/*
* This file is part of DisOrder.
- * Copyright (C) 2007, 2008 Richard Kettlewell
+ * Copyright (C) 2007-2009 Richard Kettlewell
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
#include <unistd.h>
#include <sys/mman.h>
#include <fcntl.h>
+#include <math.h>
#include "log.h"
#include "mem.h"
/** @brief Output device */
-/** @brief Minimum low watermark
- *
- * We'll stop playing if there's only this many samples in the buffer. */
-unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
+/** @brief Buffer low watermark in samples */
+unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
-/** @brief Maximum buffer size
+/** @brief Maximum buffer size in samples
*
- * We'll stop reading from the network if we have this many samples. */
+ * We'll stop reading from the network if we have this many samples.
+ */
static unsigned maxbuffer;
/** @brief Received packets
{ "pause-mode", required_argument, 0, 'P' },
{ "socket", required_argument, 0, 's' },
{ "config", required_argument, 0, 'C' },
+ { "monitor", no_argument, 0, 'M' },
{ 0, 0, 0, 0 }
};
pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
}
+#if HAVE_STUPID_GCC44
+ return NULL;
+#endif
}
/** @brief Background thread collecting samples
fatal(0, "unsupported RTP payload type %d",
header.mpt & 0x7F);
}
+ /* See if packet is silent */
+ const uint16_t *s = p->samples_raw;
+ n = p->nsamples;
+ for(; n > 0; --n)
+ if(*s++)
+ break;
+ if(!n)
+ p->flags |= SILENT;
if(logfp)
fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
seq, timestamp, p->nsamples, timestamp + p->nsamples);
*/
void playrtp_fill_buffer(void) {
/* Discard current buffer contents */
- while(nsamples)
+ while(nsamples) {
+ //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
drop_first_packet();
+ }
info("Buffering...");
/* Wait until there's at least minbuffer samples available */
while(nsamples < minbuffer) {
+ //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
pthread_cond_wait(&cond, &lock);
}
/* Start from whatever is earliest */
size_t max_samples,
void attribute((unused)) *userdata) {
size_t samples;
+ int silent = 0;
pthread_mutex_lock(&lock);
/* Get the next packet, junking any that are now in the past */
*bufptr++ = (int16_t)ntohs(*ptr++);
--i;
}
- /* We don't junk the packet here; a subsequent call to
- * playrtp_next_packet() will dispose of it (if it's actually done with). */
+ silent = !!(p->flags & SILENT);
} else {
/* There is no suitable packet. We introduce 0s up to the next packet, or
* to fill the buffer if there's no next packet or that's too many. The
samples = max_samples;
//info("infill by %zu", samples);
memset(buffer, 0, samples * uaudio_sample_size);
+ silent = 1;
}
/* Debug dump */
if(dump_buffer) {
}
/* Advance timestamp */
next_timestamp += samples;
+ /* If we're getting behind then try to drop just silent packets
+ *
+ * In theory this shouldn't be necessary. The server is supposed to send
+ * packets at the right rate and compares the number of samples sent with the
+ * time in order to ensure this.
+ *
+ * However, various things could throw this off:
+ *
+ * - the server's clock could advance at the wrong rate. This would cause it
+ * to mis-estimate the right number of samples to have sent and
+ * inappropriately throttle or speed up.
+ *
+ * - playback could happen at the wrong rate. If the playback host's sound
+ * card has a slightly incorrect clock then eventually it will get out
+ * of step.
+ *
+ * So if we play back slightly slower than the server sends for either of
+ * these reasons then eventually our buffer, and the socket's buffer, will
+ * fill, and the kernel will start dropping packets. The result is audible
+ * and not very nice.
+ *
+ * Therefore if we're getting behind, we pre-emptively drop silent packets,
+ * since a change in the duration of a silence is less noticeable than a
+ * dropped packet from the middle of continuous music.
+ *
+ * (If things go wrong the other way then eventually we run out of packets to
+ * play and are forced to play silence. This doesn't seem to happen in
+ * practice but if it does then in the same way we can artificially extend
+ * silent packets to compensate.)
+ *
+ * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
+ * track how close to target buffer occupancy we are on a once-a-minute
+ * basis.
+ */
+ if(nsamples > minbuffer && silent) {
+ info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
+ samples, nsamples, minbuffer);
+ samples = 0;
+ }
+ /* Junk obsolete packets */
+ playrtp_next_packet();
pthread_mutex_unlock(&lock);
return samples;
}
union any_sockaddr mgroup;
const char *dumpfile = 0;
pthread_t ltid;
+ int monitor = 0;
static const int one = 1;
static const struct addrinfo prefs = {
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
backend = uaudio_apis[0];
- while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:M:aocC:re:P:", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:M", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version("disorder-playrtp");
case 'r': dumpfile = optarg; break;
case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
case 'P': uaudio_set("pause-mode", optarg); break;
+ case 'M': monitor = 1; break;
default: fatal(0, "invalid option");
}
}
- if(config_read(0)) fatal(0, "cannot read configuration");
+ if(config_read(0, NULL)) fatal(0, "cannot read configuration");
if(!maxbuffer)
maxbuffer = 2 * minbuffer;
argc -= optind;
struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
memset(&in->sin_addr, 0, sizeof (struct in_addr));
- if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error binding socket to 0.0.0.0 port %d",
- ntohs(in->sin_port));
break;
}
case AF_INET6: {
rcvbuf, target_rcvbuf);
} else
info("default socket receive buffer %d", rcvbuf);
+ //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
if(logfp)
info("WARNING: -L option can impact performance");
if(control_socket) {
if((err = pthread_create(<id, 0, queue_thread, 0)))
fatal(err, "pthread_create queue_thread");
pthread_mutex_lock(&lock);
+ time_t lastlog = 0;
for(;;) {
/* Wait for the buffer to fill up a bit */
playrtp_fill_buffer();
while(nsamples >= minbuffer
|| (nsamples > 0
&& contains(pheap_first(&packets), next_timestamp))) {
+ if(monitor) {
+ time_t now = xtime(0);
+
+ if(now >= lastlog + 60) {
+ int offset = nsamples - minbuffer;
+ double offtime = (double)offset / (uaudio_rate * uaudio_channels);
+ info("%+d samples off (%d.%02ds, %d bytes)",
+ offset,
+ (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
+ (int)(fabs(offtime) * 100) % 100,
+ offset * uaudio_bits / CHAR_BIT);
+ lastlog = now;
+ }
+ }
+ //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
pthread_cond_wait(&cond, &lock);
}
+#if 0
+ if(nsamples) {
+ struct packet *p = pheap_first(&packets);
+ fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
+ nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
+ }
+#endif
/* Stop playing for a bit until the buffer re-fills */
pthread_mutex_unlock(&lock);
backend->deactivate();
~mdw / disorder / blobdiff