X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/76e72f65da97e7482da0a1eb0b110ca323f21643..ef0a7964a759f05b6c64096e4ec8e674d8283d2e:/lib/uaudio-rtp.c?ds=sidebyside diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c index bb03be4..c8e2ef2 100644 --- a/lib/uaudio-rtp.c +++ b/lib/uaudio-rtp.c @@ -59,6 +59,9 @@ static int rtp_fd; /** @brief RTP SSRC */ static uint32_t rtp_id; +/** @brief Base for timestamp */ +static uint32_t rtp_base; + /** @brief RTP sequence number */ static uint16_t rtp_sequence; @@ -68,11 +71,8 @@ static uint16_t rtp_sequence; */ static int rtp_errors; -/** @brief Delay threshold in microseconds - * - * rtp_play() never attempts to introduce a delay shorter than this. - */ -static int64_t rtp_delay_threshold; +/** @brief Set while paused */ +static volatile int rtp_paused; static const char *const rtp_options[] = { "rtp-destination", @@ -81,7 +81,6 @@ static const char *const rtp_options[] = { "rtp-source-port", "multicast-ttl", "multicast-loop", - "delay-threshold", NULL }; @@ -98,7 +97,7 @@ static void rtp_get_netconfig(const char *af, na->af = -1; else if(netaddress_parse(na, 3, vec)) - fatal(0, "invalid RTP address"); + disorder_fatal(0, "invalid RTP address"); } static void rtp_set_netconfig(const char *af, @@ -129,16 +128,28 @@ static void rtp_set_netconfig(const char *af, } } -static size_t rtp_play(void *buffer, size_t nsamples) { +static size_t rtp_play(void *buffer, size_t nsamples, unsigned flags) { struct rtp_header header; struct iovec vec[2]; - + +#if 0 + if(flags & (UAUDIO_PAUSE|UAUDIO_RESUME)) + fprintf(stderr, "rtp_play %zu samples%s%s%s%s\n", nsamples, + flags & UAUDIO_PAUSE ? " UAUDIO_PAUSE" : "", + flags & UAUDIO_RESUME ? " UAUDIO_RESUME" : "", + flags & UAUDIO_PLAYING ? " UAUDIO_PLAYING" : "", + flags & UAUDIO_PAUSED ? " UAUDIO_PAUSED" : ""); +#endif + /* We do as much work as possible before checking what time it is */ /* Fill out header */ header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ header.seq = htons(rtp_sequence++); header.ssrc = rtp_id; - header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload; + header.mpt = rtp_payload; + /* If we've come out of a pause, set the marker bit */ + if(flags & UAUDIO_RESUME) + header.mpt |= 0x80; #if !WORDS_BIGENDIAN /* Convert samples to network byte order */ uint16_t *u = buffer, *const limit = u + nsamples; @@ -151,24 +162,29 @@ static size_t rtp_play(void *buffer, size_t nsamples) { vec[0].iov_len = sizeof header; vec[1].iov_base = buffer; vec[1].iov_len = nsamples * uaudio_sample_size; - uaudio_schedule_synchronize(); - header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp); + const uint32_t timestamp = uaudio_schedule_sync(); + header.timestamp = htonl(rtp_base + (uint32_t)timestamp); + /* If we're paused don't actually end a packet, we just pretend */ + if(flags & UAUDIO_PAUSED) { + uaudio_schedule_sent(nsamples); + return nsamples; + } int written_bytes; do { written_bytes = writev(rtp_fd, vec, 2); } while(written_bytes < 0 && errno == EINTR); if(written_bytes < 0) { - error(errno, "error transmitting audio data"); + disorder_error(errno, "error transmitting audio data"); ++rtp_errors; if(rtp_errors == 10) - fatal(0, "too many audio tranmission errors"); + disorder_fatal(0, "too many audio tranmission errors"); return 0; } else rtp_errors /= 2; /* gradual decay */ - written_bytes -= sizeof (struct rtp_header); - const size_t written_samples = written_bytes / uaudio_sample_size; - uaudio_schedule_update(written_samples); - return written_samples; + /* TODO what can we sensibly do about short writes here? Really that's just + * an error and we ought to be using smaller packets. */ + uaudio_schedule_sent(nsamples); + return nsamples; } static void rtp_open(void) { @@ -187,7 +203,6 @@ static void rtp_open(void) { "rtp-source", "rtp-source-port", src); - rtp_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000")); /* ...microseconds */ /* Resolve addresses */ @@ -204,7 +219,7 @@ static void rtp_open(void) { if((rtp_fd = socket(res->ai_family, res->ai_socktype, res->ai_protocol)) < 0) - fatal(errno, "error creating broadcast socket"); + disorder_fatal(errno, "error creating broadcast socket"); if(multicast(res->ai_addr)) { /* Enable multicast options */ const int ttl = atoi(uaudio_get("multicast-ttl", "1")); @@ -213,31 +228,31 @@ static void rtp_open(void) { case PF_INET: { if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL, &ttl, sizeof ttl) < 0) - fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); + disorder_fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP, &loop, sizeof loop) < 0) - fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket"); + disorder_fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket"); break; } case PF_INET6: { if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, &ttl, sizeof ttl) < 0) - fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); + disorder_fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP, &loop, sizeof loop) < 0) - fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket"); + disorder_fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket"); break; } default: - fatal(0, "unsupported address family %d", res->ai_family); + disorder_fatal(0, "unsupported address family %d", res->ai_family); } - info("multicasting on %s TTL=%d loop=%s", - format_sockaddr(res->ai_addr), ttl, loop ? "yes" : "no"); + disorder_info("multicasting on %s TTL=%d loop=%s", + format_sockaddr(res->ai_addr), ttl, loop ? "yes" : "no"); } else { struct ifaddrs *ifs; if(getifaddrs(&ifs) < 0) - fatal(errno, "error calling getifaddrs"); + disorder_fatal(errno, "error calling getifaddrs"); while(ifs) { /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr * still a null pointer. It turns out that there's a subsequent entry @@ -250,35 +265,34 @@ static void rtp_open(void) { } if(ifs) { if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) - fatal(errno, "error setting SO_BROADCAST on broadcast socket"); - info("broadcasting on %s (%s)", + disorder_fatal(errno, "error setting SO_BROADCAST on broadcast socket"); + disorder_info("broadcasting on %s (%s)", format_sockaddr(res->ai_addr), ifs->ifa_name); } else - info("unicasting on %s", format_sockaddr(res->ai_addr)); + disorder_info("unicasting on %s", format_sockaddr(res->ai_addr)); } /* Enlarge the socket buffer */ len = sizeof sndbuf; if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, &sndbuf, &len) < 0) - fatal(errno, "error getting SO_SNDBUF"); + disorder_fatal(errno, "error getting SO_SNDBUF"); if(target_sndbuf > sndbuf) { if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, &target_sndbuf, sizeof target_sndbuf) < 0) - error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); + disorder_error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); else - info("changed socket send buffer size from %d to %d", + disorder_info("changed socket send buffer size from %d to %d", sndbuf, target_sndbuf); } else - info("default socket send buffer is %d", - sndbuf); + disorder_info("default socket send buffer is %d", sndbuf); /* We might well want to set additional broadcast- or multicast-related * options here */ if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0) - fatal(errno, "error binding broadcast socket to %s", - format_sockaddr(sres->ai_addr)); + disorder_fatal(errno, "error binding broadcast socket to %s", + format_sockaddr(sres->ai_addr)); if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0) - fatal(errno, "error connecting broadcast socket to %s", - format_sockaddr(res->ai_addr)); + disorder_fatal(errno, "error connecting broadcast socket to %s", + format_sockaddr(res->ai_addr)); } static void rtp_start(uaudio_callback *callback, @@ -293,12 +307,13 @@ static void rtp_start(uaudio_callback *callback, && uaudio_rate == 44100) rtp_payload = 11; else - fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2", - uaudio_bits, uaudio_rate, uaudio_channels); + disorder_fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2", + uaudio_bits, uaudio_rate, uaudio_channels); /* Various fields are required to have random initial values by RFC3550. The * packet contents are highly public so there's no point asking for very * strong randomness. */ gcry_create_nonce(&rtp_id, sizeof rtp_id); + gcry_create_nonce(&rtp_base, sizeof rtp_base); gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence); rtp_open(); uaudio_schedule_init(); @@ -317,15 +332,6 @@ static void rtp_stop(void) { rtp_fd = -1; } -static void rtp_activate(void) { - uaudio_schedule_reactivated = 1; - uaudio_thread_activate(); -} - -static void rtp_deactivate(void) { - uaudio_thread_deactivate(); -} - static void rtp_configure(void) { char buffer[64]; @@ -338,8 +344,6 @@ static void rtp_configure(void) { snprintf(buffer, sizeof buffer, "%ld", config->multicast_ttl); uaudio_set("multicast-ttl", buffer); uaudio_set("multicast-loop", config->multicast_loop ? "yes" : "no"); - snprintf(buffer, sizeof buffer, "%ld", config->rtp_delay_threshold); - uaudio_set("delay-threshold", buffer); } const struct uaudio uaudio_rtp = { @@ -347,8 +351,8 @@ const struct uaudio uaudio_rtp = { .options = rtp_options, .start = rtp_start, .stop = rtp_stop, - .activate = rtp_activate, - .deactivate = rtp_deactivate, + .activate = uaudio_thread_activate, + .deactivate = uaudio_thread_deactivate, .configure = rtp_configure, };