* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA
*/
+/** @file clients/playrtp.c
+ * @brief RTP player
+ *
+ * This RTP player supports Linux (ALSA) and Darwin (Core Audio) systems.
+ */
#include <config.h>
#include "types.h"
#include <sys/socket.h>
#include <netdb.h>
#include <pthread.h>
+#include <locale.h>
+#include <sys/uio.h>
+#include <string.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"
#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 */
-#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 */
+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 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 */
+
+/** @brief Maximum sample size
+ *
+ * The maximum supported size (in bytes) of one sample. */
+#define MAXSAMPLESIZE 2
+
+/** @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
+ *
+ * This is an upper bound - in practice we expxect the underlying audio API to
+ * only ask for a much smaller number of samples in any one go.
+ */
+#define INFILL_SAMPLES (44100 * 2) /* 1s */
+
+/** @brief Received packet
+ *
+ * Received packets are kept in a binary heap (see @ref pheap) ordered by
+ * timestamp.
+ */
+struct packet {
+ /** @brief Number of samples in this packet */
+ uint32_t nsamples;
+ /** @brief Timestamp from RTP packet
+ *
+ * NB that "timestamps" are really sample counters. Use lt() or lt_packet()
+ * to compare timestamps.
+ */
+ uint32_t timestamp;
+ /** @brief Raw sample data
+ *
+ * Only the first @p nsamples samples are defined; the rest is uninitialized
+ * data.
+ */
+ unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
};
-static unsigned long nsamples; /* total samples available */
+/** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic
+ *
+ * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$.
+ *
+ * See also lt_packet().
+ */
+static inline int lt(uint32_t a, uint32_t b) {
+ return (uint32_t)(a - b) & 0x80000000;
+}
+
+/** @brief Return true iff a >= b in sequence-space arithmetic */
+static inline int ge(uint32_t a, uint32_t b) {
+ return !lt(a, b);
+}
+
+/** @brief Return true iff a > b in sequence-space arithmetic */
+static inline int gt(uint32_t a, uint32_t b) {
+ return lt(b, a);
+}
+
+/** @brief Return true iff a <= b in sequence-space arithmetic */
+static inline int le(uint32_t a, uint32_t b) {
+ return !lt(b, a);
+}
+
+/** @brief Ordering for packets, used by @ref pheap */
+static inline int lt_packet(const struct packet *a, const struct packet *b) {
+ return lt(a->timestamp, b->timestamp);
+}
+
+/** @struct pheap
+ * @brief Binary heap of packets ordered by timestamp */
+HEAP_TYPE(pheap, struct packet *, lt_packet);
+
+/** @brief Binary heap of received packets */
+static struct pheap packets;
+
+/** @brief Total number of samples available */
+static unsigned long 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;
+
+/** @brief True if actively playing
+ *
+ * This is true when playing and false when just buffering. */
+static int active;
+
+/** @brief Structure of free packet list */
+union free_packet {
+ struct packet p;
+ union free_packet *next;
+};
+
+/** @brief Linked list of free packets
+ *
+ * This is a linked list of formerly used packets. For preference we re-use
+ * packets that have already been used rather than unused ones, to limit the
+ * size of the program's working set. If there are no free packets in the list
+ * we try @ref next_free_packet instead.
+ *
+ * Must hold @ref lock when accessing this.
+ */
+static union free_packet *free_packets;
+
+/** @brief Array of new free packets
+ *
+ * There are @ref count_free_packets ready to use at this address. If there
+ * are none left we allocate more memory.
+ *
+ * Must hold @ref lock when accessing this.
+ */
+static union free_packet *next_free_packet;
+
+/** @brief Count of new free packets at @ref next_free_packet
+ *
+ * Must hold @ref lock when accessing this.
+ */
+static size_t count_free_packets;
-static struct frame *frames; /* received frames in ascending order
- * of timestamp */
+/** @brief Lock protecting @ref packets
+ *
+ * This also protects the packet memory allocation infrastructure, @ref
+ * free_packets and @ref next_free_packet. */
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' },
+ { "max", required_argument, 0, 'x' },
+ { "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 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;
}
-/* 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 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 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;
+ free_packet(p);
+ 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. */
+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(!f)
- f = xmalloc(sizeof *f);
- 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:
fatal(errno, "error reading from socket");
}
}
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ /* Ignore too-short packets */
+ if((size_t)n <= sizeof (struct rtp_header)) {
+ info("ignored a short packet");
+ continue;
+ }
+ timestamp = htonl(header.timestamp);
+ seq = htons(header.seq);
+ /* Ignore packets in the past */
+ if(active && lt(timestamp, next_timestamp)) {
+ info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
+ timestamp, next_timestamp);
+ continue;
+ }
+ pthread_mutex_lock(&lock);
+ p = new_packet();
+ p->timestamp = timestamp;
/* Convert to target format */
- switch(packet.header.mtp & 0x7F) {
+ switch(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 header) / sizeof(uint16_t);
+ /* ALSA can do any necessary conversion itself (though it might be better
+ * to do any necessary conversion in the background) */
+ /* 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);
+ fatal(0, "unsupported RTP payload type %d",
+ 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 */
- 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;
+ 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) {
+ info("buffer full");
+ while(nsamples >= maxbuffer)
+ pthread_cond_wait(&cond, &lock);
+ }
+ /* Add the packet to the heap */
+ pheap_insert(&packets, p);
+ nsamples += p->nsamples;
pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
- f = 0;
}
}
+/** @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
-static OSStatus adioproc(AudioDeviceID inDevice,
- const AudioTimeStamp *inNow,
- const AudioBufferList *inInputData,
- const AudioTimeStamp *inInputTime,
- AudioBufferList *outOutputData,
- const AudioTimeStamp *inOutputTime,
- void *inClientData) {
+/** @brief Callback from Core Audio */
+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(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);
- continue;
- }
- if(samplesOutLeft == 0) {
- --nbuffers;
- ++ab;
- samplesOut = ab->data;
- samplesOutLeft = ab->mDataByteSize / sizeof (float);
- continue;
+ const struct packet *p;
+ uint32_t samples_available;
+
+ pthread_mutex_lock(&lock);
+ 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(pheap_count(&packets)) {
+ 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.) */
+ break;
+ }
+ p = pheap_count(&packets) ? pheap_first(&packets) : 0;
+ if(p && contains(p, next_timestamp)) {
+ /* This packet is ready to play */
+ 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));
+
+ 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 - that'll be dealt with next time
+ * round */
+ } else {
+ /* No packet is ready to play (and there might be no packet at all) */
+ samples_available = p ? p->timestamp - next_timestamp
+ : samplesOutLeft;
+ if(samples_available > samplesOutLeft)
+ samples_available = samplesOutLeft;
+ info("infill by %"PRIu32, samples_available);
+ /* Conveniently the buffer is 0 to start with */
+ next_timestamp += samples_available;
+ samplesOut += samples_available;
+ samplesOutLeft -= samples_available;
+ }
}
- /* 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;
+ ++ab;
+ --nbuffers;
}
pthread_mutex_unlock(&lock);
return 0;
}
#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, escape = 0;
+ time_t logged, now;
+ uint32_t packet_start, packet_end;
+
+ /* 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 */
+
+ time(&logged);
+ pthread_mutex_lock(&lock);
+ for(;;) {
+ /* Wait for the buffer to fill up a bit */
+ logged = now;
+ info("%lu samples in buffer (%lus)", nsamples,
+ nsamples / (44100 * 2));
+ 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;
+ }
+ active = 1;
+ infilling = 0;
+ escape = 0;
+ logged = now;
+ info("%lu samples in buffer (%lus)", nsamples,
+ nsamples / (44100 * 2));
+ info("Playing...");
+ /* Wait until the buffer empties out */
+ while(nsamples >= minbuffer && !escape) {
+ time(&now);
+ if(now > logged + 10) {
+ logged = now;
+ info("%lu samples in buffer (%lus)", nsamples,
+ nsamples / (44100 * 2));
+ }
+ if(packets
+ && ge(next_timestamp, packets->timestamp + packets->nsamples)) {
+ info("dropping buffered past packet %"PRIx32" < %"PRIx32,
+ packets->timestamp, next_timestamp);
+ drop_first_packet();
+ continue;
+ }
+ /* Wait for ALSA to ask us for more data */
+ pthread_mutex_unlock(&lock);
+ write(2, ".", 1); /* TODO remove me sometime */
+ switch(err = snd_pcm_wait(pcm, -1)) {
+ case 0:
+ info("snd_pcm_wait timed out");
+ break;
+ case 1:
+ break;
+ default:
+ fatal(0, "snd_pcm_wait returned %d", err);
+ }
+ pthread_mutex_lock(&lock);
+ /* ALSA is ready for more data */
+ packet_start = packets->timestamp;
+ packet_end = packets->timestamp + packets->nsamples;
+ if(ge(next_timestamp, packet_start)
+ && lt(next_timestamp, packet_end)) {
+ /* The target timestamp is somewhere in this packet */
+ const uint32_t offset = next_timestamp - packets->timestamp;
+ const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp;
+ const size_t frames_available = samples_available / 2;
+
+ frames_written = snd_pcm_writei(pcm,
+ packets->samples_raw + offset,
+ frames_available);
+ if(frames_written < 0) {
+ switch(frames_written) {
+ case -EAGAIN:
+ info("snd_pcm_wait() returned but we got -EAGAIN!");
+ break;
+ case -EPIPE:
+ error(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ escape = 1;
+ break;
+ default:
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ }
+ } else {
+ samples_written = frames_written * 2;
+ next_timestamp += samples_written;
+ if(ge(next_timestamp, packet_end))
+ drop_first_packet();
+ infilling = 0;
+ }
+ } else {
+ /* We don't have anything to play! We'd better play some 0s. */
+ static const uint16_t zeros[INFILL_SAMPLES];
+ size_t samples_available = INFILL_SAMPLES, frames_available;
+
+ /* If the maximum infill would take us past the start of the next
+ * packet then we truncate the infill to the right amount. */
+ if(lt(packets->timestamp,
+ next_timestamp + samples_available))
+ samples_available = packets->timestamp - next_timestamp;
+ if((int)samples_available < 0) {
+ info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32,
+ packets->timestamp, next_timestamp,
+ next_timestamp + INFILL_SAMPLES, samples_available);
+ }
+ frames_available = samples_available / 2;
+ if(!infilling) {
+ info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]",
+ samples_available, next_timestamp,
+ packets->timestamp, packets->timestamp + packets->nsamples);
+ //infilling++;
+ }
+ frames_written = snd_pcm_writei(pcm,
+ zeros,
+ frames_available);
+ if(frames_written < 0) {
+ switch(frames_written) {
+ case -EAGAIN:
+ info("snd_pcm_wait() returned but we got -EAGAIN!");
+ break;
+ case -EPIPE:
+ error(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ escape = 1;
+ break;
+ default:
+ 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(escape) {
+ if((err = snd_pcm_drop(pcm)))
+ fatal(0, "error calling snd_pcm_drop: %d", err);
+ escape = 0;
+ } else
+ 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;
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 */
- while(nsamples < READAHEAD)
+ info("Buffering...");
+ while(nsamples < readahead)
pthread_cond_wait(&cond, &lock);
/* Start playing now */
+ info("Playing...");
+ next_timestamp = pheap_first(&packets)->timestamp;
+ active = 1;
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);
if(status)
fatal(0, "AudioDeviceStop: %d", (int)status);
+ active = 0;
/* Go back round */
}
}
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"
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
- " --debug, -d Turn on debugging\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:x:L:", 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;
default: fatal(0, "invalid option");
}
}
+ if(!maxbuffer)
+ maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
if(argc < 1 || argc > 2)
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