nils/shairport-sync
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Ensure a new master clock is at least 1.1 seconds in mastership before using it. Remember that the mastership can be backdated.

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Mike Brady
2021-06-17 18:17:24 +01:00
parent 0ac00359f7
commit ad3412f5c2
1 changed files with 15 additions and 14 deletions
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rtp.c
+15 -14
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@@ -1284,10 +1284,11 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
// using the master clock timing
if (actual_clock_id == conn->anchor_clock) {
//if the master clock and the anchor clock are the same
//wait at least this time before using the new master clock
if (duration_of_mastership < 0000000) {
debug(1,"master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
// if the master clock and the anchor clock are the same
// wait at least this time before using the new master clock
// note that mastershipo may be backdated
if (duration_of_mastership < 1100000000) {
debug(3,"master not old enough yet: %f ms", 0.000001 * duration_of_mastership);
response = clock_not_ready;
} else if ((duration_of_mastership > 5000000000) ||
(conn->last_anchor_info_is_valid == 0)) {
@@ -1365,6 +1366,7 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
// conn->anchor_time = conn->last_anchor_local_time + actual_offset; // already done
conn->anchor_clock = actual_clock_id;
}
conn->anchor_clock_is_new = 0;
}
} else {
response = clock_not_valid; // no current clock information and no previous clock info
@@ -1442,8 +1444,8 @@ int have_ptp_timing_information(rtsp_conn_info *conn) {
int frame_to_ptp_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *conn) {
int result = -1;
uint32_t anchor_rtptime;
uint64_t anchor_local_time;
uint32_t anchor_rtptime = 0;
uint64_t anchor_local_time = 0;
if (get_ptp_anchor_local_time_info(conn, &anchor_rtptime, &anchor_local_time) == clock_ok) {
int32_t frame_difference = timestamp - anchor_rtptime;
int64_t time_difference = frame_difference;
@@ -1460,8 +1462,8 @@ int frame_to_ptp_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *
int local_ptp_time_to_frame(uint64_t time, uint32_t *frame, rtsp_conn_info *conn) {
int result = -1;
uint32_t anchor_rtptime;
uint64_t anchor_local_time;
uint32_t anchor_rtptime = 0;
uint64_t anchor_local_time = 0;
if (get_ptp_anchor_local_time_info(conn, &anchor_rtptime, &anchor_local_time) == clock_ok) {
int64_t time_difference = time - anchor_local_time;
int64_t frame_difference = time_difference;
@@ -2130,7 +2132,7 @@ void *rtp_buffered_audio_processor(void *arg) {
int dst_linesize;
int dst_bufsize;
// Prepare software resampler to convert floating point (?) to S16
// Prepare software resampler to convert floating point (?)
SwrContext *swr = swr_alloc();
if (swr == NULL) {
debug(1, "can not allocate a swr context");
@@ -2211,7 +2213,7 @@ void *rtp_buffered_audio_processor(void *arg) {
int play_enabled = 0;
uint32_t flush_from_timestamp;
double requested_lead_time = 0.3; // normal lead time minimum
reset_buffer(conn); // in case there is any garbage in the player
reset_buffer(conn); // in case there is any garbage in the player
// int not_first_time_out = 0;
do {
int flush_is_delayed = 0;
@@ -2338,13 +2340,12 @@ void *rtp_buffered_audio_processor(void *arg) {
int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
double lead_time_ms = lead_time * 0.000001;
// debug(1,"lead time in buffered_audio is %f milliseconds.", lead_time_ms);
// it seems that some garbage blocks can be left after the flush, so if they
// don't have sensible lead times, drop them
// it seems that some garbage blocks can be left after the flush, so
// only accept them if they have sensible lead times
if ((lead_time_ms < 5000.0) && (lead_time > -1000.0)) {
// if it's the very first block (thus no priming needed)
if ((blocks_read == 1) || (blocks_read_since_flush > 3)) {
//if (1) {
if ((lead_time >= (int64_t)(requested_lead_time * 1000000000)) ||
(streaming_has_started != 0)) {
if (streaming_has_started == 0)