#include <sys/socket.h>
#include <netdb.h>
#include <pthread.h>
+#include <locale.h>
#include "log.h"
#include "mem.h"
#include "addr.h"
#include "syscalls.h"
#include "rtp.h"
-#include "debug.h"
+#include "defs.h"
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
# include <CoreAudio/AudioHardware.h>
#endif
+#if API_ALSA
+#include <alsa/asoundlib.h>
+#endif
+#define readahead linux_headers_are_borked
+
+/** @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 */
+/** @brief Output device */
+static const char *device;
+
+/** @brief Maximum samples per packet we'll support
+ *
+ * NB that two channels = two samples in this program.
+ */
+#define MAXSAMPLES 2048
+
+/** @brief Minimum buffer size
+ *
+ * We'll stop playing if there's only this many samples in the buffer. */
+static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
+
+/** @brief Maximum sample size
+ *
+ * The maximum supported size (in bytes) of one sample. */
+#define MAXSAMPLESIZE 2
+
+/** @brief Buffer size
+ *
+ * We'll only start playing when this many samples are available. */
+static unsigned readahead = 4 * 2 * 44100; /* 4 seconds */
+
#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 */
+/** @brief Received packet
+ *
+ * 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
- float samples[MAXSAMPLES]; /* converted sample data */
+ /** @brief Converted sample data */
+ float samples_float[MAXSAMPLES];
+#else
+ /** @brief Raw sample data */
+ unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
#endif
};
-static unsigned long nsamples; /* total samples available */
+/** @brief Total number of samples available */
+static unsigned long nsamples;
+
+/** @brief Linked list of packets
+ *
+ * In ascending order of timestamp. */
+static struct packet *packets;
+
+/** @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;
-static struct frame *frames; /* received frames in ascending order
- * of timestamp */
+/** @brief True if actively playing
+ *
+ * This is true when playing and false when just buffering. */
+static int active;
+
+/** @brief Lock protecting @ref packets */
static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
-/* lock protecting frame list */
-static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER;
-/* signalled whenever we add a new frame */
+/** @brief Condition variable signalled whenever @ref packets is changed */
+static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
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' },
+ { "buffer", required_argument, 0, 'b' },
{ 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;
+/** @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;
}
-/* 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;
+/** @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;
+ int n;
union {
struct rtp_header header;
uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)];
+ sizeof (struct rtp_header));
for(;;) {
- if(!f)
- f = xmalloc(sizeof *f);
+ if(!p)
+ p = xmalloc(sizeof *p);
n = read(rtpfd, packet.bytes, sizeof packet.bytes);
if(n < 0) {
switch(errno) {
fatal(errno, "error reading from socket");
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ /* Ignore too-short packets */
+ if((size_t)n <= sizeof (struct rtp_header))
+ continue;
+ p->nused = 0;
+ p->timestamp = ntohl(packet.header.timestamp);
+ /* Ignore packets in the past */
+ if(active && lt(p->timestamp, next_timestamp))
+ continue;
/* Convert to target format */
- switch(packet.header.mtp & 0x7F) {
+ switch(packet.header.mpt & 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);
+ 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
break;
/* TODO support other RFC3551 media types (when the speaker does) */
default:
- fatal(0, "unsupported RTP payload type %d",
+ fatal(0, "unsupported RTP payload type %d",
packet.header.mpt & 0x7F);
}
-#endif
- f->used = 0;
- f->timestamp = ntohl(packet.header.timestamp);
pthread_mutex_lock(&lock);
- /* Stop reading if we've reached the maximum */
+ /* 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(ff = &frames; *ff && !gt(*ff, f); ff = &(*ff)->next)
+ for(pp = &packets;
+ *pp && lt((*pp)->timestamp, p->timestamp);
+ pp = &(*pp)->next)
;
- f->next = *ff;
- *ff = f;
- nsamples += f->nsamples;
- pthread_cond_broadcast(&cond);
+ /* 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 */
+ }
pthread_mutex_unlock(&lock);
- f = 0;
}
}
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+/** @brief Callback from Core Audio */
static OSStatus adioproc(AudioDeviceID inDevice,
const AudioTimeStamp *inNow,
const AudioBufferList *inInputData,
size_t samplesInLeft;
size_t samplesToCopy;
- pthread_mutex_lock(&lock);
+ pthread_mutex_lock(&lock);
samplesOut = ab->data;
samplesOutLeft = ab->mDataByteSize / sizeof (float);
- while(frames && nbuffers > 0) {
- if(frames->used == frames->nsamples) {
+ while(packets && nbuffers > 0) {
+ if(packets->used == packets->nsamples) {
/* TODO if we dropped a packet then we should introduce a gap here */
- struct frame *const f = frames;
- frames = f->next;
- free(f);
+ struct packet *const p = packets;
+ packets = p->next;
+ free(p);
pthread_cond_broadcast(&cond);
continue;
}
}
/* Now: (1) there is some data left to read
* (2) there is some space to put it */
- samplesInLeft = frames->nsamples - frames->used;
+ samplesInLeft = packets->nsamples - packets->used;
samplesToCopy = (samplesInLeft < samplesOutLeft
? samplesInLeft : samplesOutLeft);
- memcpy(samplesOut, frame->samples + frames->used, samplesToCopy);
- frames->used += samplesToCopy;
+ memcpy(samplesOut, packet->samples + packets->used, samplesToCopy);
+ packets->used += samplesToCopy;
samplesOut += samplesToCopy;
samesOutLeft -= samplesToCopy;
}
}
#endif
-void play_rtp(void) {
- pthread_t lt;
+/** @brief Play an RTP stream
+ *
+ * This is the guts of the program. It is responsible for:
+ * - starting the listening thread
+ * - opening the audio device
+ * - reading ahead to build up a buffer
+ * - arranging for audio to be played
+ * - detecting when the buffer has got too small and re-buffering
+ */
+static void play_rtp(void) {
+ pthread_t ltid;
/* We receive and convert audio data in a background thread */
- pthread_create(<, 0, listen_thread, 0);
+ pthread_create(<id, 0, listen_thread, 0);
#if API_ALSA
- assert(!"implemented");
+ {
+ 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) {
+ if(frames_written != -EAGAIN)
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ } else {
+ 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) {
+ if(frames_written != -EAGAIN)
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ } else {
+ samples_written = frames_written * 2;
+ 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;
pthread_mutex_lock(&lock);
for(;;) {
/* Wait for the buffer to fill up a bit */
- while(nsamples < READAHEAD)
+ 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)
+ while(nsamples >= minbuffer)
pthread_cond_wait(&cond, &lock);
/* Stop playing for a bit until the buffer re-fills */
status = AudioDeviceStop(adid, adioproc);
"Options:\n"
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
- " --debug, -d Turn on debugging\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");
xfclose(stdout);
exit(0);
}
int n;
struct addrinfo *res;
struct stringlist sl;
- const char *sockname;
+ char *sockname;
- static const struct addrinfo prefbind = {
+ static const struct addrinfo prefs = {
AI_PASSIVE,
PF_INET,
SOCK_DGRAM,
mem_init();
if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVd", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVdD:m:b:", 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;
default: fatal(0, "invalid option");
}
}
sl.n = argc;
sl.s = argv;
/* Listen for inbound audio data */
- if(!(res = get_address(&sl, &pref, &sockname)))
+ if(!(res = get_address(&sl, &prefs, &sockname)))
exit(1);
if((rtpfd = socket(res->ai_family,
res->ai_socktype,
~mdw / disorder / blobdiff