X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/e9b635a3223a51d0c30867e42ec4f9b54b62e591..0e72bf84b9d3a45de98bb3dbb30ef2d2aaabb4ca:/clients/playrtp.c
diff --git a/clients/playrtp.c b/clients/playrtp.c
index 1263d7a..a5542db 100644
--- a/clients/playrtp.c
+++ b/clients/playrtp.c
@@ -1,21 +1,19 @@
/*
* This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007, 2008 Richard Kettlewell
*
- * This program is free software; you can redistribute it and/or modify
+ * This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
* You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
- * USA
+ * along with this program. If not, see .
*/
/** @file clients/playrtp.c
* @brief RTP player
@@ -26,20 +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:
+ *
+ * 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.
+ * control_thread() accepts commands from Disobedience (or anything else).
*
- * 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.
+ * 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
@@ -49,12 +47,9 @@
* - it is safe to read uint32_t values without a lock protecting them
*/
-#include
-#include "types.h"
+#include "common.h"
#include
-#include
-#include
#include
#include
#include
@@ -62,8 +57,6 @@
#include
#include
#include
-#include
-#include
#include
#include
#include
@@ -85,8 +78,8 @@
#include "client.h"
#include "playrtp.h"
#include "inputline.h"
-
-#define readahead linux_headers_are_borked
+#include "version.h"
+#include "uaudio.h"
/** @brief Obsolete synonym */
#ifndef IPV6_JOIN_GROUP
@@ -100,18 +93,12 @@ static int rtpfd;
static FILE *logfp;
/** @brief Output device */
-const char *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 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. */
@@ -174,27 +161,34 @@ 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 || EMPEG_HOST
-# 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;
+static const struct uaudio *backend;
HEAP_DEFINE(pheap, struct packet *, lt_packet);
/** @brief Control socket or NULL */
const char *control_socket;
+/** @brief Buffer for debugging dump
+ *
+ * The debug dump is enabled by the @c --dump option. It records the last 20s
+ * of audio to the specified file (which will be about 3.5Mbytes). The file is
+ * written as as ring buffer, so the start point will progress through it.
+ *
+ * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
+ * into (e.g.) Audacity for further inspection.
+ *
+ * All three backends (ALSA, OSS, Core Audio) now support this option.
+ *
+ * The idea is to allow the user a few seconds to react to an audible artefact.
+ */
int16_t *dump_buffer;
+
+/** @brief Current index within debugging dump */
size_t dump_index;
-size_t dump_size = 44100 * 2 * 20; /* 20s */
+
+/** @brief Size of debugging dump in samples */
+size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
@@ -203,7 +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' },
@@ -215,6 +208,8 @@ static const struct option options[] = {
{ "core-audio", no_argument, 0, 'c' },
#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' },
{ 0, 0, 0, 0 }
@@ -312,8 +307,9 @@ static void *queue_thread(void attribute((unused)) *arg) {
for(;;) {
/* Get the next packet */
pthread_mutex_lock(&receive_lock);
- while(!received_packets)
+ while(!received_packets) {
pthread_cond_wait(&receive_cond, &receive_lock);
+ }
p = received_packets;
received_packets = p->next;
if(!received_packets)
@@ -387,6 +383,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
timestamp, next_timestamp);
continue;
}
+ /* Ignore packets with the extension bit set. */
+ if(header.vpxcc & 0x10)
+ continue;
p->next = 0;
p->flags = 0;
p->timestamp = timestamp;
@@ -394,7 +393,7 @@ static void *listen_thread(void attribute((unused)) *arg) {
if(header.mpt & 0x80)
p->flags |= IDLE;
switch(header.mpt & 0x7F) {
- case 10:
+ case 10: /* L16 */
p->nsamples = (n - sizeof header) / sizeof(uint16_t);
break;
/* TODO support other RFC3551 media types (when the speaker does) */
@@ -411,8 +410,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
* out of order then we guarantee dropouts. But for now... */
if(nsamples >= maxbuffer) {
pthread_mutex_lock(&lock);
- while(nsamples >= maxbuffer)
+ while(nsamples >= maxbuffer) {
pthread_cond_wait(&cond, &lock);
+ }
pthread_mutex_unlock(&lock);
}
/* Add the packet to the receive queue */
@@ -432,11 +432,15 @@ static void *listen_thread(void attribute((unused)) *arg) {
* Must be called with @ref lock held.
*/
void playrtp_fill_buffer(void) {
+ /* Discard current buffer contents */
while(nsamples)
drop_first_packet();
info("Buffering...");
- while(nsamples < readahead)
+ /* Wait until there's at least minbuffer samples available */
+ while(nsamples < minbuffer) {
pthread_cond_wait(&cond, &lock);
+ }
+ /* Start from whatever is earliest */
next_timestamp = pheap_first(&packets)->timestamp;
active = 1;
}
@@ -464,37 +468,13 @@ struct packet *playrtp_next_packet(void) {
return 0;
}
-/** @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;
- int err;
-
- /* We receive and convert audio data in a background thread */
- if((err = pthread_create(<id, 0, listen_thread, 0)))
- fatal(err, "pthread_create listen_thread");
- /* We have a second thread to add received packets to the queue */
- if((err = pthread_create(<id, 0, queue_thread, 0)))
- fatal(err, "pthread_create queue_thread");
- /* The rest of the work is backend-specific */
- backend();
-}
-
/* display usage message and terminate */
static void help(void) {
xprintf("Usage:\n"
- " disorder-playrtp [OPTIONS] ADDRESS [PORT]\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"
" --config, -C PATH Set configuration file\n"
@@ -507,6 +487,9 @@ 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"
+ " --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"
);
@@ -514,11 +497,64 @@ static void help(void) {
exit(0);
}
-/* display version number and terminate */
-static void version(void) {
- xprintf("disorder-playrtp version %s\n", disorder_version_string);
- xfclose(stdout);
- exit(0);
+static size_t playrtp_callback(void *buffer,
+ size_t max_samples,
+ void attribute((unused)) *userdata) {
+ size_t samples;
+
+ pthread_mutex_lock(&lock);
+ /* Get the next packet, junking any that are now in the past */
+ const struct packet *p = playrtp_next_packet();
+ if(p && contains(p, next_timestamp)) {
+ /* This packet is ready to play; the desired next timestamp points
+ * somewhere into it. */
+
+ /* Timestamp of end of packet */
+ const uint32_t packet_end = p->timestamp + p->nsamples;
+
+ /* Offset of desired next timestamp into current packet */
+ const uint32_t offset = next_timestamp - p->timestamp;
+
+ /* Pointer to audio data */
+ const uint16_t *ptr = (void *)(p->samples_raw + offset);
+
+ /* Compute number of samples left in packet, limited to output buffer
+ * size */
+ samples = packet_end - next_timestamp;
+ if(samples > max_samples)
+ samples = max_samples;
+
+ /* Copy into buffer, converting to native endianness */
+ size_t i = samples;
+ int16_t *bufptr = buffer;
+ while(i > 0) {
+ *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
+ * comparison with max_samples deals with the otherwise troubling overflow
+ * case. */
+ samples = p ? p->timestamp - next_timestamp : max_samples;
+ if(samples > max_samples)
+ samples = max_samples;
+ //info("infill by %zu", samples);
+ memset(buffer, 0, samples * uaudio_sample_size);
+ }
+ /* Debug dump */
+ if(dump_buffer) {
+ for(size_t i = 0; i < samples; ++i) {
+ dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
+ dump_index %= dump_size;
+ }
+ }
+ /* Advance timestamp */
+ next_timestamp += samples;
+ pthread_mutex_unlock(&lock);
+ return samples;
}
int main(int argc, char **argv) {
@@ -526,7 +562,7 @@ 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;
@@ -540,49 +576,52 @@ int main(int argc, char **argv) {
};
union any_sockaddr mgroup;
const char *dumpfile = 0;
+ pthread_t ltid;
+ static const int one = 1;
static const struct addrinfo prefs = {
- AI_PASSIVE,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
+ .ai_flags = AI_PASSIVE,
+ .ai_family = PF_INET,
+ .ai_socktype = SOCK_DGRAM,
+ .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");
- while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
+ backend = uaudio_apis[0];
+ while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:M:aocC:re:P:", options, 0)) >= 0) {
switch(n) {
case 'h': help();
- case 'V': version();
+ case 'V': version("disorder-playrtp");
case 'd': debugging = 1; break;
- case 'D': device = optarg; 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;
#if HAVE_ALSA_ASOUNDLIB_H
- case 'a': backend = playrtp_alsa; break;
+ case 'a': backend = &uaudio_alsa; break;
#endif
#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
- case 'o': backend = playrtp_oss; break;
+ case 'o': backend = &uaudio_oss; break;
#endif
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- case 'c': backend = playrtp_coreaudio; break;
+ case 'c': backend = &uaudio_coreaudio; break;
#endif
case 'C': configfile = optarg; break;
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;
default: fatal(0, "invalid option");
}
}
if(config_read(0)) fatal(0, "cannot read configuration");
if(!maxbuffer)
- maxbuffer = 4 * readahead;
+ maxbuffer = 2 * minbuffer;
argc -= optind;
argv += optind;
switch(argc) {
@@ -613,8 +652,14 @@ int main(int argc, char **argv) {
res->ai_socktype,
res->ai_protocol)) < 0)
fatal(errno, "error creating socket");
- /* Stash the multicast group address */
- if((is_multicast = multicast(res->ai_addr))) {
+ /* Allow multiple listeners */
+ xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
+ is_multicast = multicast(res->ai_addr);
+ /* The multicast and unicast/broadcast cases are different enough that they
+ * are totally split. Trying to find commonality between them causes more
+ * trouble that it's worth. */
+ if(is_multicast) {
+ /* Stash the multicast group address */
memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
switch(res->ai_addr->sa_family) {
case AF_INET:
@@ -623,24 +668,13 @@ int main(int argc, char **argv) {
case AF_INET6:
mgroup.in6.sin6_port = 0;
break;
+ default:
+ fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
}
- }
- /* Bind to 0/port */
- switch(res->ai_addr->sa_family) {
- case AF_INET:
- memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
- sizeof (struct in_addr));
- break;
- case AF_INET6:
- memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
- sizeof (struct in6_addr));
- break;
- default:
- fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
- }
- if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error binding socket to %s", sockname);
- if(is_multicast) {
+ /* Bind to to the multicast group address */
+ if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
+ /* Add multicast group membership */
switch(mgroup.sa.sa_family) {
case PF_INET:
mreq.imr_multiaddr = mgroup.in.sin_addr;
@@ -659,10 +693,35 @@ int main(int argc, char **argv) {
default:
fatal(0, "unsupported address family %d", res->ai_family);
}
+ /* Report what we did */
info("listening on %s multicast group %s",
format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
- } else
+ } else {
+ /* Bind to 0/port */
+ switch(res->ai_addr->sa_family) {
+ case AF_INET: {
+ struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
+
+ memset(&in->sin_addr, 0, sizeof (struct in_addr));
+ if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error binding socket to 0.0.0.0 port %d",
+ ntohs(in->sin_port));
+ break;
+ }
+ case AF_INET6: {
+ struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)res->ai_addr;
+
+ memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
+ break;
+ }
+ default:
+ fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
+ }
+ if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
+ /* Report what we did */
info("listening on %s", format_sockaddr(res->ai_addr));
+ }
len = sizeof rcvbuf;
if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
fatal(errno, "error calling getsockopt SO_RCVBUF");
@@ -706,7 +765,49 @@ int main(int argc, char **argv) {
fatal(errno, "mapping %s", dumpfile);
info("dumping to %s", dumpfile);
}
- play_rtp();
+ /* Set up output. Currently we only support L16 so there's no harm setting
+ * the format before we know what it is! */
+ uaudio_set_format(44100/*Hz*/, 2/*channels*/,
+ 16/*bits/channel*/, 1/*signed*/);
+ backend->start(playrtp_callback, NULL);
+ /* We receive and convert audio data in a background thread */
+ if((err = pthread_create(<id, 0, listen_thread, 0)))
+ fatal(err, "pthread_create listen_thread");
+ /* We have a second thread to add received packets to the queue */
+ if((err = pthread_create(<id, 0, queue_thread, 0)))
+ fatal(err, "pthread_create queue_thread");
+ pthread_mutex_lock(&lock);
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ playrtp_fill_buffer();
+ /* Start playing now */
+ info("Playing...");
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+ pthread_mutex_unlock(&lock);
+ backend->activate();
+ pthread_mutex_lock(&lock);
+ /* 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))) {
+ pthread_cond_wait(&cond, &lock);
+ }
+ /* Stop playing for a bit until the buffer re-fills */
+ pthread_mutex_unlock(&lock);
+ backend->deactivate();
+ pthread_mutex_lock(&lock);
+ active = 0;
+ /* Go back round */
+ }
return 0;
}