X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/d76bbdea349bf3856172ce9599e657d566e91162..b061950146f73c9fde005df8969f8c6e0e9b2a64:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index d97864b..fe750b2 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -24,20 +24,20 @@ * systems. There is no support for Microsoft Windows yet, and that will in * fact probably an entirely separate program. * - * The program runs (at least) three threads. listen_thread() is responsible - * for reading RTP packets off the wire and adding them to the linked list @ref - * received_packets, assuming they are basically sound. queue_thread() takes - * packets off this linked list and adds them to @ref packets (an operation - * which might be much slower due to contention for @ref lock). + * The program runs (at least) three threads: * - * The main thread is responsible for actually playing audio. In ALSA this - * means it waits until ALSA says it's ready for more audio which it then - * plays. See @ref clients/playrtp-alsa.c. + * listen_thread() is responsible for reading RTP packets off the wire and + * adding them to the linked list @ref received_packets, assuming they are + * basically sound. * - * In Core Audio the main thread is only responsible for starting and stopping - * play: the system does the actual playback in its own private thread, and - * calls adioproc() to fetch the audio data. See @ref - * clients/playrtp-coreaudio.c. + * queue_thread() takes packets off this linked list and adds them to @ref + * packets (an operation which might be much slower due to contention for @ref + * lock). + * + * control_thread() accepts commands from Disobedience (or anything else). + * + * The main thread activates and deactivates audio playing via the @ref + * lib/uaudio.h API (which probably implies at least one further thread). * * Sometimes it happens that there is no audio available to play. This may * because the server went away, or a packet was dropped, or the server @@ -64,6 +64,7 @@ #include #include #include +#include #include "log.h" #include "mem.h" @@ -81,8 +82,6 @@ #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 +95,13 @@ 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 +197,6 @@ 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' }, @@ -217,8 +209,10 @@ static const struct option options[] = { #endif { "dump", required_argument, 0, 'r' }, { "command", required_argument, 0, 'e' }, + { "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 } }; @@ -439,12 +433,18 @@ 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; } @@ -479,7 +479,6 @@ 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" @@ -492,7 +491,9 @@ static void help(void) { #if HAVE_COREAUDIO_AUDIOHARDWARE_H " --core-audio, -c Use Core Audio to play audio\n" #endif - " --command, -e COMMAND Pipe audio to command\n" + " --command, -e COMMAND Pipe audio to command.\n" + " --pause-mode, -P silence For -e: pauses send silence (default)\n" + " --pause-mode, -P suspend For -e: pauses suspend writes\n" " --help, -h Display usage message\n" " --version, -V Display version number\n" ); @@ -534,8 +535,6 @@ 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 @@ -556,6 +555,8 @@ static size_t playrtp_callback(void *buffer, } /* Advance timestamp */ next_timestamp += samples; + /* Junk obsolete packets */ + playrtp_next_packet(); pthread_mutex_unlock(&lock); return samples; } @@ -565,7 +566,7 @@ 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; @@ -580,6 +581,7 @@ int main(int argc, char **argv) { union any_sockaddr mgroup; const char *dumpfile = 0; pthread_t ltid; + int monitor = 0; static const int one = 1; static const struct addrinfo prefs = { @@ -589,17 +591,19 @@ 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:", 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 '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; @@ -616,12 +620,14 @@ int main(int argc, char **argv) { case 's': control_socket = optarg; break; 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 = 4 * readahead; + maxbuffer = 2 * minbuffer; argc -= optind; argv += optind; switch(argc) { @@ -736,6 +742,7 @@ 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) { @@ -777,6 +784,7 @@ int main(int argc, char **argv) { 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(); @@ -787,12 +795,42 @@ 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))) { + if(monitor) { + time_t now = time(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();