Add stall detection to the alsa subsystem.

diff --git a/audio_alsa.c b/audio_alsa.c
index 07ee0cab2dfdc4593e4b780ca45849fd61ee2a5a..3eff6fbed80cf63874dc6d756d533fd25baec379 100644 (file)
--- a/audio_alsa.c
+++ b/audio_alsa.c
@@ -74,6 +74,11 @@ audio_output audio_alsa = {
 
 static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
 
+// for tracking how long the output device has stalled
+uint64_t stall_monitor_start_time; // zero if not initialised / not started / zeroed by flush
+long stall_monitor_frame_count; // set to delay at start of time, incremented by any writes
+int stall_monitor_error_notified; 
+
 static snd_output_t *output = NULL;
 static unsigned int desired_sample_rate;
 static enum sps_format_t sample_format;
@@ -935,6 +940,11 @@ static void start(int i_sample_rate, int i_sample_format) {
 
   frame_index = 0;
   measurement_data_is_valid = 0;
+  
+  stall_monitor_start_time = 0;
+  stall_monitor_frame_count = 0;
+  stall_monitor_error_notified = 0;
+  
 }
 
 // assuming pthread cancellation is disabled
@@ -1051,8 +1061,32 @@ int delay(long *the_delay) {
     // reply = -EIO; // pretend something bad has happened
     pthread_setcancelstate(oldState, NULL);
   }
-  debug(1, "Total playing time to get delay of %d frames: %8.2f us.", *the_delay,
+  debug(3, "Total playing time to get delay of %d frames: %8.2f us.", *the_delay,
     (1000000.0 * (get_absolute_time_in_fp() - overall_time_before) / (uint64_t)0x100000000));
+  
+  if ((reply == 0) && (*the_delay != 0)) {
+    uint64_t stall_monitor_time_now = get_absolute_time_in_fp();
+    if ((stall_monitor_start_time != 0) && (stall_monitor_frame_count == *the_delay)) {
+      // hasn't outputted anything since the last call to delay()
+      int64_t time_stalled = stall_monitor_time_now - stall_monitor_start_time;
+      int64_t stall_monitor_error_threshold = 1000; // microseconds;
+      stall_monitor_error_threshold = (stall_monitor_error_threshold << 32) / 1000000; // now in fp form
+      if (time_stalled > stall_monitor_error_threshold) {
+        if (stall_monitor_error_notified == 0) {
+          debug(1, "alsa output device appears to be stalled -- no output in %8.2 us.",(1000000.0 * stall_monitor_error_threshold) / (uint64_t)0x100000000);
+          stall_monitor_error_notified = 1;
+        }
+      }
+    } else {
+      // is outputting stuff, so restart the monitoring here
+      stall_monitor_start_time = stall_monitor_time_now;
+      stall_monitor_frame_count = *the_delay;
+    }
+  } else {
+    // if there is an error or the delay is zero (from which it is assumed there is no output)
+    stall_monitor_start_time = 0; // zero if not initialised / not started / zeroed by flush
+    stall_monitor_frame_count = 0; // set to delay at start of time, incremented by any writes
+  }
   return reply;
 }
 
@@ -1080,7 +1114,7 @@ static int play(void *buf, int samples) {
     pthread_cleanup_debug_mutex_lock(&alsa_mutex, 10000, 1);
     overall_time_before = get_absolute_time_in_fp();
     ret = actual_open_alsa_device();
-    debug(1, "Opening time: %8.2f us.", (1000000.0 * (get_absolute_time_in_fp() - overall_time_before ) / (uint64_t)0x100000000));
+    debug(3, "Opening time: %8.2f us.", (1000000.0 * (get_absolute_time_in_fp() - overall_time_before ) / (uint64_t)0x100000000));
     if (ret == 0) {
       if (audio_alsa.volume)
         do_volume(set_volume);
@@ -1120,6 +1154,7 @@ static int play(void *buf, int samples) {
         overall_time_before = get_absolute_time_in_fp();
         err = alsa_pcm_write(alsa_handle, buf, samples);
         uint64_t writing_time = get_absolute_time_in_fp() - overall_time_before;
+        stall_monitor_frame_count += samples;
 
         if (err < 0) {
           frame_index = 0;
@@ -1196,7 +1231,7 @@ static int play(void *buf, int samples) {
     pthread_cleanup_pop(0); // release the mutex
   }
   pthread_setcancelstate(oldState, NULL);
-  debug(1, "Total playing time to write %d samples: %8.2f us.",
+  debug(3, "Total playing time to write %d samples: %8.2f us.",
     samples, (1000000.0 * (get_absolute_time_in_fp() - overall_time_before) / (uint64_t)0x100000000));
 
   return ret;
@@ -1212,7 +1247,7 @@ static void flush(void) {
   do_mute(1);
 
   if (alsa_handle) {
-
+    stall_monitor_start_time = 0;
     if ((derr = snd_pcm_drop(alsa_handle)))
       debug(1, "Error %d (\"%s\") dropping output device.", derr, snd_strerror(derr));
 
@@ -1229,7 +1264,7 @@ static void flush(void) {
   debug_mutex_unlock(&alsa_mutex, 3);
   pthread_cleanup_pop(0); // release the mutex
   pthread_setcancelstate(oldState, NULL);
-  debug(1, "Total playing time to flush: %8.2f us.",
+  debug(3, "Total playing time to flush: %8.2f us.",
     (1000000.0 * (get_absolute_time_in_fp() - overall_time_before) / (uint64_t)0x100000000));
 }
 

diff --git a/player.c b/player.c
index 3d3e8daa430de79ea1959df366eb5ffa33828735..5d60687bcc416c8fd23b9053a6d2fd25f8efd7bb 100644 (file)
--- a/player.c
+++ b/player.c
@@ -449,12 +449,6 @@ void player_put_packet(seq_t seqno, uint32_t actual_timestamp, uint8_t *data, in
   conn->time_of_last_audio_packet = get_absolute_time_in_fp();
   if (conn->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);
-
-    // now, if a flush_rtp_timestamp has been defined and the incoming timestamp is "before" it,
-    // drop itÉ
-
     if ((conn->flush_rtp_timestamp != 0) && (actual_timestamp != conn->flush_rtp_timestamp) &&
         (modulo_32_offset(actual_timestamp, conn->flush_rtp_timestamp) <
          conn->input_rate * 10)) { // if it's less than 10 seconds
@@ -2035,7 +2029,7 @@ void *player_thread_func(void *arg) {
                 debug(2, "Large positive sync error: %" PRId64 ".", sync_error);
                 int64_t local_frames_to_drop = sync_error / conn->output_sample_ratio;
                 uint32_t frames_to_drop_sized = local_frames_to_drop;
-                do_flush(inframe->given_timestamp + frames_to_drop_sized, conn);
+                do_flush(inframe->given_timestamp + frames_to_drop_sized, conn); // this will reset_input_flow_metrics anyway
               } else if ((sync_error < 0) && ((-sync_error) > filler_length)) {
                 debug(2,
                       "Large negative sync error: %" PRId64 " with should_be_frame_32 of %" PRIu32