Add experimental negotiated latencies setting and associated logic.

common.h

@@ -69,7 +69,7 @@ typedef struct {
   uint32_t AirPlayLatency; // supplied with --AirPlayLatency option
   uint32_t ForkedDaapdLatency; // supplied with --ForkedDaapdLatency option
   int daemonise;
-  int statistics_requested;
+  int statistics_requested,use_negotiated_latencies;
   char *cmd_start, *cmd_stop;
   int cmd_blocking;
   int tolerance; // allow this much drift before attempting to correct it

configure.ac

@@ -2,7 +2,7 @@
 # Process this file with autoconf to produce a configure script.
 
 AC_PREREQ([2.50])
-AC_INIT([shairport-sync], [2.5.0.5], [mikebrady@eircom.net])
+AC_INIT([shairport-sync], [2.5.0.6], [mikebrady@eircom.net])
 AM_INIT_AUTOMAKE
 AC_CONFIG_SRCDIR([shairport.c])
 AC_CONFIG_HEADERS([config.h])

player.c

@@ -299,11 +299,10 @@ void player_put_packet(seq_t seqno, uint32_t timestamp, uint8_t *data, int len)
   time_of_last_audio_packet = get_absolute_time_in_fp();
   if (connection_state_to_output) { // if we are supposed to be processing these packets
 
-    if (flush_rtp_timestamp != 0)
-    	debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
+//    if (flush_rtp_timestamp != 0)
+//    	debug(1,"Flush_rtp_timestamp is %u",flush_rtp_timestamp);
 
-//    if ((flush_rtp_timestamp != 0) &&
-    if ((0) &&
+    if ((flush_rtp_timestamp != 0) &&
         ((timestamp == flush_rtp_timestamp) || seq32_order(timestamp, flush_rtp_timestamp))) {
       debug(2, "Dropping flushed packet in player_put_packet, seqno %u, timestamp %u, flushing to "
                "timestamp: %u.",
@@ -1247,7 +1246,7 @@ void player_volume(double f) {
 }
 
 void player_flush(uint32_t timestamp) {
-  debug(1,"Flush requested up to %u. It seems as if 0 is special.",timestamp);
+  // debug(1,"Flush requested up to %u. It seems as if 0 is special.",timestamp);
   pthread_mutex_lock(&flush_mutex);
   flush_requested = 1;
   // if (timestamp!=0)

rtp.c

@@ -194,8 +194,14 @@ static void *rtp_control_receiver(void *arg) {
         rtp_timestamp_less_latency = ntohl(*((uint32_t *)&packet[4]));
         sync_rtp_timestamp = ntohl(*((uint32_t *)&packet[16]));
         
-        debug(1,"Latency is %u frames.",sync_rtp_timestamp-rtp_timestamp_less_latency);
-
+        if (config.use_negotiated_latencies) {
+          uint32_t la = sync_rtp_timestamp-rtp_timestamp_less_latency+11025;
+          if (la!=config.latency) {
+            config.latency = la;
+            debug(1,"Using negotiated latency of %u frames.",config.latency);
+          }
+        }
+        
         if (packet[0] & 0x10) {
           // if it's a packet right after a flush or resume
           sync_rtp_timestamp += 352; // add frame_size -- can't see a reference to this anywhere,

rtsp.c

@@ -582,7 +582,7 @@ static void msg_write_response(int fd, rtsp_message *resp) {
 }
 
 static void handle_record(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) {
-  debug(1,"Handle Record");
+  //debug(1,"Handle Record");
   resp->respcode = 200;
    // I think this is for telling the client what the asbsolute minimum latency actually is,
    // and when the client specifies a latency, it should be added to this figure.
@@ -590,7 +590,7 @@ static void handle_record(rtsp_conn_info *conn, rtsp_message *req, rtsp_message
    // Thus, AirPlay's latency figure of 77175, when added to 11025 gives you exactly 88200
    // and iTunes' latency figure of 88553, when added to 11025 gives you 99578, pretty close to the 99400 we guessed.
    
-  msg_add_header(resp, "Audio-Latency", "88200");
+  msg_add_header(resp, "Audio-Latency", "11025");
   
   char *p;
   uint32_t rtptime = 0;
@@ -650,7 +650,7 @@ static void handle_flush(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *
 }
 
 static void handle_setup(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) {
-  debug(1,"Handle Setup");
+  // debug(1,"Handle Setup");
   int cport, tport;
   int lsport, lcport, ltport;
   uint32_t active_remote = 0;

scripts/shairport-sync.conf

@@ -23,6 +23,7 @@ general =
 // It's probably better to compensate for a delay in the output device using the alsa "audio_backend_latency_offset" setting -- see below.
 latencies =
 {
+//  use_negotiated_latencies = "no"; // set this to "yes" to negotiate latencies with the client, and ignore the static settings listed below 
 //	default = 88200; // used for unrecognised sources and for iTunes up to and including iTunes 9.X.
 //	itunes = 99400; // used for iTunes 10 or later
 //	airplay = 88200;

shairport.c

@@ -399,6 +399,16 @@ int parse_options(int argc, char **argv) {
           die("Invalid ignore_volume_control option choice \"%s\". It should be \"yes\" or \"no\"");
       }
 
+      /* Get the dynamic latencies setting. */
+      if (config_lookup_string(config.cfg, "latencies.use_negotiated_latencies", &str)) {
+        if (strcasecmp(str, "no") == 0)
+          config.use_negotiated_latencies = 0;
+        else if (strcasecmp(str, "yes") == 0)
+          config.use_negotiated_latencies = 1;
+        else
+          die("Invalid use_negotiated_latencies option choice \"%s\". It should be \"yes\" or \"no\"");
+      }
+
       /* Get the default latency. */
       if (config_lookup_int(config.cfg, "latencies.default", &value))
         config.latency = value;
@@ -850,6 +860,7 @@ int main(int argc, char **argv) {
   debug(2, "audio backend desired buffer length is %d.",
         config.audio_backend_buffer_desired_length);
   debug(2, "audio backend latency offset is %d.", config.audio_backend_latency_offset);
+  debug(2, "use_negotiated_latencies %d.", config.use_negotiated_latencies);
   
   char *realConfigPath = realpath(config.configfile,NULL);
   if (realConfigPath) {