int32_t channelId,
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
+
+  SetLocalSSRC(random_.Rand<uint32_t>());
 }
 
 Channel::~Channel() {