diff --git a/common/backends.h b/common/backends.h
index 560f9a59..f3a59310 100644
--- a/common/backends.h
+++ b/common/backends.h
@@ -27,6 +27,7 @@ GNU General Public License for more details.
 #define SOUND_NULL 0
 #define SOUND_SDL 1
 #define SOUND_OPENSLES 2
+#define SOUND_ALSA 3
 
 // crash handler (XASH_CRASHHANDLER)
 #define CRASHHANDLER_NULL 0
diff --git a/common/defaults.h b/common/defaults.h
index 9e2a8631..d2d58a4b 100644
--- a/common/defaults.h
+++ b/common/defaults.h
@@ -81,7 +81,7 @@ SETUP BACKENDS DEFINITIONS
 		#endif
 
 		#ifndef XASH_SOUND
-			#define XASH_SOUND SOUND_NULL
+			#define XASH_SOUND SOUND_ALSA
 		#endif
 		#define XASH_USE_EVDEV
 	#endif // android case
diff --git a/engine/platform/linux/s_alsa.c b/engine/platform/linux/s_alsa.c
new file mode 100644
index 00000000..c0fb8f39
--- /dev/null
+++ b/engine/platform/linux/s_alsa.c
@@ -0,0 +1,295 @@
+/*
+s_alsa.c - alsa sound hardware output
+Copyright (C) 2019 mittorn
+
+This program is free software: you can redistribute it and/or modify
+it under the terms of the GNU General Public License as published by
+the Free Software Foundation, either version 3 of the License, or
+(at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+*/
+
+#include "platform/platform.h"
+#if XASH_SOUND == SOUND_ALSA
+#include <alsa/asoundlib.h>
+#include "sound.h"
+
+#define BUFFER_SAMPLES 4096
+#define SUBMISSION_CHUNK BUFFER_SAMPLES / 2
+
+static struct s_alsa_t
+{
+	snd_pcm_t *pcm_handle;
+	snd_pcm_hw_params_t *hw_params;
+
+	struct sndinfo * si;
+
+	int period_size;
+} s_alsa;
+
+/*
+==================
+SNDDMA_Init
+
+Initialize ALSA pcm device, and bind it to sndinfo.
+==================
+*/
+qboolean SNDDMA_Init( void )
+{
+	int err, dir = 0;
+	unsigned int r;
+	snd_pcm_uframes_t p = 0;
+	string device = "default";
+
+	Sys_GetParmFromCmdLine( "-alsadev", device );
+
+	if( ( err = snd_pcm_open( &s_alsa.pcm_handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) < 0)
+	{
+		Con_Printf( "ALSA: cannot open device %s(%s)\n", device, snd_strerror( err ) );
+		return false;
+	}
+
+	if( ( err = snd_pcm_hw_params_malloc( &s_alsa.hw_params )) < 0 )
+	{
+		Con_Printf( "ALSA: cannot allocate hw params(%s)\n", snd_strerror( err ) );
+		snd_pcm_close(s_alsa.pcm_handle);
+		return false;
+	}
+
+	if( ( err = snd_pcm_hw_params_any( s_alsa.pcm_handle, s_alsa.hw_params ) ) < 0 )
+	{
+		Con_Printf( "ALSA: cannot init hw params(%s)\n", snd_strerror( err ) );
+		snd_pcm_hw_params_free(	s_alsa.hw_params );
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+
+	if( ( err = snd_pcm_hw_params_set_access( s_alsa.pcm_handle, s_alsa.hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
+	{
+		Con_Printf( "ALSA: cannot set access(%s)\n", snd_strerror( err ) );
+		snd_pcm_hw_params_free( s_alsa.hw_params );
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+
+	if( ( err = snd_pcm_hw_params_set_format( s_alsa.pcm_handle, s_alsa.hw_params, SND_PCM_FORMAT_S16 ) ) < 0 )
+	{
+		Con_Printf("ALSA: 16 bit not supported\n");
+		snd_pcm_hw_params_free( s_alsa.hw_params );
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+
+	dma.format.speed = SOUND_DMA_SPEED;
+	r = dma.format.speed;
+
+	if( ( err = snd_pcm_hw_params_set_rate_near( s_alsa.pcm_handle, s_alsa.hw_params, &r, &dir ) ) < 0 )
+	{
+		Con_Printf( "ALSA: cannot set rate %d(%s)\n", r, snd_strerror( err ) );
+		snd_pcm_hw_params_free(s_alsa.hw_params);
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+	else
+	{
+		// rate succeeded, but is perhaps slightly different
+		if( dir != 0 )
+		{
+			Con_Printf( "ALSA: rate %d not supported, using %d\n", SOUND_DMA_SPEED, r );
+			dma.format.speed = r;
+			dir = 0;
+		}
+	}
+
+	dma.format.channels = 2;
+
+	if( ( err = snd_pcm_hw_params_set_channels(s_alsa.pcm_handle, s_alsa.hw_params, dma.format.channels ) ) < 0 )
+	{
+		Con_Printf( "ALSA: cannot set channels %d(%s)\n", 2, snd_strerror( err ) );
+		snd_pcm_hw_params_free( s_alsa.hw_params );
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+
+	p = BUFFER_SAMPLES / dma.format.channels;
+	if( ( err = snd_pcm_hw_params_set_period_size_near( s_alsa.pcm_handle, s_alsa.hw_params, &p, &dir ) ) < 0 )
+	{
+		Con_Printf("ALSA: cannot set period size (%s)\n", snd_strerror(err));
+		snd_pcm_hw_params_free(s_alsa.hw_params);
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+	else
+	{
+		// period succeeded, but is perhaps slightly different
+		if( dir != 0 )
+		{
+			snd_pcm_hw_params_get_period_size(s_alsa.hw_params, &p, NULL);
+			Con_Printf( "ALSA: period %d not supported, using %lu\n", ( BUFFER_SAMPLES / dma.format.channels ), p );
+			dir = 0;
+		}
+	}
+
+	// set params
+	if( ( err = snd_pcm_hw_params( s_alsa.pcm_handle, s_alsa.hw_params ) ) < 0 )
+	{
+		Con_Printf( "ALSA: cannot set params(%s)\n", snd_strerror( err ) );
+		snd_pcm_hw_params_free(s_alsa.hw_params);
+		snd_pcm_close( s_alsa.pcm_handle );
+		return false;
+	}
+
+	// request period size again
+	snd_pcm_hw_params_get_period_size( s_alsa.hw_params, &p, NULL );
+	s_alsa.period_size = p;
+	Con_Printf( "ALSA: period size %lu\n", p );
+
+	dma.buffer = Z_Malloc( BUFFER_SAMPLES * 2 );  //allocate pcm frame buffer
+
+	dma.samplepos = 0;
+
+	dma.samples = BUFFER_SAMPLES;
+	dma.format.width = 2;
+	dma.initialized = 1;
+	snd_pcm_prepare( s_alsa.pcm_handle );
+
+	return true;
+}
+
+/*
+==============
+SNDDMA_GetDMAPos
+
+return the current sample position (in mono samples read)
+inside the recirculating dma buffer, so the mixing code will know
+how many sample are required to fill it up.
+===============
+*/
+int SNDDMA_GetDMAPos( void )
+{
+	return dma.samplepos;
+}
+
+/*
+==============
+SNDDMA_GetSoundtime
+
+update global soundtime
+===============
+*/
+int SNDDMA_GetSoundtime( void )
+{
+	static int buffers, oldsamplepos;
+	int samplepos, fullsamples;
+
+	fullsamples = dma.samples / 2;
+
+	// it is possible to miscount buffers
+	// if it has wrapped twice between
+	// calls to S_Update.  Oh well.
+	samplepos = SNDDMA_GetDMAPos( );
+
+	if( samplepos < oldsamplepos )
+	{
+		buffers++; // buffer wrapped
+
+		if( paintedtime > 0x40000000 )
+		{
+			// time to chop things off to avoid 32 bit limits
+			buffers     = 0;
+			paintedtime = fullsamples;
+			S_StopAllSounds( true );
+		}
+	}
+
+	oldsamplepos = samplepos;
+
+	return ( buffers * fullsamples + samplepos / 2 );
+}
+
+
+/*
+==============
+SNDDMA_Shutdown
+
+Closes the ALSA pcm device and frees the dma buffer.
+===============
+*/
+void SNDDMA_Shutdown(void)
+{
+	Con_Printf( "Shutting down audio.\n" );
+	dma.initialized = false;
+
+	if( dma.buffer )
+	{
+	  snd_pcm_drop( s_alsa.pcm_handle );
+	  snd_pcm_close( s_alsa.pcm_handle );
+	  Mem_Free( dma.buffer );
+	  dma.buffer = NULL;
+	}
+}
+
+/*
+==============
+SNDDMA_Submit
+
+Send sound to device if buffer isn't really the dma buffer
+===============
+*/
+void SNDDMA_Submit( void )
+{
+	int avail = snd_pcm_avail_update( s_alsa.pcm_handle );
+
+	while( avail >= s_alsa.period_size )
+	{
+		int size    = dma.samples << 1;
+		int pos     = dma.samplepos << 1;
+		unsigned long  len = s_alsa.period_size;
+		int wrapped = pos + len - size;
+
+		if( wrapped < 0 )
+		{
+			snd_pcm_writei( s_alsa.pcm_handle, dma.buffer + pos, len / 4 );
+			dma.samplepos += len >> 1;
+		}
+		else
+		{
+			int remaining = size - pos;
+			snd_pcm_writei( s_alsa.pcm_handle, dma.buffer + pos, remaining / 4 );
+			snd_pcm_writei( s_alsa.pcm_handle, dma.buffer, wrapped / 4 );
+			dma.samplepos = wrapped >> 1;
+		}
+
+		avail = snd_pcm_avail_update( s_alsa.pcm_handle );
+	}
+}
+
+/*
+==============
+SNDDMA_BeginPainting
+
+Callback provided by the engine in case we need it.  We don't.
+==============
+*/
+void SNDDMA_BeginPainting(void)
+{
+}
+
+/*
+===========
+SNDDMA_Activate
+Called when the main window gains or loses focus.
+The window have been destroyed and recreated
+between a deactivate and an activate.
+===========
+*/
+void SNDDMA_Activate( qboolean active )
+{
+	snd_pcm_pause( s_alsa.pcm_handle, active );
+}
+
+#endif
diff --git a/engine/wscript b/engine/wscript
index 190c2cbf..cc762621 100644
--- a/engine/wscript
+++ b/engine/wscript
@@ -28,7 +28,7 @@ def configure(conf):
 			conf.check_cc(lib = i)
 	elif conf.options.FBDEV_SW:
 		conf.define('XASH_FBDEV', 1)
-		conf.env.FBDEV
+		conf.check_cc(lib = 'asound')
 	else:
 		conf.load('sdl2')
 		if not conf.env.HAVE_SDL2:
@@ -57,7 +57,7 @@ def build(bld):
 
 	# basic build: dedicated only, no dependencies
 	if bld.env.DEST_OS != 'win32':
-		libs += [ 'DL' , 'M', 'PTHREAD' ]
+		libs += [ 'DL' , 'M' , 'RT', 'PTHREAD']
 		source += bld.path.ant_glob(['platform/posix/*.c'])
 	else:
 		libs += ['USER32', 'SHELL32', 'GDI32', 'ADVAPI32', 'DBGHELP', 'PSAPI', 'WS2_32' ]
@@ -82,9 +82,12 @@ def build(bld):
 			'client/*.c',
 			'client/vgui/*.c',
 			'client/avi/*.c'])
-	else:
-		if bld.env.DEST_OS == 'linux':
-			libs.append('RT')
+
+	if bld.env.DEST_OS == 'linux' and not bld.env.HAVE_SDL2:
+		libs.append('RT')
+		if not bld.env.DEDICATED:
+			libs.append('ASOUND')
+	
 
 	# HACK: public headers must be put before common, so we don't get wrong mathlib included
 	includes = ['common', 'server', 'client', 'client/vgui', '.', '../public', '../common', '../pm_shared' ]