/*
* This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007-2009, 2011, 2013 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 <http://www.gnu.org/licenses/>.
+ */
+/** @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).
+ *
+ * control_thread() accepts commands from Disobedience (or anything else).
+ *
+ * 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
+ * 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 "common.h"
#include <getopt.h>
-#include <stdio.h>
-#include <stdlib.h>
#include <sys/socket.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netdb.h>
#include <pthread.h>
+#include <locale.h>
+#include <sys/uio.h>
+#include <errno.h>
+#include <netinet/in.h>
+#include <sys/time.h>
+#include <sys/un.h>
+#include <unistd.h>
+#include <sys/mman.h>
+#include <fcntl.h>
+#include <math.h>
+#include <arpa/inet.h>
+#include <ifaddrs.h>
+#include <net/if.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 "client.h"
+#include "playrtp.h"
+#include "inputline.h"
+#include "version.h"
+#include "uaudio.h"
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-# include <CoreAudio/AudioHardware.h>
+/** @brief Obsolete synonym */
+#ifndef IPV6_JOIN_GROUP
+# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
#endif
+/** @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;
+
+/** @brief Output device */
+
+/** @brief Buffer low watermark in samples */
+unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
+
+/** @brief Maximum buffer size in samples
+ *
+ * We'll stop reading from the network if we have this many samples.
+ */
+static unsigned maxbuffer;
-static unsigned long nsamples; /* total samples available */
+/** @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;
-static struct frame *frames; /* received frames in ascending order
- * of timestamp */
-static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
-/* lock protecting frame list */
+/** @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;
-static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER;
-/* signalled whenever we add a new frame */
+/** @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;
+
+/** @brief Backend to play with */
+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;
+
+/** @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' },
{ "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' },
+ { "rcvbuf", required_argument, 0, 'R' },
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ { "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
+ { "api", required_argument, 0, 'A' },
+ { "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' },
+ { "monitor", no_argument, 0, 'M' },
{ 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;
-}
-
-/* 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 Control thread
+ *
+ * This thread is responsible for accepting control commands from Disobedience
+ * (or other controllers) over an AF_UNIX stream socket with a path specified
+ * by the @c --socket option. The protocol uses simple string commands and
+ * replies:
+ *
+ * - @c stop will shut the player down
+ * - @c query will send back the reply @c running
+ * - anything else is ignored
+ *
+ * Commands and response strings terminated by shutting down the connection or
+ * by a newline. No attempt is made to multiplex multiple clients so it is
+ * important that the command be sent as soon as the connection is made - it is
+ * assumed that both parties to the protocol are entirely cooperating with one
+ * another.
+ */
+static void *control_thread(void attribute((unused)) *arg) {
+ struct sockaddr_un sa;
+ int sfd, cfd;
+ char *line;
+ socklen_t salen;
+ FILE *fp;
+ assert(control_socket);
+ unlink(control_socket);
+ memset(&sa, 0, sizeof sa);
+ sa.sun_family = AF_UNIX;
+ strcpy(sa.sun_path, control_socket);
+ sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
+ if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
+ disorder_fatal(errno, "error binding to %s", control_socket);
+ if(listen(sfd, 128) < 0)
+ disorder_fatal(errno, "error calling listen on %s", control_socket);
+ disorder_info("listening on %s", control_socket);
for(;;) {
- if(!f)
- f = xmalloc(sizeof *f);
- n = read(rtpfd, packet.bytes, sizeof packet.bytes);
- if(n < 0) {
+ salen = sizeof sa;
+ cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
+ if(cfd < 0) {
switch(errno) {
case EINTR:
- continue;
+ case EAGAIN:
+ break;
default:
- fatal(errno, "error reading from socket");
+ disorder_fatal(errno, "error calling accept on %s", control_socket);
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- /* Convert to target format */
- switch(packet.header.mtp & 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);
- break;
- /* TODO support other RFC3551 media types (when the speaker does) */
- default:
- fatal(0, "unsupported RTP payload type %d",
- packet.header.mpt & 0x7F);
+ if(!(fp = fdopen(cfd, "r+"))) {
+ disorder_error(errno, "error calling fdopen for %s connection", control_socket);
+ close(cfd);
+ continue;
}
-#endif
- f->used = 0;
- f->timestamp = ntohl(packet.header.timestamp);
+ if(!inputline(control_socket, fp, &line, '\n')) {
+ if(!strcmp(line, "stop")) {
+ disorder_info("stopped via %s", control_socket);
+ exit(0); /* terminate immediately */
+ }
+ if(!strcmp(line, "query"))
+ fprintf(fp, "running");
+ xfree(line);
+ }
+ if(fclose(fp) < 0)
+ disorder_error(errno, "error closing %s connection", control_socket);
+ }
+}
+
+/** @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);
- /* 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;
+ pheap_insert(&packets, p);
+ nsamples += p->nsamples;
pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
- f = 0;
}
+#if HAVE_STUPID_GCC44
+ return NULL;
+#endif
}
-#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);
+/** @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(!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:
+ continue;
+ default:
+ disorder_fatal(errno, "error reading from socket");
+ }
+ }
+ /* Ignore too-short packets */
+ if((size_t)n <= sizeof (struct rtp_header)) {
+ disorder_info("ignored a short packet");
continue;
}
- if(samplesOutLeft == 0) {
- --nbuffers;
- ++ab;
- samplesOut = ab->data;
- samplesOutLeft = ab->mDataByteSize / sizeof (float);
+ timestamp = htonl(header.timestamp);
+ seq = htons(header.seq);
+ /* Ignore packets in the past */
+ if(active && lt(timestamp, next_timestamp)) {
+ disorder_info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
+ timestamp, next_timestamp);
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;
+ /* Ignore packets with the extension bit set. */
+ if(header.vpxcc & 0x10)
+ continue;
+ p->next = 0;
+ p->flags = 0;
+ p->timestamp = timestamp;
+ /* Convert to target format */
+ if(header.mpt & 0x80)
+ p->flags |= IDLE;
+ switch(header.mpt & 0x7F) {
+ case 10: /* L16 */
+ p->nsamples = (n - sizeof header) / sizeof(uint16_t);
+ break;
+ /* TODO support other RFC3551 media types (when the speaker does) */
+ default:
+ disorder_fatal(0, "unsupported RTP payload type %d", header.mpt & 0x7F);
+ }
+ /* See if packet is silent */
+ const uint16_t *s = p->samples_raw;
+ n = p->nsamples;
+ for(; n > 0; --n)
+ if(*s++)
+ break;
+ if(!n)
+ p->flags |= SILENT;
+ 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;
}
- pthread_mutex_unlock(&lock);
- return 0;
}
-#endif
-void play_rtp(void) {
- pthread_t lt;
+/** @brief Wait until the buffer is adequately full
+ *
+ * Must be called with @ref lock held.
+ */
+void playrtp_fill_buffer(void) {
+ /* Discard current buffer contents */
+ while(nsamples) {
+ //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
+ drop_first_packet();
+ }
+ disorder_info("Buffering...");
+ /* Wait until there's at least minbuffer samples available */
+ while(nsamples < minbuffer) {
+ //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
+ pthread_cond_wait(&cond, &lock);
+ }
+ /* Start from whatever is earliest */
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
+}
- /* 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 */
- }
+/** @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;
}
-#else
-# error No known audio API
-#endif
+ return 0;
}
/* 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"
+ " --max, -x FRAMES Buffer maximum size\n"
+ " --rcvbuf, -R BYTES Socket receive buffer size\n"
+ " --config, -C PATH Set configuration file\n"
+ " --api, -A API Select audio API. Possibilities:\n"
+ " ");
+ int first = 1;
+ for(int n = 0; uaudio_apis[n]; ++n) {
+ if(uaudio_apis[n]->flags & UAUDIO_API_CLIENT) {
+ if(first)
+ first = 0;
+ else
+ xprintf(", ");
+ xprintf("%s", uaudio_apis[n]->name);
+ }
+ }
+ xprintf("\n"
+ " --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"
- " --debug, -d Turn on debugging\n");
+ );
xfclose(stdout);
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;
+ int silent = 0;
+
+ 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;
+ }
+ silent = !!(p->flags & SILENT);
+ } 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);
+ silent = 1;
+ }
+ /* 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;
+ /* If we're getting behind then try to drop just silent packets
+ *
+ * In theory this shouldn't be necessary. The server is supposed to send
+ * packets at the right rate and compares the number of samples sent with the
+ * time in order to ensure this.
+ *
+ * However, various things could throw this off:
+ *
+ * - the server's clock could advance at the wrong rate. This would cause it
+ * to mis-estimate the right number of samples to have sent and
+ * inappropriately throttle or speed up.
+ *
+ * - playback could happen at the wrong rate. If the playback host's sound
+ * card has a slightly incorrect clock then eventually it will get out
+ * of step.
+ *
+ * So if we play back slightly slower than the server sends for either of
+ * these reasons then eventually our buffer, and the socket's buffer, will
+ * fill, and the kernel will start dropping packets. The result is audible
+ * and not very nice.
+ *
+ * Therefore if we're getting behind, we pre-emptively drop silent packets,
+ * since a change in the duration of a silence is less noticeable than a
+ * dropped packet from the middle of continuous music.
+ *
+ * (If things go wrong the other way then eventually we run out of packets to
+ * play and are forced to play silence. This doesn't seem to happen in
+ * practice but if it does then in the same way we can artificially extend
+ * silent packets to compensate.)
+ *
+ * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
+ * track how close to target buffer occupancy we are on a once-a-minute
+ * basis.
+ */
+ if(nsamples > minbuffer && silent) {
+ disorder_info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
+ samples, nsamples, minbuffer);
+ samples = 0;
+ }
+ /* Junk obsolete packets */
+ playrtp_next_packet();
+ pthread_mutex_unlock(&lock);
+ return samples;
+}
+
+static int compare_family(const struct ifaddrs *a,
+ const struct ifaddrs *b,
+ int family) {
+ int afamily = a->ifa_addr->sa_family;
+ int bfamily = b->ifa_addr->sa_family;
+ if(afamily != bfamily) {
+ /* Preferred family wins */
+ if(afamily == family) return 1;
+ if(bfamily == family) return -1;
+ /* Either there's no preference or it doesn't help. Prefer IPv4 */
+ if(afamily == AF_INET) return 1;
+ if(bfamily == AF_INET) return -1;
+ /* Failing that prefer IPv6 */
+ if(afamily == AF_INET6) return 1;
+ if(bfamily == AF_INET6) return -1;
+ }
+ return 0;
+}
+
+static int compare_flags(const struct ifaddrs *a,
+ const struct ifaddrs *b) {
+ unsigned aflags = a->ifa_flags, bflags = b->ifa_flags;
+ /* Up interfaces are better than down ones */
+ unsigned aup = aflags & IFF_UP, bup = bflags & IFF_UP;
+ if(aup != bup)
+ return aup > bup ? 1 : -1;
+ /* Static addresses are better than dynamic */
+ unsigned adynamic = aflags & IFF_DYNAMIC, bdynamic = bflags & IFF_DYNAMIC;
+ if(adynamic != bdynamic)
+ return adynamic < bdynamic ? 1 : -1;
+ unsigned aloopback = aflags & IFF_LOOPBACK, bloopback = bflags & IFF_LOOPBACK;
+ /* Static addresses are better than dynamic */
+ if(aloopback != bloopback)
+ return aloopback < bloopback ? 1 : -1;
+ return 0;
+}
+
+static int compare_interfaces(const struct ifaddrs *a,
+ const struct ifaddrs *b,
+ int family) {
+ int c;
+ if((c = compare_family(a, b, family))) return c;
+ return compare_flags(a, b);
}
int main(int argc, char **argv) {
- int n;
+ int n, err;
struct addrinfo *res;
struct stringlist sl;
- const char *sockname;
-
- static const struct addrinfo prefbind = {
- AI_PASSIVE,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
+ char *sockname;
+ int rcvbuf, target_rcvbuf = 0;
+ socklen_t len;
+ struct ip_mreq mreq;
+ struct ipv6_mreq mreq6;
+ disorder_client *c = NULL;
+ char *address, *port;
+ int is_multicast;
+ union any_sockaddr {
+ struct sockaddr sa;
+ struct sockaddr_in in;
+ struct sockaddr_in6 in6;
};
+ union any_sockaddr mgroup;
+ const char *dumpfile = 0;
+ pthread_t ltid;
+ int monitor = 0;
+ static const int one = 1;
+ static const struct addrinfo prefs = {
+ .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, "hVd", options, 0)) >= 0) {
+ if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale");
+ while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:MA:", options, 0)) >= 0) {
switch(n) {
case 'h': help();
- case 'V': version();
+ case 'V': version("disorder-playrtp");
case 'd': debugging = 1; break;
- default: fatal(0, "invalid option");
+ case 'D': uaudio_set("device", optarg); break;
+ case 'm': minbuffer = 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':
+ disorder_error(0, "deprecated option; use --api alsa instead");
+ backend = &uaudio_alsa; break;
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ case 'o':
+ disorder_error(0, "deprecated option; use --api oss instead");
+ backend = &uaudio_oss;
+ break;
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ case 'c':
+ disorder_error(0, "deprecated option; use --api coreaudio instead");
+ backend = &uaudio_coreaudio;
+ break;
+#endif
+ case 'A': backend = uaudio_find(optarg); break;
+ 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;
+ case 'M': monitor = 1; break;
+ default: disorder_fatal(0, "invalid option");
}
}
+ if(config_read(0, NULL)) disorder_fatal(0, "cannot read configuration");
+ if(!backend) {
+ backend = uaudio_default(uaudio_apis, UAUDIO_API_CLIENT);
+ if(!backend)
+ disorder_fatal(0, "no default uaudio API found");
+ disorder_info("default audio API %s", backend->name);
+ }
+ if(backend == &uaudio_rtp) {
+ /* This means that you have NO local sound output. This can happen if you
+ * use a non-Apple GCC on a Mac (because it doesn't know how to compile
+ * CoreAudio/AudioHardware.h). */
+ disorder_fatal(0, "cannot play RTP through RTP");
+ }
+ if(!maxbuffer)
+ maxbuffer = 2 * minbuffer;
argc -= optind;
argv += optind;
- if(argc < 1 || argc > 2)
- fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
- sl.n = argc;
- sl.s = argv;
- /* Listen for inbound audio data */
- if(!(res = get_address(&sl, &pref, &sockname)))
- exit(1);
- if((rtpfd = socket(res->ai_family,
- res->ai_socktype,
- res->ai_protocol)) < 0)
- fatal(errno, "error creating socket");
- if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error binding socket to %s", sockname);
- play_rtp();
+ switch(argc) {
+ case 0:
+ /* Get configuration from server */
+ if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
+ if(disorder_connect(c)) exit(EXIT_FAILURE);
+ if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
+ sl.n = 2;
+ sl.s = xcalloc(2, sizeof *sl.s);
+ sl.s[0] = address;
+ sl.s[1] = port;
+ break;
+ case 1:
+ case 2:
+ /* Use command-line ADDRESS+PORT or just PORT */
+ sl.n = argc;
+ sl.s = argv;
+ break;
+ default:
+ disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
+ }
+ disorder_info("version "VERSION" process ID %lu",
+ (unsigned long)getpid());
+ struct sockaddr *addr;
+ socklen_t addr_len;
+ if(!strcmp(sl.s[0], "-")) {
+ /* Pick address family to match known-working connectivity to the server */
+ int family = disorder_client_af(c);
+ /* Get a list of interfaces */
+ struct ifaddrs *ifa, *bestifa = NULL;
+ if(getifaddrs(&ifa) < 0)
+ disorder_fatal(errno, "error calling getifaddrs");
+ /* Try to pick a good one */
+ for(; ifa; ifa = ifa->ifa_next) {
+ if(bestifa == NULL
+ || compare_interfaces(ifa, bestifa, family) > 0)
+ bestifa = ifa;
+ }
+ if(!bestifa)
+ disorder_fatal(0, "failed to select a network interface");
+ family = bestifa->ifa_addr->sa_family;
+ if((rtpfd = socket(family,
+ SOCK_DGRAM,
+ IPPROTO_UDP)) < 0)
+ disorder_fatal(errno, "error creating socket (family %d)", family);
+ /* Bind the address */
+ if(bind(rtpfd, bestifa->ifa_addr,
+ family == AF_INET
+ ? sizeof (struct sockaddr_in) : sizeof (struct sockaddr_in6)) < 0)
+ disorder_fatal(errno, "error binding socket");
+ static struct sockaddr_storage bound_address;
+ addr = (struct sockaddr *)&bound_address;
+ addr_len = sizeof bound_address;
+ if(getsockname(rtpfd, addr, &addr_len) < 0)
+ disorder_fatal(errno, "error getting socket address");
+ /* Convert to string */
+ char addrname[128], portname[32];
+ if(getnameinfo(addr, addr_len,
+ addrname, sizeof addrname,
+ portname, sizeof portname,
+ NI_NUMERICHOST|NI_NUMERICSERV) < 0)
+ disorder_fatal(errno, "getnameinfo");
+ /* Ask for audio data */
+ if(disorder_rtp_request(c, addrname, portname)) exit(EXIT_FAILURE);
+ /* Report what we did */
+ disorder_info("listening on %s", format_sockaddr(addr));
+ } else {
+ /* Look up address and port */
+ if(!(res = get_address(&sl, &prefs, &sockname)))
+ exit(1);
+ addr = res->ai_addr;
+ addr_len = res->ai_addrlen;
+ /* Create the socket */
+ if((rtpfd = socket(res->ai_family,
+ res->ai_socktype,
+ res->ai_protocol)) < 0)
+ disorder_fatal(errno, "error creating socket");
+ /* Allow multiple listeners */
+ xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
+ is_multicast = multicast(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, addr, addr_len);
+ switch(res->ai_addr->sa_family) {
+ case AF_INET:
+ mgroup.in.sin_port = 0;
+ break;
+ case AF_INET6:
+ mgroup.in6.sin6_port = 0;
+ break;
+ default:
+ disorder_fatal(0, "unsupported address family %d",
+ (int)addr->sa_family);
+ }
+ /* Bind to to the multicast group address */
+ if(bind(rtpfd, addr, addr_len) < 0)
+ disorder_fatal(errno, "error binding socket to %s",
+ format_sockaddr(addr));
+ /* Add multicast group membership */
+ switch(mgroup.sa.sa_family) {
+ case PF_INET:
+ mreq.imr_multiaddr = mgroup.in.sin_addr;
+ mreq.imr_interface.s_addr = 0; /* use primary interface */
+ if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
+ &mreq, sizeof mreq) < 0)
+ disorder_fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
+ break;
+ case PF_INET6:
+ mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
+ memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
+ if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
+ &mreq6, sizeof mreq6) < 0)
+ disorder_fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
+ break;
+ default:
+ disorder_fatal(0, "unsupported address family %d", res->ai_family);
+ }
+ /* Report what we did */
+ disorder_info("listening on %s multicast group %s",
+ format_sockaddr(addr), format_sockaddr(&mgroup.sa));
+ } else {
+ /* Bind to 0/port */
+ switch(addr->sa_family) {
+ case AF_INET: {
+ struct sockaddr_in *in = (struct sockaddr_in *)addr;
+
+ memset(&in->sin_addr, 0, sizeof (struct in_addr));
+ break;
+ }
+ case AF_INET6: {
+ struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)addr;
+
+ memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
+ break;
+ }
+ default:
+ disorder_fatal(0, "unsupported family %d", (int)addr->sa_family);
+ }
+ if(bind(rtpfd, addr, addr_len) < 0)
+ disorder_fatal(errno, "error binding socket to %s",
+ format_sockaddr(addr));
+ /* Report what we did */
+ disorder_info("listening on %s", format_sockaddr(addr));
+ }
+ }
+ len = sizeof rcvbuf;
+ if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
+ disorder_fatal(errno, "error calling getsockopt SO_RCVBUF");
+ if(target_rcvbuf > rcvbuf) {
+ if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
+ &target_rcvbuf, sizeof target_rcvbuf) < 0)
+ disorder_error(errno, "error calling setsockopt SO_RCVBUF %d",
+ target_rcvbuf);
+ /* We try to carry on anyway */
+ else
+ disorder_info("changed socket receive buffer from %d to %d",
+ rcvbuf, target_rcvbuf);
+ } else
+ disorder_info("default socket receive buffer %d", rcvbuf);
+ //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
+ if(logfp)
+ disorder_info("WARNING: -L option can impact performance");
+ if(control_socket) {
+ pthread_t tid;
+
+ if((err = pthread_create(&tid, 0, control_thread, 0)))
+ disorder_fatal(err, "pthread_create control_thread");
+ }
+ if(dumpfile) {
+ int fd;
+ unsigned char buffer[65536];
+ size_t written;
+
+ if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
+ disorder_fatal(errno, "opening %s", dumpfile);
+ /* Fill with 0s to a suitable size */
+ memset(buffer, 0, sizeof buffer);
+ for(written = 0; written < dump_size * sizeof(int16_t);
+ written += sizeof buffer) {
+ if(write(fd, buffer, sizeof buffer) < 0)
+ disorder_fatal(errno, "clearing %s", dumpfile);
+ }
+ /* Map the buffer into memory for convenience */
+ dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
+ MAP_SHARED, fd, 0);
+ if(dump_buffer == (void *)-1)
+ disorder_fatal(errno, "mapping %s", dumpfile);
+ disorder_info("dumping to %s", dumpfile);
+ }
+ /* 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*/);
+ uaudio_set("application", "disorder-playrtp");
+ backend->start(playrtp_callback, NULL);
+ /* We receive and convert audio data in a background thread */
+ if((err = pthread_create(<id, 0, listen_thread, 0)))
+ disorder_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)))
+ disorder_fatal(err, "pthread_create queue_thread");
+ pthread_mutex_lock(&lock);
+ time_t lastlog = 0;
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ playrtp_fill_buffer();
+ /* Start playing now */
+ disorder_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))) {
+ if(monitor) {
+ time_t now = xtime(0);
+
+ if(now >= lastlog + 60) {
+ int offset = nsamples - minbuffer;
+ double offtime = (double)offset / (uaudio_rate * uaudio_channels);
+ disorder_info("%+d samples off (%d.%02ds, %d bytes)",
+ offset,
+ (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
+ (int)(fabs(offtime) * 100) % 100,
+ offset * uaudio_bits / CHAR_BIT);
+ lastlog = now;
+ }
+ }
+ //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
+ pthread_cond_wait(&cond, &lock);
+ }
+#if 0
+ if(nsamples) {
+ struct packet *p = pheap_first(&packets);
+ fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
+ nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
+ }
+#endif
+ /* 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;
}
~mdw / disorder / blobdiff