X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/b64efe7e78086710c0196e2a9cd46ea03e925e90..655cae6737903aac20835588f82ca592cbdbde99:/clients/playrtp.c
diff --git a/clients/playrtp.c b/clients/playrtp.c
index 11df0e0..8931b6e 100644
--- a/clients/playrtp.c
+++ b/clients/playrtp.c
@@ -20,7 +20,32 @@
/** @file clients/playrtp.c
* @brief RTP player
*
- * This RTP player supports Linux (ALSA) and Darwin (Core Audio) systems.
+ * This player supports Linux (ALSA)
+ * and Apple Mac (Core Audio)
+ * 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.
+ *
+ * InCore 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
@@ -47,6 +72,7 @@
#include "defs.h"
#include "vector.h"
#include "heap.h"
+#include "timeval.h"
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
# include
@@ -100,6 +126,9 @@ static unsigned maxbuffer;
* timestamp.
*/
struct packet {
+ /** @brief Next packet in @ref next_free_packet or @ref received_packets */
+ struct packet *next;
+
/** @brief Number of samples in this packet */
uint32_t nsamples;
@@ -113,10 +142,11 @@ struct packet {
/** @brief Flags
*
* Valid values are:
- * - @ref IDLE: the idle bit was set in the RTP packet
+ * - @ref IDLE - the idle bit was set in the RTP packet
*/
unsigned flags;
-#define IDLE 0x0001 /**< idle bit set in RTP packet */
+/** @brief idle bit set in RTP packet*/
+#define IDLE 0x0001
/** @brief Raw sample data
*
@@ -156,6 +186,35 @@ static inline int lt_packet(const struct packet *a, const struct packet *b) {
return lt(a->timestamp, b->timestamp);
}
+/** @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.
+ */
+static struct packet *received_packets;
+
+/** @brief Tail of @ref received_packets
+ * Protected by @ref receive_lock
+ */
+static 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. */
+static 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. */
+static pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
+
+/** @brief Length of @ref received_packets */
+static uint32_t nreceived;
+
/** @struct pheap
* @brief Binary heap of packets ordered by timestamp */
HEAP_TYPE(pheap, struct packet *, lt_packet);
@@ -163,8 +222,12 @@ HEAP_TYPE(pheap, struct packet *, lt_packet);
/** @brief Binary heap of received packets */
static 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.
+ */
+static volatile uint32_t nsamples;
/** @brief Timestamp of next packet to play.
*
@@ -178,6 +241,12 @@ static uint32_t next_timestamp;
* This is true when playing and false when just buffering. */
static int active;
+/** @brief Lock protecting @ref packets */
+static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
+
+/** @brief Condition variable signalled whenever @ref packets is changed */
+static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+
/** @brief Structure of free packet list */
union free_packet {
struct packet p;
@@ -210,14 +279,8 @@ static union free_packet *next_free_packet;
*/
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 Condition variable signalled whenever @ref packets is changed */
-static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+/** @brief Lock protecting packet allocator */
+static pthread_mutex_t mem_lock = PTHREAD_MUTEX_INITIALIZER;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
@@ -227,15 +290,16 @@ static const struct option options[] = {
{ "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. */
+/** @brief Return a new packet */
static struct packet *new_packet(void) {
struct packet *p;
-
+
+ pthread_mutex_lock(&mem_lock);
if(free_packets) {
p = &free_packets->p;
free_packets = free_packets->next;
@@ -247,16 +311,17 @@ static struct packet *new_packet(void) {
p = &(next_free_packet++)->p;
--count_free_packets;
}
+ pthread_mutex_unlock(&mem_lock);
return p;
}
-/** @brief Free a packet
- *
- * Assumes that @ref lock is held. */
+/** @brief Free a packet */
static void free_packet(struct packet *p) {
union free_packet *u = (union free_packet *)p;
+ pthread_mutex_lock(&mem_lock);
u->next = free_packets;
free_packets = u;
+ pthread_mutex_unlock(&mem_lock);
}
/** @brief Drop the first packet
@@ -272,10 +337,57 @@ static void drop_first_packet(void) {
}
}
+/** @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,
- * and adds them to the packet list. */
+ * 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 new_packet().
+ */
static void *listen_thread(void attribute((unused)) *arg) {
struct packet *p = 0;
int n;
@@ -285,11 +397,8 @@ static void *listen_thread(void attribute((unused)) *arg) {
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;
@@ -316,7 +425,7 @@ static void *listen_thread(void attribute((unused)) *arg) {
timestamp, next_timestamp);
continue;
}
- pthread_mutex_lock(&lock);
+ p->next = 0;
p->flags = 0;
p->timestamp = timestamp;
/* Convert to target format */
@@ -325,9 +434,6 @@ static void *listen_thread(void attribute((unused)) *arg) {
switch(header.mpt & 0x7F) {
case 10:
p->nsamples = (n - sizeof header) / sizeof(uint16_t);
- /* ALSA can do any necessary conversion itself (though it might be better
- * to do any necessary conversion in the background) */
- /* TODO we could readv into the buffer */
break;
/* TODO support other RFC3551 media types (when the speaker does) */
default:
@@ -342,21 +448,27 @@ static void *listen_thread(void attribute((unused)) *arg) {
* 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);
}
}
-/** @brief Return true if @p p contains @p timestamp */
+/** @brief Return true if @p p contains @p timestamp
+ *
+ * Containment implies that a sample @p timestamp exists within the packet.
+ */
static inline int contains(const struct packet *p, uint32_t timestamp) {
const uint32_t packet_start = p->timestamp;
const uint32_t packet_end = p->timestamp + p->nsamples;
@@ -365,6 +477,43 @@ static inline int contains(const struct packet *p, uint32_t timestamp) {
&& lt(timestamp, packet_end));
}
+/** @brief Wait until the buffer is adequately full
+ *
+ * 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);
+ 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.
+ */
+static struct packet *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;
+ }
+ return 0;
+}
+
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
/** @brief Callback from Core Audio */
static OSStatus adioproc
@@ -377,33 +526,16 @@ static OSStatus adioproc
void attribute((unused)) *inClientData) {
UInt32 nbuffers = outOutputData->mNumberBuffers;
AudioBuffer *ab = outOutputData->mBuffers;
- const struct packet *p;
uint32_t samples_available;
- struct timeval in, out;
- gettimeofday(&in, 0);
pthread_mutex_lock(&lock);
while(nbuffers > 0) {
float *samplesOut = ab->mData;
size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
while(samplesOutLeft > 0) {
- /* Look for a suitable packet, dropping any unsuitable ones along the
- * way. Unsuitable packets are ones that are in the past. */
- while(pheap_count(&packets)) {
- 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.) */
- break;
- }
- p = pheap_count(&packets) ? pheap_first(&packets) : 0;
+ const struct packet *p = next_packet();
if(p && contains(p, next_timestamp)) {
- if(p->flags & IDLE)
- fprintf(stderr, "\nIDLE\n");
/* This packet is ready to play */
const uint32_t packet_end = p->timestamp + p->nsamples;
const uint32_t offset = next_timestamp - p->timestamp;
@@ -418,7 +550,6 @@ static OSStatus adioproc
*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
@@ -430,20 +561,12 @@ static OSStatus adioproc
next_timestamp += samples_available;
samplesOut += samples_available;
samplesOutLeft -= samples_available;
- write(2, "?", 1);
}
}
++ab;
--nbuffers;
}
pthread_mutex_unlock(&lock);
- gettimeofday(&out, 0);
- {
- static double max;
- double thistime = (out.tv_sec - in.tv_sec) + (out.tv_usec - in.tv_usec) / 1000000.0;
- if(thistime > max)
- fprintf(stderr, "adioproc: %8.8fs\n", max = thistime);
- }
return 0;
}
#endif
@@ -531,7 +654,7 @@ static void wait_alsa(void) {
}
}
-/** @brief Play some sound
+/** @brief Play some sound via ALSA
* @param s Pointer to sample data
* @param n Number of samples
* @return 0 on success, -1 on non-fatal error
@@ -564,7 +687,6 @@ static int alsa_writei(const void *s, size_t n) {
* @return 0 on success, -1 on non-fatal error
*/
static int alsa_play(const struct packet *p) {
- write(2, ".", 1);
return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
(p->timestamp + p->nsamples) - next_timestamp);
}
@@ -579,7 +701,6 @@ static int alsa_infill(const struct packet *p) {
if(p && samples_available > p->timestamp - next_timestamp)
samples_available = p->timestamp - next_timestamp;
- write(2, "?", 1);
return alsa_writei(zeros, samples_available);
}
@@ -601,41 +722,6 @@ static void alsa_reset(int hard_reset) {
}
#endif
-/** @brief Wait until the buffer is adequately full
- *
- * Must be called with @ref lock held.
- */
-static void fill_buffer(void) {
- 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.
- */
-static struct packet *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;
- }
- return 0;
-}
-
/** @brief Play an RTP stream
*
* This is the guts of the program. It is responsible for:
@@ -650,6 +736,8 @@ static void play_rtp(void) {
/* 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;
@@ -670,7 +758,10 @@ static void play_rtp(void) {
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();
@@ -741,7 +832,9 @@ static void play_rtp(void) {
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);
@@ -765,6 +858,8 @@ static void help(void) {
" --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"
);
@@ -784,6 +879,11 @@ int main(int argc, char **argv) {
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,
@@ -798,7 +898,7 @@ int main(int argc, char **argv) {
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();
@@ -808,6 +908,8 @@ int main(int argc, char **argv) {
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");
}
}
@@ -828,6 +930,44 @@ int main(int argc, char **argv) {
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;
}