diff --git a/Alc/backends/alsa.c b/Alc/backends/alsa.c
index 9a443c0..97723ca 100644
--- a/Alc/backends/alsa.c
+++ b/Alc/backends/alsa.c
@@ -770,11 +770,8 @@ static ALCboolean ALCplaybackAlsa_reset(ALCplaybackAlsa *self)
     }
     CHECK(snd_pcm_hw_params_set_channels(self->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)));
     /* set rate (implicitly constrains period/buffer parameters) */
-    if(GetConfigValueBool(al_string_get_cstr(device->DeviceName), "alsa", "allow-resampler", 0))
-    {
-        if(snd_pcm_hw_params_set_rate_resample(self->pcmHandle, hp, 0) < 0)
-            ERR("Failed to disable ALSA resampler\n");
-    }
+    if(snd_pcm_hw_params_set_rate_resample(self->pcmHandle, hp, 0) < 0)
+        ERR("Failed to disable ALSA resampler\n");
     CHECK(snd_pcm_hw_params_set_rate_near(self->pcmHandle, hp, &rate, NULL));
     /* set buffer time (implicitly constrains period/buffer parameters) */
     if((err=snd_pcm_hw_params_set_buffer_time_near(self->pcmHandle, hp, &bufferLen, NULL)) < 0)
diff --git a/Alc/backends/pulseaudio.c b/Alc/backends/pulseaudio.c
index 9ad04a7..ffbf04b 100644
--- a/Alc/backends/pulseaudio.c
+++ b/Alc/backends/pulseaudio.c
@@ -950,8 +950,7 @@ static ALCboolean ALCpulsePlayback_reset(ALCpulsePlayback *self)
                                          ALCpulsePlayback_sinkInfoCallback, self);
     wait_for_operation(o, self->loop);
 
-    if(GetConfigValueBool(al_string_get_cstr(device->DeviceName), "pulse", "fix-rate", 0) ||
-       !(device->Flags&DEVICE_FREQUENCY_REQUEST))
+    if(!(device->Flags&DEVICE_FREQUENCY_REQUEST))
         flags |= PA_STREAM_FIX_RATE;
     flags |= PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE;
     flags |= PA_STREAM_ADJUST_LATENCY;
diff --git a/alsoftrc.sample b/alsoftrc.sample
index eccc88b..5b98c2d 100644
--- a/alsoftrc.sample
+++ b/alsoftrc.sample
@@ -132,7 +132,7 @@
 #  sinc4 - extrapolates samples using a 4-point Sinc filter
 #  sinc8 - extrapolates samples using an 8-point Sinc filter
 #  bsinc - extrapolates samples using a band-limited Sinc filter (varying
-#          between 12 and 24 points, with anti-aliasing)
+#          between 12 and 24 points)
 #  Specifying other values will result in using the default (linear).
 #resampler = linear
 
@@ -227,6 +227,7 @@
 #  etc).
 #allow-moves = false
 
+<<<<<<< HEAD
 ## fix-rate:
 #  Specifies whether to match the playback stream's sample rate to the device's
 #  sample rate. Enabling this forces OpenAL Soft to mix sources and effects
@@ -234,6 +235,8 @@
 #  PulseAudio server.
 #fix-rate = false
 
+=======
+>>>>>>> parent of 9891d61... Add options to disable Pulse's and ALSA's resamplers
 ##
 ## ALSA backend stuff
 ##
@@ -280,13 +283,6 @@
 #  and anything else will force mmap off.
 #mmap = true
 
-## allow-resampler:
-#  Specifies whether to allow ALSA's built-in resampler. Enabling this will
-#  allow the playback device to be set to a different sample rate than the
-#  actual output, causing ALSA to apply its own resampling pass after OpenAL
-#  Soft resamples and mixes the sources and effects for output.
-#allow-resampler = false
-
 ##
 ## OSS backend stuff
 ##