diff --git a/akode/plugins/alsa_sink/alsa_sink.cpp b/akode/plugins/alsa_sink/alsa_sink.cpp
index 501b8ca..c226519 100644
--- a/akode/plugins/alsa_sink/alsa_sink.cpp
+++ b/akode/plugins/alsa_sink/alsa_sink.cpp
@@ -18,7 +18,8 @@
     Boston, MA 02110-1301, USA.
 */
 
-// #define DEBUG
+#define DEBUG
+#define AKODE_MODULE "alsa_sink"
 
 #include "akode_debug.h"
 
@@ -172,6 +173,9 @@ found_format:
     else
         return -1;
 
+    // Disable ALSA resampling, since we do that.
+    snd_pcm_hw_params_set_rate_resample(m_data->pcm_playback, hw, 0);
+
     unsigned int rate = config->sample_rate;
     snd_pcm_hw_params_set_rate_near(m_data->pcm_playback, hw, &rate, 0);
     if (m_data->config.sample_rate != rate) {
@@ -181,25 +185,49 @@ found_format:
 
     snd_pcm_hw_params_set_channels(m_data->pcm_playback, hw, config->channels);
 
+    // Set up buffer parameters.
+    // This mainly fixes a delay with PulseAudio.
+    unsigned bufferTime = 100'000; // us (100ms)
+    unsigned periods = 4;
 
-    m_data->fragmentSize = 1024;
-    snd_pcm_uframes_t period_size = m_data->fragmentSize / (wid*config->channels);
-    snd_pcm_hw_params_set_period_size_near(m_data->pcm_playback, hw, &period_size, 0);
+    int err = snd_pcm_hw_params_set_buffer_time_near(m_data->pcm_playback, hw, &bufferTime, nullptr);
+    if (err < 0)
+    {
+        AKODE_WARN("unable to set buffer time: " << snd_strerror(err));
+    }
+    if (err >= 0)
+    {
+        err = snd_pcm_hw_params_set_periods_near(m_data->pcm_playback, hw, &periods, nullptr);
+        if (err < 0)
+        {
+            AKODE_WARN("unable to set periods near: " << snd_strerror(err));
+        }
+    }
 
-    m_data->fragmentSize = period_size * (wid*config->channels);
-//     std::cerr << "akode: ALSA fragment-size: " << m_data->fragmentSize << "\n";
+    err = snd_pcm_hw_params(m_data->pcm_playback, hw);
+    if (err < 0)
+    {
+        AKODE_WARN("unable to set hw params: " << snd_strerror(err));
+        m_data->error = true;
+        return -1;
+    }
+
+    snd_pcm_uframes_t bufferSize;
+    snd_pcm_hw_params_get_buffer_size(hw, &bufferSize);
+    m_data->fragmentSize = snd_pcm_frames_to_bytes(m_data->pcm_playback, bufferSize) * 2;
 
     delete m_data->buffer;
     m_data->buffer = new char [m_data->fragmentSize];
     m_data->filled = 0;
 
-    if (snd_pcm_hw_params(m_data->pcm_playback, hw) < 0) {
-        return -1;
-    }
-    else {
-        m_data->can_pause = (snd_pcm_hw_params_can_pause(hw) == 1);
-        return res;
-    }
+    m_data->can_pause = (snd_pcm_hw_params_can_pause(hw) == 1);
+
+    AKODE_DEBUG("hw can-pause:  " << (m_data->can_pause ? "true" : "false"));
+    AKODE_DEBUG("buffer-size:   " << bufferSize << " frames");
+    AKODE_DEBUG("fragment-size: " <<
+        snd_pcm_bytes_to_frames(m_data->pcm_playback, m_data->fragmentSize) << " frames");
+
+    return res;
 }
 
 const AudioConfiguration* ALSASink::audioConfiguration() const