From: Mike Brady <mikebrady@eircom.net>
Date: Thu, 19 Apr 2018 09:16:28 +0000 (+0100)
Subject: Send resend requests via control channel rather than the audio channel. This makes... 
X-Git-Tag: 3.2RC4~4
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=3e72bb216e990aae11757bf0a641af763afb2ebd;p=thirdparty%2Fshairport-sync.git

Send resend requests via control channel rather than the audio channel. This makes it compatilbe iwht AirAudio and confformant with the unofficial guide to the airplay protocol. Brown bag time.
---

3e72bb216e990aae11757bf0a641af763afb2ebd
diff --cc player.c
index 153ae227,c333e318..704a23dc
--- a/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 0f8d3dcf,e0718148..446ba53f
--- a/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 {