*/
#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 Linked list of packets
*
- * In ascending order of timestamp. */
+ * In ascending order of timestamp. Really this should be a heap for more
+ * efficient access. */
static struct packet *packets;
/** @brief Timestamp of next packet to play.
{ "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 }
};
return !lt(b, a);
}
+/** @brief Drop the packet at the head of the queue */
+static void drop_first_packet(void) {
+ struct packet *const p = packets;
+ packets = p->next;
+ nsamples -= p->nsamples;
+ free(p);
+ pthread_cond_broadcast(&cond);
+}
+
/** @brief Background thread collecting samples
*
* This function collects samples, perhaps converts them to the target format,
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);
+ {
+ size_t i;
+
+ for(i = 0; i < p->nsamples; ++i)
+ p->samples_float[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767);
+ }
#else
/* ALSA can do any necessary conversion itself (though it might be better
* to do any necessary conversion in the background) */
*
* This is rather unsatisfactory: it means that if packets get heavily
* out of order then we guarantee dropouts. But for now... */
- while(nsamples >= MAXBUFFER)
+ while(nsamples >= maxbuffer)
pthread_cond_wait(&cond, &lock);
for(pp = &packets;
*pp && lt((*pp)->timestamp, p->timestamp);
#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;
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) {
+ if(packets) {
+ /* There's a packet */
+ const uint32_t packet_start = packets->timestamp;
+ const uint32_t packet_end = packets->timestamp + packets->nsamples;
+
+ if(le(packet_end, next_timestamp)) {
+ /* This packet is in the past */
+ info("dropping buffered past packet %"PRIx32" < %"PRIx32,
+ packet_start, next_timestamp);
+ drop_first_packet();
+ continue;
+ }
+ if(ge(next_timestamp, packet_start)
+ && lt(next_timestamp, packet_end)) {
+ /* This packet is suitable */
+ const uint32_t offset = next_timestamp - packet_start;
+ uint32_t samples_available = packet_end - next_timestamp;
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ memcpy(samplesOut,
+ packets->samples_float + offset,
+ samples_available * sizeof(float));
+ samplesOut += samples_available;
+ next_timestamp += samples_available;
+ samplesOutLeft -= samples_available;
+ if(ge(next_timestamp, packet_end))
+ drop_first_packet();
+ continue;
+ }
+ }
+ /* We didn't find a suitable packet (though there might still be
+ * unsuitable ones). We infill with 0s. */
+ if(packets) {
+ /* There is a next packet, only infill up to that point */
+ uint32_t samples_available = packets->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;
+ /* TODO log infill */
+ } else {
+ /* There's no next packet at all */
+ info("infilled by %zu", samplesOutLeft);
+ next_timestamp += samplesOutLeft;
+ samplesOut += samplesOutLeft;
+ samplesOutLeft = 0;
+ /* TODO log infill */
+ }
}
- /* 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;
}
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->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:", 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;
default: fatal(0, "invalid option");
}
}
+ if(!maxbuffer)
+ maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
if(argc < 1 || argc > 2)