Send resend requests via control channel rather than the audio channel. This makes...

diff --cc player.c
index 153ae227ca3f2a53de9fb14d3be6662ce09b4897,c333e3187bacd2dd85f53b54df8377571e92a995..704a23dcd351a20fe6307e0236343654c3f5a8e5
--- 1/player.c
--- 2/player.c
+++ b/player.c
@@@ -1694,7 -1694,7 +1694,7 @@@ static void *player_thread_func(void *a
          //          debug(3, "Play frame %d.", play_number);
          conn->play_number_after_flush++;
          if (inframe->timestamp == 0) {
--          debug(1, "Player has supplied a silent frame, (possibly frame %u) for play number %d.",
++          debug(3, "Player has supplied a silent frame, (possibly frame %u) for play number %d.",
                  SUCCESSOR(conn->last_seqno_read),play_number);
            conn->last_seqno_read = (SUCCESSOR(conn->last_seqno_read) &
                                     0xffff); // manage the packet out of sequence minder
@@@ -1848,7 -1848,7 +1848,7 @@@
                  SUCCESSOR(conn->last_seqno_read); // int32_t from seq_t, i.e. uint16_t, so okay.
              if (inframe->sequence_number !=
                  conn->last_seqno_read) { // seq_t, ei.e. uint16_t and int32_t, so okay
--              debug(1, "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u "
++              debug(2, "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u "
                         "and ab_write: %u.",
                      conn->last_seqno_read, inframe->sequence_number, conn->ab_read, conn->ab_write);
                conn->last_seqno_read = inframe->sequence_number; // reset warning...

diff --cc rtp.c
index 0f8d3dcfa5b3d3f0e716181b5b2f48b234c5e4e9,e0718148d489a3553705c064f79528509589630f..446ba53f2017673fe391633eaa55267de464307d
--- 1/rtp.c
--- 2/rtp.c
+++ b/rtp.c
@@@ -147,7 -147,7 +147,7 @@@ void *rtp_audio_receiver(void *arg) 
              last_seqno = seqno; // reset warning...
            }
          } else {
--          debug(2, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
++          debug(3, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
          }
  
          uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
@@@ -165,7 -165,7 +165,7 @@@
                (drand48() > config.diagnostic_drop_packet_fraction))
              player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
            else
--            debug(2, "Dropping audio packet %u to simulate a bad connection.", seqno);
++            debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
            continue;
          }
          if (type == 0x56 && seqno == 0) {
@@@ -319,7 -319,7 +319,7 @@@ void *rtp_control_receiver(void *arg) 
  
                  if (la != conn->latency) {
                    conn->latency = la;
--                  debug(2, "New latency detected: %" PRId64 ", sync latency: %" PRId64
++                  debug(3, "New latency detected: %" PRId64 ", sync latency: %" PRId64
                             ", minimum latency: %" PRId64 ", maximum "
                             "latency: %" PRId64 ", fixed offset: %" PRId64 ".",
                          la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
@@@ -366,13 -366,13 +366,13 @@@
              // debug(1,"Sync Time is %lld us late (local
              // times).",((local_time_now-reference_timestamp_time)*1000000)>>32);
            } else {
--            debug(1, "Sync packet received before we got a timing packet back.");
++            debug(2, "Sync packet received before we got a timing packet back.");
            }
          } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
            pktp = packet + 4;
            plen -= 4;
            seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
--          debug(2, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
++          debug(3, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
  
            uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
            int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
@@@ -391,7 -391,7 +391,7 @@@
            debug(1, "Control Receiver -- Unknown RTP packet of type 0x%02X length %d, ignored.",
                  packet[1], nread);
        } else {
--        debug(2, "Control Receiver -- dropping a packet to simulate a bad network.");
++        debug(3, "Control Receiver -- dropping a packet to simulate a bad network.");
        }
      } else {
        debug(1, "Control Receiver -- error receiving a packet.");
@@@ -469,9 -469,8 +469,9 @@@ void *rtp_timing_sender(void *arg) 
        }
  
      } else {
--      debug(2, "Timing Sender Thread -- dropping outgoing packet to simulate bad network.");
++      debug(3, "Timing Sender Thread -- dropping outgoing packet to simulate bad network.");
      }
 +
      request_number++;
  
      // this is to deal with the possibility of missing a timing_sender_stop signal.
@@@ -717,14 -718,14 +717,14 @@@ void *rtp_timing_receiver(void *arg) 
              // buffer_occupancy,
              //(return_time*1000000)>>32);
            } else {
--            debug(1, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
++            debug(2, "Time ping turnaround time: %lld us -- it looks like a timing ping was lost.",
                    rtus);
            }
          } else {
            debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
          }
        } else {
--        debug(2, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
++        debug(3, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
        }
      } else {
        debug(1, "Timing receiver -- error receiving a packet.");
@@@ -817,7 -818,7 +817,7 @@@ void rtp_setup(SOCKADDR *local, SOCKADD
           "SETUP call?");
    else {
  
--    debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport);
++    debug(3, "rtp_setup: cport=%d tport=%d.", cport, tport);
  
      // print out what we know about the client
      void *client_addr = NULL, *self_addr = NULL;
@@@ -853,7 -854,7 +853,7 @@@
      inet_ntop(conn->connection_ip_family, self_addr, conn->self_ip_string,
                sizeof(conn->self_ip_string));
  
--    debug(2, "Set up play connection from %s to self at %s on RTSP conversation thread %d.",
++    debug(2, "SETUP connection from %s to self at %s on RTSP conversation thread %d.",
            conn->client_ip_string, conn->self_ip_string, conn->connection_number);
  
      // set up a the record of the remote's control socket
@@@ -911,7 -912,7 +911,7 @@@
      conn->local_audio_port = bind_port(conn->connection_ip_family, conn->self_ip_string,
                                         conn->self_scope_id, &conn->audio_socket);
  
--    debug(2, "listening for audio, control and timing on ports %d, %d, %d.", conn->local_audio_port,
++    debug(3, "listening for audio, control and timing on ports %d, %d, %d.", conn->local_audio_port,
            conn->local_control_port, conn->local_timing_port);
  
      conn->reference_timestamp = 0;
@@@ -985,7 -986,7 +985,7 @@@ void rtp_request_resend(seq_t first, ui
            conn->rtp_time_of_last_resend_request_error_fp = 0;
          }
        } else {
--        debug(2, "Dropping resend request packet to simulate a bad network.");
++        debug(3, "Dropping resend request packet to simulate a bad network.");
        }
      }
    } else {