Allow audio packets as well as resends on the control port. Add some debug messages.

diff --git a/dacp.c b/dacp.c
index 6300111373a8cc3295c8ff8219bed70abae268fa..4ecc2064da8295b82eac62c7281f35187e2bd72c 100644 (file)
--- a/dacp.c
+++ b/dacp.c
@@ -223,7 +223,7 @@ int dacp_send_command(const char *command, char **body, ssize_t *bodysize) {
               if (setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, (const char *)&tv, sizeof tv) == -1)
                 debug(1, "Error %d setting receive timeout for DACP service.", errno);
               int ndata = recv(sockfd, buffer, sizeof(buffer), 0);
-              debug(3, "Received %d bytes: \"%s\".", ndata, buffer);
+              // debug(3, "Received %d bytes: \"%s\".", ndata, buffer);
               if (ndata <= 0) {
                 debug(1, "dacp_send_command -- error receiving response for command \"%s\".",
                       command);

diff --git a/player.c b/player.c
index 3552410a07bf56a3199cb320b95fa1fa3337e998..4f652802a64e3d228f5fc63f7cc79249695d2104 100644 (file)
--- a/player.c
+++ b/player.c
@@ -1173,7 +1173,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
 
   if (!conn->ab_buffering) {
     // check once, after a short period of has elapsed, assuming 352 frames per packet
-    i = ((250 * 44100) / 352) / 1000; // approx 250 ms
+    i = ((125 * 44100) / 352) / 1000; // approx 125 ms
     if (i < seq_diff(conn->ab_read, conn->ab_write, conn->ab_read)) {
       seq_t next = seq_sum(conn->ab_read, i);
       abuf = conn->audio_buffer + BUFIDX(next);
@@ -1686,7 +1686,7 @@ static void *player_thread_func(void *arg) {
           config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio);
         } else if (frames_to_drop) {
           if (frames_to_drop > 3 * config.output_rate)
-            warn("Very large number of frames to drop: %" PRId64 ".",frames_to_drop);
+            warn("Very large number of frames to drop: %" PRId64 ".", frames_to_drop);
           debug(3, "%" PRId64 " frames to drop.", frames_to_drop);
           frames_to_drop -= inframe->length;
           if (frames_to_drop < 0)
@@ -1888,7 +1888,10 @@ static void *player_thread_func(void *arg) {
                 (!conn->player_thread_please_stop) && (config.resyncthreshold > 0.0) &&
                 (abs_sync_error > config.resyncthreshold * config.output_rate)) {
               if (abs_sync_error > 3 * config.output_rate) {
-                warn("Very large sync error: %" PRId64 " frames, with a delay of %" PRId64 "frames.",sync_error,delay);
+                warn("Very large sync error: %" PRId64 " frames, with delay: %" PRId64
+                     ", td_in_frames: %" PRId64 ", rt: %" PRId64 ", nt: %" PRId64
+                     ", current_delay: %" PRId64 ", frames.",
+                     sync_error, delay, td_in_frames, rt, nt, current_delay);
               }
               sync_error_out_of_bounds++;
             } else {

diff --git a/rtp.c b/rtp.c
index 8168de145beca00af90668d541c3e56c0e752002..f402080433aeccb96c97398235c8fdb15261e311 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -134,6 +134,7 @@ void *rtp_audio_receiver(void *arg) {
       if (type == 0x56) {
         pktp += 4;
         plen -= 4;
+        debug(1, "resent packet %u received in audio port.", ntohs(*(uint16_t *)(pktp + 2)));
       }
       seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
       // increment last_seqno and see if it's the same as the incoming seqno
@@ -344,27 +345,42 @@ void *rtp_control_receiver(void *arg) {
       } else {
         debug(1, "Sync packet received before we got a timing packet back.");
       }
-    } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
-      // debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
-      pktp = packet + 4;
-      plen -= 4;
-      seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
+    } else {
+      uint8_t type = packet[1] & ~0x80;
+      if ((type == 0x60) ||
+          (type == 0x56)) { // audio data / resent audio data in the control path -- whaale only?
+        pktp = packet;
+
+        if (type == 0x56) { // resent audio data in the control path -- whaale only?
+          // debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
+          pktp = packet + 4;
+          plen -= 4;
+        }
 
-      int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
+        seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
+        if (type == 0x56)
+          debug(1, "Resent audio packet %u received in the control port.", seqno);
+        else
+          debug(1, "Audio packet %u received in the control port.", seqno);
 
-      pktp += 12;
-      plen -= 12;
+        int64_t timestamp = monotonic_timestamp(ntohl(*(uint32_t *)(pktp + 4)), conn);
 
-      // check if packet contains enough content to be reasonable
-      if (plen >= 16) {
-        player_put_packet(seqno, timestamp, pktp, plen, conn);
-        continue;
+        pktp += 12;
+        plen -= 12;
+
+        // check if packet contains enough content to be reasonable
+
+        if (plen >= 16) {
+          player_put_packet(seqno, timestamp, pktp, plen, conn);
+          continue;
+        } else {
+          debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
+        }
       } else {
-        debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
+        debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
+              nread);
       }
-    } else
-      debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
-            nread);
+    }
   }
 
   debug(3, "Control RTP thread interrupted. terminating.");