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Add code to check divisions and mods for a potrntial divide by zero problem. MAy cause SPS to terminate but leave a log message.

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This commit is contained in:
Mike Brady
2021-11-21 21:47:32 +00:00
parent 881a5ca25e
commit 14bfba2724
10 changed files with 208 additions and 131 deletions
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rtp.c
+15 -1
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@@ -154,7 +154,7 @@ void *rtp_audio_receiver(void *arg) {
float stat_delta = time_interval_us - stat_mean;
stat_mean += stat_delta / stat_n;
stat_M2 += stat_delta * (time_interval_us - stat_mean);
if (stat_n % 2500 == 0) {
if ((stat_n != 1) && (stat_n % 2500 == 0)) {
debug(2,
"Packet reception interval stats: mean, standard deviation and max for the last "
"2,500 packets in microseconds: %10.1f, %10.1f, %10.1f.",
@@ -650,6 +650,8 @@ void *rtp_timing_receiver(void *arg) {
double log_of_multiplier = log10(diffusion_expansion_factor) / time_ping_history;
double multiplier = pow(10, log_of_multiplier);
uint64_t dispersion_factor = (uint64_t)(multiplier * 100);
if (dispersion_factor == 0)
die("dispersion factor is zero!");
// debug(1,"dispersion factor is %" PRIu64 ".", dispersion_factor);
// uint64_t first_local_to_remote_time_difference_time;
@@ -1029,6 +1031,8 @@ const int use_nominal_rate = 0; // specify whether to use the nominal input rate
int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_conn_info *conn) {
int result = 1;
if (conn->input_rate == 0)
die("conn->input_rate is zero!");
uint32_t fs = conn->input_rate;
*frames = fs; // default value to return
*time = 1000000000; // default value to return
@@ -1051,7 +1055,11 @@ int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_con
result = 1;
} else {
*frames = local_frames;
if (local_frames == 0)
die("local_frames is zero!");
*time = local_time;
if (local_time == 0)
die("local_time is zero!");
result = 0;
}
}
@@ -1469,6 +1477,8 @@ int frame_to_ptp_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *
int32_t frame_difference = timestamp - anchor_rtptime;
int64_t time_difference = frame_difference;
time_difference = time_difference * 1000000000;
if (conn->input_rate == 0)
die("conn->input_rate is zero!");
time_difference = time_difference / conn->input_rate;
uint64_t ltime = anchor_local_time + time_difference;
*time = ltime;
@@ -2246,6 +2256,10 @@ void *rtp_buffered_audio_processor(void *arg) {
double requested_lead_time = 0.3; // normal lead time minimum
reset_buffer(conn); // in case there is any garbage in the player
// int not_first_time_out = 0;
// quick check of parameters
if (conn->input_bytes_per_frame == 0)
die("conn->input_bytes_per_frame is zero!");
do {
int flush_is_delayed = 0;
int flush_newly_requested = 0;