Go for a 0.250 second buffer length.

diff --git a/audio_alsa.c b/audio_alsa.c
index 4ae2d029f4854ddf2024c79c979777705b695787..b7b3e7e125a4f2679ad6874606642006a5b25f27 100644 (file)
--- a/audio_alsa.c
+++ b/audio_alsa.c
@@ -629,12 +629,12 @@ static int init(int argc, char **argv) {
   config.alsa_use_hardware_mute = 0; // don't use it by default
 
   config.audio_backend_latency_offset = 0;
-  config.audio_backend_buffer_desired_length = 0.15;
+  config.audio_backend_buffer_desired_length = 0.25;
   config.audio_backend_buffer_interpolation_threshold_in_seconds =
-      0.100; // below this, basic interpolation will be used to save time.
+      0.175; // below this, basic interpolation will be used to save time.
   config.alsa_maximum_stall_time = 0.200; // 200 milliseconds -- if it takes longer, it's a problem
-  config.audio_backend_silence_threshold = 0.040; //start sending silent frames if the delay goes below this time
-  config.audio_backend_silence_scan_interval = 0.003; //check silence threshold this often
+  config.audio_backend_silence_threshold = 0.60; //start sending silent frames if the delay goes below this time
+  config.audio_backend_silence_scan_interval = 0.004; //check silence threshold this often
 
   stall_monitor_error_threshold =
       (uint64_t)1000000 * config.alsa_maximum_stall_time; // stall time max to microseconds;
@@ -1472,7 +1472,8 @@ void *alsa_buffer_monitor_thread_code(void *arg) {
   sleep_time_in_fp = sleep_time_in_fp << 32;
   sleep_time_in_fp = sleep_time_in_fp / 1000;
   // debug(1,"alsa: sleep_time: %d ms or 0x%" PRIx64 " in fp form.",sleep_time_ms,sleep_time_in_fp);
-  int frames_of_silence = (desired_sample_rate * sleep_time_ms * 2) / 1000;
+  // int frames_of_silence = (desired_sample_rate * sleep_time_ms * 2) / 1000;
+  int frames_of_silence = 1024;
   size_t size_of_silence_buffer = frames_of_silence * frame_size;
   // debug(1,"Silence buffer length: %u bytes.",size_of_silence_buffer);
   void *silence = malloc(size_of_silence_buffer);

diff --git a/player.c b/player.c
index 724b5cab2dde9586c3b861b85df6019f3a437ea9..ac2226d96c8050fa53d3551800092c75f94d2bd9 100644 (file)
--- a/player.c
+++ b/player.c
@@ -4,7 +4,7 @@
  * All rights reserved.
  *
  * Modifications for audio synchronisation
- * and related work, copyright (c) Mike Brady 2014 -- 2018
+ * and related work, copyright (c) Mike Brady 2014 -- 2019
  * All rights reserved.
  *
  * Permission is hereby granted, free of charge, to any person
@@ -624,6 +624,16 @@ int32_t rand_in_range(int32_t exclusive_range_limit) {
 
 static inline void process_sample(int32_t sample, char **outp, enum sps_format_t format, int volume,
                                   int dither, rtsp_conn_info *conn) {
+  /*
+  {                                
+               static int old_volume = 0;
+               if (volume != old_volume) {
+                       debug(1,"Volume is now %d.",volume);
+                       old_volume = volume;
+               }
+  }
+  */
+  
   int64_t hyper_sample = sample;
   int result = 0;
 
@@ -1554,7 +1564,7 @@ void *player_thread_func(void *arg) {
 
   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.",
+  debug(3, "dac_buffer_queue_minimum_length is %" PRId64 " frames.",
         conn->dac_buffer_queue_minimum_length);
 
   conn->session_corrections = 0;
@@ -2220,9 +2230,6 @@ void *player_thread_func(void *arg) {
                 }
               }
 
-                                                if (config.output->preflight)
-                                                       config.output->preflight(conn->outbuf,play_samples);
-
              if ((current_delay < conn->dac_buffer_queue_minimum_length) ||
                   (config.packet_stuffing == ST_basic)) {
                 play_samples =