Use a 100ms capture buffer minimum for PulseAudio, with 50ms fragments

This better matches what is given with the WaveIn backend

1 files changed, 5 insertions, 3 deletions

diff --git a/Alc/pulseaudio.c b/Alc/pulseaudio.c
index 594126cf..dabcbaf0 100644
--- a/Alc/pulseaudio.c
+++ b/Alc/pulseaudio.c
@@ -977,6 +977,8 @@ static ALCboolean pulse_open_capture(ALCdevice *device, const ALCchar *device_na
 
     data->samples = device->UpdateSize * device->NumUpdates;
     data->frame_size = aluFrameSizeFromFormat(device->Format);
+    if(data->samples < 100 * device->Frequency / 1000)
+        data->samples = 100 * device->Frequency / 1000;
 
     if(!(data->ring = CreateRingBuffer(data->frame_size, data->samples)))
     {
@@ -986,10 +988,10 @@ static ALCboolean pulse_open_capture(ALCdevice *device, const ALCchar *device_na
 
     data->attr.minreq = -1;
     data->attr.prebuf = -1;
-    data->attr.maxlength = data->frame_size * data->samples;
+    data->attr.maxlength = data->samples * data->frame_size;
     data->attr.tlength = -1;
-    data->attr.fragsize = min(data->frame_size * data->samples,
-                              10 * device->Frequency / 1000);
+    data->attr.fragsize = min(data->samples, 50 * device->Frequency / 1000) *
+                          data->frame_size;
 
     data->spec.rate = device->Frequency;
     data->spec.channels = aluChannelsFromFormat(device->Format);