Add code for calculating and recognising the interpolation threshold.

diff --git a/player.c b/player.c
index 5885568bd1774200fbe68e6eadd6c13bbc72ff18..f17cf6852fb1840a92316827e59f674ceb9b1d5a 100644 (file)
--- a/player.c
+++ b/player.c
@@ -1491,7 +1491,7 @@ void *player_thread_func(void *arg) {
   conn->first_packet_timestamp = 0;
   conn->flush_requested = 0;
   conn->fix_volume = 0x10000;
-
+  
   if (conn->latency == 0) {
     debug(3, "No latency has (yet) been specified. Setting 88,200 (2 seconds) frames "
              "as a default.");
@@ -1551,6 +1551,10 @@ void *player_thread_func(void *arg) {
 
   debug(3, "Output frame bytes is %d.", conn->output_bytes_per_frame);
 
+  conn->dac_buffer_queue_minimum_length = (int64_t)(config.audio_backend_buffer_interpolation_threshold_in_seconds *
+                                           config.output_rate);
+  debug(1,"dac_buffer_queue_minimum_length is %" PRId64 " frames.", conn->dac_buffer_queue_minimum_length);
+
   conn->session_corrections = 0;
   // conn->play_segment_reference_frame = 0; // zero signals that we are not in a play segment
 
@@ -2214,7 +2218,7 @@ void *player_thread_func(void *arg) {
                 }
               }
 
-              if ((current_delay < (DAC_BUFFER_QUEUE_MINIMUM_LENGTH * conn->output_sample_ratio)) ||
+              if ((current_delay < conn->dac_buffer_queue_minimum_length) ||
                   (config.packet_stuffing == ST_basic)) {
                 play_samples =
                     stuff_buffer_basic_32((int32_t *)conn->tbuf, inbuflength, config.output_format,