예제 #1
0
void shairport_shutdown() {
  if (shutting_down)
    return;
  shutting_down = 1;
  mdns_unregister();
  rtsp_request_shutdown_stream();
  if (config.output)
    config.output->deinit();
}
예제 #2
0
// get the next frame, when available. return 0 if underrun/stream reset.
static abuf_t *buffer_get_frame(void) {
  int16_t buf_fill;
  uint64_t local_time_now;
  // struct timespec tn;
  abuf_t *abuf = 0;
  int i;
  abuf_t *curframe;

  pthread_mutex_lock(&ab_mutex);
  int wait;
  int32_t dac_delay = 0;
  do {
    // get the time
    local_time_now = get_absolute_time_in_fp();

    // if config.timeout (default 120) seconds have elapsed since the last audio packet was
    // received, then we should stop.
    // config.timeout of zero means don't check..., but iTunes may be confused by a long gap
    // followed by a resumption...

    if ((time_of_last_audio_packet != 0) && (shutdown_requested == 0) &&
        (config.dont_check_timeout == 0)) {
      uint64_t ct = config.timeout; // go from int to 64-bit int
      if ((local_time_now > time_of_last_audio_packet) &&
          (local_time_now - time_of_last_audio_packet >= ct << 32)) {
        debug(1, "As Yeats almost said, \"Too long a silence / can make a stone of the heart\"");
        rtsp_request_shutdown_stream();
        shutdown_requested = 1;
      }
    }
    int rco = get_requested_connection_state_to_output();

    if (connection_state_to_output != rco) {
      connection_state_to_output = rco;
      // change happening
      if (connection_state_to_output == 0) { // going off
        pthread_mutex_lock(&flush_mutex);
        flush_requested = 1;
        pthread_mutex_unlock(&flush_mutex);
      }
    }

    pthread_mutex_lock(&flush_mutex);
    if (flush_requested == 1) {
      if (config.output->flush)
        config.output->flush();
      ab_resync();
      first_packet_timestamp = 0;
      first_packet_time_to_play = 0;
      time_since_play_started = 0;
      flush_requested = 0;
    }
    pthread_mutex_unlock(&flush_mutex);
    uint32_t flush_limit = 0;
    if (ab_synced) {
      do {
        curframe = audio_buffer + BUFIDX(ab_read);
        if (curframe->ready) {

          if (curframe->sequence_number != ab_read) {
            // some kind of sync problem has occurred.
            if (BUFIDX(curframe->sequence_number) == BUFIDX(ab_read)) {
              // it looks like some kind of aliasing has happened
              if (seq_order(ab_read, curframe->sequence_number)) {
                ab_read = curframe->sequence_number;
                debug(1, "Aliasing of buffer index -- reset.");
              }
            } else {
              debug(1, "Inconsistent sequence numbers detected");
            }
          }

          if ((flush_rtp_timestamp != 0) &&
              ((curframe->timestamp == flush_rtp_timestamp) ||
               seq32_order(curframe->timestamp, flush_rtp_timestamp))) {
            debug(1, "Dropping flushed packet seqno %u, timestamp %u", curframe->sequence_number,
                  curframe->timestamp);
            curframe->ready = 0;
            flush_limit++;
            ab_read = SUCCESSOR(ab_read);
          }
          if ((flush_rtp_timestamp != 0) &&
              (!seq32_order(curframe->timestamp,
                            flush_rtp_timestamp))) // if we have gone past the flush boundary time
            flush_rtp_timestamp = 0;
        }
      } while ((flush_rtp_timestamp != 0) && (flush_limit <= 8820) && (curframe->ready == 0));

      if (flush_limit == 8820) {
        debug(1, "Flush hit the 8820 frame limit!");
        flush_limit = 0;
      }

      curframe = audio_buffer + BUFIDX(ab_read);

      if (curframe->ready) {
        if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player
          if (first_packet_timestamp == 0) { // if this is the very first packet
            // debug(1,"First frame seen, time %u, with %d
            // frames...",curframe->timestamp,seq_diff(ab_read, ab_write));
            uint32_t reference_timestamp;
            uint64_t reference_timestamp_time,remote_reference_timestamp_time;
            get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
            if (reference_timestamp) { // if we have a reference time
              // debug(1,"First frame seen with timestamp...");
              first_packet_timestamp = curframe->timestamp; // we will keep buffering until we are
                                                            // supposed to start playing this

              // Here, calculate when we should start playing. We need to know when to allow the
              // packets to be sent to the player.
              // We will send packets of silence from now until that time and then we will send the
              // first packet, which will be followed by the subsequent packets.

              // we will get a fix every second or so, which will be stored as a pair consisting of
              // the time when the packet with a particular timestamp should be played, neglecting
              // latencies, etc.

              // It probably won't  be the timestamp of our first packet, however, so we might have
              // to do some calculations.

              // To calculate when the first packet will be played, we figure out the exact time the
              // packet should be played according to its timestamp and the reference time.
              // We then need to add the desired latency, typically 88200 frames.

              // Then we need to offset this by the backend latency offset. For example, if we knew
              // that the audio back end has a latency of 100 ms, we would
              // ask for the first packet to be emitted 100 ms earlier than it should, i.e. -4410
              // frames, so that when it got through the audio back end,
              // if would be in sync. To do this, we would give it a latency offset of -100 ms, i.e.
              // -4410 frames.

              int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp);

              first_packet_time_to_play =
                  reference_timestamp_time +
                  ((delta + (int64_t)config.latency + (int64_t)config.audio_backend_latency_offset)
                   << 32) /
                      44100;

              if (local_time_now >= first_packet_time_to_play) {
                debug(
                    1,
                    "First packet is late! It should have played before now. Flushing 0.1 seconds");
                player_flush(first_packet_timestamp + 4410);
              }
            }
          }

          if (first_packet_time_to_play != 0) {

            uint32_t filler_size = frame_size;
            uint32_t max_dac_delay = 4410;
            filler_size = 4410; // 0.1 second -- the maximum we'll add to the DAC

            if (local_time_now >= first_packet_time_to_play) {
              // we've gone past the time...
              // debug(1,"Run past the exact start time by %llu frames, with time now of %llx, fpttp
              // of %llx and dac_delay of %d and %d packets;
              // flush.",(((tn-first_packet_time_to_play)*44100)>>32)+dac_delay,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
              // ab_write));

              if (config.output->flush)
                config.output->flush();
              ab_resync();
              first_packet_timestamp = 0;
              first_packet_time_to_play = 0;
              time_since_play_started = 0;
            } else {
              if (config.output->delay) {
                dac_delay = config.output->delay();
                if (dac_delay == -1) {
                  debug(1, "Error getting dac_delay in buffer_get_frame.");
                  dac_delay = 0;
                }
              } else
                dac_delay = 0;
              uint64_t gross_frame_gap =
                  ((first_packet_time_to_play - local_time_now) * 44100) >> 32;
              int64_t exact_frame_gap = gross_frame_gap - dac_delay;
              if (exact_frame_gap <= 0) {
                // we've gone past the time...
                // debug(1,"Run a bit past the exact start time by %lld frames, with time now of
                // %llx, fpttp of %llx and dac_delay of %d and %d packets;
                // flush.",-exact_frame_gap,tn,first_packet_time_to_play,dac_delay,seq_diff(ab_read,
                // ab_write));
                if (config.output->flush)
                  config.output->flush();
                ab_resync();
                first_packet_timestamp = 0;
                first_packet_time_to_play = 0;
              } else {
                uint32_t fs = filler_size;
                if (fs > (max_dac_delay - dac_delay))
                  fs = max_dac_delay - dac_delay;
                if ((exact_frame_gap <= fs) || (exact_frame_gap <= frame_size * 2)) {
                  fs = exact_frame_gap;
                  // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
                  // with %d packets, ab_read is %04x, ab_write is
                  // %04x.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read,
                  // ab_write),ab_read,ab_write);
                  ab_buffering = 0;
                }
                signed short *silence;
                silence = malloc(FRAME_BYTES(fs));
                memset(silence, 0, FRAME_BYTES(fs));
                // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
                // with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write));
                config.output->play(silence, fs);
                free(silence);
                if (ab_buffering == 0) {
                  uint64_t reference_timestamp_time; // don't need this...
                  get_reference_timestamp_stuff(&play_segment_reference_frame, &reference_timestamp_time, &play_segment_reference_frame_remote_time);
#ifdef CONFIG_METADATA
                  send_ssnc_metadata('prsm', NULL, 0, 0); // "resume", but don't wait if the queue is locked
#endif
                }
              }
            }
          }
        }
      }
    }

    // Here, we work out whether to release a packet or wait
    // We release a buffer when the time is right.

    // To work out when the time is right, we need to take account of (1) the actual time the packet
    // should be released,
    // (2) the latency requested, (3) the audio backend latency offset and (4) the desired length of
    // the audio backend's buffer

    // The time is right if the current time is later or the same as
    // The packet time + (latency + latency offset - backend_buffer_length).
    // Note: the last three items are expressed in frames and must be converted to time.

    int do_wait = 1;
    if ((ab_synced) && (curframe) && (curframe->ready) && (curframe->timestamp)) {
      uint32_t reference_timestamp;
      uint64_t reference_timestamp_time,remote_reference_timestamp_time;
      get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
      if (reference_timestamp) { // if we have a reference time
        uint32_t packet_timestamp = curframe->timestamp;
        int64_t delta = ((int64_t)packet_timestamp - (int64_t)reference_timestamp);
        int64_t offset = (int64_t)config.latency + config.audio_backend_latency_offset -
                         (int64_t)config.audio_backend_buffer_desired_length;
        int64_t net_offset = delta + offset;
        int64_t time_to_play = reference_timestamp_time;
        int64_t net_offset_fp_sec;
        if (net_offset >= 0) {
          net_offset_fp_sec = (net_offset << 32) / 44100;
          time_to_play += net_offset_fp_sec; // using the latency requested...
          // debug(2,"Net Offset: %lld, adjusted: %lld.",net_offset,net_offset_fp_sec);
        } else {
          net_offset_fp_sec = ((-net_offset) << 32) / 44100;
          time_to_play -= net_offset_fp_sec;
          // debug(2,"Net Offset: %lld, adjusted: -%lld.",net_offset,net_offset_fp_sec);
        }

        if (local_time_now >= time_to_play) {
          do_wait = 0;
        }
      }
    }
    wait = (ab_buffering || (do_wait != 0) || (!ab_synced)) && (!please_stop);

    if (wait) {
      uint64_t time_to_wait_for_wakeup_fp =
          ((uint64_t)1 << 32) / 44100;       // this is time period of one frame
      time_to_wait_for_wakeup_fp *= 4 * 352; // four full 352-frame packets
      time_to_wait_for_wakeup_fp /= 3; // four thirds of a packet time

#ifdef COMPILE_FOR_LINUX_AND_FREEBSD
      uint64_t time_of_wakeup_fp = local_time_now + time_to_wait_for_wakeup_fp;
      uint64_t sec = time_of_wakeup_fp >> 32;
      uint64_t nsec = ((time_of_wakeup_fp & 0xffffffff) * 1000000000) >> 32;

      struct timespec time_of_wakeup;
      time_of_wakeup.tv_sec = sec;
      time_of_wakeup.tv_nsec = nsec;

      pthread_cond_timedwait(&flowcontrol, &ab_mutex, &time_of_wakeup);
// int rc = pthread_cond_timedwait(&flowcontrol,&ab_mutex,&time_of_wakeup);
// if (rc!=0)
//  debug(1,"pthread_cond_timedwait returned error code %d.",rc);
#endif
#ifdef COMPILE_FOR_OSX
      uint64_t sec = time_to_wait_for_wakeup_fp >> 32;
      ;
      uint64_t nsec = ((time_to_wait_for_wakeup_fp & 0xffffffff) * 1000000000) >> 32;
      struct timespec time_to_wait;
      time_to_wait.tv_sec = sec;
      time_to_wait.tv_nsec = nsec;
      pthread_cond_timedwait_relative_np(&flowcontrol, &ab_mutex, &time_to_wait);
#endif
    }
  } while (wait);