2 * This file is part of DisOrder.
3 * Copyright (C) 2007 Richard Kettlewell
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
27 #include <sys/socket.h>
28 #include <sys/types.h>
29 #include <sys/socket.h>
37 #include "configuration.h"
43 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
44 # include <CoreAudio/AudioHardware.h>
47 #include <alsa/asoundlib.h>
50 #define readahead linux_headers_are_borked
52 /** @brief RTP socket */
55 /** @brief Log output */
58 /** @brief Output device */
59 static const char *device
;
61 /** @brief Maximum samples per packet we'll support
63 * NB that two channels = two samples in this program.
65 #define MAXSAMPLES 2048
67 /** @brief Minimum low watermark
69 * We'll stop playing if there's only this many samples in the buffer. */
70 static unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
72 /** @brief Maximum sample size
74 * The maximum supported size (in bytes) of one sample. */
75 #define MAXSAMPLESIZE 2
77 /** @brief Buffer high watermark
79 * We'll only start playing when this many samples are available. */
80 static unsigned readahead
= 2 * 2 * 44100;
82 /** @brief Maximum buffer size
84 * We'll stop reading from the network if we have this many samples. */
85 static unsigned maxbuffer
;
87 /** @brief Number of samples to infill by in one go */
88 #define INFILL_SAMPLES (44100 * 2) /* 1s */
90 /** @brief Received packet */
92 /** @brief Number of samples in this packet */
94 /** @brief Timestamp from RTP packet
96 * NB that "timestamps" are really sample counters.*/
98 /** @brief Raw sample data */
99 unsigned char samples_raw
[MAXSAMPLES
* MAXSAMPLESIZE
];
102 /** @brief Total number of samples available */
103 static unsigned long nsamples
;
105 /** @brief Mapping of sequence numbers to packets
107 * This isn't very efficient - 256KB on 32-bit machines, 512KB if you do a
108 * 64-bit build for some reason. It can be optimized later if need be. */
109 static struct packet
*packets
[65536];
111 /** @brief Total number of packets */
112 static unsigned npackets
;
114 /** @brief Timestamp of next packet to play.
116 * This is set to the timestamp of the last packet, plus the number of
117 * samples it contained. Only valid if @ref active is nonzero.
119 static uint32_t next_timestamp
;
121 /** @brief True if actively playing
123 * This is true when playing and false when just buffering. */
126 /** @brief Sequence number of next packet we expxect to play */
127 static uint16_t sequence
;
129 /** @brief Structure of free packet list */
132 union free_packet
*next
;
135 /** @brief Linked list of free packets */
136 static union free_packet
*free_packets
;
138 /** @brief Array of new free packets */
139 static union free_packet
*next_free_packet
;
141 /** @brief Count of new free packets */
142 static size_t count_free_packets
;
144 /** @brief Lock protecting @ref packets */
145 static pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
147 /** @brief Condition variable signalled whenever @ref packets is changed */
148 static pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
150 static const struct option options
[] = {
151 { "help", no_argument
, 0, 'h' },
152 { "version", no_argument
, 0, 'V' },
153 { "debug", no_argument
, 0, 'd' },
154 { "device", required_argument
, 0, 'D' },
155 { "min", required_argument
, 0, 'm' },
156 { "max", required_argument
, 0, 'x' },
157 { "buffer", required_argument
, 0, 'b' },
161 /** @brief Return a new packet
163 * Assumes that @ref lock is held. */
164 static struct packet
*new_packet(void) {
168 p
= &free_packets
->p
;
169 free_packets
= free_packets
->next
;
171 if(!count_free_packets
) {
172 next_free_packet
= xcalloc(1024, sizeof (union free_packet
));
173 count_free_packets
= 1024;
175 p
= &(next_free_packet
++)->p
;
176 --count_free_packets
;
181 /** @brief Free a packet
183 * Assumes that @ref lock is held. */
184 static void free_packet(struct packet
*p
) {
185 union free_packet
*u
= (union free_packet
*)p
;
186 u
->next
= free_packets
;
190 /** @brief Return true iff a < b in sequence-space arithmetic */
191 static inline int lt(uint32_t a
, uint32_t b
) {
192 return (uint32_t)(a
- b
) & 0x80000000;
195 /** @brief Return true iff a >= b in sequence-space arithmetic */
196 static inline int ge(uint32_t a
, uint32_t b
) {
200 /** @brief Return true iff a > b in sequence-space arithmetic */
201 static inline int gt(uint32_t a
, uint32_t b
) {
205 /** @brief Return true iff a <= b in sequence-space arithmetic */
206 static inline int le(uint32_t a
, uint32_t b
) {
210 /** @brief Drop the packet at the head of the queue */
211 static void drop_packet(unsigned sequence
) {
212 if(packets
[sequence
]) {
213 nsamples
-= packets
[sequence
]->nsamples
;
214 free_packet(packets
[sequence
]);
215 packets
[sequence
] = 0;
216 pthread_cond_broadcast(&cond
);
221 /** @brief Background thread collecting samples
223 * This function collects samples, perhaps converts them to the target format,
224 * and adds them to the packet list. */
225 static void *listen_thread(void attribute((unused
)) *arg
) {
226 struct packet
*p
= 0;
228 struct rtp_header header
;
235 pthread_mutex_lock(&lock
);
237 pthread_mutex_unlock(&lock
);
239 iov
[0].iov_base
= &header
;
240 iov
[0].iov_len
= sizeof header
;
241 iov
[1].iov_base
= p
->samples_raw
;
242 iov
[1].iov_len
= sizeof p
->samples_raw
;
243 n
= readv(rtpfd
, iov
, 2);
249 fatal(errno
, "error reading from socket");
252 /* Ignore too-short packets */
253 if((size_t)n
<= sizeof (struct rtp_header
)) {
254 info("ignored a short packet");
257 timestamp
= htonl(header
.timestamp
);
258 seq
= htons(header
.seq
);
259 /* Ignore packets in the past */
260 if(active
&& lt(timestamp
, next_timestamp
)) {
261 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
262 timestamp
, next_timestamp
);
265 pthread_mutex_lock(&lock
);
267 p
->timestamp
= timestamp
;
268 /* Convert to target format */
269 switch(header
.mpt
& 0x7F) {
271 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
272 /* ALSA can do any necessary conversion itself (though it might be better
273 * to do any necessary conversion in the background) */
274 /* TODO we could readv into the buffer */
276 /* TODO support other RFC3551 media types (when the speaker does) */
278 fatal(0, "unsupported RTP payload type %d",
282 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
283 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
284 /* Stop reading if we've reached the maximum.
286 * This is rather unsatisfactory: it means that if packets get heavily
287 * out of order then we guarantee dropouts. But for now... */
288 if(nsamples
>= maxbuffer
) {
290 while(nsamples
>= maxbuffer
)
291 pthread_cond_wait(&cond
, &lock
);
293 /* If there's a packet there already we overwrite it; perhaps it is left
294 * over from an earlier stage. */
296 /* Record this packet */
298 /* If we currently have no idea where to start playing, this is it */
302 nsamples
+= p
->nsamples
;
303 pthread_cond_broadcast(&cond
);
304 pthread_mutex_unlock(&lock
);
308 /** @brief Return true if @p p contains @p timestamp */
309 static inline int contains(const struct packet
*p
, uint32_t timestamp
) {
310 const uint32_t packet_start
= p
->timestamp
;
311 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
313 return (ge(timestamp
, packet_start
)
314 && lt(timestamp
, packet_end
));
317 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
318 /** @brief Callback from Core Audio */
319 static OSStatus adioproc
320 (AudioDeviceID
attribute((unused
)) inDevice
,
321 const AudioTimeStamp
attribute((unused
)) *inNow
,
322 const AudioBufferList
attribute((unused
)) *inInputData
,
323 const AudioTimeStamp
attribute((unused
)) *inInputTime
,
324 AudioBufferList
*outOutputData
,
325 const AudioTimeStamp
attribute((unused
)) *inOutputTime
,
326 void attribute((unused
)) *inClientData
) {
327 UInt32 nbuffers
= outOutputData
->mNumberBuffers
;
328 AudioBuffer
*ab
= outOutputData
->mBuffers
;
329 const struct packet
*p
;
331 pthread_mutex_lock(&lock
);
332 while(nbuffers
> 0) {
333 float *samplesOut
= ab
->mData
;
334 size_t samplesOutLeft
= ab
->mDataByteSize
/ sizeof (float);
336 while(samplesOutLeft
> 0) {
337 /* Look for a suitable packet, dropping any unsuitable ones along the
338 * way. Unsuitable packets are ones that are in the past. */
340 && (!packets
[sequence
]
341 || le(packets
[sequence
]->timestamp
342 + packets
[sequence
]->nsamples
,
344 drop_packet(sequence
++);
345 p
= packets
[sequence
];
347 if(contains(p
, next_timestamp
)) {
348 /* This packet is suitable */
349 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
350 const uint32_t offset
= next_timestamp
- p
->timestamp
;
351 const uint16_t *ptr
=
352 (void *)(p
->samples_raw
+ offset
* sizeof (uint16_t));
353 uint32_t samples_available
= packet_end
- next_timestamp
;
354 if(samples_available
> samplesOutLeft
)
355 samples_available
= samplesOutLeft
;
356 next_timestamp
+= samples_available
;
357 samplesOutLeft
-= samples_available
;
358 while(samples_available
-- > 0)
359 *samplesOut
++ = (int16_t)ntohs(*ptr
++) * (0.5 / 32767);
360 /* We don't bother junking the packet or advancing sequence - that'll
361 * be dealt with next time round */
365 /* We didn't find a suitable packet (though there might still be
366 * unsuitable ones). We infill with 0s. */
368 /* There is a next packet, only infill up to that point */
369 uint32_t samples_available
= p
->timestamp
- next_timestamp
;
371 if(samples_available
> samplesOutLeft
)
372 samples_available
= samplesOutLeft
;
373 info("infill by %"PRIu32
, samples_available
);
374 /* Convniently the buffer is 0 to start with */
375 next_timestamp
+= samples_available
;
376 samplesOut
+= samples_available
;
377 samplesOutLeft
-= samples_available
;
379 /* There's no next packet at all */
380 info("infilled by %zu", samplesOutLeft
);
381 next_timestamp
+= samplesOutLeft
;
382 samplesOut
+= samplesOutLeft
;
389 pthread_mutex_unlock(&lock
);
394 /** @brief Play an RTP stream
396 * This is the guts of the program. It is responsible for:
397 * - starting the listening thread
398 * - opening the audio device
399 * - reading ahead to build up a buffer
400 * - arranging for audio to be played
401 * - detecting when the buffer has got too small and re-buffering
403 static void play_rtp(void) {
406 /* We receive and convert audio data in a background thread */
407 pthread_create(<id
, 0, listen_thread
, 0);
411 snd_pcm_hw_params_t
*hwparams
;
412 snd_pcm_sw_params_t
*swparams
;
413 /* Only support one format for now */
414 const int sample_format
= SND_PCM_FORMAT_S16_BE
;
415 unsigned rate
= 44100;
416 const int channels
= 2;
417 const int samplesize
= channels
* sizeof(uint16_t);
418 snd_pcm_uframes_t pcm_bufsize
= MAXSAMPLES
* samplesize
* 3;
419 /* If we can write more than this many samples we'll get a wakeup */
420 const int avail_min
= 256;
421 snd_pcm_sframes_t frames_written
;
422 size_t samples_written
;
425 int infilling
= 0, escape
= 0;
427 uint32_t packet_start
, packet_end
;
430 if((err
= snd_pcm_open(&pcm
,
431 device ? device
: "default",
432 SND_PCM_STREAM_PLAYBACK
,
434 fatal(0, "error from snd_pcm_open: %d", err
);
435 /* Set up 'hardware' parameters */
436 snd_pcm_hw_params_alloca(&hwparams
);
437 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
438 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
439 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
440 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
441 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
442 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
444 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
446 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0)
447 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
449 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
451 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
453 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
455 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
456 MAXSAMPLES
* samplesize
* 3, err
);
457 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
458 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
459 /* Set up 'software' parameters */
460 snd_pcm_sw_params_alloca(&swparams
);
461 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
462 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
463 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, avail_min
)) < 0)
464 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
466 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
467 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
472 pthread_mutex_lock(&lock
);
474 /* Wait for the buffer to fill up a bit */
476 info("%lu samples in buffer (%lus)", nsamples
,
477 nsamples
/ (44100 * 2));
478 info("Buffering...");
479 while(nsamples
< readahead
)
480 pthread_cond_wait(&cond
, &lock
);
482 if((err
= snd_pcm_prepare(pcm
)))
483 fatal(0, "error calling snd_pcm_prepare: %d", err
);
486 assert(sequence
!= -1);
487 /* Start at the first available packet */
488 next_timestamp
= packets
[sequence
]->timestamp
;
493 info("%lu samples in buffer (%lus)", nsamples
,
494 nsamples
/ (44100 * 2));
496 /* Wait until the buffer empties out */
497 while(nsamples
>= minbuffer
&& !escape
) {
499 if(now
> logged
+ 10) {
501 info("%lu samples in buffer (%lus)", nsamples
,
502 nsamples
/ (44100 * 2));
505 && ge(next_timestamp
, packets
->timestamp
+ packets
->nsamples
)) {
506 info("dropping buffered past packet %"PRIx32
" < %"PRIx32
,
507 packets
->timestamp
, next_timestamp
);
511 /* Wait for ALSA to ask us for more data */
512 pthread_mutex_unlock(&lock
);
513 write(2, ".", 1); /* TODO remove me sometime */
514 switch(err
= snd_pcm_wait(pcm
, -1)) {
516 info("snd_pcm_wait timed out");
521 fatal(0, "snd_pcm_wait returned %d", err
);
523 pthread_mutex_lock(&lock
);
524 /* ALSA is ready for more data */
525 packet_start
= packets
->timestamp
;
526 packet_end
= packets
->timestamp
+ packets
->nsamples
;
527 if(ge(next_timestamp
, packet_start
)
528 && lt(next_timestamp
, packet_end
)) {
529 /* The target timestamp is somewhere in this packet */
530 const uint32_t offset
= next_timestamp
- packets
->timestamp
;
531 const uint32_t samples_available
= (packets
->timestamp
+ packets
->nsamples
) - next_timestamp
;
532 const size_t frames_available
= samples_available
/ 2;
534 frames_written
= snd_pcm_writei(pcm
,
535 packets
->samples_raw
+ offset
,
537 if(frames_written
< 0) {
538 switch(frames_written
) {
540 info("snd_pcm_wait() returned but we got -EAGAIN!");
543 error(0, "error calling snd_pcm_writei: %ld",
544 (long)frames_written
);
548 fatal(0, "error calling snd_pcm_writei: %ld",
549 (long)frames_written
);
552 samples_written
= frames_written
* 2;
553 next_timestamp
+= samples_written
;
554 if(ge(next_timestamp
, packet_end
))
559 /* We don't have anything to play! We'd better play some 0s. */
560 static const uint16_t zeros
[INFILL_SAMPLES
];
561 size_t samples_available
= INFILL_SAMPLES
, frames_available
;
563 /* If the maximum infill would take us past the start of the next
564 * packet then we truncate the infill to the right amount. */
565 if(lt(packets
->timestamp
,
566 next_timestamp
+ samples_available
))
567 samples_available
= packets
->timestamp
- next_timestamp
;
568 if((int)samples_available
< 0) {
569 info("packets->timestamp: %"PRIx32
" next_timestamp: %"PRIx32
" next+max: %"PRIx32
" available: %"PRIx32
,
570 packets
->timestamp
, next_timestamp
,
571 next_timestamp
+ INFILL_SAMPLES
, samples_available
);
573 frames_available
= samples_available
/ 2;
575 info("Infilling %d samples, next=%"PRIx32
" packet=[%"PRIx32
",%"PRIx32
"]",
576 samples_available
, next_timestamp
,
577 packets
->timestamp
, packets
->timestamp
+ packets
->nsamples
);
580 frames_written
= snd_pcm_writei(pcm
,
583 if(frames_written
< 0) {
584 switch(frames_written
) {
586 info("snd_pcm_wait() returned but we got -EAGAIN!");
589 error(0, "error calling snd_pcm_writei: %ld",
590 (long)frames_written
);
594 fatal(0, "error calling snd_pcm_writei: %ld",
595 (long)frames_written
);
598 samples_written
= frames_written
* 2;
599 next_timestamp
+= samples_written
;
604 /* We stop playing for a bit until the buffer re-fills */
605 pthread_mutex_unlock(&lock
);
606 if((err
= snd_pcm_nonblock(pcm
, 0)))
607 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
609 if((err
= snd_pcm_drop(pcm
)))
610 fatal(0, "error calling snd_pcm_drop: %d", err
);
613 if((err
= snd_pcm_drain(pcm
)))
614 fatal(0, "error calling snd_pcm_drain: %d", err
);
615 if((err
= snd_pcm_nonblock(pcm
, 1)))
616 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
618 pthread_mutex_lock(&lock
);
622 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
627 AudioStreamBasicDescription asbd
;
629 /* If this looks suspiciously like libao's macosx driver there's an
630 * excellent reason for that... */
632 /* TODO report errors as strings not numbers */
633 propertySize
= sizeof adid
;
634 status
= AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice
,
635 &propertySize
, &adid
);
637 fatal(0, "AudioHardwareGetProperty: %d", (int)status
);
638 if(adid
== kAudioDeviceUnknown
)
639 fatal(0, "no output device");
640 propertySize
= sizeof asbd
;
641 status
= AudioDeviceGetProperty(adid
, 0, false,
642 kAudioDevicePropertyStreamFormat
,
643 &propertySize
, &asbd
);
645 fatal(0, "AudioHardwareGetProperty: %d", (int)status
);
646 D(("mSampleRate %f", asbd
.mSampleRate
));
647 D(("mFormatID %08lx", asbd
.mFormatID
));
648 D(("mFormatFlags %08lx", asbd
.mFormatFlags
));
649 D(("mBytesPerPacket %08lx", asbd
.mBytesPerPacket
));
650 D(("mFramesPerPacket %08lx", asbd
.mFramesPerPacket
));
651 D(("mBytesPerFrame %08lx", asbd
.mBytesPerFrame
));
652 D(("mChannelsPerFrame %08lx", asbd
.mChannelsPerFrame
));
653 D(("mBitsPerChannel %08lx", asbd
.mBitsPerChannel
));
654 D(("mReserved %08lx", asbd
.mReserved
));
655 if(asbd
.mFormatID
!= kAudioFormatLinearPCM
)
656 fatal(0, "audio device does not support kAudioFormatLinearPCM");
657 status
= AudioDeviceAddIOProc(adid
, adioproc
, 0);
659 fatal(0, "AudioDeviceAddIOProc: %d", (int)status
);
660 pthread_mutex_lock(&lock
);
662 /* Wait for the buffer to fill up a bit */
663 info("Buffering...");
664 while(nsamples
< readahead
)
665 pthread_cond_wait(&cond
, &lock
);
666 /* Start playing now */
668 next_timestamp
= packets
[sequence
]->timestamp
;
670 status
= AudioDeviceStart(adid
, adioproc
);
672 fatal(0, "AudioDeviceStart: %d", (int)status
);
673 /* Wait until the buffer empties out */
674 while(nsamples
>= minbuffer
)
675 pthread_cond_wait(&cond
, &lock
);
676 /* Stop playing for a bit until the buffer re-fills */
677 status
= AudioDeviceStop(adid
, adioproc
);
679 fatal(0, "AudioDeviceStop: %d", (int)status
);
685 # error No known audio API
689 /* display usage message and terminate */
690 static void help(void) {
692 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
694 " --device, -D DEVICE Output device\n"
695 " --min, -m FRAMES Buffer low water mark\n"
696 " --buffer, -b FRAMES Buffer high water mark\n"
697 " --max, -x FRAMES Buffer maximum size\n"
698 " --help, -h Display usage message\n"
699 " --version, -V Display version number\n"
705 /* display version number and terminate */
706 static void version(void) {
707 xprintf("disorder-playrtp version %s\n", disorder_version_string
);
712 int main(int argc
, char **argv
) {
714 struct addrinfo
*res
;
715 struct stringlist sl
;
718 static const struct addrinfo prefs
= {
730 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
731 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:", options
, 0)) >= 0) {
735 case 'd': debugging
= 1; break;
736 case 'D': device
= optarg
; break;
737 case 'm': minbuffer
= 2 * atol(optarg
); break;
738 case 'b': readahead
= 2 * atol(optarg
); break;
739 case 'x': maxbuffer
= 2 * atol(optarg
); break;
740 case 'L': logfp
= fopen(optarg
, "w"); break;
741 default: fatal(0, "invalid option");
745 maxbuffer
= 4 * readahead
;
748 if(argc
< 1 || argc
> 2)
749 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
752 /* Listen for inbound audio data */
753 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
755 if((rtpfd
= socket(res
->ai_family
,
757 res
->ai_protocol
)) < 0)
758 fatal(errno
, "error creating socket");
759 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
760 fatal(errno
, "error binding socket to %s", sockname
);