* systems. There is no support for Microsoft Windows yet, and that will in
* fact probably an entirely separate program.
*
- * The program runs (at least) two threads. listen_thread() is responsible for
- * reading RTP packets off the wire and adding them to the binary heap @ref
- * packets, assuming they are basically sound.
+ * 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.
*
- * InCore Audio the main thread is only responsible for starting and stopping
+ * 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.
*
* 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 <locale.h>
#include <sys/uio.h>
#include <string.h>
+#include <assert.h>
#include "log.h"
#include "mem.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>
static FILE *logfp;
/** @brief Output device */
-static const char *device;
-
-/** @brief Maximum samples per packet we'll support
- *
- * NB that two channels = two samples in this program.
- */
-#define MAXSAMPLES 2048
+const char *device;
/** @brief Minimum low watermark
*
* We'll stop reading from the network if we have this many samples. */
static unsigned maxbuffer;
-/** @brief Number of samples to infill by in one go
+/** @brief Received packets
+ * Protected by @ref receive_lock
*
- * This is an upper bound - in practice we expect the underlying audio API to
- * only ask for a much smaller number of samples in any one go.
+ * 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.
*/
-#define INFILL_SAMPLES (44100 * 2) /* 1s */
+struct packet *received_packets;
-/** @brief Received packet
- *
- * Received packets are kept in a binary heap (see @ref pheap) ordered by
- * timestamp.
+/** @brief Tail of @ref received_packets
+ * Protected by @ref receive_lock
*/
-struct packet {
- /** @brief Number of samples in this packet */
- uint32_t nsamples;
-
- /** @brief Timestamp from RTP packet
- *
- * NB that "timestamps" are really sample counters. Use lt() or lt_packet()
- * to compare timestamps.
- */
- uint32_t timestamp;
-
- /** @brief Flags
- *
- * Valid values are:
- * - @ref IDLE - the idle bit was set in the RTP packet
- */
- unsigned flags;
-/** @brief idle bit set in RTP packet*/
-#define IDLE 0x0001
-
- /** @brief Raw sample data
- *
- * Only the first @p nsamples samples are defined; the rest is uninitialized
- * data.
- */
- uint16_t samples_raw[MAXSAMPLES];
-};
+struct packet **received_tail = &received_packets;
-/** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic
+/** @brief Lock protecting @ref received_packets
*
- * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$.
- *
- * See also lt_packet().
- */
-static inline int lt(uint32_t a, uint32_t b) {
- return (uint32_t)(a - b) & 0x80000000;
-}
-
-/** @brief Return true iff a >= b in sequence-space arithmetic */
-static inline int ge(uint32_t a, uint32_t b) {
- return !lt(a, b);
-}
-
-/** @brief Return true iff a > b in sequence-space arithmetic */
-static inline int gt(uint32_t a, uint32_t b) {
- return lt(b, a);
-}
+ * 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 Return true iff a <= b in sequence-space arithmetic */
-static inline int le(uint32_t a, uint32_t b) {
- return !lt(b, a);
-}
-
-/** @brief Ordering for packets, used by @ref pheap */
-static inline int lt_packet(const struct packet *a, const struct packet *b) {
- return lt(a->timestamp, b->timestamp);
-}
+/** @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;
-/** @struct pheap
- * @brief Binary heap of packets ordered by timestamp */
-HEAP_TYPE(pheap, struct packet *, lt_packet);
+/** @brief Length of @ref received_packets */
+uint32_t nreceived;
/** @brief Binary heap of received packets */
-static struct pheap packets;
+struct pheap packets;
-/** @brief Total number of samples available */
-static unsigned long nsamples;
+/** @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.
*/
-static uint32_t next_timestamp;
+uint32_t next_timestamp;
/** @brief True if actively playing
*
* This is true when playing and false when just buffering. */
-static int active;
+int active;
-/** @brief Structure of free packet list */
-union free_packet {
- struct packet p;
- union free_packet *next;
-};
-
-/** @brief Linked list of free packets
- *
- * This is a linked list of formerly used packets. For preference we re-use
- * packets that have already been used rather than unused ones, to limit the
- * size of the program's working set. If there are no free packets in the list
- * we try @ref next_free_packet instead.
- *
- * Must hold @ref lock when accessing this.
- */
-static union free_packet *free_packets;
-
-/** @brief Array of new free packets
- *
- * There are @ref count_free_packets ready to use at this address. If there
- * are none left we allocate more memory.
- *
- * Must hold @ref lock when accessing this.
- */
-static union free_packet *next_free_packet;
-
-/** @brief Count of new free packets at @ref next_free_packet
- *
- * Must hold @ref lock when accessing this.
- */
-static size_t count_free_packets;
-
-/** @brief Lock protecting @ref packets
- *
- * This also protects the packet memory allocation infrastructure, @ref
- * free_packets and @ref next_free_packet. */
-static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
+/** @brief Lock protecting @ref packets */
+pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
/** @brief Condition variable signalled whenever @ref packets is changed */
-static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+
+HEAP_DEFINE(pheap, struct packet *, lt_packet);
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "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' },
{ 0, 0, 0, 0 }
};
-/** @brief Return a new packet
- *
- * Assumes that @ref lock is held. */
-static struct packet *new_packet(void) {
- struct packet *p;
-
- if(free_packets) {
- p = &free_packets->p;
- free_packets = free_packets->next;
- } else {
- if(!count_free_packets) {
- next_free_packet = xcalloc(1024, sizeof (union free_packet));
- count_free_packets = 1024;
- }
- p = &(next_free_packet++)->p;
- --count_free_packets;
- }
- return p;
-}
-
-/** @brief Free a packet
- *
- * Assumes that @ref lock is held. */
-static void free_packet(struct packet *p) {
- union free_packet *u = (union free_packet *)p;
- u->next = free_packets;
- free_packets = u;
-}
-
/** @brief Drop the first packet
*
* Assumes that @ref lock is held.
}
}
+/** @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);
+ }
+}
+
/** @brief Background thread collecting samples
*
* This function collects samples, perhaps converts them to the target format,
struct iovec iov[2];
for(;;) {
- if(!p) {
- pthread_mutex_lock(&lock);
+ if(!p)
p = new_packet();
- pthread_mutex_unlock(&lock);
- }
iov[0].iov_base = &header;
iov[0].iov_len = sizeof header;
iov[1].iov_base = p->samples_raw;
timestamp, next_timestamp);
continue;
}
- pthread_mutex_lock(&lock);
+ p->next = 0;
p->flags = 0;
p->timestamp = timestamp;
/* Convert to target format */
* This is rather unsatisfactory: it means that if packets get heavily
* out of order then we guarantee dropouts. But for now... */
if(nsamples >= maxbuffer) {
- info("Buffer full");
+ pthread_mutex_lock(&lock);
while(nsamples >= maxbuffer)
pthread_cond_wait(&cond, &lock);
+ pthread_mutex_unlock(&lock);
}
- /* Add the packet to the heap */
- pheap_insert(&packets, p);
- nsamples += p->nsamples;
+ /* 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;
- pthread_cond_broadcast(&cond);
- pthread_mutex_unlock(&lock);
}
}
* Must be called with @ref lock held.
*/
static void fill_buffer(void) {
+ while(nsamples)
+ drop_first_packet();
info("Buffering...");
while(nsamples < readahead)
pthread_cond_wait(&cond, &lock);
while(samplesOutLeft > 0) {
const struct packet *p = next_packet();
if(p && contains(p, next_timestamp)) {
- if(p->flags & IDLE)
- write(2, "I", 1);
/* This packet is ready to play */
const uint32_t packet_end = p->timestamp + p->nsamples;
const uint32_t offset = next_timestamp - p->timestamp;
*samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
/* We don't bother junking the packet - that'll be dealt with next time
* round */
- write(2, ".", 1);
} else {
/* No packet is ready to play (and there might be no packet at all) */
samples_available = p ? p->timestamp - next_timestamp
next_timestamp += samples_available;
samplesOut += samples_available;
samplesOutLeft -= samples_available;
- write(2, "?", 1);
}
}
++ab;
/* Something went wrong */
switch(frames_written) {
case -EAGAIN:
- write(2, "#", 1);
return 0;
case -EPIPE:
error(0, "error calling snd_pcm_writei: %ld",
* @return 0 on success, -1 on non-fatal error
*/
static int alsa_play(const struct packet *p) {
- if(p->flags & IDLE)
- write(2, "I", 1);
- write(2, ".", 1);
return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
(p->timestamp + p->nsamples) - next_timestamp);
}
if(p && samples_available > p->timestamp - next_timestamp)
samples_available = p->timestamp - next_timestamp;
- write(2, "?", 1);
return alsa_writei(zeros, samples_available);
}
/* We receive and convert audio data in a background thread */
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);
#if API_ALSA
{
struct packet *p;
info("Playing...");
/* Keep playing until the buffer empties out, or ALSA tells us to get
* lost */
- while(nsamples >= minbuffer && !escape) {
+ while((nsamples >= minbuffer
+ || (nsamples > 0
+ && contains(pheap_first(&packets), next_timestamp)))
+ && !escape) {
/* Wait for ALSA to ask us for more data */
pthread_mutex_unlock(&lock);
wait_alsa();
if(status)
fatal(0, "AudioDeviceStart: %d", (int)status);
/* Wait until the buffer empties out */
- while(nsamples >= minbuffer)
+ while(nsamples >= minbuffer
+ || (nsamples > 0
+ && contains(pheap_first(&packets), next_timestamp)))
pthread_cond_wait(&cond, &lock);
/* Stop playing for a bit until the buffer re-fills */
status = AudioDeviceStop(adid, adioproc);
" --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"
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
);
struct addrinfo *res;
struct stringlist sl;
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 prefs = {
AI_PASSIVE,
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
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;
default: fatal(0, "invalid option");
}
}
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