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Try to fix modulo calculations

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This commit is contained in:
Mike Brady
2018-10-18 17:44:50 +01:00
parent 5e16fdbbf9
commit de071aeffc
5 changed files with 119 additions and 141 deletions
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rtp.c
+19 -21
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@@ -179,7 +179,6 @@ void *rtp_audio_receiver(void *arg) {
}
uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
// if (packet[1]&0x10)
// debug(1,"Audio packet Extension bit set.");
@@ -191,7 +190,7 @@ void *rtp_audio_receiver(void *arg) {
if (plen >= 16) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction))
player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
player_put_packet(seqno, actual_timestamp, pktp, plen, conn);
else
debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
continue;
@@ -238,7 +237,7 @@ void *rtp_control_receiver(void *arg) {
uint8_t packet[2048], *pktp;
// struct timespec tn;
uint64_t remote_time_of_sync;
int64_t sync_rtp_timestamp;
uint32_t sync_rtp_timestamp;
ssize_t nread;
while (1) {
nread = recv(conn->control_socket, packet, sizeof(packet), 0);
@@ -304,14 +303,14 @@ void *rtp_control_receiver(void *arg) {
// debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync);
sync_rtp_timestamp = monotonic_timestamp(nctohl(&packet[16]), conn);
int64_t rtp_timestamp_less_latency = monotonic_timestamp(nctohl(&packet[4]), conn);
sync_rtp_timestamp = nctohl(&packet[16]);
uint32_t rtp_timestamp_less_latency = nctohl(&packet[4]);
// debug(1,"Sync timestamp is %u.",ntohl(*((uint32_t *)&packet[16])));
if (config.userSuppliedLatency) {
if (config.userSuppliedLatency != conn->latency) {
debug(1, "Using the user-supplied latency: %" PRId64 ".",
debug(1, "Using the user-supplied latency: %" PRIu32 ".",
config.userSuppliedLatency);
}
conn->latency = config.userSuppliedLatency;
@@ -330,8 +329,8 @@ void *rtp_control_receiver(void *arg) {
// Sigh, it would be nice to have a published protocol...
uint16_t flags = nctohs(&packet[2]);
int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency;
// debug(3, "Latency derived just from the sync packet is %" PRId64 " frames.", la);
uint32_t la = sync_rtp_timestamp - rtp_timestamp_less_latency; // note, this might loop around in modulo. Not sure if you'll get an error!
// debug(3, "Latency derived just from the sync packet is %" PRIu32 " frames.", la);
if ((flags == 7) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion <= 353)) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion >= 371))) {
la += config.fixedLatencyOffset;
@@ -347,20 +346,20 @@ void *rtp_control_receiver(void *arg) {
if ((conn->minimum_latency) && (conn->minimum_latency > la))
la = conn->minimum_latency;
const int max_frames = ((3 * BUFFER_FRAMES * 352) / 4) - 11025;
const uint32_t max_frames = ((3 * BUFFER_FRAMES * 352) / 4) - 11025;
if ((la < 0) || (la > max_frames)) {
warn("An out-of-range latency request of %" PRId64
" frames was ignored. Must be %d frames or less (44,100 frames per second). "
"Latency remains at %" PRId64 " frames.",
if (la > max_frames) {
warn("An out-of-range latency request of %" PRIu32
" frames was ignored. Must be %" PRIu32 " frames or less (44,100 frames per second). "
"Latency remains at %" PRIu32 " frames.",
la, max_frames, conn->latency);
} else {
if (la != conn->latency) {
conn->latency = la;
debug(3, "New latency detected: %" PRId64 ", sync latency: %" PRId64
", minimum latency: %" PRId64 ", maximum "
"latency: %" PRId64 ", fixed offset: %" PRId64 ".",
debug(3, "New latency detected: %" PRIu32 ", sync latency: %" PRIu32
", minimum latency: %" PRIu32 ", maximum "
"latency: %" PRIu32 ", fixed offset: %" PRIu32 ".",
la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
conn->maximum_latency, config.fixedLatencyOffset);
}
@@ -380,7 +379,7 @@ void *rtp_control_receiver(void *arg) {
if (remote_frame_time_interval) {
conn->remote_frame_rate =
(1.0 * (conn->reference_timestamp - conn->initial_reference_timestamp)) /
remote_frame_time_interval; // an IEEE double calculation with two 64-bit
remote_frame_time_interval; // an IEEE double calculation with a 32-bit numerator and 64-bit denominator
// integers
conn->remote_frame_rate = conn->remote_frame_rate *
(uint64_t)0x100000000; // this should just change the
@@ -395,7 +394,7 @@ void *rtp_control_receiver(void *arg) {
// this is for debugging
uint64_t old_remote_reference_time = conn->remote_reference_timestamp_time;
int64_t old_reference_timestamp = conn->reference_timestamp;
uint32_t old_reference_timestamp = conn->reference_timestamp;
// int64_t old_latency_delayed_timestamp = conn->latency_delayed_timestamp;
conn->remote_reference_timestamp_time = remote_time_of_sync;
@@ -444,14 +443,13 @@ void *rtp_control_receiver(void *arg) {
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);
pktp += 12;
plen -= 12;
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
player_put_packet(seqno, actual_timestamp, pktp, plen, conn);
continue;
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
@@ -1004,7 +1002,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport
}
}
void get_reference_timestamp_stuff(int64_t *timestamp, uint64_t *timestamp_time,
void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time,
uint64_t *remote_timestamp_time, rtsp_conn_info *conn) {
// types okay
debug_mutex_lock(&conn->reference_time_mutex, 1000, 1);