nils/shairport-sync
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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.

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This commit is contained in:
Mike Brady
2018-04-19 10:16:28 +01:00
parent ad48056686 18c3b3afe8
commit 3e72bb216e
2 changed files with 17 additions and 17 deletions
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player.c
+2 -2
View File
@@ -1694,7 +1694,7 @@ static void *player_thread_func(void *arg) {
// 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 @@ static void *player_thread_func(void *arg) {
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...
rtp.c
+15 -15
View File
@@ -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 @@ void *rtp_audio_receiver(void *arg) {
(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 @@ 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 @@ void *rtp_control_receiver(void *arg) {
// 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 @@ void *rtp_control_receiver(void *arg) {
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.");
@@ -469,7 +469,7 @@ 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++;
@@ -717,14 +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 +817,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
"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 +853,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
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 +911,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
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;
@@ -959,7 +959,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
char req[8]; // *not* a standard RTCP NACK
req[0] = 0x80;
req[1] = (char)0x55 | (char)0x80; // Apple 'resend'
*(unsigned short *)(req + 2) = htons(1); // our seqnum
*(unsigned short *)(req + 2) = htons(1); // our sequence number
*(unsigned short *)(req + 4) = htons(first); // missed seqnum
*(unsigned short *)(req + 6) = htons(count); // count
socklen_t msgsize = sizeof(struct sockaddr_in);
@@ -975,7 +975,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
resend_error_backoff_time)) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
if (sendto(conn->audio_socket, req, sizeof(req), 0,
if (sendto(conn->control_socket, req, sizeof(req), 0,
(struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
char em[1024];
strerror_r(errno, em, sizeof(em));
@@ -985,7 +985,7 @@ void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
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 {