Use custom sound DMA instead of SDL for audio

MakePHAT

@@ -29,6 +29,7 @@ COMMON_OBJS = \
 	source/psp/input.o \
 	source/psp/main.o \
 	source/psp/math.o \
+	source/psp/sound.o \
 	source/psp/system.o \
 	source/psp/module.o \
 	source/psp/network.o \
@@ -64,7 +65,6 @@ COMMON_OBJS = \
 	source/pr_cmds.o \
 	source/pr_edict.o \
 	source/pr_exec.o \
-	source/snd_sdl.o \
 	source/snd_dma.o \
 	source/snd_mem.o \
 	source/snd_mix.o \
@@ -108,14 +108,13 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
 
 OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
 
-SDL_LIBS = -lSDL2 -lSDL2main -lGL
 GU_LIBS	= -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
-AUDIO_LIBS	= -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
+AUDIO_LIBS	= -lpspaudiolib -lpspaudio -lpspaudiocodec -lpspmp3 source/psp/m33libs/libpspkubridge.a
 
 MISC_LIBS	= -lpsppower -lpspmath -lpsphprm -ljpeg -lpng -lz
 NET_LIBS    = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
 STD_LIBS	= -lstdc++ -lm -lc
-LIBS		= $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
+LIBS		= $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
 
 CFLAGS = -ffast-math -O3 -Ofast -G0 -Wall $(GPROF_FLAGS) -Did386="0" -DPSP $(MODE) $(HARDWARE_VIDEO_ONLY_FLAGS) -DSWIZZLE32 -DPSP_MP3_HWDECODE -DFULLBRIGHT -DHL_RENDER -Wno-strict-aliasing -DPSP_VFPU
 CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing

MakeSLIM

@@ -29,6 +29,7 @@ COMMON_OBJS = \
 	source/psp/input.o \
 	source/psp/main.o \
 	source/psp/math.o \
+	source/psp/sound.o \
 	source/psp/system.o \
 	source/psp/module.o \
 	source/psp/network.o \
@@ -64,7 +65,6 @@ COMMON_OBJS = \
 	source/pr_cmds.o \
 	source/pr_edict.o \
 	source/pr_exec.o \
-	source/snd_sdl.o \
 	source/snd_dma.o \
 	source/snd_mem.o \
 	source/snd_mix.o \
@@ -108,14 +108,13 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
 
 OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
 
-SDL_LIBS = -lSDL2 -lSDL2main -lGL
 GU_LIBS	= -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
-AUDIO_LIBS	= -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
+AUDIO_LIBS	= -lpspaudiolib -lpspaudio -lpspaudiocodec -lpspmp3 source/psp/m33libs/libpspkubridge.a
 
 MISC_LIBS	= -lpsppower -lpspmath -lpsphprm -ljpeg -lpng -lz
 NET_LIBS    = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
 STD_LIBS	= -lstdc++ -lm -lc
-LIBS		= $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
+LIBS		= $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
 
 CFLAGS = -ffast-math -O3 -Ofast -G0 -Wall $(GPROF_FLAGS) -Did386="0" -DPSP $(MODE) $(HARDWARE_VIDEO_ONLY_FLAGS) -DSWIZZLE32 -DSLIM -DPSP_MP3_HWDECODE -DFULLBRIGHT -DHL_RENDER -Wno-strict-aliasing -DPSP_VFPU
 CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing

source/psp/main.cpp

@@ -36,9 +36,6 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #include <pspge.h>
 #include <pspsysevent.h>
 
-#include <SDL2/SDL.h>
-#include <SDL2/SDL_mixer.h>
-
 extern "C"
 {
 #include "../quakedef.h"
@@ -534,13 +531,6 @@ int user_main(SceSize argc, void* argp)
 	// Initialize the Common module.
     InitExtModules ();
 
-	// Initialize SDL
-    if (SDL_Init(SDL_INIT_AUDIO) < 0)
-    {
-        Sys_Error("SDL2: Could not initialize!\n");
-        return 0;
-    }
-
 	ramClockSpeed = cpuClockSpeed = scePowerGetCpuClockFrequencyInt();
 	busClockSpeed = scePowerGetBusClockFrequencyInt();
 

source/psp/module.cpp

@@ -23,4 +23,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #include "module.h"
 
 // Set up the module info.
+// Set up the module info.
+PSP_MODULE_INFO("nzportable", 0, 1, 1);
+PSP_MAIN_THREAD_ATTR(PSP_THREAD_ATTR_USER | PSP_THREAD_ATTR_VFPU);
 PSP_HEAP_SIZE_KB(-1024);

source/psp/sound.cpp

@@ -0,0 +1,183 @@
+/*
+Copyright (C) 1996-1997 Id Software, Inc.
+Copyright (C) 2007 Peter Mackay and Chris Swindle.
+
+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 2
+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.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
+
+*/
+
+#include <pspaudiolib.h>
+#include <pspdebug.h>
+#include <pspkernel.h>
+
+extern "C"
+{
+#include "../quakedef.h"
+}
+
+namespace quake
+{
+	namespace sound
+	{
+		struct Sample
+		{
+			short left;
+			short right;
+		};
+
+		static const unsigned int		channelCount				= 2;
+		static const unsigned int		inputBufferSize				= 16384;
+
+#if 1 //def NORMAL_MODE
+		static const unsigned int		inputFrequency				= 11025;
+#else
+		static const unsigned int		inputFrequency				= 22050;
+#endif
+		static const unsigned int		outputFrequency				= 44100;
+		static const unsigned int		inputSamplesPerOutputSample	= outputFrequency / inputFrequency;
+		static Sample					inputBuffer[inputBufferSize];
+		static volatile unsigned int	samplesRead;
+
+		static inline void copySamples(const Sample* first, const Sample* last, Sample* destination)
+		{
+			switch (inputSamplesPerOutputSample)
+			{
+			case 1:
+				memcpy(destination, first, (last - first) * sizeof(Sample));
+				break;
+
+			case 2:
+				for (const Sample* source = first; source != last; ++source)
+				{
+					const Sample sample = *source;
+					*destination++ = sample;
+					*destination++ = sample;
+				}
+				break;
+
+			case 4:
+				for (const Sample* source = first; source != last; ++source)
+				{
+					const Sample sample = *source;
+					*destination++ = sample;
+					*destination++ = sample;
+					*destination++ = sample;
+					*destination++ = sample;
+				}
+				break;
+
+			default:
+				break;
+			}
+		}
+
+		static void fillOutputBuffer(void* buffer, unsigned int samplesToWrite, void* userData)
+		{
+			// Where are we writing to?
+			Sample* const destination = static_cast<Sample*> (buffer);
+
+			// Where are we reading from?
+			const Sample* const firstSampleToRead = &inputBuffer[samplesRead];
+
+			// How many samples to read?
+			const unsigned int samplesToRead = samplesToWrite / inputSamplesPerOutputSample;
+
+			// Going to wrap past the end of the input buffer?
+			const unsigned int samplesBeforeEndOfInput = inputBufferSize - samplesRead;
+			if (samplesToRead > samplesBeforeEndOfInput)
+			{
+				// Yes, so write the first chunk from the end of the input buffer.
+				copySamples(
+					firstSampleToRead,
+					firstSampleToRead + samplesBeforeEndOfInput,
+					&destination[0]);
+
+				// Write the second chunk from the start of the input buffer.
+				const unsigned int samplesToReadFromBeginning = samplesToRead - samplesBeforeEndOfInput;
+				copySamples(
+					&inputBuffer[0],
+					&inputBuffer[samplesToReadFromBeginning],
+					&destination[samplesBeforeEndOfInput * inputSamplesPerOutputSample]);
+			}
+			else
+			{
+				// No wrapping, just copy.
+				copySamples(
+					firstSampleToRead,
+					firstSampleToRead + samplesToRead,
+					&destination[0]);
+			}
+
+			// Update the read offset.
+			samplesRead = (samplesRead + samplesToRead) % inputBufferSize;
+		}
+	}
+}
+
+using namespace quake;
+using namespace quake::sound;
+
+qboolean SNDDMA_Init(void)
+{
+	// Set up Quake's audio.
+	shm = &sn;
+	shm->channels			= channelCount;
+	shm->samplebits			= 16;
+	shm->speed				= inputFrequency;
+	shm->soundalive			= qtrue;
+	shm->splitbuffer		= qfalse;
+	shm->samples			= inputBufferSize * channelCount;
+	shm->samplepos			= 0;
+	shm->submission_chunk	= 1;
+	shm->buffer				= (unsigned char *) inputBuffer;
+
+	// Initialise the audio system. This initialises it for the CD audio module
+	// too.
+	pspAudioInit();
+
+	// Set the channel callback.
+	// Sound effects use channel 0, CD audio uses channel 1.
+	pspAudioSetChannelCallback(0, fillOutputBuffer, 0);
+
+	return qtrue;
+}
+
+void SNDDMA_Shutdown(void)
+{
+	// Clear the mixing buffer so we don't get any noise during cleanup.
+	memset(inputBuffer, 0, sizeof(inputBuffer));
+
+	// Clear the channel callback.
+	pspAudioSetChannelCallback(0, 0, 0);
+
+	// Stop the audio system?
+	pspAudioEndPre();
+
+	// Insert a false delay so the thread can be cleaned up.
+	sceKernelDelayThread(50 * 1000);
+
+	// Shut down the audio system.
+	pspAudioEnd();
+}
+
+int SNDDMA_GetDMAPos(void)
+{
+	return samplesRead * channelCount;
+}
+
+void SNDDMA_Submit(void)
+{
+}

source/snd_dma.c

@@ -8,7 +8,7 @@ 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.  
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
 
 See the GNU General Public License for more details.
 
@@ -21,12 +21,16 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 
 #include "quakedef.h"
 
-void S_Play(void);
+
-void S_PlayVol(void);
+
-void S_SoundList(void);
+void S_Play_f(void);
+void S_PlayVol_f(void);
+void S_SoundList_f(void);
 void S_Update_();
 void S_StopAllSounds(qboolean clear);
-void S_StopAllSoundsC(void);
+void S_StopAllSoundsC_f(void);
+void S_VolumeDown_f (void); // Baker 3.60 - from JoeQuake 0.15
+void S_VolumeUp_f (void);  // Baker 3.60 - from JoeQuake 0.15
 
 // =======================================================================
 // Internal sound data & structures
@@ -47,49 +51,43 @@ vec3_t		listener_origin;
 vec3_t		listener_forward;
 vec3_t		listener_right;
 vec3_t		listener_up;
-vec_t		sound_nominal_clip_dist=1000.0;
+vec_t		sound_nominal_clip_dist=1500.0; // JPG - changed this from 1000 to 15000 (I'm 99% sure that's what it was in 1.06)
 
 int			soundtime;		// sample PAIRS
 int   		paintedtime; 	// sample PAIRS
 
 
-#define	MAX_SFX		512
 sfx_t		*known_sfx;		// hunk allocated [MAX_SFX]
 int			num_sfx;
 
 sfx_t		*ambient_sfx[NUM_AMBIENTS];
 
-int 		desired_speed = 44100; //11025;
+int 		desired_speed = 11025;
 int 		desired_bits = 16;
 
 int sound_started=0;
 
 cvar_t bgmvolume = {"bgmvolume", "1", true};
 cvar_t bgmtype = {"bgmtype", "cd", true};   // cd or none
-
 cvar_t volume = {"volume", "0.7", true};
 
 cvar_t nosound = {"nosound", "0"};
 cvar_t precache = {"precache", "1"};
 cvar_t loadas8bit = {"loadas8bit", "0"};
 cvar_t bgmbuffer = {"bgmbuffer", "4096"};
-cvar_t ambient_level = {"ambient_level", "0.3"};
+cvar_t ambient_level = {"ambient_level", "0.3", true}; // Baker 3.60 - Save to config
 cvar_t ambient_fade = {"ambient_fade", "100"};
 cvar_t snd_noextraupdate = {"snd_noextraupdate", "0"};
 cvar_t snd_show = {"snd_show", "0"};
 cvar_t _snd_mixahead = {"_snd_mixahead", "0.1", true};
 
-
 // ====================================================================
 // User-setable variables
 // ====================================================================
 
-
-//
 // Fake dma is a synchronous faking of the DMA progress used for
 // isolating performance in the renderer.  The fakedma_updates is
 // number of times S_Update() is called per second.
-//
 
 qboolean fakedma = false;
 int fakedma_updates = 15;
@@ -114,7 +112,7 @@ void S_SoundInfo_f(void)
 		Con_Printf ("sound system not started\n");
 		return;
 	}
-	
+
     Con_Printf("%5d stereo\n", shm->channels - 1);
     Con_Printf("%5d samples\n", shm->samples);
     Con_Printf("%5d samplepos\n", shm->samplepos);
@@ -141,9 +139,7 @@ void S_Startup (void)
 
 	if (!fakedma)
 	{
-		rc = SNDDMA_Init();
+		if (!(rc = SNDDMA_Init()))
-
-		if (!rc)
 		{
 			Con_Printf("S_Startup: SNDDMA_Init failed.\n");
 			sound_started = 0;
@@ -160,22 +156,24 @@ void S_Startup (void)
 S_Init
 ================
 */
+void CDAudioSetVolume (void);
 void S_Init (void)
 {
-
-	Con_Printf("\nSound Initialization\n");
-
 	if (COM_CheckParm("-nosound"))
 		return;
 
+	Con_Printf("\nSound Initialization\n");
+
 	if (COM_CheckParm("-simsound"))
 		fakedma = true;
 
-	Cmd_AddCommand("play", S_Play);
+	Cmd_AddCommand("play", S_Play_f);
-	Cmd_AddCommand("playvol", S_PlayVol);
+	Cmd_AddCommand("playvol", S_PlayVol_f);
-	Cmd_AddCommand("stopsound", S_StopAllSoundsC);
+	Cmd_AddCommand("stopsound", S_StopAllSoundsC_f);
-	Cmd_AddCommand("soundlist", S_SoundList);
+	Cmd_AddCommand("soundlist", S_SoundList_f);
 	Cmd_AddCommand("soundinfo", S_SoundInfo_f);
+	Cmd_AddCommand ("volumedown", S_VolumeDown_f); // Baker 3.60 - from JoeQuake 0.15
+	Cmd_AddCommand ("volumeup", S_VolumeUp_f); // Baker 3.60 - from JoeQuake 0.15
 
 	Cvar_RegisterVariable(&nosound);
 	Cvar_RegisterVariable(&volume);
@@ -196,7 +194,6 @@ void S_Init (void)
 	}
 
 
-
 	snd_initialized = true;
 
 	S_Startup ();
@@ -223,9 +220,7 @@ void S_Init (void)
 		shm->buffer = Hunk_AllocName(1<<16, "shmbuf");
 	}
 
-	if ( shm ) {
+	Con_Printf ("Sound sampling rate: %i Hz\n", shm->speed);
-		Con_Printf ("Sound sampling rate: %i\n", shm->speed);
-	}
 
 	// provides a tick sound until washed clean
 
@@ -242,7 +237,6 @@ void S_Init (void)
 
 void S_Shutdown(void)
 {
-
 	if (!sound_started)
 		return;
 
@@ -253,9 +247,7 @@ void S_Shutdown(void)
 	sound_started = 0;
 
 	if (!fakedma)
-	{
 		SNDDMA_Shutdown();
-	}
 }
 
 
@@ -277,24 +269,22 @@ sfx_t *S_FindName (char *name)
 	if (!name)
 		Sys_Error ("S_FindName: NULL\n");
 
-	if (Q_strlen(name) >= MAX_QPATH)
+	if (strlen(name) >= MAX_QPATH)
 		Sys_Error ("Sound name too long: %s", name);
 
 // see if already loaded
 	for (i=0 ; i < num_sfx ; i++)
 		if (!strcmp(known_sfx[i].name, name))
-		{
 			return &known_sfx[i];
-		}
 
 	if (num_sfx == MAX_SFX)
 		Sys_Error ("S_FindName: out of sfx_t");
-	
+
 	sfx = &known_sfx[i];
 	strcpy (sfx->name, name);
 
 	num_sfx++;
-	
+
 	return sfx;
 }
 
@@ -308,7 +298,7 @@ S_TouchSound
 void S_TouchSound (char *name)
 {
 	sfx_t	*sfx;
-	
+
 	if (!sound_started)
 		return;
 
@@ -330,11 +320,11 @@ sfx_t *S_PrecacheSound (char *name)
 		return NULL;
 
 	sfx = S_FindName (name);
-	
+
 // cache it in
 	if (precache.value)
 		S_LoadSound (sfx);
-	
+
 	return sfx;
 }
 
@@ -360,7 +350,7 @@ channel_t *SND_PickChannel(int entnum, int entchannel)
 		if (entchannel != 0		// channel 0 never overrides
 		&& channels[ch_idx].entnum == entnum
 		&& (channels[ch_idx].entchannel == entchannel || entchannel == -1) )
-		{	// allways override sound from same entity
+		{	// always override sound from same entity
 			first_to_die = ch_idx;
 			break;
 		}
@@ -382,8 +372,8 @@ channel_t *SND_PickChannel(int entnum, int entchannel)
 	if (channels[first_to_die].sfx)
 		channels[first_to_die].sfx = NULL;
 
-    return &channels[first_to_die];    
+    return &channels[first_to_die];
-}       
+}
 
 /*
 =================
@@ -393,12 +383,12 @@ SND_Spatialize
 void SND_Spatialize(channel_t *ch)
 {
     vec_t dot;
-    vec_t ldist, rdist, dist;
+    vec_t dist;
     vec_t lscale, rscale, scale;
     vec3_t source_vec;
 	sfx_t *snd;
 
-// anything coming from the view entity will allways be full volume
+// anything coming from the view entity will always be full volume
 	if (ch->entnum == cl.viewentity)
 	{
 		ch->leftvol = ch->master_vol;
@@ -410,9 +400,9 @@ void SND_Spatialize(channel_t *ch)
 
 	snd = ch->sfx;
 	VectorSubtract(ch->origin, listener_origin, source_vec);
-	
+
 	dist = VectorNormalize(source_vec) * ch->dist_mult;
-	
+
 	dot = DotProduct(listener_right, source_vec);
 
 	if (shm->channels == 1)
@@ -436,7 +426,7 @@ void SND_Spatialize(channel_t *ch)
 	ch->leftvol = (int) (ch->master_vol * scale);
 	if (ch->leftvol < 0)
 		ch->leftvol = 0;
-}           
+}
 
 
 // =======================================================================
@@ -466,7 +456,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
 	target_chan = SND_PickChannel(entnum, entchannel);
 	if (!target_chan)
 		return;
-		
+
 // spatialize
 	memset (target_chan, 0, sizeof(*target_chan));
 	VectorCopy(origin, target_chan->origin);
@@ -480,8 +470,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
 		return;		// not audible at all
 
 // new channel
-	sc = S_LoadSound (sfx);
+	if (!(sc = S_LoadSound (sfx)))
-	if (!sc)
 	{
 		target_chan->sfx = NULL;
 		return;		// couldn't load the sound's data
@@ -489,7 +478,7 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
 
 	target_chan->sfx = sfx;
 	target_chan->pos = 0.0;
-    target_chan->end = paintedtime + sc->length;	
+    target_chan->end = paintedtime + sc->length;
 
 // if an identical sound has also been started this frame, offset the pos
 // a bit to keep it from just making the first one louder
@@ -507,7 +496,6 @@ void S_StartSound(int entnum, int entchannel, sfx_t *sfx, vec3_t origin, float f
 			target_chan->end -= skip;
 			break;
 		}
-		
 	}
 }
 
@@ -517,8 +505,7 @@ void S_StopSound(int entnum, int entchannel)
 
 	for (i=0 ; i<MAX_DYNAMIC_CHANNELS ; i++)
 	{
-		if (channels[i].entnum == entnum
+		if (channels[i].entnum == entnum && channels[i].entchannel == entchannel)
-			&& channels[i].entchannel == entchannel)
 		{
 			channels[i].end = 0;
 			channels[i].sfx = NULL;
@@ -540,13 +527,13 @@ void S_StopAllSounds(qboolean clear)
 		if (channels[i].sfx)
 			channels[i].sfx = NULL;
 
-	Q_memset(channels, 0, MAX_CHANNELS * sizeof(channel_t));
+	memset(channels, 0, MAX_CHANNELS * sizeof(channel_t));
 
 	if (clear)
 		S_ClearBuffer ();
 }
 
-void S_StopAllSoundsC (void)
+void S_StopAllSoundsC_f (void)
 {
 	S_StopAllSounds (true);
 }
@@ -554,7 +541,7 @@ void S_StopAllSoundsC (void)
 void S_ClearBuffer (void)
 {
 	int		clear;
-		
+
 	if (!sound_started || !shm || !shm->buffer)
 		return;
 
@@ -563,7 +550,9 @@ void S_ClearBuffer (void)
 	else
 		clear = 0;
 
-	Q_memset(shm->buffer, clear, shm->samples * shm->samplebits/8);
+	{
+		memset(shm->buffer, clear, shm->samples * shm->samplebits/8);
+	}
 }
 
 
@@ -589,8 +578,7 @@ void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
 	ss = &channels[total_channels];
 	total_channels++;
 
-	sc = S_LoadSound (sfx);
+	if (!(sc = S_LoadSound (sfx)))
-	if (!sc)
 		return;
 
 	if (sc->loopstart == -1)
@@ -598,13 +586,13 @@ void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
 		Con_Printf ("Sound %s not looped\n", sfx->name);
 		return;
 	}
-	
+
 	ss->sfx = sfx;
 	VectorCopy (origin, ss->origin);
 	ss->master_vol = vol;
 	ss->dist_mult = (attenuation/64) / sound_nominal_clip_dist;
-    ss->end = paintedtime + sc->length;	
+    ss->end = paintedtime + sc->length;
-	
+
 	SND_Spatialize (ss);
 }
 
@@ -640,9 +628,9 @@ void S_UpdateAmbientSounds (void)
 
 	for (ambient_channel = 0 ; ambient_channel< NUM_AMBIENTS ; ambient_channel++)
 	{
-		chan = &channels[ambient_channel];	
+		chan = &channels[ambient_channel];
 		chan->sfx = ambient_sfx[ambient_channel];
-	
+
 		vol = ambient_level.value * l->ambient_sound_level[ambient_channel];
 		if (vol < 8)
 			vol = 0;
@@ -660,7 +648,7 @@ void S_UpdateAmbientSounds (void)
 			if (chan->master_vol < vol)
 				chan->master_vol = vol;
 		}
-		
+
 		chan->leftvol = chan->rightvol = chan->master_vol;
 	}
 }
@@ -687,13 +675,13 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
 	VectorCopy(forward, listener_forward);
 	VectorCopy(right, listener_right);
 	VectorCopy(up, listener_up);
-	
+
 // update general area ambient sound sources
 	S_UpdateAmbientSounds ();
 
 	combine = NULL;
 
-// update spatialization for static and dynamic sounds	
+// update spatialization for static and dynamic sounds
 	ch = channels+NUM_AMBIENTS;
 	for (i=NUM_AMBIENTS ; i<total_channels; i++, ch++)
 	{
@@ -705,7 +693,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
 
 	// try to combine static sounds with a previous channel of the same
 	// sound effect so we don't mix five torches every frame
-	
+
 		if (i >= MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS)
 		{
 		// see if it can just use the last one
@@ -721,7 +709,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
 			for (j=MAX_DYNAMIC_CHANNELS + NUM_AMBIENTS ; j<i; j++, combine++)
 				if (combine->sfx == ch->sfx)
 					break;
-					
+
 			if (j == total_channels)
 			{
 				combine = NULL;
@@ -737,13 +725,9 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
 				continue;
 			}
 		}
-		
-		
 	}
 
-//
 // debugging output
-//
 	if (snd_show.value)
 	{
 		total = 0;
@@ -754,7 +738,7 @@ void S_Update(vec3_t origin, vec3_t forward, vec3_t right, vec3_t up)
 				//Con_Printf ("%3i %3i %s\n", ch->leftvol, ch->rightvol, ch->sfx->name);
 				total++;
 			}
-		
+
 		Con_Printf ("----(%i)----\n", total);
 	}
 
@@ -768,18 +752,18 @@ void GetSoundtime(void)
 	static	int		buffers;
 	static	int		oldsamplepos;
 	int		fullsamples;
-	
+
+
 	fullsamples = shm->samples / shm->channels;
 
 // 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;
@@ -796,16 +780,15 @@ void S_ExtraUpdate (void)
 {
 	if (snd_noextraupdate.value)
 		return;		// don't pollute timings
+
 	S_Update_();
 }
 
 void S_Update_(void)
 {
-/*#ifndef SDL
-
 	unsigned        endtime;
 	int				samps;
-	
+
 	if (!sound_started || (snd_blocked > 0))
 		return;
 
@@ -819,6 +802,8 @@ void S_Update_(void)
 		paintedtime = soundtime;
 	}
 
+	//OutputDebugString(va("paintedtime: %i, soundtime: %i\n", paintedtime, soundtime));
+
 // mix ahead of current position
 	endtime = soundtime + _snd_mixahead.value * shm->speed;
 	samps = shm->samples >> (shm->channels-1);
@@ -828,7 +813,6 @@ void S_Update_(void)
 	S_PaintChannels (endtime);
 
 	SNDDMA_Submit ();
-#endif*/
 }
 
 /*
@@ -839,55 +823,61 @@ console functions
 ===============================================================================
 */
 
-void S_Play(void)
+void S_Play_f(void)
 {
 	static int hash=345;
 	int 	i;
 	char name[256];
 	sfx_t	*sfx;
-	
+
 	i = 1;
 	while (i<Cmd_Argc())
 	{
-		if (!Q_strrchr(Cmd_Argv(i), '.'))
+		if (!strrchr(Cmd_Argv(i), '.'))
 		{
-			Q_strcpy(name, Cmd_Argv(i));
+			strcpy(name, Cmd_Argv(i));
-			Q_strcat(name, ".wav");
+			strlcat (name, ".wav", sizeof(name));
 		}
 		else
-			Q_strcpy(name, Cmd_Argv(i));
+			strcpy(name, Cmd_Argv(i));
 		sfx = S_PrecacheSound(name);
 		S_StartSound(hash++, 0, sfx, listener_origin, 1.0, 1.0);
 		i++;
 	}
 }
 
-void S_PlayVol(void)
+void S_PlayVol_f(void)
 {
 	static int hash=543;
 	int i;
 	float vol;
 	char name[256];
 	sfx_t	*sfx;
-	
+
+	if (Cmd_Argc() != 2)
+	{
+		Con_Printf ("Usage: playvol <filename>\n");
+		return;
+	}
+
 	i = 1;
 	while (i<Cmd_Argc())
 	{
-		if (!Q_strrchr(Cmd_Argv(i), '.'))
+		if (!strrchr(Cmd_Argv(i), '.'))
 		{
-			Q_strcpy(name, Cmd_Argv(i));
+			strcpy(name, Cmd_Argv(i));
-			Q_strcat(name, ".wav");
+			strlcat (name, ".wav", sizeof(name));
 		}
 		else
-			Q_strcpy(name, Cmd_Argv(i));
+			strcpy(name, Cmd_Argv(i));
 		sfx = S_PrecacheSound(name);
-		vol = Q_atof(Cmd_Argv(i+1));
+		vol = atof(Cmd_Argv(i+1));
 		S_StartSound(hash++, 0, sfx, listener_origin, vol, 1.0);
 		i+=2;
 	}
 }
 
-void S_SoundList(void)
+void S_SoundList_f(void)
 {
 	int		i;
 	sfx_t	*sfx;
@@ -897,8 +887,7 @@ void S_SoundList(void)
 	total = 0;
 	for (sfx=known_sfx, i=0 ; i<num_sfx ; i++, sfx++)
 	{
-		sc = Cache_Check (&sfx->cache);
+		if (!(sc = Cache_Check (&sfx->cache)))
-		if (!sc)
 			continue;
 		size = sc->length*sc->width*(sc->stereo+1);
 		total += size;
@@ -911,6 +900,25 @@ void S_SoundList(void)
 	Con_Printf ("Total resident: %i\n", total);
 }
 
+qboolean	volume_changed;
+
+void S_VolumeDown_f (void)
+{
+	//S_LocalSound ("misc/menu3.wav");
+	volume.value -= 0.1;
+	volume.value = bound(0, volume.value, 1);
+	//Cvar_SetValueByRef (&volume, volume.value);
+	volume_changed = true;
+}
+
+void S_VolumeUp_f (void)
+{
+	//S_LocalSound ("misc/menu3.wav");
+	volume.value += 0.1;
+	volume.value = bound(0, volume.value, 1);
+	//Cvar_SetValueByRef (&volume, volume.value);
+	volume_changed = true;
+}
 
 void S_LocalSound (char *sound)
 {
@@ -920,9 +928,8 @@ void S_LocalSound (char *sound)
 		return;
 	if (!sound_started)
 		return;
-		
+
-	sfx = S_PrecacheSound (sound);
+	if (!(sfx = S_PrecacheSound (sound)))
-	if (!sfx)
 	{
 		Con_Printf ("S_LocalSound: can't cache %s\n", sound);
 		return;
@@ -943,4 +950,5 @@ void S_BeginPrecaching (void)
 
 void S_EndPrecaching (void)
 {
-}
+}
+

source/snd_mem.c

@@ -105,8 +105,7 @@ sfxcache_t *S_LoadSound (sfx_t *s)
 	byte	stackbuf[1*1024];		// avoid dirtying the cache heap
 
 // see if still in memory
-	sc = Cache_Check (&s->cache);
+	if ((sc = Cache_Check (&s->cache)))
-	if (sc)
 		return sc;
 
 //Con_Printf ("S_LoadSound: %x\n", (int)stackbuf);
@@ -116,9 +115,7 @@ sfxcache_t *S_LoadSound (sfx_t *s)
 
 //	Con_Printf ("loading %s\n",namebuffer);
 
-	data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf));
+	if (!(data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf))))
-
-	if (!data)
 	{
 		Con_Printf ("Couldn't load %s\n", namebuffer);
 		return NULL;
@@ -212,7 +209,7 @@ void FindNextChunk(char *name)
 //			Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
 		data_p -= 8;
 		last_chunk = data_p + 8 + ( (iff_chunk_len + 1) & ~1 );
-		if (!Q_strncmp((char *)data_p, name, 4))
+		if (!strncmp((char*) data_p, name, 4))
 			return;
 	}
 }
@@ -262,7 +259,7 @@ wavinfo_t GetWavinfo (char *name, byte *wav, int wavlength)
 
 // find "RIFF" chunk
 	FindChunk("RIFF");
-	if (!(data_p && !Q_strncmp((char *)data_p+8, "WAVE", 4)))
+	if (!(data_p && !strncmp((char*) data_p+8, "WAVE", 4)))
 	{
 		Con_Printf("Missing RIFF/WAVE chunks\n");
 		return info;
@@ -303,7 +300,7 @@ wavinfo_t GetWavinfo (char *name, byte *wav, int wavlength)
 		FindNextChunk ("LIST");
 		if (data_p)
 		{
-			if (!strncmp ((char *)data_p + 28, "mark", 4))
+			if (!strncmp ((char*) data_p + 28, "mark", 4))
 			{	// this is not a proper parse, but it works with cooledit...
 				data_p += 24;
 				i = GetLittleLong ();	// samples in loop

source/snd_mix.c

@@ -34,8 +34,7 @@ void Snd_WriteLinearBlastStereo16 (void);
 #if	!id386
 void Snd_WriteLinearBlastStereo16 (void)
 {
-	int		i;
+	int	i, val;
-	int		val;
 
 	for (i=0 ; i<snd_linear_count ; i+=2)
 	{
@@ -60,8 +59,7 @@ void Snd_WriteLinearBlastStereo16 (void)
 
 void S_TransferStereo16 (int endtime)
 {
-	int		lpos;
+	int		lpos, lpaintedtime;
-	int		lpaintedtime;
 	DWORD	*pbuf;
 
 	snd_vol = volume.value*256;
@@ -96,13 +94,7 @@ void S_TransferStereo16 (int endtime)
 
 void S_TransferPaintBuffer(int endtime)
 {
-	int 	out_idx;
+	int 	out_idx, count, out_mask, *p, step, val, snd_vol;
-	int 	count;
-	int 	out_mask;
-	int 	*p;
-	int 	step;
-	int		val;
-	int		snd_vol;
 	DWORD	*pbuf;
 
 	if (shm->samplebits == 16 && shm->channels == 2)
@@ -153,6 +145,7 @@ void S_TransferPaintBuffer(int endtime)
 			out_idx = (out_idx + 1) & out_mask;
 		}
 	}
+
 }
 
 
@@ -169,11 +162,9 @@ void SND_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int endtime);
 
 void S_PaintChannels(int endtime)
 {
-	int 	i;
+	int		i, end, ltime, count;
-	int 	end;
 	channel_t *ch;
 	sfxcache_t	*sc;
-	int		ltime, count;
 
 	while (paintedtime < endtime)
 	{
@@ -183,7 +174,7 @@ void S_PaintChannels(int endtime)
 			end = paintedtime + PAINTBUFFER_SIZE;
 
 	// clear the paint buffer
-		Q_memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
+		memset(paintbuffer, 0, (end - paintedtime) * sizeof(portable_samplepair_t));
 
 	// paint in the channels.
 		ch = channels;
@@ -193,8 +184,7 @@ void S_PaintChannels(int endtime)
 				continue;
 			if (!ch->leftvol && !ch->rightvol)
 				continue;
-			sc = S_LoadSound (ch->sfx);
+			if (!(sc = S_LoadSound (ch->sfx)))
-			if (!sc)
 				continue;
 
 			ltime = paintedtime;
@@ -254,10 +244,8 @@ void SND_InitScaletable (void)
 
 void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count)
 {
-	int 	data;
+	int 		i, data, *lscale, *rscale;
-	int		*lscale, *rscale;
 	unsigned char *sfx;
-	int		i;
 
 	if (ch->leftvol > 255)
 		ch->leftvol = 255;
@@ -283,9 +271,7 @@ void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count)
 
 void SND_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int count)
 {
-	int data;
+	int data, left, right, leftvol, rightvol;
-	int left, right;
-	int leftvol, rightvol;
 	signed short *sfx;
 	int	i;
 

source/snd_sdl.c

@@ -1,102 +0,0 @@
-
-#include <stdio.h>
-#include <SDL2/SDL_audio.h>
-#include "quakedef.h"
-
-static dma_t the_shm;
-static int snd_inited;
-
-extern int desired_speed;
-extern int desired_bits;
-
-static void paint_audio(void *unused, Uint8 *stream, int len)
-{
-	if ( shm ) {
-		shm->buffer = stream;
-		shm->samplepos += len/(shm->samplebits/8)/2;
-		// Check for samplepos overflow?
-		S_PaintChannels (shm->samplepos);
-	}
-}
-
-qboolean SNDDMA_Init(void)
-{
-	SDL_AudioSpec desired, obtained;
-
-	snd_inited = 0;
-
-	/* Set up the desired format */
-	desired.freq = desired_speed;
-	switch (desired_bits) {
-		case 8:
-			desired.format = AUDIO_U8;
-			break;
-		case 16:
-			desired.format = AUDIO_S16LSB;
-			break;
-		default:
-        		Con_Printf("Unknown number of audio bits: %d\n",
-								desired_bits);
-			return 0;
-	}
-	desired.channels = 2;
-	desired.samples = 48;
-	desired.callback = paint_audio;
-
-	/* Open the audio device */
-	if ( SDL_OpenAudio(&desired, &obtained) < 0 ) {
-        	Con_Printf("Couldn't open SDL audio: %s\n", SDL_GetError());
-		return 0;
-	}
-
-	/* Make sure we can support the audio format */
-	switch (obtained.format) {
-		case AUDIO_U8:
-			/* Supported */
-			break;
-		case AUDIO_S16LSB:
-		case AUDIO_S16MSB:
-			/* Supported */
-			break;
-		/* Unsupported, fall through */;
-		default:
-			/* Not supported -- force SDL to do our bidding */
-			SDL_CloseAudio();
-			if ( SDL_OpenAudio(&desired, NULL) < 0 ) {
-        			Con_Printf("Couldn't open SDL audio: %s\n",
-							SDL_GetError());
-				return 0;
-			}
-			memcpy(&obtained, &desired, sizeof(desired));
-			break;
-	}
-	SDL_PauseAudio(0);
-
-	/* Fill the audio DMA information block */
-	shm = &the_shm;
-	shm->splitbuffer = 0;
-	shm->samplebits = (obtained.format & 0xFF);
-	shm->speed = obtained.freq;
-	shm->channels = obtained.channels;
-	shm->samples = obtained.samples*shm->channels;
-	shm->samplepos = 0;
-	shm->submission_chunk = 1;
-	shm->buffer = NULL;
-
-	snd_inited = 1;
-	return 1;
-}
-
-int SNDDMA_GetDMAPos(void)
-{
-	return shm->samplepos;
-}
-
-void SNDDMA_Shutdown(void)
-{
-	if (snd_inited)
-	{
-		SDL_CloseAudio();
-		snd_inited = 0;
-	}
-}

source/sound.h

@@ -130,6 +130,8 @@ void SNDDMA_Shutdown(void);
 #define	MAX_CHANNELS			128
 #define	MAX_DYNAMIC_CHANNELS	8
 
+#define	MAX_SFX		512
+
 
 extern	channel_t   channels[MAX_CHANNELS];
 // 0 to MAX_DYNAMIC_CHANNELS-1	= normal entity sounds