* 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) two threads. listen_thread() is responsible for
+ * reading RTP packets off the wire and adding them to the binary heap @ref
+ * packets, assuming they are basically sound.
+ *
+ * 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.
+ *
+ * InCore 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.
+ *
+ * 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.
+ */
#include <config.h>
#include "types.h"
#include <netdb.h>
#include <pthread.h>
#include <locale.h>
+#include <sys/uio.h>
+#include <string.h>
#include "log.h"
#include "mem.h"
#include "syscalls.h"
#include "rtp.h"
#include "defs.h"
+#include "vector.h"
+#include "heap.h"
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
# include <CoreAudio/AudioHardware.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 */
-/** @brief Maximum sample size
+/** @brief Buffer high watermark
*
- * The maximum supported size (in bytes) of one sample. */
-#define MAXSAMPLESIZE 2
+ * We'll only start playing when this many samples are available. */
+static unsigned readahead = 2 * 2 * 44100;
-/** @brief Buffer size
+/** @brief Maximum buffer size
*
- * We'll only start playing when this many samples are available. */
-static unsigned readahead = 4 * 2 * 44100; /* 4 seconds */
+ * We'll stop reading from the network if we have this many samples. */
+static unsigned maxbuffer;
-#define MAXBUFFER (3 * 88200) /* maximum buffer contents */
+/** @brief Number of samples to infill by in one go
+ *
+ * This is an upper bound - in practice we expect 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
*
- * Packets are recorded in an ordered linked list. */
+ * Received packets are kept in a binary heap (see @ref pheap) ordered by
+ * timestamp.
+ */
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;
+ uint32_t nsamples;
+
/** @brief Timestamp from RTP packet
*
- * NB that "timestamps" are really sample counters.*/
+ * NB that "timestamps" are really sample counters. Use lt() or lt_packet()
+ * to compare timestamps.
+ */
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 Flags
+ *
+ * Valid values are:
+ * - @ref IDLE - the idle bit was set in the RTP packet
+ */
+ unsigned flags;
+/** @brief idle bit set in RTP packet*/
+#define IDLE 0x0001
+
+ /** @brief Raw sample data
+ *
+ * Only the first @p nsamples samples are defined; the rest is uninitialized
+ * data.
+ */
+ uint16_t samples_raw[MAXSAMPLES];
};
+/** @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 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
* This is true when playing and false when just buffering. */
static int active;
-/** @brief Lock protecting @ref packets */
+/** @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;
+
+/** @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;
/** @brief Condition variable signalled whenever @ref packets is changed */
{ "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 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 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 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. */
+ * 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 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);
+ 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 / 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;
+ }
+ pthread_mutex_lock(&lock);
+ 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) {
+ //info("Buffer full");
+ write(2, "B", 1);
+ while(nsamples >= maxbuffer)
+ pthread_cond_wait(&cond, &lock);
}
+ /* Add the packet to the heap */
+ pheap_insert(&packets, p);
+ nsamples += p->nsamples;
+ /* We'll need a new packet */
+ p = 0;
+ pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
}
}
+/** @brief Return true if @p p contains @p timestamp
+ *
+ * Containment implies that a sample @p timestamp exists within the packet.
+ */
+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));
+}
+
+/** @brief Wait until the buffer is adequately full
+ *
+ * Must be called with @ref lock held.
+ */
+static void fill_buffer(void) {
+ 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.
+ */
+static struct packet *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;
+ }
+ return 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) {
+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;
+ uint32_t samples_available;
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) {
+ const struct packet *p = next_packet();
+ if(p && contains(p, next_timestamp)) {
+ if(p->flags & IDLE)
+ write(2, "I", 1);
+ /* 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);
+
+ 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 */
+ write(2, ".", 1);
+ } 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;
+ write(2, "?", 1);
+ }
}
- /* 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;
}
#endif
+
+#if API_ALSA
+/** @brief PCM handle */
+static snd_pcm_t *pcm;
+
+/** @brief True when @ref pcm is up and running */
+static int alsa_prepared = 1;
+
+/** @brief Initialize @ref pcm */
+static void setup_alsa(void) {
+ 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;
+ int err;
+
+ /* 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);
+}
+
+/** @brief Wait until ALSA wants some audio */
+static void wait_alsa(void) {
+ struct pollfd fds[64];
+ int nfds, err;
+ unsigned short events;
+
+ for(;;) {
+ do {
+ if((nfds = snd_pcm_poll_descriptors(pcm,
+ fds, sizeof fds / sizeof *fds)) < 0)
+ fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds);
+ } while(poll(fds, nfds, -1) < 0 && errno == EINTR);
+ if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events)))
+ fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
+ if(events & POLLOUT)
+ return;
+ }
+}
+
+/** @brief Play some sound via ALSA
+ * @param s Pointer to sample data
+ * @param n Number of samples
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_writei(const void *s, size_t n) {
+ /* Do the write */
+ const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2);
+ if(frames_written < 0) {
+ /* Something went wrong */
+ switch(frames_written) {
+ case -EAGAIN:
+ write(2, "#", 1);
+ return 0;
+ case -EPIPE:
+ error(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ return -1;
+ default:
+ fatal(0, "error calling snd_pcm_writei: %ld",
+ (long)frames_written);
+ }
+ } else {
+ /* Success */
+ next_timestamp += frames_written * 2;
+ return 0;
+ }
+}
+
+/** @brief Play the relevant part of a packet
+ * @param p Packet to play
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_play(const struct packet *p) {
+ if(p->flags & IDLE)
+ write(2, "I", 1);
+ write(2, ".", 1);
+ return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
+ (p->timestamp + p->nsamples) - next_timestamp);
+}
+
+/** @brief Play some silence
+ * @param p Next packet or NULL
+ * @return 0 on success, -1 on non-fatal error
+ */
+static int alsa_infill(const struct packet *p) {
+ static const uint16_t zeros[INFILL_SAMPLES];
+ size_t samples_available = INFILL_SAMPLES;
+
+ if(p && samples_available > p->timestamp - next_timestamp)
+ samples_available = p->timestamp - next_timestamp;
+ write(2, "?", 1);
+ return alsa_writei(zeros, samples_available);
+}
+
+/** @brief Reset ALSA state after we lost synchronization */
+static void alsa_reset(int hard_reset) {
+ int err;
+
+ if((err = snd_pcm_nonblock(pcm, 0)))
+ fatal(0, "error calling snd_pcm_nonblock: %d", err);
+ if(hard_reset) {
+ if((err = snd_pcm_drop(pcm)))
+ fatal(0, "error calling snd_pcm_drop: %d", err);
+ } 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);
+ alsa_prepared = 0;
+}
+#endif
+
/** @brief Play an RTP stream
*
* This is the guts of the program. It is responsible for:
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 */
+ struct packet *p;
+ int escape, err;
+ /* Open the sound device */
+ setup_alsa();
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) {
+ fill_buffer();
+ if(!alsa_prepared) {
if((err = snd_pcm_prepare(pcm)))
fatal(0, "error calling snd_pcm_prepare: %d", err);
- prepared = 1;
+ alsa_prepared = 1;
}
- /* Start at the first available packet */
- next_timestamp = packets->timestamp;
- active = 1;
- infilling = 0;
+ escape = 0;
info("Playing...");
- /* Wait until the buffer empties out */
- while(nsamples >= minbuffer) {
+ /* Keep playing until the buffer empties out, or ALSA tells us to get
+ * lost */
+ while(nsamples >= minbuffer && !escape) {
/* Wait for ALSA to ask us for more data */
pthread_mutex_unlock(&lock);
- snd_pcm_wait(pcm, -1);
+ wait_alsa();
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
- next_timestamp += samples_written;
- }
+ /* ALSA is ready for more data, find something to play */
+ p = next_packet();
+ /* Play it or play some silence */
+ if(contains(p, next_timestamp))
+ escape = alsa_play(p);
+ else
+ escape = alsa_infill(p);
}
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;
+ alsa_reset(escape);
pthread_mutex_lock(&lock);
}
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)
- pthread_cond_wait(&cond, &lock);
+ fill_buffer();
/* 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);
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