From 8b2c68af0ef72631ddaf5f9e49ce18846e68aa07 Mon Sep 17 00:00:00 2001
From: ejurgensen <espenjurgensen@gmail.com>
Date: Wed, 16 Mar 2016 22:54:27 +0100
Subject: [PATCH] [alsa] Try to fix issue where devices with small buffers
 would overrun, rendering the device unusable because snd_pcm_writei() starts
 blocking. This is just a poor temporary fix until we either start using
 SND_PCM_NONBLOCK or put alsa in it's own thread or implement Pulseaudio...

---
 src/laudio_alsa.c | 39 +++++++++++++++++++++++++++++----------
 1 file changed, 29 insertions(+), 10 deletions(-)

diff --git a/src/laudio_alsa.c b/src/laudio_alsa.c
index 17b0b79f..731332af 100644
--- a/src/laudio_alsa.c
+++ b/src/laudio_alsa.c
@@ -52,6 +52,7 @@ static uint64_t pcm_start_pos;
 static int pcm_last_error;
 static int pcm_recovery;
 static int pcm_buf_threshold;
+static int pcm_period_size;
 
 static struct pcm_packet *pcm_pkt_head;
 static struct pcm_packet *pcm_pkt_tail;
@@ -213,7 +214,7 @@ laudio_alsa_write(uint8_t *buf, uint64_t rtptime)
 
       return;
     }
-  else if ((pcm_status != LAUDIO_RUNNING) && (pcm_pos + pcm_buf_threshold >= pcm_start_pos))
+  else if ((pcm_status != LAUDIO_RUNNING) && (pcm_pos >= pcm_start_pos))
     {
       /* Kill threshold */
       ret = laudio_alsa_set_start_threshold(0);
@@ -283,9 +284,8 @@ laudio_alsa_write(uint8_t *buf, uint64_t rtptime)
 	}
 
       /* Don't let ALSA fill up the buffer too much */
-// Disabled - seems to cause buffer underruns
-//      if (nsamp == AIRTUNES_V2_PACKET_SAMPLES)
-//	return;
+      if (nsamp == AIRTUNES_V2_PACKET_SAMPLES)
+	return;
     }
 }
 
@@ -358,17 +358,16 @@ laudio_alsa_start(uint64_t cur_pos, uint64_t next_pkt)
     }
 
   DPRINTF(E_DBG, L_LAUDIO, "Start local audio curpos %" PRIu64 ", next_pkt %" PRIu64 "\n", cur_pos, next_pkt);
-  DPRINTF(E_DBG, L_LAUDIO, "PCM will start after %d samples (%d packets)\n", pcm_buf_threshold, pcm_buf_threshold / AIRTUNES_V2_PACKET_SAMPLES);
 
   /* Make pcm_pos the rtptime of the packet containing cur_pos */
   pcm_pos = next_pkt;
   while (pcm_pos > cur_pos)
     pcm_pos -= AIRTUNES_V2_PACKET_SAMPLES;
 
-  pcm_start_pos = next_pkt + pcm_buf_threshold;
+  pcm_start_pos = next_pkt + pcm_period_size;
 
-  /* Compensate threshold, as it's taken into account by snd_pcm_delay() */
-  //pcm_pos += pcm_buf_threshold;
+  /* Compensate period size, otherwise get_pos won't be correct */
+  pcm_pos += pcm_period_size;
 
   DPRINTF(E_DBG, L_LAUDIO, "PCM pos %" PRIu64 ", start pos %" PRIu64 "\n", pcm_pos, pcm_start_pos);
 
@@ -378,6 +377,7 @@ laudio_alsa_start(uint64_t cur_pos, uint64_t next_pkt)
   pcm_last_error = 0;
   pcm_recovery = 0;
 
+  // alsa doesn't actually seem to wait for this threshold?
   ret = laudio_alsa_set_start_threshold(pcm_buf_threshold);
   if (ret < 0)
     {
@@ -532,6 +532,7 @@ laudio_alsa_open(void)
 {
   snd_pcm_hw_params_t *hw_params;
   snd_pcm_uframes_t bufsize;
+  snd_pcm_uframes_t period_size;
   int ret;
 
   hw_params = NULL;
@@ -611,7 +612,24 @@ laudio_alsa_open(void)
       goto out_fail;
     }
 
-  DPRINTF(E_DBG, L_LAUDIO, "Buffer size is %lu samples\n", bufsize);
+  // With a small period size we seem to get underruns because the period time
+  // passes before we manage to feed with samples (if the player is slightly
+  // behind - especially critical during startup when the buffer is low)
+  // Internet suggests period_size should be /2 bufsize, but default seems to be
+  // much lower, so compromise on /4 (but not more than 65536 frames = almost 2 sec).
+  period_size = bufsize / 4;
+  if (period_size > 65536)
+    period_size = 65536;
+
+  ret = snd_pcm_hw_params_set_period_size_near(hdl, hw_params, &period_size, NULL);
+  if (ret < 0)
+    {
+      DPRINTF(E_LOG, L_LAUDIO, "Could not set period size: %s\n", snd_strerror(ret));
+
+      goto out_fail;
+    }
+
+  DPRINTF(E_DBG, L_LAUDIO, "Buffer size is %lu samples, period size is %lu samples\n", bufsize, period_size);
 
   ret = snd_pcm_hw_params(hdl, hw_params);
   if (ret < 0)
@@ -627,7 +645,8 @@ laudio_alsa_open(void)
   pcm_pos = 0;
   pcm_last_error = 0;
   pcm_recovery = 0;
-  pcm_buf_threshold = (bufsize / AIRTUNES_V2_PACKET_SAMPLES) * AIRTUNES_V2_PACKET_SAMPLES;
+  pcm_buf_threshold = ((bufsize - period_size) / AIRTUNES_V2_PACKET_SAMPLES) * AIRTUNES_V2_PACKET_SAMPLES;
+  pcm_period_size = period_size;
 
   ret = mixer_open();
   if (ret < 0)