X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/8d251217b5cfeae9d277054a456c4064eabde569..e99a9fd614f9bd1d205e00b47e3e669b7735b91f:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index c6f0ad9..f5f538d 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -81,8 +81,6 @@ #include "version.h" #include "uaudio.h" -#define readahead linux_headers_are_borked - /** @brief Obsolete synonym */ #ifndef IPV6_JOIN_GROUP # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP @@ -96,19 +94,13 @@ static FILE *logfp; /** @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 Buffer high watermark +/** @brief Maximum buffer size in samples * - * We'll only start playing when this many samples are available. */ -static unsigned readahead = 44100; /* 0.5 seconds */ - -/** @brief Maximum buffer size - * - * 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 @@ -204,7 +196,6 @@ static const struct option options[] = { { "device", required_argument, 0, 'D' }, { "min", required_argument, 0, 'm' }, { "max", required_argument, 0, 'x' }, - { "buffer", required_argument, 0, 'b' }, { "rcvbuf", required_argument, 0, 'R' }, #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST { "oss", no_argument, 0, 'o' }, @@ -440,12 +431,18 @@ static void *listen_thread(void attribute((unused)) *arg) { * Must be called with @ref lock held. */ void playrtp_fill_buffer(void) { - while(nsamples) + /* Discard current buffer contents */ + while(nsamples) { + //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer); drop_first_packet(); + } info("Buffering..."); - while(nsamples < readahead) { + /* 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 */ next_timestamp = pheap_first(&packets)->timestamp; active = 1; } @@ -480,7 +477,6 @@ static void help(void) { "Options:\n" " --device, -D DEVICE Output device\n" " --min, -m FRAMES Buffer low water mark\n" - " --buffer, -b FRAMES Buffer high water mark\n" " --max, -x FRAMES Buffer maximum size\n" " --rcvbuf, -R BYTES Socket receive buffer size\n" " --config, -C PATH Set configuration file\n" @@ -537,8 +533,6 @@ static size_t playrtp_callback(void *buffer, *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). */ } 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 @@ -559,6 +553,8 @@ static size_t playrtp_callback(void *buffer, } /* Advance timestamp */ next_timestamp += samples; + /* Junk obsolete packets */ + playrtp_next_packet(); pthread_mutex_unlock(&lock); return samples; } @@ -568,7 +564,7 @@ int main(int argc, char **argv) { struct addrinfo *res; struct stringlist sl; char *sockname; - int rcvbuf, target_rcvbuf = 131072; + int rcvbuf, target_rcvbuf = 0; socklen_t len; struct ip_mreq mreq; struct ipv6_mreq mreq6; @@ -592,17 +588,19 @@ int main(int argc, char **argv) { .ai_protocol = IPPROTO_UDP }; + /* Timing information is often important to debugging playrtp, so we include + * timestamps in the logs */ + logdate = 1; mem_init(); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); backend = uaudio_apis[0]; - while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:re:P:", options, 0)) >= 0) { + while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:M:aocC:re:P:", options, 0)) >= 0) { switch(n) { case 'h': help(); case 'V': version("disorder-playrtp"); case 'd': debugging = 1; break; case 'D': uaudio_set("device", optarg); break; case 'm': minbuffer = 2 * atol(optarg); break; - case 'b': readahead = 2 * atol(optarg); break; case 'x': maxbuffer = 2 * atol(optarg); break; case 'L': logfp = fopen(optarg, "w"); break; case 'R': target_rcvbuf = atoi(optarg); break; @@ -625,7 +623,7 @@ int main(int argc, char **argv) { } if(config_read(0)) fatal(0, "cannot read configuration"); if(!maxbuffer) - maxbuffer = 4 * readahead; + maxbuffer = 2 * minbuffer; argc -= optind; argv += optind; switch(argc) { @@ -740,6 +738,7 @@ int main(int argc, char **argv) { 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) { @@ -791,12 +790,28 @@ int main(int argc, char **argv) { pthread_mutex_unlock(&lock); backend->activate(); pthread_mutex_lock(&lock); - /* Wait until the buffer empties out */ + /* Wait until the buffer empties out + * + * If there's a packet that we can play right now then we definitely + * continue. + * + * Also if there's at least minbuffer samples we carry on regardless and + * insert silence. The assumption is there's been a pause but more data + * is now available. + */ while(nsamples >= minbuffer || (nsamples > 0 && contains(pheap_first(&packets), next_timestamp))) { + //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();