X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/662887576254386b8e3481f6f3e6f0289f2500aa..ec57f6c97b41d54ade912f7e3b9f727b40e38e16:/lib/uaudio-rtp.c?ds=inline

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 9980ba0..ea1d1ca 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -59,28 +59,6 @@ static uint32_t rtp_id;
 /** @brief RTP sequence number */
 static uint16_t rtp_sequence;
 
-/** @brief RTP timestamp
- *
- * This is the timestamp that will be used on the next outbound packet.
- *
- * The timestamp in the packet header is only 32 bits wide.  With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes.  Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_timestamp;
-
-/** @brief Actual time corresponding to @ref rtp_timestamp
- *
- * This is the time, on this machine, at which the sample at @ref rtp_timestamp
- * ought to be sent, interpreted as the time the last packet was sent plus the
- * time length of the packet. */
-static struct timeval rtp_timeval;
-
-/** @brief Set when we (re-)activate, to provoke timestamp resync */
-static int rtp_reactivated;
-
 /** @brief Network error count
  *
  * If too many errors occur in too short a time, we give up.
@@ -100,21 +78,20 @@ static const char *const rtp_options[] = {
   "rtp-source-port",
   "multicast-ttl",
   "multicast-loop",
-  "rtp-delay-threshold",
+  "delay-threshold",
   NULL
 };
 
 static size_t rtp_play(void *buffer, size_t nsamples) {
   struct rtp_header header;
   struct iovec vec[2];
-  struct timeval now;
   
   /* We do as much work as possible before checking what time it is */
   /* Fill out header */
   header.vpxcc = 2 << 6;              /* V=2, P=0, X=0, CC=0 */
   header.seq = htons(rtp_sequence++);
   header.ssrc = rtp_id;
-  header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload;
+  header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
 #if !WORDS_BIGENDIAN
   /* Convert samples to network byte order */
   uint16_t *u = buffer, *const limit = u + nsamples;
@@ -127,61 +104,8 @@ static size_t rtp_play(void *buffer, size_t nsamples) {
   vec[0].iov_len = sizeof header;
   vec[1].iov_base = buffer;
   vec[1].iov_len = nsamples * uaudio_sample_size;
-retry:
-  xgettimeofday(&now, NULL);
-  if(rtp_reactivated) {
-    /* We've been deactivated for some unknown interval.  We need to advance
-     * rtp_timestamp to account for the dead air. */
-    /* On the first run through we'll set the start time. */
-    if(!rtp_timeval.tv_sec)
-      rtp_timeval = now;
-    /* See how much time we missed.
-     *
-     * This will be 0 on the first run through, in which case we'll not modify
-     * anything.
-     *
-     * It'll be negative in the (rare) situation where the deactivation
-     * interval is shorter than the last packet we sent.  In this case we wait
-     * for that much time and then return having sent no samples, which will
-     * cause uaudio_play_thread_fn() to retry.
-     *
-     * In the normal case it will be positive.
-     */
-    const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */
-    if(delay < 0) {
-      usleep(-delay);
-      goto retry;
-    }
-    /* Advance the RTP timestamp to the present.  With 44.1KHz stereo this will
-     * overflow the intermediate value with a delay of a bit over 6 years.
-     * This seems acceptable. */
-    uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
-    /* Don't throw off channel synchronization */
-    update -= update % uaudio_channels;
-    /* We log nontrivial changes */
-    if(update)
-      info("advancing rtp_time by %"PRIu64" samples", update);
-      rtp_timestamp += update;
-    rtp_timeval = now;
-    rtp_reactivated = 0;
-  } else {
-    /* Chances are we've been called right on the heels of the previous packet.
-     * If we just sent packets as fast as we got audio data we'd get way ahead
-     * of the player and some buffer somewhere would fill (or at least become
-     * unreasonably large).
-     *
-     * First find out how far ahead of the target time we are.
-     */
-    const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */
-    /* Only delay at all if we are nontrivially ahead. */
-    if(ahead > rtp_delay_threshold) {
-      /* Don't delay by the full amount */
-      usleep(ahead - rtp_delay_threshold / 2);
-      /* Refetch time (so we don't get out of step with reality) */
-      xgettimeofday(&now, NULL);
-    }
-  }
-  header.timestamp = htonl((uint32_t)rtp_timestamp);
+  uaudio_schedule_synchronize();
+  header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
   int written_bytes;
   do {
     written_bytes = writev(rtp_fd, vec, 2);
@@ -195,17 +119,8 @@ retry:
   } else
     rtp_errors /= 2;                    /* gradual decay */
   written_bytes -= sizeof (struct rtp_header);
-  size_t written_samples = written_bytes / uaudio_sample_size;
-  /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample
-   * of the next packet */
-  rtp_timestamp += written_samples;
-  const unsigned usec = (rtp_timeval.tv_usec
-                         + 1000000 * written_samples / (uaudio_rate
-                                                            * uaudio_channels));
-  /* ...will only overflow 32 bits if one packet is more than about half an
-   * hour long, which is not plausible. */
-  rtp_timeval.tv_sec += usec / 1000000;
-  rtp_timeval.tv_usec = usec % 1000000;
+  const size_t written_samples = written_bytes / uaudio_sample_size;
+  uaudio_schedule_update(written_samples);
   return written_samples;
 }
 
@@ -340,14 +255,6 @@ static void rtp_open(void) {
     fatal(errno, "error binding broadcast socket to %s", ssockname);
   if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
     fatal(errno, "error connecting broadcast socket to %s", sockname);
-  /* Various fields are required to have random initial values by RFC3550.  The
-   * packet contents are highly public so there's no point asking for very
-   * strong randomness. */
-  gcry_create_nonce(&rtp_id, sizeof rtp_id);
-  gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
-  gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp);
-  /* rtp_play() will spot this and choose an initial value */
-  rtp_timeval.tv_sec = 0;
 }
 
 static void rtp_start(uaudio_callback *callback,
@@ -364,7 +271,13 @@ static void rtp_start(uaudio_callback *callback,
   else
     fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
           uaudio_bits, uaudio_rate, uaudio_channels); 
+  /* Various fields are required to have random initial values by RFC3550.  The
+   * packet contents are highly public so there's no point asking for very
+   * strong randomness. */
+  gcry_create_nonce(&rtp_id, sizeof rtp_id);
+  gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
   rtp_open();
+  uaudio_schedule_init();
   uaudio_thread_start(callback,
                       userdata,
                       rtp_play,
@@ -380,7 +293,7 @@ static void rtp_stop(void) {
 }
 
 static void rtp_activate(void) {
-  rtp_reactivated = 1;
+  uaudio_schedule_reactivated = 1;
   uaudio_thread_activate();
 }