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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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common.h
+2 -2
View File
@@ -134,8 +134,8 @@ typedef struct {
char *mdns_name;
mdns_backend *mdns;
int buffer_start_fill;
int64_t userSuppliedLatency; // overrides all other latencies -- use with caution
int64_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual
uint32_t userSuppliedLatency; // overrides all other latencies -- use with caution
uint32_t fixedLatencyOffset; // add this to all automatic latencies supplied to get the actual
// total latency
// the total latency will be limited to the min and max-latency values, if supplied
int daemonise;
player.c
+86 -106
View File
@@ -99,60 +99,15 @@
// static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)
void do_flush(int64_t timestamp, rtsp_conn_info *conn);
// make timestamps and seqnos definitely monotonic
// add an epoch to the timestamp. The monotonic timestamp guaranteed to start between 2^32 and 2^33
// frames and continue up to 2^63-1 frames
// if should never get into the negative range
// which is about 2*10^8 * 1,000 seconds at 384,000 frames per second -- about 2 trillion seconds or
// over 50,000 years.
// also, it won't reach zero until then, if ever, so we can safely say that a null monotonic
// timestamp can mean something special
int64_t monotonic_timestamp(uint32_t timestamp, rtsp_conn_info *conn) {
int64_t previous_value;
int64_t return_value;
if (conn->timestamp_epoch == 0) {
if (timestamp > conn->maximum_timestamp_interval)
conn->timestamp_epoch = 1;
else
conn->timestamp_epoch = 2;
previous_value = conn->timestamp_epoch;
previous_value <<= 32;
previous_value += timestamp;
} else {
previous_value = conn->timestamp_epoch;
previous_value <<= 32;
previous_value += conn->last_timestamp;
if (timestamp < conn->last_timestamp) {
// the incoming timestamp is less than the last one.
// if the difference is more than a minute, assume it's really from the next epoch
if ((conn->last_timestamp - timestamp) > conn->maximum_timestamp_interval)
conn->timestamp_epoch++;
} else {
// the incoming timestamp is greater than the last one.
// if the difference is more than a minute, assume it's really from the previous epoch
if ((timestamp - conn->last_timestamp) > conn->maximum_timestamp_interval)
conn->timestamp_epoch--;
}
}
return_value = conn->timestamp_epoch;
return_value <<= 32;
return_value += timestamp;
if (previous_value > return_value) {
if ((previous_value - return_value) > conn->maximum_timestamp_interval)
debug(2, "interval between successive rtptimes greater than allowed!");
} else {
if ((return_value - previous_value) > conn->maximum_timestamp_interval)
debug(2, "interval between successive rtptimes greater than allowed!");
}
if (return_value < 0)
debug(1, "monotonic rtptime is negative!");
conn->last_timestamp = timestamp;
return return_value;
uint32_t rtp_frame_offset(uint32_t base, uint32_t frame_in_question) {
if (base <= frame_in_question)
return frame_in_question - base;
else
return UINT32_MAX - base + frame_in_question + 1;
}
void do_flush(uint32_t timestamp, rtsp_conn_info *conn);
static void ab_resync(rtsp_conn_info *conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++) {
@@ -165,11 +120,36 @@ static void ab_resync(rtsp_conn_info *conn) {
conn->ab_buffering = 1;
}
// the sequence number is a 16-bit unsigned number which wraps pretty often
// to work out if one seqno is 'after' another therefore depends whether wrap has occurred
// this function works out the actual ordinate of the seqno, i.e. the distance up from
// the zeroth element, at ab_read, taking due account of wrap.
// given starting and ending points as unsigned 32-bit integers running modulo 2^32, returns the position of x in the interval in *pos
// returns true if x is actually within the buffer
int position_in_modulo_uint32_t_buffer(uint32_t x, uint32_t start, uint32_t end, uint32_t *pos ) {
int response = 0; // not in the buffer
if (start <= end ) {
if (x < start) {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
} else {
if (pos)
*pos = x - start;
if (x < end)
response = 1;
}
} else if ((x >= start) && (x <= UINT32_MAX)) {
response = 1;
if (pos)
*pos = x - start;
} else {
if (pos)
*pos = UINT32_MAX - start + 1 + x;
if (x < end) {
response = 1;
}
}
return response;
}
// this is used.
static inline seq_t SUCCESSOR(seq_t x) {
uint32_t p = x & 0xffff;
p += 1;
@@ -177,6 +157,7 @@ static inline seq_t SUCCESSOR(seq_t x) {
return p;
}
// this is not used
static inline seq_t PREDECESSOR(seq_t x) {
uint32_t p = (x & 0xffff) + 0x10000;
p -= 1;
@@ -184,6 +165,8 @@ static inline seq_t PREDECESSOR(seq_t x) {
return p;
}
// used in seq_diff and seq_order
// anything with ORDINATE in it must be proctected by the ab_mutex
static inline int32_t ORDINATE(seq_t x, seq_t base) {
int32_t p = x; // int32_t from seq_t, i.e. uint16_t, so okay
@@ -458,14 +441,8 @@ static void free_audio_buffers(rtsp_conn_info *conn) {
free(conn->audio_buffer[i].data);
}
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data,
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
int len, rtsp_conn_info *conn) {
// all timestamps are done at the output rate
// the "actual_timestamp" is the one that comes in the packet, and is carried over for
// debugging
// and checking only.
int64_t ltimestamp = timestamp * conn->output_sample_ratio;
// ignore a request to flush that has been made before the first packet...
if (conn->packet_count == 0) {
@@ -483,22 +460,26 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
// if (flush_rtp_timestamp != 0)
// debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
if ((conn->flush_rtp_timestamp != 0) && (ltimestamp <= conn->flush_rtp_timestamp)) {
debug(3, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRId64
", flushing to "
"timestamp: %" PRId64 ".",
seqno, ltimestamp, conn->flush_rtp_timestamp);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
// now, if a flush_rtp_timestamp has been defined and the incoming timestamp is "before" it, drop it…
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(actual_timestamp,conn->flush_rtp_timestamp) < 8820)) {// if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
seqno, actual_timestamp, conn->flush_rtp_timestamp);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
} else {
if ((conn->flush_rtp_timestamp != 0x0) &&
(ltimestamp > conn->flush_rtp_timestamp)) // if we have gone past the flush boundary time
conn->flush_rtp_timestamp = 0x0;
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(conn->flush_rtp_timestamp,actual_timestamp) > 8820) && (rtp_frame_offset(conn->flush_rtp_timestamp,actual_timestamp) < 44100)) {
debug(2,"Dropping flush request");
conn->flush_rtp_timestamp = 0;
}
abuf_t *abuf = 0;
if (!conn->ab_synced) {
// if this is the first packet…
debug(3, "syncing to seqno %u.", seqno);
conn->ab_write = seqno;
conn->ab_read = seqno;
@@ -519,21 +500,21 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
conn->resend_interval = resend_interval;
}
if (conn->ab_write == seqno) { // expected packet
if (conn->ab_write == seqno) { // if this is the expected packet (which could be the first packet…)
uint64_t reception_time = get_absolute_time_in_fp();
if (conn->input_frame_rate_starting_point_is_valid == 0) {
if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) {
conn->frames_inward_measurement_start_time = reception_time;
conn->frames_inward_frames_received_at_measurement_start_time = timestamp;
conn->frames_inward_frames_received_at_measurement_start_time = actual_timestamp;
conn->input_frame_rate_starting_point_is_valid = 1; // valid now
}
}
conn->frames_inward_measurement_time = reception_time;
conn->frames_inward_frames_received_at_measurement_time = timestamp;
conn->frames_inward_frames_received_at_measurement_time = actual_timestamp;
abuf = conn->audio_buffer + BUFIDX(seqno);
conn->ab_write = SUCCESSOR(seqno);
conn->ab_write = SUCCESSOR(seqno); // move the write pointer to the next free space
} else if (seq_order(conn->ab_write, seqno, conn->ab_read)) { // newer than expected
// if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
// debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and
@@ -547,7 +528,7 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i));
abuf->ready = 0; // to be sure, to be sure
abuf->resend_level = 0;
abuf->timestamp = 0;
// abuf->timestamp = 0;
abuf->given_timestamp = 0;
abuf->sequence_number = 0;
}
@@ -580,14 +561,14 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
if (alac_decode(abuf->data, &datalen, data, len, conn) == 0) {
abuf->ready = 1;
abuf->length = datalen;
abuf->timestamp = ltimestamp;
// abuf->timestamp = ltimestamp;
abuf->given_timestamp = actual_timestamp;
abuf->sequence_number = seqno;
} else {
debug(1, "Bad audio packet detected and discarded.");
abuf->ready = 0;
abuf->resend_level = 0;
abuf->timestamp = 0;
// abuf->timestamp = 0;
abuf->given_timestamp = 0;
abuf->sequence_number = 0;
}
@@ -621,12 +602,6 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp
j)) { // prevent multiple requests from the same level of lookback
check_buf->resend_level = j;
if (config.disable_resend_requests == 0) {
if (((int)(resend_interval * pow(j + 1, step_exponent)) + k) >=
seq_diff(conn->ab_read, conn->ab_write, conn->ab_read))
debug(3, "Last-ditch (#%d) resend request for packet %u in range %u to %u. "
"Looking back %d packets.",
j, next, conn->ab_read, conn->ab_write, back_step + k);
debug_mutex_unlock(&conn->ab_mutex, 3);
rtp_request_resend(next, 1, conn);
conn->resend_requests++;
debug_mutex_lock(&conn->ab_mutex, 20000, 1);
@@ -877,20 +852,23 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
debug(1, "Inconsistent sequence numbers detected");
}
}
if ((conn->flush_rtp_timestamp != 0) &&
(curframe->timestamp <= conn->flush_rtp_timestamp)) {
debug(2, "Dropping flushed packet in buffer_get_frame seqno %u, timestamp %" PRId64 ".",
curframe->sequence_number, curframe->timestamp);
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(curframe->given_timestamp,conn->flush_rtp_timestamp) < 8820)) {// if it's less than 0.2 seconds
debug(2, "Dropping flushed packet in buffer_get_frame, seqno %u, timestamp %" PRIu32
", flushing to "
"timestamp: %" PRIu32 ".",
curframe->sequence_number, curframe->given_timestamp, conn->flush_rtp_timestamp);
curframe->ready = 0;
curframe->resend_level = 0;
flush_limit++;
flush_limit += curframe->length;
conn->ab_read = SUCCESSOR(conn->ab_read);
conn->initial_reference_time = 0;
conn->initial_reference_timestamp = 0;
}
if (curframe->timestamp > conn->flush_rtp_timestamp)
conn->flush_rtp_timestamp = 0;
if ((conn->flush_rtp_timestamp != 0) && (rtp_frame_offset(conn->flush_rtp_timestamp,curframe->given_timestamp) > 8820) && (rtp_frame_offset(conn->flush_rtp_timestamp,curframe->given_timestamp) < 44100)) {
debug(2,"Dropping flush request");
conn->flush_rtp_timestamp = 0;
}
}
} while ((conn->flush_rtp_timestamp != 0) && (flush_limit <= 8820) && (curframe->ready == 0));
@@ -921,7 +899,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (have_timestamp_timing_information(conn)) { // if we have a reference time
// debug(1,"First frame seen with timestamp...");
conn->first_packet_timestamp =
curframe->timestamp; // we will keep buffering until we are
curframe->given_timestamp; // we will keep buffering until we are
// supposed to start playing this
have_sent_prefiller_silence = 0;
conn->packet_stream_established = 1;
@@ -1174,7 +1152,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
// Note: the last three items are expressed in frames and must be converted to time.
int do_wait = 0; // don't wait unless we can really prove we must
if ((conn->ab_synced) && (curframe) && (curframe->ready) && (curframe->timestamp)) {
if ((conn->ab_synced) && (curframe) && (curframe->ready) && (curframe->given_timestamp)) {
do_wait =
1; // if the current frame exists and is ready, then wait unless it's time to let it go...
@@ -1184,7 +1162,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
uint64_t time_to_play;
frame_to_local_time(
curframe->timestamp + conn->latency * conn->output_sample_ratio +
(curframe->given_timestamp + conn->latency) * conn->output_sample_ratio +
(int64_t)(config.audio_backend_latency_offset * config.output_rate) -
config.audio_backend_buffer_desired_length * config.output_rate,
&time_to_play, conn);
@@ -1242,7 +1220,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (!curframe->ready) {
// debug(1, "Supplying a silent frame for frame %u", read);
conn->missing_packets++;
curframe->timestamp = 0; // indicate a silent frame should be substituted
curframe->given_timestamp = 0; // indicate a silent frame should be substituted
}
curframe->ready = 0;
curframe->resend_level = 0;
@@ -1791,7 +1769,7 @@ void *player_thread_func(void *arg) {
// if (play_number % 100 == 0)
// debug(3, "Play frame %d.", play_number);
conn->play_number_after_flush++;
if (inframe->timestamp == 0) {
if (inframe->given_timestamp == 0) {
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) &
@@ -1925,14 +1903,16 @@ void *player_thread_func(void *arg) {
// now, go back as far as the total latency less, say, 100 ms, and check the presence of
// frames from then onwards
int64_t reference_timestamp;
uint32_t reference_timestamp;
uint64_t reference_timestamp_time, remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time,
&remote_reference_timestamp_time, conn); // types okay
reference_timestamp *= conn->output_sample_ratio;
int64_t rt, nt;
rt = reference_timestamp; // uint32_t to int64_t
nt = inframe->timestamp; // uint32_t to int64_t
nt = inframe->given_timestamp; // uint32_t to int64_t
rt *= conn->output_sample_ratio;
nt *= conn->output_sample_ratio;
uint64_t local_time_now = get_absolute_time_in_fp(); // types okay
// struct timespec tn;
@@ -2056,7 +2036,7 @@ void *player_thread_func(void *arg) {
if (abs_sync_error < 0)
abs_sync_error = -abs_sync_error;
if ((config.no_sync == 0) && (inframe->timestamp != 0) &&
if ((config.no_sync == 0) && (inframe->given_timestamp != 0) &&
(config.resyncthreshold > 0.0) &&
(abs_sync_error > config.resyncthreshold * config.output_rate)) {
if (abs_sync_error > 3 * config.output_rate) {
@@ -2282,7 +2262,7 @@ void *player_thread_func(void *arg) {
}
// mark the frame as finished
inframe->timestamp = 0;
inframe->given_timestamp = 0;
inframe->sequence_number = 0;
// debug(1,"Sync error %lld frames. Amount to stuff %d." ,sync_error,amount_to_stuff);
@@ -2726,7 +2706,7 @@ void player_volume(double airplay_volume, rtsp_conn_info *conn) {
player_volume_without_notification(airplay_volume, conn);
}
void do_flush(int64_t timestamp, rtsp_conn_info *conn) {
void do_flush(uint32_t timestamp, rtsp_conn_info *conn) {
debug(3, "Flush requested up to %u. It seems as if 0 is special.", timestamp);
debug_mutex_lock(&conn->flush_mutex, 1000, 1);
@@ -2755,7 +2735,7 @@ void do_flush(int64_t timestamp, rtsp_conn_info *conn) {
debug(3, "Flush request made.");
}
void player_flush(int64_t timestamp, rtsp_conn_info *conn) {
void player_flush(uint32_t timestamp, rtsp_conn_info *conn) {
debug(3, "player_flush");
do_flush(timestamp, conn);
}
player.h
+11 -11
View File
@@ -40,7 +40,7 @@ typedef uint16_t seq_t;
typedef struct audio_buffer_entry { // decoded audio packets
int ready;
int resend_level;
int64_t timestamp;
// int64_t timestamp;
seq_t sequence_number;
uint32_t given_timestamp; // for debugging and checking
signed short *data;
@@ -71,10 +71,10 @@ typedef struct {
int connection_number; // for debug ID purposes, nothing else...
int resend_interval; // this is really just for debugging
int AirPlayVersion; // zero if not an AirPlay session. Used to help calculate latency
int64_t latency; // the actual latency used for this play session
int64_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero
uint32_t latency; // the actual latency used for this play session
uint32_t minimum_latency; // set if an a=min-latency: line appears in the ANNOUNCE message; zero
// otherwise
int64_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero
uint32_t maximum_latency; // set if an a=max-latency: line appears in the ANNOUNCE message; zero
// otherwise
int fd;
@@ -103,10 +103,10 @@ typedef struct {
int input_frame_rate_starting_point_is_valid;
uint64_t frames_inward_measurement_start_time;
uint64_t frames_inward_frames_received_at_measurement_start_time;
uint32_t frames_inward_frames_received_at_measurement_start_time;
uint64_t frames_inward_measurement_time;
uint64_t frames_inward_frames_received_at_measurement_time;
uint32_t frames_inward_frames_received_at_measurement_time;
// other stuff...
pthread_t *player_thread;
@@ -136,7 +136,7 @@ typedef struct {
int ab_buffering, ab_synced;
int64_t first_packet_timestamp;
int flush_requested;
int64_t flush_rtp_timestamp;
uint32_t flush_rtp_timestamp;
uint64_t time_of_last_audio_packet;
seq_t ab_read, ab_write;
@@ -187,7 +187,7 @@ typedef struct {
// this is what connects an rtp timestamp to the remote time
int64_t reference_timestamp;
uint32_t reference_timestamp;
uint64_t remote_reference_timestamp_time;
int packet_stream_established; // true if a stream of packets is flowing, made true by a first
@@ -195,7 +195,7 @@ typedef struct {
// used as the initials values for calculating the rate at which the source thinks it's sending
// frames
int64_t initial_reference_timestamp;
uint32_t initial_reference_timestamp;
uint64_t initial_reference_time;
double remote_frame_rate;
@@ -247,8 +247,8 @@ int player_stop(rtsp_conn_info *conn);
void player_volume(double f, rtsp_conn_info *conn);
void player_volume_without_notification(double f, rtsp_conn_info *conn);
void player_flush(int64_t timestamp, rtsp_conn_info *conn);
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data,
void player_flush(uint32_t timestamp, rtsp_conn_info *conn);
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
int len, rtsp_conn_info *conn);
int64_t monotonic_timestamp(uint32_t timestamp,
rtsp_conn_info *conn); // add an epoch to the timestamp. The monotonic
rtp.c
+19 -21
View File
@@ -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);
rtp.h
+1 -1
View File
@@ -17,7 +17,7 @@ void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t controlport, uint16_t
void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn);
void rtp_request_client_pause(rtsp_conn_info *conn); // ask the client to pause
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);
void clear_reference_timestamp(rtsp_conn_info *conn);