* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA
*/
+/** @file clients/playrtp.c
+ * @brief RTP player
+ *
+ * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
+ * and Apple Mac (<a
+ * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
+ * 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 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.
+ *
+ * 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.
+ *
+ * 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
+ * deliberately did not send any sound because it encountered a silence.
+ *
+ * Assumptions:
+ * - it is safe to read uint32_t values without a lock protecting them
+ */
#include <config.h>
#include "types.h"
#include <sys/socket.h>
#include <netdb.h>
#include <pthread.h>
+#include <locale.h>
+#include <sys/uio.h>
+#include <string.h>
+#include <assert.h>
+#include <errno.h>
#include "log.h"
#include "mem.h"
#include "addr.h"
#include "syscalls.h"
#include "rtp.h"
-#include "debug.h"
+#include "defs.h"
+#include "vector.h"
+#include "heap.h"
+#include "timeval.h"
+#include "playrtp.h"
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-# include <CoreAudio/AudioHardware.h>
-#endif
+#define readahead linux_headers_are_borked
+/** @brief RTP socket */
static int rtpfd;
-#define MAXSAMPLES 2048 /* max samples/frame we'll support */
-/* NB two channels = two samples in this program! */
-#define MINBUFFER 8820 /* when to stop playing */
-#define READAHEAD 88200 /* how far to read ahead */
-#define MAXBUFFER (3 * 88200) /* maximum buffer contents */
-
-struct frame {
- struct frame *next; /* another frame */
- int nsamples; /* number of samples */
- int nused; /* number of samples used so far */
- uint32_t timestamp; /* timestamp from packet */
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- float samples[MAXSAMPLES]; /* converted sample data */
-#endif
-};
+/** @brief Log output */
+static FILE *logfp;
-static unsigned long nsamples; /* total samples available */
+/** @brief Output device */
+const char *device;
-static struct frame *frames; /* received frames in ascending order
- * of timestamp */
-static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
-/* lock protecting frame list */
+/** @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 */
-static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER;
-/* signalled whenever we add a new frame */
+/** @brief Buffer high watermark
+ *
+ * We'll only start playing when this many samples are available. */
+static unsigned readahead = 2 * 2 * 44100;
+
+/** @brief Maximum buffer size
+ *
+ * We'll stop reading from the network if we have this many samples. */
+static unsigned maxbuffer;
+
+/** @brief Received packets
+ * Protected by @ref receive_lock
+ *
+ * Received packets are added to this list, and queue_thread() picks them off
+ * it and adds them to @ref packets. Whenever a packet is added to it, @ref
+ * receive_cond is signalled.
+ */
+struct packet *received_packets;
+
+/** @brief Tail of @ref received_packets
+ * Protected by @ref receive_lock
+ */
+struct packet **received_tail = &received_packets;
+
+/** @brief Lock protecting @ref received_packets
+ *
+ * Only listen_thread() and queue_thread() ever hold this lock. It is vital
+ * that queue_thread() not hold it any longer than it strictly has to. */
+pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** @brief Condition variable signalled when @ref received_packets is updated
+ *
+ * Used by listen_thread() to notify queue_thread() that it has added another
+ * packet to @ref received_packets. */
+pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
+
+/** @brief Length of @ref received_packets */
+uint32_t nreceived;
+
+/** @brief Binary heap of received packets */
+struct pheap packets;
+
+/** @brief Total number of samples available
+ *
+ * We make this volatile because we inspect it without a protecting lock,
+ * so the usual pthread_* guarantees aren't available.
+ */
+volatile uint32_t nsamples;
+
+/** @brief Timestamp of next packet to play.
+ *
+ * This is set to the timestamp of the last packet, plus the number of
+ * samples it contained. Only valid if @ref active is nonzero.
+ */
+uint32_t next_timestamp;
+
+/** @brief True if actively playing
+ *
+ * This is true when playing and false when just buffering. */
+int active;
+
+/** @brief Lock protecting @ref packets */
+pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** @brief Condition variable signalled whenever @ref packets is changed */
+pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+
+#if HAVE_ALSA_ASOUNDLIB_H
+# define DEFAULT_BACKEND playrtp_alsa
+#elif HAVE_SYS_SOUNDCARD_H
+# define DEFAULT_BACKEND playrtp_oss
+#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
+# define DEFAULT_BACKEND playrtp_coreaudio
+#else
+# error No known backend
+#endif
+
+/** @brief Backend to play with */
+static void (*backend)(void) = &DEFAULT_BACKEND;
+
+HEAP_DEFINE(pheap, struct packet *, lt_packet);
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "version", no_argument, 0, 'V' },
{ "debug", no_argument, 0, 'd' },
+ { "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' },
+ { "multicast", required_argument, 0, 'M' },
+#if HAVE_SYS_SOUNDCARD_H
+ { "oss", no_argument, 0, 'o' },
+#endif
+#if HAVE_ALSA_ASOUNDLIB_H
+ { "alsa", no_argument, 0, 'a' },
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ { "core-audio", no_argument, 0, 'c' },
+#endif
{ 0, 0, 0, 0 }
};
-/* Return true iff a > b in sequence-space arithmetic */
-static inline int gt(const struct frame *a, const struct frame *b) {
- return (uint32_t)(a->timestamp - b->timestamp) < 0x80000000;
+/** @brief Drop the first packet
+ *
+ * Assumes that @ref lock is held.
+ */
+static void drop_first_packet(void) {
+ if(pheap_count(&packets)) {
+ struct packet *const p = pheap_remove(&packets);
+ nsamples -= p->nsamples;
+ playrtp_free_packet(p);
+ pthread_cond_broadcast(&cond);
+ }
+}
+
+/** @brief Background thread adding packets to heap
+ *
+ * This just transfers packets from @ref received_packets to @ref packets. It
+ * is important that it holds @ref receive_lock for as little time as possible,
+ * in order to minimize the interval between calls to read() in
+ * listen_thread().
+ */
+static void *queue_thread(void attribute((unused)) *arg) {
+ struct packet *p;
+
+ for(;;) {
+ /* Get the next packet */
+ pthread_mutex_lock(&receive_lock);
+ while(!received_packets)
+ pthread_cond_wait(&receive_cond, &receive_lock);
+ p = received_packets;
+ received_packets = p->next;
+ if(!received_packets)
+ received_tail = &received_packets;
+ --nreceived;
+ pthread_mutex_unlock(&receive_lock);
+ /* Add it to the heap */
+ pthread_mutex_lock(&lock);
+ pheap_insert(&packets, p);
+ nsamples += p->nsamples;
+ pthread_cond_broadcast(&cond);
+ pthread_mutex_unlock(&lock);
+ }
}
-/* Background thread that reads frames over the network and add them to the
- * list */
-static listen_thread(void attribute((unused)) *arg) {
- struct frame *f = 0, **ff;
- int n, i;
- union {
- struct rtp_header header;
- uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)];
- } packet;
- const uint16_t *const samples = (uint16_t *)(packet.bytes
- + sizeof (struct rtp_header));
+/** @brief Background thread collecting samples
+ *
+ * This function collects samples, perhaps converts them to the target format,
+ * and adds them to the packet list.
+ *
+ * It is crucial that the gap between successive calls to read() is as small as
+ * possible: otherwise packets will be dropped.
+ *
+ * We use a binary heap to ensure that the unavoidable effort is at worst
+ * logarithmic in the total number of packets - in fact if packets are mostly
+ * received in order then we will largely do constant work per packet since the
+ * newest packet will always be last.
+ *
+ * Of more concern is that we must acquire the lock on the heap to add a packet
+ * to it. If this proves a problem in practice then the answer would be
+ * (probably doubly) linked list with new packets added the end and a second
+ * thread which reads packets off the list and adds them to the heap.
+ *
+ * We keep memory allocation (mostly) very fast by keeping pre-allocated
+ * packets around; see @ref playrtp_new_packet().
+ */
+static void *listen_thread(void attribute((unused)) *arg) {
+ struct packet *p = 0;
+ int n;
+ struct rtp_header header;
+ uint16_t seq;
+ uint32_t timestamp;
+ struct iovec iov[2];
for(;;) {
- if(!f)
- f = xmalloc(sizeof *f);
- n = read(rtpfd, packet.bytes, sizeof packet.bytes);
+ if(!p)
+ p = playrtp_new_packet();
+ iov[0].iov_base = &header;
+ iov[0].iov_len = sizeof header;
+ iov[1].iov_base = p->samples_raw;
+ iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
+ n = readv(rtpfd, iov, 2);
if(n < 0) {
switch(errno) {
case EINTR:
fatal(errno, "error reading from socket");
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ /* Ignore too-short packets */
+ if((size_t)n <= sizeof (struct rtp_header)) {
+ info("ignored a short packet");
+ continue;
+ }
+ timestamp = htonl(header.timestamp);
+ seq = htons(header.seq);
+ /* Ignore packets in the past */
+ if(active && lt(timestamp, next_timestamp)) {
+ info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
+ timestamp, next_timestamp);
+ continue;
+ }
+ p->next = 0;
+ p->flags = 0;
+ p->timestamp = timestamp;
/* Convert to target format */
- switch(packet.header.mtp & 0x7F) {
+ if(header.mpt & 0x80)
+ p->flags |= IDLE;
+ switch(header.mpt & 0x7F) {
case 10:
- f->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t);
- for(i = 0; i < f->nsamples; ++i)
- f->samples[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767);
+ p->nsamples = (n - sizeof header) / sizeof(uint16_t);
break;
/* TODO support other RFC3551 media types (when the speaker does) */
default:
- fatal(0, "unsupported RTP payload type %d",
- packet.header.mpt & 0x7F);
+ fatal(0, "unsupported RTP payload type %d",
+ header.mpt & 0x7F);
}
-#endif
- f->used = 0;
- f->timestamp = ntohl(packet.header.timestamp);
- pthread_mutex_lock(&lock);
- /* Stop reading if we've reached the maximum */
- while(nsamples >= MAXBUFFER)
- pthread_cond_wait(&cond, &lock);
- for(ff = &frames; *ff && !gt(*ff, f); ff = &(*ff)->next)
- ;
- f->next = *ff;
- *ff = f;
- nsamples += f->nsamples;
- pthread_cond_broadcast(&cond);
- pthread_mutex_unlock(&lock);
- f = 0;
+ if(logfp)
+ fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
+ seq, timestamp, p->nsamples, timestamp + p->nsamples);
+ /* Stop reading if we've reached the maximum.
+ *
+ * This is rather unsatisfactory: it means that if packets get heavily
+ * out of order then we guarantee dropouts. But for now... */
+ if(nsamples >= maxbuffer) {
+ pthread_mutex_lock(&lock);
+ while(nsamples >= maxbuffer)
+ pthread_cond_wait(&cond, &lock);
+ pthread_mutex_unlock(&lock);
+ }
+ /* Add the packet to the receive queue */
+ pthread_mutex_lock(&receive_lock);
+ *received_tail = p;
+ received_tail = &p->next;
+ ++nreceived;
+ pthread_cond_signal(&receive_cond);
+ pthread_mutex_unlock(&receive_lock);
+ /* We'll need a new packet */
+ p = 0;
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-static OSStatus adioproc(AudioDeviceID inDevice,
- const AudioTimeStamp *inNow,
- const AudioBufferList *inInputData,
- const AudioTimeStamp *inInputTime,
- AudioBufferList *outOutputData,
- const AudioTimeStamp *inOutputTime,
- void *inClientData) {
- UInt32 nbuffers = outOutputData->mNumberBuffers;
- AudioBuffer *ab = outOutputData->mBuffers;
- float *samplesOut; /* where to write samples to */
- size_t samplesOutLeft; /* space left */
- size_t samplesInLeft;
- size_t samplesToCopy;
-
- pthread_mutex_lock(&lock);
- samplesOut = ab->data;
- samplesOutLeft = ab->mDataByteSize / sizeof (float);
- while(frames && nbuffers > 0) {
- if(frames->used == frames->nsamples) {
- /* TODO if we dropped a packet then we should introduce a gap here */
- struct frame *const f = frames;
- frames = f->next;
- free(f);
- pthread_cond_broadcast(&cond);
- continue;
- }
- if(samplesOutLeft == 0) {
- --nbuffers;
- ++ab;
- samplesOut = ab->data;
- samplesOutLeft = ab->mDataByteSize / sizeof (float);
- continue;
- }
- /* Now: (1) there is some data left to read
- * (2) there is some space to put it */
- samplesInLeft = frames->nsamples - frames->used;
- samplesToCopy = (samplesInLeft < samplesOutLeft
- ? samplesInLeft : samplesOutLeft);
- memcpy(samplesOut, frame->samples + frames->used, samplesToCopy);
- frames->used += samplesToCopy;
- samplesOut += samplesToCopy;
- samesOutLeft -= samplesToCopy;
+/** @brief Wait until the buffer is adequately full
+ *
+ * Must be called with @ref lock held.
+ */
+void playrtp_fill_buffer(void) {
+ while(nsamples)
+ drop_first_packet();
+ info("Buffering...");
+ while(nsamples < readahead)
+ pthread_cond_wait(&cond, &lock);
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+}
+
+/** @brief Find next packet
+ * @return Packet to play or NULL if none found
+ *
+ * The return packet is merely guaranteed not to be in the past: it might be
+ * the first packet in the future rather than one that is actually suitable to
+ * play.
+ *
+ * Must be called with @ref lock held.
+ */
+struct packet *playrtp_next_packet(void) {
+ while(pheap_count(&packets)) {
+ struct packet *const p = pheap_first(&packets);
+ if(le(p->timestamp + p->nsamples, next_timestamp)) {
+ /* This packet is in the past. Drop it and try another one. */
+ drop_first_packet();
+ } else
+ /* This packet is NOT in the past. (It might be in the future
+ * however.) */
+ return p;
}
- pthread_mutex_unlock(&lock);
return 0;
}
-#endif
-void play_rtp(void) {
- pthread_t lt;
+/** @brief Play an RTP stream
+ *
+ * This is the guts of the program. It is responsible for:
+ * - starting the listening thread
+ * - opening the audio device
+ * - reading ahead to build up a buffer
+ * - arranging for audio to be played
+ * - detecting when the buffer has got too small and re-buffering
+ */
+static void play_rtp(void) {
+ pthread_t ltid;
/* We receive and convert audio data in a background thread */
- pthread_create(<, 0, listen_thread, 0);
-#if API_ALSA
- assert(!"implemented");
-#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
- {
- OSStatus status;
- UInt32 propertySize;
- AudioDeviceID adid;
- AudioStreamBasicDescription asbd;
-
- /* If this looks suspiciously like libao's macosx driver there's an
- * excellent reason for that... */
-
- /* TODO report errors as strings not numbers */
- propertySize = sizeof adid;
- status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
- &propertySize, &adid);
- if(status)
- fatal(0, "AudioHardwareGetProperty: %d", (int)status);
- if(adid == kAudioDeviceUnknown)
- fatal(0, "no output device");
- propertySize = sizeof asbd;
- status = AudioDeviceGetProperty(adid, 0, false,
- kAudioDevicePropertyStreamFormat,
- &propertySize, &asbd);
- if(status)
- fatal(0, "AudioHardwareGetProperty: %d", (int)status);
- D(("mSampleRate %f", asbd.mSampleRate));
- D(("mFormatID %08"PRIx32, asbd.mFormatID));
- D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags));
- D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket));
- D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket));
- D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame));
- D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame));
- D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel));
- D(("mReserved %08"PRIx32, asbd.mReserved));
- if(asbd.mFormatID != kAudioFormatLinearPCM)
- fatal(0, "audio device does not support kAudioFormatLinearPCM");
- status = AudioDeviceAddIOProc(adid, adioproc, 0);
- if(status)
- fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
- pthread_mutex_lock(&lock);
- for(;;) {
- /* Wait for the buffer to fill up a bit */
- while(nsamples < READAHEAD)
- pthread_cond_wait(&cond, &lock);
- /* Start playing now */
- status = AudioDeviceStart(adid, adioproc);
- if(status)
- fatal(0, "AudioDeviceStart: %d", (int)status);
- /* Wait until the buffer empties out */
- while(nsamples >= MINBUFFER)
- pthread_cond_wait(&cond, &lock);
- /* Stop playing for a bit until the buffer re-fills */
- status = AudioDeviceStop(adid, adioproc);
- if(status)
- fatal(0, "AudioDeviceStop: %d", (int)status);
- /* Go back round */
- }
- }
-#else
-# error No known audio API
-#endif
+ pthread_create(<id, 0, listen_thread, 0);
+ /* We have a second thread to add received packets to the queue */
+ pthread_create(<id, 0, queue_thread, 0);
+ /* The rest of the work is backend-specific */
+ backend();
}
/* display usage message and terminate */
xprintf("Usage:\n"
" disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
"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"
+ " --multicast, -M GROUP Join multicast group\n"
+#if HAVE_ALSA_ASOUNDLIB_H
+ " --alsa, -a Use ALSA to play audio\n"
+#endif
+#if HAVE_SYS_SOUNDCARD_H
+ " --oss, -o Use OSS to play audio\n"
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ " --core-audio, -c Use Core Audio to play audio\n"
+#endif
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
- " --debug, -d Turn on debugging\n");
+ );
xfclose(stdout);
exit(0);
}
int n;
struct addrinfo *res;
struct stringlist sl;
- const char *sockname;
+ char *sockname;
+ int rcvbuf, target_rcvbuf = 131072;
+ socklen_t len;
+ char *multicast_group = 0;
+ struct ip_mreq mreq;
+ struct ipv6_mreq mreq6;
- static const struct addrinfo prefbind = {
+ static const struct addrinfo prefs = {
AI_PASSIVE,
PF_INET,
SOCK_DGRAM,
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVd", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aoc", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
case 'd': debugging = 1; break;
+ case 'D': 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;
+ case 'M': multicast_group = optarg; break;
+#if HAVE_ALSA_ASOUNDLIB_H
+ case 'a': backend = playrtp_alsa; break;
+#endif
+#if HAVE_SYS_SOUNDCARD_H
+ case 'o': backend = playrtp_oss; break;
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ case 'c': backend = playrtp_coreaudio; break;
+#endif
default: fatal(0, "invalid option");
}
}
+ if(!maxbuffer)
+ maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
if(argc < 1 || argc > 2)
sl.n = argc;
sl.s = argv;
/* Listen for inbound audio data */
- if(!(res = get_address(&sl, &pref, &sockname)))
+ if(!(res = get_address(&sl, &prefs, &sockname)))
exit(1);
if((rtpfd = socket(res->ai_family,
res->ai_socktype,
fatal(errno, "error creating socket");
if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
fatal(errno, "error binding socket to %s", sockname);
+ if(multicast_group) {
+ if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
+ fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
+ switch(res->ai_family) {
+ case PF_INET:
+ mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
+ mreq.imr_interface.s_addr = 0; /* use primary interface */
+ if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
+ &mreq, sizeof mreq) < 0)
+ fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
+ break;
+ case PF_INET6:
+ mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
+ memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
+ if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
+ &mreq6, sizeof mreq6) < 0)
+ fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
+ break;
+ default:
+ fatal(0, "unsupported address family %d", res->ai_family);
+ }
+ }
+ len = sizeof rcvbuf;
+ if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
+ fatal(errno, "error calling getsockopt SO_RCVBUF");
+ if(target_rcvbuf > rcvbuf) {
+ if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
+ &target_rcvbuf, sizeof target_rcvbuf) < 0)
+ error(errno, "error calling setsockopt SO_RCVBUF %d",
+ target_rcvbuf);
+ /* We try to carry on anyway */
+ else
+ info("changed socket receive buffer from %d to %d",
+ rcvbuf, target_rcvbuf);
+ } else
+ info("default socket receive buffer %d", rcvbuf);
+ if(logfp)
+ info("WARNING: -L option can impact performance");
play_rtp();
return 0;
}
~mdw / disorder / blobdiff