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Add an extra field to a buffer for checking purposes.

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This commit is contained in:
Mike Brady
2018-04-12 08:52:47 +01:00
parent d392f75c1c
commit ad450d745b
3 changed files with 15 additions and 8 deletions
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player.c
+7 -3
View File
@@ -462,10 +462,11 @@ static void free_audio_buffers(rtsp_conn_info *conn) {
free(conn->audio_buffer[i].data);
}
void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t 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;
@@ -518,6 +519,7 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
abuf = conn->audio_buffer + BUFIDX(seq_sum(conn->ab_write, i));
abuf->ready = 0; // to be sure, to be sure
abuf->timestamp = 0;
abuf->given_timestamp = 0;
abuf->sequence_number = 0;
}
// debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
@@ -550,11 +552,13 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len,
abuf->ready = 1;
abuf->length = datalen;
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->timestamp = 0;
abuf->given_timestamp = 0;
abuf->sequence_number = 0;
}
}
@@ -1890,8 +1894,8 @@ static void *player_thread_func(void *arg) {
if (abs_sync_error > 3 * config.output_rate) {
warn("Very large sync error: %" PRId64 " frames, with delay: %" PRId64
", td_in_frames: %" PRId64 ", rt: %" PRId64 ", nt: %" PRId64
", current_delay: %" PRId64 ", frames.",
sync_error, delay, td_in_frames, rt, nt, current_delay);
", current_delay: %" PRId64 ", seqno: %u, given timestamp: %" PRIu32 ".",
sync_error, delay, td_in_frames, rt, nt, current_delay, inframe->sequence_number, inframe->given_timestamp);
}
sync_error_out_of_bounds++;
} else {
player.h
+2 -1
View File
@@ -39,6 +39,7 @@ typedef struct audio_buffer_entry { // decoded audio packets
int ready;
int64_t timestamp;
seq_t sequence_number;
uint32_t given_timestamp; // for debugging and checking
signed short *data;
int length; // the length of the decoded data
} abuf_t;
@@ -193,7 +194,7 @@ void 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, int64_t timestamp, uint8_t *data, int len,
void player_put_packet(seq_t seqno, uint32_t actual_timestamp, int64_t timestamp, uint8_t *data, int len,
rtsp_conn_info *conn);
int64_t monotonic_timestamp(uint32_t timestamp,
rtp.c
+6 -4
View File
@@ -147,7 +147,8 @@ void *rtp_audio_receiver(void *arg) {
// debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
last_seqno = seqno; // reset warning...
}
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
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.");
@@ -157,7 +158,7 @@ void *rtp_audio_receiver(void *arg) {
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, timestamp, pktp, plen, conn);
player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
continue;
}
if (type == 0x56 && seqno == 0) {
@@ -361,8 +362,9 @@ void *rtp_control_receiver(void *arg) {
debug(3,"Resent audio packet %u received in the control port.",seqno);
else
debug(1,"Audio packet %u received in the control port.",seqno);
uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
int64_t timestamp = monotonic_timestamp(actual_timestamp, conn);
pktp += 12;
plen -= 12;
@@ -370,7 +372,7 @@ void *rtp_control_receiver(void *arg) {
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, timestamp, pktp, plen, conn);
player_put_packet(seqno, actual_timestamp, timestamp, pktp, plen, conn);
continue;
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");