#include <netdb.h>
#include <pthread.h>
#include <locale.h>
+#include <sys/uio.h>
#include "log.h"
#include "mem.h"
/** @brief RTP socket */
static int rtpfd;
+/** @brief Log output */
+static FILE *logfp;
+
/** @brief Output device */
static const char *device;
*/
#define MAXSAMPLES 2048
-/** @brief Minimum buffer size
+/** @brief Minimum low watermark
*
* We'll stop playing if there's only this many samples in the buffer. */
static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
* The maximum supported size (in bytes) of one sample. */
#define MAXSAMPLESIZE 2
-/** @brief Buffer size
+/** @brief Buffer high watermark
*
* We'll only start playing when this many samples are available. */
static unsigned readahead = 2 * 2 * 44100;
+/** @brief Maximum buffer size
+ *
+ * We'll stop reading from the network if we have this many samples. */
+static unsigned maxbuffer;
+
/** @brief Number of samples to infill by in one go */
#define INFILL_SAMPLES (44100 * 2) /* 1s */
-#define MAXBUFFER (3 * 88200) /* maximum buffer contents */
-
-/** @brief Received packet
- *
- * Packets are recorded in an ordered linked list. */
+/** @brief Received packet */
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 */
uint32_t nsamples;
/** @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
};
/** @brief Total number of samples available */
static unsigned long nsamples;
-/** @brief Linked list of packets
+/** @brief Mapping of sequence numbers to packets
*
- * In ascending order of timestamp. */
-static struct packet *packets;
+ * This isn't very efficient - 256KB on 32-bit machines, 512KB if you do a
+ * 64-bit build for some reason. It can be optimized later if need be. */
+static struct packet *packets[65536];
+
+/** @brief Total number of packets */
+static unsigned npackets;
/** @brief Timestamp of next packet to play.
*
* This is true when playing and false when just buffering. */
static int active;
+/** @brief Sequence number of next packet we expxect to play */
+static uint16_t sequence;
+
+/** @brief Structure of free packet list */
+union free_packet {
+ struct packet p;
+ union free_packet *next;
+};
+
+/** @brief Linked list of free packets */
+static union free_packet *free_packets;
+
+/** @brief Array of new free packets */
+static union free_packet *next_free_packet;
+
+/** @brief Count of new free packets */
+static size_t count_free_packets;
+
/** @brief Lock protecting @ref packets */
static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
{ "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' },
{ 0, 0, 0, 0 }
};
+/** @brief Return a new packet
+ *
+ * Assumes that @ref lock is held. */
+static struct packet *new_packet(void) {
+ struct packet *p;
+
+ if(free_packets) {
+ p = &free_packets->p;
+ free_packets = free_packets->next;
+ } else {
+ if(!count_free_packets) {
+ next_free_packet = xcalloc(1024, sizeof (union free_packet));
+ count_free_packets = 1024;
+ }
+ p = &(next_free_packet++)->p;
+ --count_free_packets;
+ }
+ return p;
+}
+
+/** @brief Free a packet
+ *
+ * Assumes that @ref lock is held. */
+static void free_packet(struct packet *p) {
+ union free_packet *u = (union free_packet *)p;
+ u->next = free_packets;
+ free_packets = u;
+}
+
/** @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;
return !lt(b, a);
}
+/** @brief Drop the packet at the head of the queue */
+static void drop_packet(unsigned sequence) {
+ if(packets[sequence]) {
+ nsamples -= packets[sequence]->nsamples;
+ free_packet(packets[sequence]);
+ packets[sequence] = 0;
+ pthread_cond_broadcast(&cond);
+ --npackets;
+ }
+}
+
/** @brief Background thread collecting samples
*
* This function collects samples, perhaps converts them to the target format,
* and adds them to the packet list. */
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);
+ if(!p) {
+ pthread_mutex_lock(&lock);
+ p = new_packet();
+ pthread_mutex_unlock(&lock);
+ }
+ iov[0].iov_base = &header;
+ iov[0].iov_len = sizeof header;
+ iov[1].iov_base = p->samples_raw;
+ iov[1].iov_len = 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->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,
- p->timestamp, next_timestamp);
+ timestamp, next_timestamp);
continue;
}
+ pthread_mutex_lock(&lock);
+ p = new_packet();
+ p->timestamp = timestamp;
/* Convert to target format */
- switch(packet.header.mpt & 0x7F) {
+ 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
+ p->nsamples = (n - sizeof header) / sizeof(uint16_t);
/* ALSA can do any necessary conversion itself (though it might be better
* to do any necessary conversion in the background) */
- memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
-#endif
+ /* TODO we could readv into the buffer */
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. */
- info("dropped a duplicated");
- } else {
- if(*pp)
- info("receiving packets out of order");
- p->next = *pp;
- *pp = p;
- nsamples += p->nsamples;
- pthread_cond_broadcast(&cond);
- p = 0; /* we've consumed this packet */
+ if(nsamples >= maxbuffer) {
+ info("buffer full");
+ while(nsamples >= maxbuffer)
+ pthread_cond_wait(&cond, &lock);
}
+ /* If there's a packet there already we overwrite it; perhaps it is left
+ * over from an earlier stage. */
+ drop_packet(seq);
+ /* Record this packet */
+ packets[seq] = p;
+ /* If we currently have no idea where to start playing, this is it */
+ if(!npackets)
+ sequence = seq;
+ ++npackets;
+ nsamples += p->nsamples;
+ pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
}
}
+/** @brief Return true if @p p contains @p timestamp */
+static inline int contains(const struct packet *p, uint32_t timestamp) {
+ const uint32_t packet_start = p->timestamp;
+ const uint32_t packet_end = p->timestamp + p->nsamples;
+
+ return (ge(timestamp, packet_start)
+ && lt(timestamp, packet_end));
+}
+
#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) {
+static OSStatus adioproc
+ (AudioDeviceID attribute((unused)) inDevice,
+ const AudioTimeStamp attribute((unused)) *inNow,
+ const AudioBufferList attribute((unused)) *inInputData,
+ const AudioTimeStamp attribute((unused)) *inInputTime,
+ AudioBufferList *outOutputData,
+ const AudioTimeStamp attribute((unused)) *inOutputTime,
+ void attribute((unused)) *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;
+ const struct packet *p;
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;
+ while(nbuffers > 0) {
+ float *samplesOut = ab->mData;
+ size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
+
+ while(samplesOutLeft > 0) {
+ /* Look for a suitable packet, dropping any unsuitable ones along the
+ * way. Unsuitable packets are ones that are in the past. */
+ while(npackets
+ && (!packets[sequence]
+ || le(packets[sequence]->timestamp
+ + packets[sequence]->nsamples,
+ next_timestamp)))
+ drop_packet(sequence++);
+ p = packets[sequence];
+ if(p) {
+ if(contains(p, next_timestamp)) {
+ /* This packet is suitable */
+ const uint32_t packet_end = p->timestamp + p->nsamples;
+ const uint32_t offset = next_timestamp - p->timestamp;
+ const uint16_t *ptr =
+ (void *)(p->samples_raw + offset * sizeof (uint16_t));
+ uint32_t samples_available = packet_end - next_timestamp;
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ next_timestamp += samples_available;
+ samplesOutLeft -= samples_available;
+ while(samples_available-- > 0)
+ *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
+ /* We don't bother junking the packet or advancing sequence - that'll
+ * be dealt with next time round */
+ continue;
+ }
+ }
+ /* We didn't find a suitable packet (though there might still be
+ * unsuitable ones). We infill with 0s. */
+ if(p) {
+ /* There is a next packet, only infill up to that point */
+ uint32_t samples_available = p->timestamp - next_timestamp;
+
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ info("infill by %"PRIu32, samples_available);
+ /* Convniently the buffer is 0 to start with */
+ next_timestamp += samples_available;
+ samplesOut += samples_available;
+ samplesOutLeft -= samples_available;
+ } else {
+ /* There's no next packet at all */
+ info("infilled by %zu", samplesOutLeft);
+ next_timestamp += samplesOutLeft;
+ samplesOut += samplesOutLeft;
+ samplesOutLeft = 0;
+ }
}
- /* 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;
+ ++ab;
+ --nbuffers;
}
pthread_mutex_unlock(&lock);
return 0;
fatal(0, "error calling snd_pcm_prepare: %d", err);
prepared = 1;
}
+ assert(sequence != -1);
/* Start at the first available packet */
- next_timestamp = packets->timestamp;
+ next_timestamp = packets[sequence]->timestamp;
active = 1;
infilling = 0;
escape = 0;
}
if(packets
&& ge(next_timestamp, packets->timestamp + packets->nsamples)) {
- struct packet *p = packets;
-
info("dropping buffered past packet %"PRIx32" < %"PRIx32,
packets->timestamp, next_timestamp);
-
- packets = p->next;
- if(packets)
- assert(lt(p->timestamp, packets->timestamp));
- nsamples -= p->nsamples;
- free(p);
- pthread_cond_broadcast(&cond);
+ drop_first_packet();
continue;
}
/* Wait for ALSA to ask us for more data */
} else {
samples_written = frames_written * 2;
next_timestamp += samples_written;
- if(ge(next_timestamp, packet_end)) {
- /* We're done with this packet */
- struct packet *p = packets;
-
- packets = p->next;
- if(packets)
- assert(lt(p->timestamp, packets->timestamp));
- nsamples -= p->nsamples;
- free(p);
- pthread_cond_broadcast(&cond);
- }
+ if(ge(next_timestamp, packet_end))
+ drop_first_packet();
infilling = 0;
}
} else {
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));
+ D(("mFormatID %08lx", asbd.mFormatID));
+ D(("mFormatFlags %08lx", asbd.mFormatFlags));
+ D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
+ D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
+ D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
+ D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
+ D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
+ D(("mReserved %08lx", asbd.mReserved));
if(asbd.mFormatID != kAudioFormatLinearPCM)
fatal(0, "audio device does not support kAudioFormatLinearPCM");
status = AudioDeviceAddIOProc(adid, adioproc, 0);
pthread_mutex_lock(&lock);
for(;;) {
/* Wait for the buffer to fill up a bit */
+ info("Buffering...");
while(nsamples < readahead)
pthread_cond_wait(&cond, &lock);
/* Start playing now */
+ info("Playing...");
+ next_timestamp = packets[sequence]->timestamp;
+ active = 1;
status = AudioDeviceStart(adid, adioproc);
if(status)
fatal(0, "AudioDeviceStart: %d", (int)status);
status = AudioDeviceStop(adid, adioproc);
if(status)
fatal(0, "AudioDeviceStop: %d", (int)status);
+ active = 0;
/* Go back round */
}
}
xprintf("Usage:\n"
" disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
"Options:\n"
- " --help, -h Display usage message\n"
- " --version, -V Display version number\n"
- " --debug, -d Turn on debugging\n"
" --device, -D DEVICE Output device\n"
" --min, -m FRAMES Buffer low water mark\n"
- " --buffer, -b FRAMES Buffer high water mark\n");
+ " --buffer, -b FRAMES Buffer high water mark\n"
+ " --max, -x FRAMES Buffer maximum size\n"
+ " --help, -h Display usage message\n"
+ " --version, -V Display version number\n"
+ );
xfclose(stdout);
exit(0);
}
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVdD:m:b:", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
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;
default: fatal(0, "invalid option");
}
}
+ if(!maxbuffer)
+ maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
if(argc < 1 || argc > 2)
~mdw / disorder / blobdiff