nils/shairport-sync
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Allow audio packets as well as resends on the control port. Add some debug messages.

Browse Source
This commit is contained in:
Mike Brady
2018-04-11 20:05:21 +01:00
parent 70d51b6722
commit fd7c2d7954
3 changed files with 39 additions and 20 deletions
Download Patch File Download Diff File Expand all files Collapse all files
dacp.c
+1 -1
View File
@@ -223,7 +223,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) {
if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1)
debug(1, "Error %d setting receive timeout for DACP service.", errno);
int ndata = recv(sockfd, buffer, sizeof(buffer), 0);
debug(3, "Received %d bytes: \"%s\".", ndata, buffer);
// debug(3, "Received %d bytes: \"%s\".", ndata, buffer);
if (ndata <= 0) {
debug(1, "dacp_send_command -- error receiving response for command \"%s\".",
command);
player.c
+6 -3
View File
@@ -1173,7 +1173,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
if (!conn->ab_buffering) {
// check once, after a short period of has elapsed, assuming 352 frames per packet
i = ((250 * 44100) / 352) / 1000; // approx 250 ms
i = ((125 * 44100) / 352) / 1000; // approx 125 ms
if (i < seq_diff(conn->ab_read, conn->ab_write, conn->ab_read)) {
seq_t next = seq_sum(conn->ab_read, i);
abuf = conn->audio_buffer + BUFIDX(next);
@@ -1686,7 +1686,7 @@ static void *player_thread_func(void *arg) {
config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
} else if (frames_to_drop) {
if (frames_to_drop > 3 * config.output_rate)
warn("Very large number of frames to drop: %" PRId64 ".",frames_to_drop);
warn("Very large number of frames to drop: %" PRId64 ".", frames_to_drop);
debug(3, "%" PRId64 " frames to drop.", frames_to_drop);
frames_to_drop -= inframe->length;
if (frames_to_drop < 0)
@@ -1888,7 +1888,10 @@ static void *player_thread_func(void *arg) {
(!conn->player_thread_please_stop) && (config.resyncthreshold > 0.0) &&
(abs_sync_error > config.resyncthreshold * config.output_rate)) {
if (abs_sync_error > 3 * config.output_rate) {
warn("Very large sync error: %" PRId64 " frames, with a delay of %" PRId64 "frames.",sync_error,delay);
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);
}
sync_error_out_of_bounds++;
} else {
rtp.c
+32 -16
View File
@@ -134,6 +134,7 @@ void *rtp_audio_receiver(void *arg) {
if (type == 0x56) {
pktp += 4;
plen -= 4;
debug(1, "resent packet %u received in audio port.", ntohs(*(uint16_t *)(pktp + 2)));
}
seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
// increment last_seqno and see if it's the same as the incoming seqno
@@ -344,27 +345,42 @@ void *rtp_control_receiver(void *arg) {
} else {
debug(1, "Sync packet received before we got a timing packet back.");
}
} else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
// debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
pktp = packet + 4;
plen -= 4;
seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
} else {
uint8_t type = packet[1] & ~0x80;
if ((type == 0x60) ||
(type == 0x56)) { // audio data / resent audio data in the control path -- whaale only?
pktp = packet;
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
if (type == 0x56) { // resent audio data in the control path -- whaale only?
// debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
pktp = packet + 4;
plen -= 4;
}
pktp += 12;
plen -= 12;
seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
if (type == 0x56)
debug(1, "Resent audio packet %u received in the control port.", seqno);
else
debug(1, "Audio packet %u received in the control port.", seqno);
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, timestamp, pktp, plen, conn);
continue;
int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
pktp += 12;
plen -= 12;
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, timestamp, pktp, plen, conn);
continue;
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
}
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
nread);
}
} else
debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
nread);
}
}
debug(3, "Control RTP thread interrupted. terminating.");