chiark / gitweb /
abolish linked list of packets. (linux still to do.)
authorRichard Kettlewell <rjk@greenend.org.uk>
Tue, 18 Sep 2007 23:37:49 +0000 (00:37 +0100)
committerRichard Kettlewell <rjk@greenend.org.uk>
Tue, 18 Sep 2007 23:37:49 +0000 (00:37 +0100)
clients/playrtp.c

index fe9d1e287fa330bcb5ad94dcbeb432e66bac6678..398f47b181b3b63c1d4ce7fb1fa67e50bb810a50 100644 (file)
@@ -30,6 +30,7 @@
 #include <netdb.h>
 #include <pthread.h>
 #include <locale.h>
+#include <sys/uio.h>
 
 #include "log.h"
 #include "mem.h"
@@ -86,37 +87,29 @@ static unsigned maxbuffer;
 /** @brief Number of samples to infill by in one go */
 #define INFILL_SAMPLES (44100 * 2)      /* 1s */
 
-/** @brief Received packet
- *
- * Packets are recorded in an ordered linked list. */
+/** @brief Received packet */
 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 */
   uint32_t nsamples;
   /** @brief Timestamp from RTP packet
    *
    * NB that "timestamps" are really sample counters.*/
   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 Total number of samples available */
 static unsigned long nsamples;
 
-/** @brief Linked list of packets
+/** @brief Mapping of sequence numbers to packets
  *
- * In ascending order of timestamp.  Really this should be a heap for more
- * efficient access. */
-static struct packet *packets;
+ * This isn't very efficient - 256KB on 32-bit machines, 512KB if you do a
+ * 64-bit build for some reason.  It can be optimized later if need be. */
+static struct packet *packets[65536];
+
+/** @brief Total number of packets */
+static unsigned npackets;
 
 /** @brief Timestamp of next packet to play.
  *
@@ -130,6 +123,24 @@ static uint32_t next_timestamp;
  * This is true when playing and false when just buffering. */
 static int active;
 
+/** @brief Sequence number of next packet we expxect to play */
+static uint16_t sequence;
+
+/** @brief Structure of free packet list */
+union free_packet {
+  struct packet p;
+  union free_packet *next;
+};
+
+/** @brief Linked list of free packets */
+static union free_packet *free_packets;
+
+/** @brief Array of new free packets */
+static union free_packet *next_free_packet;
+
+/** @brief Count of new free packets */
+static size_t count_free_packets;
+
 /** @brief Lock protecting @ref packets */
 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
 
@@ -147,6 +158,35 @@ static const struct option options[] = {
   { 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 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;
@@ -168,12 +208,14 @@ static inline int le(uint32_t a, uint32_t b) {
 }
 
 /** @brief Drop the packet at the head of the queue */
-static void drop_first_packet(void) {
-  struct packet *const p = packets;
-  packets = p->next;
-  nsamples -= p->nsamples;
-  free(p);
-  pthread_cond_broadcast(&cond);
+static void drop_packet(unsigned sequence) {
+  if(packets[sequence]) {
+    nsamples -= packets[sequence]->nsamples;
+    free_packet(packets[sequence]);
+    packets[sequence] = 0;
+    pthread_cond_broadcast(&cond);
+    --npackets;
+  }
 }
 
 /** @brief Background thread collecting samples
@@ -181,19 +223,24 @@ static void drop_first_packet(void) {
  * 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, **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;
+    n = readv(rtpfd, iov, 2);
     if(n < 0) {
       switch(errno) {
       case EINTR:
@@ -207,41 +254,33 @@ static void *listen_thread(void attribute((unused)) *arg) {
       info("ignored a short packet");
       continue;
     }
-    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,
-           p->timestamp, next_timestamp);
+           timestamp, next_timestamp);
       continue;
     }
+    pthread_mutex_lock(&lock);
+    p = new_packet();
+    p->timestamp = timestamp;
     /* Convert to target format */
-    switch(packet.header.mpt & 0x7F) {
+    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 */
-      {
-        size_t i;
-
-        for(i = 0; i < p->nsamples; ++i)
-          p->samples_float[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767);
-      }
-#else
+      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) */
-      memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
-#endif
+      /* 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);
+            header.mpt & 0x7F);
     }
     if(logfp)
       fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
-              ntohs(packet.header.seq), 
-              p->timestamp, p->nsamples, p->timestamp + p->nsamples);
-    pthread_mutex_lock(&lock);
+              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
@@ -251,28 +290,30 @@ static void *listen_thread(void attribute((unused)) *arg) {
       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. */
-      info("dropped a duplicated");
-    } else {
-      if(*pp)
-        info("receiving packets out of order");
-      p->next = *pp;
-      *pp = p;
-      nsamples += p->nsamples;
-      pthread_cond_broadcast(&cond);
-      p = 0;                            /* we've consumed this packet */
-    }
+    /* If there's a packet there already we overwrite it; perhaps it is left
+     * over from an earlier stage. */
+    drop_packet(seq);
+    /* Record this packet */
+    packets[seq] = p;
+    /* If we currently have no idea where to start playing, this is it */
+    if(!npackets)
+      sequence = seq;
+    ++npackets;
+    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
@@ -285,48 +326,47 @@ static OSStatus adioproc
      void attribute((unused)) *inClientData) {
   UInt32 nbuffers = outOutputData->mNumberBuffers;
   AudioBuffer *ab = outOutputData->mBuffers;
+  const struct packet *p;
 
   pthread_mutex_lock(&lock);
   while(nbuffers > 0) {
     float *samplesOut = ab->mData;
     size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
-    
+
     while(samplesOutLeft > 0) {
-      if(packets) {
-        /* There's a packet */
-        const uint32_t packet_start = packets->timestamp;
-        const uint32_t packet_end = packets->timestamp + packets->nsamples;
-        
-        if(le(packet_end, next_timestamp)) {
-          /* This packet is in the past */
-          info("dropping buffered past packet %"PRIx32" < %"PRIx32,
-               packet_start, next_timestamp);
-          drop_first_packet();
-          continue;
-        }
-        if(ge(next_timestamp, packet_start)
-           && lt(next_timestamp, packet_end)) {
+      /* Look for a suitable packet, dropping any unsuitable ones along the
+       * way.  Unsuitable packets are ones that are in the past. */
+      while(npackets
+            && (!packets[sequence]
+                || le(packets[sequence]->timestamp
+                         + packets[sequence]->nsamples,
+                      next_timestamp)))
+        drop_packet(sequence++);
+      p = packets[sequence];
+      if(p) {
+        if(contains(p, next_timestamp)) {
           /* This packet is suitable */
-          const uint32_t offset = next_timestamp - packet_start;
+          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));
           uint32_t samples_available = packet_end - next_timestamp;
           if(samples_available > samplesOutLeft)
             samples_available = samplesOutLeft;
-          memcpy(samplesOut,
-                 packets->samples_float + offset,
-                 samples_available * sizeof(float));
-          samplesOut += samples_available;
           next_timestamp += samples_available;
           samplesOutLeft -= samples_available;
-          if(ge(next_timestamp, packet_end))
-            drop_first_packet();
+          while(samples_available-- > 0)
+            *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
+          /* We don't bother junking the packet or advancing sequence - that'll
+           * be dealt with next time round */
           continue;
         }
       }
       /* We didn't find a suitable packet (though there might still be
        * unsuitable ones).  We infill with 0s. */
-      if(packets) {
+      if(p) {
         /* There is a next packet, only infill up to that point */
-        uint32_t samples_available = packets->timestamp - next_timestamp;
+        uint32_t samples_available = p->timestamp - next_timestamp;
         
         if(samples_available > samplesOutLeft)
           samples_available = samplesOutLeft;
@@ -443,8 +483,9 @@ static void play_rtp(void) {
           fatal(0, "error calling snd_pcm_prepare: %d", err);
         prepared = 1;
       }
+      assert(sequence != -1);
       /* Start at the first available packet */
-      next_timestamp = packets->timestamp;
+      next_timestamp = packets[sequence]->timestamp;
       active = 1;
       infilling = 0;
       escape = 0;
@@ -624,7 +665,7 @@ static void play_rtp(void) {
         pthread_cond_wait(&cond, &lock);
       /* Start playing now */
       info("Playing...");
-      next_timestamp = packets->timestamp;
+      next_timestamp = packets[sequence]->timestamp;
       active = 1;
       status = AudioDeviceStart(adid, adioproc);
       if(status)