playrtp now uses heap.h

[disorder] / clients / playrtp.c

/*
 * This file is part of DisOrder.
 * Copyright (C) 2007 Richard Kettlewell
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful, but
 * WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * 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 <getopt.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/socket.h>
#include <sys/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 "configuration.h"
#include "addr.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>
#endif
#if API_ALSA
#include <alsa/asoundlib.h>
#endif

#define readahead linux_headers_are_borked

/** @brief RTP socket */
static int rtpfd;

/** @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];
};

/** @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;

/** @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 */
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 }
};

/** @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. */
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(!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:
        continue;
      default:
        fatal(errno, "error reading from socket");
      }
    }
    /* 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(header.mpt & 0x7F) {
    case 10:
      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",
            header.mpt & 0x7F);
    }
    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);
  }
}

/** @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 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;
  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;
      }
    }
    ++ab;
    --nbuffers;
  }
  pthread_mutex_unlock(&lock);
  return 0;
}
#endif

/** @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(&ltid, 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, 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;
    UInt32 propertySize;
    AudioDeviceID adid;
    AudioStreamBasicDescription asbd;

    /* If this looks suspiciously like libao's macosx driver there's an
     * excellent reason for that... */

    /* TODO report errors as strings not numbers */
    propertySize = sizeof adid;
    status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
                                      &propertySize, &adid);
    if(status)
      fatal(0, "AudioHardwareGetProperty: %d", (int)status);
    if(adid == kAudioDeviceUnknown)
      fatal(0, "no output device");
    propertySize = sizeof asbd;
    status = AudioDeviceGetProperty(adid, 0, false,
                                    kAudioDevicePropertyStreamFormat,
                                    &propertySize, &asbd);
    if(status)
      fatal(0, "AudioHardwareGetProperty: %d", (int)status);
    D(("mSampleRate       %f", asbd.mSampleRate));
    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);
    if(status)
      fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
    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 = 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)
        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 */
    }
  }
#else
# error No known audio API
#endif
}

/* display usage message and terminate */
static void help(void) {
  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"
          );
  xfclose(stdout);
  exit(0);
}

/* display version number and terminate */
static void version(void) {
  xprintf("disorder-playrtp version %s\n", disorder_version_string);
  xfclose(stdout);
  exit(0);
}

int main(int argc, char **argv) {
  int n;
  struct addrinfo *res;
  struct stringlist sl;
  char *sockname;

  static const struct addrinfo prefs = {
    AI_PASSIVE,
    PF_INET,
    SOCK_DGRAM,
    IPPROTO_UDP,
    0,
    0,
    0,
    0
  };

  mem_init();
  if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
  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)
    fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
  sl.n = argc;
  sl.s = argv;
  /* Listen for inbound audio data */
  if(!(res = get_address(&sl, &prefs, &sockname)))
    exit(1);
  if((rtpfd = socket(res->ai_family,
                     res->ai_socktype,
                     res->ai_protocol)) < 0)
    fatal(errno, "error creating socket");
  if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
    fatal(errno, "error binding socket to %s", sockname);
  play_rtp();
  return 0;
}

/*
Local Variables:
c-basic-offset:2
comment-column:40
fill-column:79
indent-tabs-mode:nil
End:
*/