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收到未知SSRC 包处理逻辑

if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
    return;
  }

  // Create an unsignaled receive stream for this previously not received ssrc.
  // If there already is N unsignaled receive streams, delete the oldest.
  // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
  uint32_t ssrc = 0;
  if (!GetRtpSsrc(packet.cdata(), packet.size(), &ssrc)) {
    return;
  }
  RTC_DCHECK(!absl::c_linear_search(unsignaled_recv_ssrcs_, ssrc));

  // Add new stream.
  StreamParams sp = unsignaled_stream_params_;
  sp.ssrcs.push_back(ssrc);
  RTC_LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
  if (!AddRecvStream(sp)) {
    RTC_LOG(LS_WARNING) << "Could not create unsignaled receive stream.";
    return;
  }
  unsignaled_recv_ssrcs_.push_back(ssrc);
  RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.NumOfUnsignaledStreams",
                              unsignaled_recv_ssrcs_.size(), 1, 100, 101);

  // Remove oldest unsignaled stream, if we have too many.
  if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
    uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
    RTC_DLOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
                      << remove_ssrc;
    RemoveRecvStream(remove_ssrc);
  }
  RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());

  SetOutputVolume(ssrc, default_recv_volume_);
  SetBaseMinimumPlayoutDelayMs(ssrc, default_recv_base_minimum_delay_ms_);

  // The default sink can only be attached to one stream at a time, so we hook
  // it up to the *latest* unsignaled stream we've seen, in order to support the
  // case where the SSRC of one unsignaled stream changes.
  if (default_sink_) {
    for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
      auto it = recv_streams_.find(drop_ssrc);
      it->second->SetRawAudioSink(nullptr);
    }
    std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
        new ProxySink(default_sink_.get()));
    SetRawAudioSink(ssrc, std::move(proxy_sink));
  }

  delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                                     packet, packet_time_us);
  RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);