* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA
*/
+/** @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).
+ *
+ * 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.
+ *
+ * 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.
+ *
+ * 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 <netdb.h>
#include <pthread.h>
#include <locale.h>
+#include <sys/uio.h>
+#include <string.h>
+#include <assert.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 "log.h"
#include "mem.h"
#include "syscalls.h"
#include "rtp.h"
#include "defs.h"
-
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-# include <CoreAudio/AudioHardware.h>
-#endif
-#if API_ALSA
-#include <alsa/asoundlib.h>
+#include "vector.h"
+#include "heap.h"
+#include "timeval.h"
+#include "client.h"
+#include "playrtp.h"
+#include "inputline.h"
+
+#define readahead linux_headers_are_borked
+
+/** @brief Obsolete synonym */
+#ifndef IPV6_JOIN_GROUP
+# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
#endif
/** @brief RTP socket */
static int rtpfd;
+/** @brief Log output */
+static FILE *logfp;
+
/** @brief Output device */
-static const char *device;
+const char *device;
-/** @brief Maximum samples per packet we'll support
+/** @brief Minimum low watermark
*
- * NB that two channels = two samples in this program.
- */
-#define MAXSAMPLES 2048
+ * We'll stop playing if there's only this many samples in the buffer. */
+unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
-/** @brief Minimum buffer size
+/** @brief Buffer high watermark
*
- * We'll stop playing if there's only this many samples in the buffer. */
-#define MINBUFFER 8820
+ * We'll only start playing when this many samples are available. */
+static unsigned readahead = 2 * 2 * 44100;
-/** @brief Maximum sample size
+/** @brief Maximum buffer size
*
- * The maximum supported size (in bytes) of one sample. */
-#define MAXSAMPLESIZE 2
+ * We'll stop reading from the network if we have this many samples. */
+static unsigned maxbuffer;
-#define READAHEAD 88200 /* how far to read ahead */
+/** @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;
-#define MAXBUFFER (3 * 88200) /* maximum buffer contents */
+/** @brief Tail of @ref received_packets
+ * Protected by @ref receive_lock
+ */
+struct packet **received_tail = &received_packets;
-/** @brief Received packet
+/** @brief Lock protecting @ref received_packets
*
- * Packets are recorded in an ordered linked list. */
-struct packet {
- /** @brief Pointer to next packet
- * The next packet might not be immediately next: if packets are dropped
- * or mis-ordered there may be gaps at any given moment. */
- struct packet *next;
- /** @brief Number of samples in this packet */
- int nsamples;
- /** @brief Number of samples used from this packet */
- int nused;
- /** @brief Timestamp from RTP packet
- *
- * NB that "timestamps" are really sample counters.*/
- uint32_t timestamp;
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- /** @brief Converted sample data */
- float samples_float[MAXSAMPLES];
-#else
- /** @brief Raw sample data */
- unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
-#endif
-};
+ * Only listen_thread() and queue_thread() ever hold this lock. It is vital
+ * that queue_thread() not hold it any longer than it strictly has to. */
+pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
-/** @brief Total number of samples available */
-static unsigned long nsamples;
+/** @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 Linked list of packets
+/** @brief Length of @ref received_packets */
+uint32_t nreceived;
+
+/** @brief Binary heap of received packets */
+struct pheap packets;
+
+/** @brief Total number of samples available
*
- * In ascending order of timestamp. */
-static struct packet *packets;
+ * We make this volatile because we inspect it without a protecting lock,
+ * so the usual pthread_* guarantees aren't available.
+ */
+volatile uint32_t nsamples;
/** @brief Timestamp of next packet to play.
*
* This is set to the timestamp of the last packet, plus the number of
* samples it contained. Only valid if @ref active is nonzero.
*/
-static uint32_t next_timestamp;
+uint32_t next_timestamp;
/** @brief True if actively playing
*
* This is true when playing and false when just buffering. */
-static int active;
+int active;
/** @brief Lock protecting @ref packets */
-static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
+pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
/** @brief Condition variable signalled whenever @ref packets is changed */
-static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
+
+#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;
+
+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' },
+ { "buffer", required_argument, 0, 'b' },
+ { "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
+ { "dump", required_argument, 0, 'r' },
+ { "socket", required_argument, 0, 's' },
+ { "config", required_argument, 0, 'C' },
{ 0, 0, 0, 0 }
};
-/** @brief Return true iff a < b in sequence-space arithmetic */
-static inline int lt(uint32_t a, uint32_t b) {
- return (uint32_t)(a - b) & 0x80000000;
+/** @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)
+ fatal(errno, "error binding to %s", control_socket);
+ if(listen(sfd, 128) < 0)
+ fatal(errno, "error calling listen on %s", control_socket);
+ info("listening on %s", control_socket);
+ for(;;) {
+ salen = sizeof sa;
+ cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
+ if(cfd < 0) {
+ switch(errno) {
+ case EINTR:
+ case EAGAIN:
+ break;
+ default:
+ fatal(errno, "error calling accept on %s", control_socket);
+ }
+ }
+ if(!(fp = fdopen(cfd, "r+"))) {
+ error(errno, "error calling fdopen for %s connection", control_socket);
+ close(cfd);
+ continue;
+ }
+ if(!inputline(control_socket, fp, &line, '\n')) {
+ if(!strcmp(line, "stop")) {
+ info("stopped via %s", control_socket);
+ exit(0); /* terminate immediately */
+ }
+ if(!strcmp(line, "query"))
+ fprintf(fp, "running");
+ xfree(line);
+ }
+ if(fclose(fp) < 0)
+ 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);
+ 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 playrtp_new_packet().
+ */
static void *listen_thread(void attribute((unused)) *arg) {
- struct packet *p = 0, **pp;
+ struct packet *p = 0;
int n;
- 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));
+ struct rtp_header header;
+ uint16_t seq;
+ uint32_t timestamp;
+ struct iovec iov[2];
for(;;) {
if(!p)
- p = xmalloc(sizeof *p);
- n = read(rtpfd, packet.bytes, sizeof packet.bytes);
+ 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:
}
}
/* Ignore too-short packets */
- if((size_t)n <= sizeof (struct rtp_header))
+ if((size_t)n <= sizeof (struct rtp_header)) {
+ info("ignored a short packet");
continue;
- p->nused = 0;
- p->timestamp = ntohl(packet.header.timestamp);
+ }
+ timestamp = htonl(header.timestamp);
+ seq = htons(header.seq);
/* Ignore packets in the past */
- if(active && lt(p->timestamp, next_timestamp))
+ if(active && lt(timestamp, next_timestamp)) {
+ info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
+ timestamp, next_timestamp);
continue;
+ }
+ p->next = 0;
+ p->flags = 0;
+ p->timestamp = timestamp;
/* Convert to target format */
- switch(packet.header.mpt & 0x7F) {
+ if(header.mpt & 0x80)
+ p->flags |= IDLE;
+ switch(header.mpt & 0x7F) {
case 10:
- p->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t);
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- /* Convert to what Core Audio expects */
- for(n = 0; n < p->nsamples; ++n)
- p->samples_float[n] = (int16_t)ntohs(samples[n]) * (0.5f / 32767);
-#else
- /* ALSA can do any necessary conversion itself (though it might be better
- * to do any necessary conversion in the background) */
- memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
-#endif
+ p->nsamples = (n - sizeof header) / sizeof(uint16_t);
break;
/* TODO support other RFC3551 media types (when the speaker does) */
default:
fatal(0, "unsupported RTP payload type %d",
- packet.header.mpt & 0x7F);
+ header.mpt & 0x7F);
}
- pthread_mutex_lock(&lock);
+ 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... */
- while(nsamples >= MAXBUFFER)
- pthread_cond_wait(&cond, &lock);
- for(pp = &packets;
- *pp && lt((*pp)->timestamp, p->timestamp);
- pp = &(*pp)->next)
- ;
- /* So now either !*pp or *pp >= p */
- if(*pp && p->timestamp == (*pp)->timestamp) {
- /* *pp == p; a duplicate. Ideally we avoid the translation step here,
- * but we'll worry about that another time. */
- } else {
- p->next = *pp;
- *pp = p;
- nsamples += p->nsamples;
- pthread_cond_broadcast(&cond);
- p = 0; /* we've consumed this packet */
+ if(nsamples >= maxbuffer) {
+ pthread_mutex_lock(&lock);
+ while(nsamples >= maxbuffer)
+ pthread_cond_wait(&cond, &lock);
+ pthread_mutex_unlock(&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;
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
-/** @brief Callback from Core Audio */
-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(packets && nbuffers > 0) {
- if(packets->used == packets->nsamples) {
- /* TODO if we dropped a packet then we should introduce a gap here */
- struct packet *const p = packets;
- packets = p->next;
- free(p);
- pthread_cond_broadcast(&cond);
- continue;
- }
- if(samplesOutLeft == 0) {
- --nbuffers;
- ++ab;
- samplesOut = ab->data;
- samplesOutLeft = ab->mDataByteSize / sizeof (float);
- continue;
- }
- /* Now: (1) there is some data left to read
- * (2) there is some space to put it */
- samplesInLeft = packets->nsamples - packets->used;
- samplesToCopy = (samplesInLeft < samplesOutLeft
- ? samplesInLeft : samplesOutLeft);
- memcpy(samplesOut, packet->samples + packets->used, samplesToCopy);
- packets->used += samplesToCopy;
- samplesOut += samplesToCopy;
- samesOutLeft -= samplesToCopy;
+/** @brief Wait until the buffer is adequately full
+ *
+ * Must be called with @ref lock held.
+ */
+void playrtp_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.
+ */
+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;
}
- pthread_mutex_unlock(&lock);
return 0;
}
-#endif
/** @brief Play an RTP stream
*
*/
static void play_rtp(void) {
pthread_t ltid;
+ int err;
/* We receive and convert audio data in a background thread */
- pthread_create(<id, 0, listen_thread, 0);
-#if API_ALSA
- {
- snd_pcm_t *pcm;
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- /* Only support one format for now */
- const int sample_format = SND_PCM_FORMAT_S16_BE;
- unsigned rate = 44100;
- const int channels = 2;
- const int samplesize = channels * sizeof(uint16_t);
- snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
- /* If we can write more than this many samples we'll get a wakeup */
- const int avail_min = 256;
- snd_pcm_sframes_t frames_written;
- size_t samples_written;
- int prepared = 1;
- int err;
- int infilling = 0;
-
- /* Open ALSA */
- if((err = snd_pcm_open(&pcm,
- device ? device : "default",
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK)))
- fatal(0, "error from snd_pcm_open: %d", err);
- /* Set up 'hardware' parameters */
- snd_pcm_hw_params_alloca(&hwparams);
- if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
- fatal(0, "error from snd_pcm_hw_params_any: %d", err);
- if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
- SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
- if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
- sample_format)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
- sample_format, err);
- if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
- rate, err);
- if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
- channels)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
- channels, err);
- if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
- &pcm_bufsize)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
- MAXSAMPLES * samplesize * 3, err);
- if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
- fatal(0, "error calling snd_pcm_hw_params: %d", err);
- /* Set up 'software' parameters */
- snd_pcm_sw_params_alloca(&swparams);
- if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
- if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
- avail_min, err);
- if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params: %d", err);
-
- /* Ready to go */
-
- pthread_mutex_lock(&lock);
- for(;;) {
- /* Wait for the buffer to fill up a bit */
- info("Buffering...");
- while(nsamples < READAHEAD)
- pthread_cond_wait(&cond, &lock);
- if(!prepared) {
- if((err = snd_pcm_prepare(pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- prepared = 1;
- }
- /* Start at the first available packet */
- next_timestamp = packets->timestamp;
- active = 1;
- infilling = 0;
- info("Playing...");
- /* Wait until the buffer empties out */
- while(nsamples >= MINBUFFER) {
- /* Wait for ALSA to ask us for more data */
- pthread_mutex_unlock(&lock);
- snd_pcm_wait(pcm, -1);
- pthread_mutex_lock(&lock);
- /* ALSA is ready for more data */
- if(packets && packets->timestamp + packets->nused == next_timestamp) {
- /* Hooray, we have a packet we can play */
- const size_t samples_available = packets->nsamples - packets->nused;
- const size_t frames_available = samples_available / 2;
-
- frames_written = snd_pcm_writei(pcm,
- packets->samples_raw + packets->nused,
- frames_available);
- if(frames_written < 0)
- fatal(0, "error calling snd_pcm_writei: %ld",
- (long)frames_written);
- samples_written = frames_written * 2;
- packets->nused += samples_written;
- next_timestamp += samples_written;
- if(packets->nused == packets->nsamples) {
- /* We're done with this packet */
- struct packet *p = packets;
-
- packets = p->next;
- nsamples -= p->nsamples;
- free(p);
- pthread_cond_broadcast(&cond);
- }
- infilling = 0;
- } else {
- /* We don't have anything to play! We'd better play some 0s. */
- static const uint16_t zeros[1024];
- size_t samples_available = 1024, frames_available;
-
- if(!infilling) {
- info("Infilling...");
- infilling = 1;
- }
- if(packets && next_timestamp + samples_available > packets->timestamp)
- samples_available = packets->timestamp - next_timestamp;
- frames_available = samples_available / 2;
- frames_written = snd_pcm_writei(pcm,
- zeros,
- frames_available);
- if(frames_written < 0)
- fatal(0, "error calling snd_pcm_writei: %ld",
- (long)frames_written);
- next_timestamp += samples_written;
- }
- }
- active = 0;
- /* We stop playing for a bit until the buffer re-fills */
- pthread_mutex_unlock(&lock);
- if((err = snd_pcm_nonblock(pcm, 0)))
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- if((err = snd_pcm_drain(pcm)))
- fatal(0, "error calling snd_pcm_drain: %d", err);
- if((err = snd_pcm_nonblock(pcm, 1)))
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- prepared = 0;
- pthread_mutex_lock(&lock);
- }
-
- }
-#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 */
- }
- }
-#else
-# error No known audio API
-#endif
+ 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 */
xprintf("Usage:\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"
+#if HAVE_ALSA_ASOUNDLIB_H
+ " --alsa, -a Use ALSA to play audio\n"
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ " --oss, -o Use OSS to play audio\n"
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ " --core-audio, -c Use Core Audio to play audio\n"
+#endif
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
- " --debug, -d Turn on debugging\n"
- " --device, -D DEVICE Output device\n");
+ );
xfclose(stdout);
exit(0);
}
/* display version number and terminate */
static void version(void) {
- xprintf("disorder-playrtp version %s\n", disorder_version_string);
+ xprintf("%s", disorder_version_string);
xfclose(stdout);
exit(0);
}
int main(int argc, char **argv) {
- int n;
+ int n, err;
struct addrinfo *res;
struct stringlist sl;
char *sockname;
+ int rcvbuf, target_rcvbuf = 131072;
+ socklen_t len;
+ struct ip_mreq mreq;
+ struct ipv6_mreq mreq6;
+ disorder_client *c;
+ 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;
static const struct addrinfo prefs = {
AI_PASSIVE,
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVdD", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
case 'd': debugging = 1; break;
case 'D': 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;
+#endif
+#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
+ case 'o': backend = playrtp_oss; break;
+#endif
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ case 'c': backend = playrtp_coreaudio; break;
+#endif
+ case 'C': configfile = optarg; break;
+ case 's': control_socket = optarg; break;
+ case 'r': dumpfile = optarg; break;
default: fatal(0, "invalid option");
}
}
+ if(config_read(0)) fatal(0, "cannot read configuration");
+ if(!maxbuffer)
+ maxbuffer = 4 * readahead;
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 */
+ 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:
+ fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
+ }
+ /* Look up address and port */
if(!(res = get_address(&sl, &prefs, &sockname)))
exit(1);
+ /* Create the socket */
if((rtpfd = socket(res->ai_family,
res->ai_socktype,
res->ai_protocol)) < 0)
fatal(errno, "error creating socket");
+ /* Stash the multicast group address */
+ if((is_multicast = multicast(res->ai_addr))) {
+ memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
+ switch(res->ai_addr->sa_family) {
+ case AF_INET:
+ mgroup.in.sin_port = 0;
+ break;
+ case AF_INET6:
+ mgroup.in6.sin6_port = 0;
+ break;
+ }
+ }
+ /* 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) {
+ 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)
+ 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)
+ fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
+ break;
+ default:
+ fatal(0, "unsupported address family %d", res->ai_family);
+ }
+ info("listening on %s multicast group %s",
+ format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
+ } else
+ 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");
+ 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");
+ if(control_socket) {
+ pthread_t tid;
+
+ if((err = pthread_create(&tid, 0, control_thread, 0)))
+ 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)
+ 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)
+ 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)
+ fatal(errno, "mapping %s", dumpfile);
+ info("dumping to %s", dumpfile);
+ }
play_rtp();
return 0;
}
~mdw / disorder / blobdiff