Use custom sound DMA instead of SDL for audio
This commit is contained in:
cypress
parent
debcddf2cc
commit
cfbf7f5098
7
MakePHAT
7
MakePHAT
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@ -29,6 +29,7 @@ COMMON_OBJS = \
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source/psp/input.o \
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source/psp/main.o \
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source/psp/math.o \
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source/psp/sound.o \
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source/psp/system.o \
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source/psp/module.o \
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source/psp/network.o \
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@ -64,7 +65,6 @@ COMMON_OBJS = \
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source/pr_cmds.o \
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source/pr_edict.o \
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source/pr_exec.o \
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source/snd_sdl.o \
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source/snd_dma.o \
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source/snd_mem.o \
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source/snd_mix.o \
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@ -108,14 +108,13 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
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OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
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SDL_LIBS = -lSDL2 -lSDL2main -lGL
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GU_LIBS = -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
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AUDIO_LIBS = -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
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AUDIO_LIBS = -lpspaudiolib -lpspaudio -lpspaudiocodec -lpspmp3 source/psp/m33libs/libpspkubridge.a
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MISC_LIBS = -lpsppower -lpspmath -lpsphprm -ljpeg -lpng -lz
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NET_LIBS = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
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STD_LIBS = -lstdc++ -lm -lc
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LIBS = $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
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LIBS = $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
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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
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CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing
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7
MakeSLIM
7
MakeSLIM
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@ -29,6 +29,7 @@ COMMON_OBJS = \
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source/psp/input.o \
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source/psp/main.o \
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source/psp/math.o \
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source/psp/sound.o \
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source/psp/system.o \
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source/psp/module.o \
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source/psp/network.o \
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@ -64,7 +65,6 @@ COMMON_OBJS = \
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source/pr_cmds.o \
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source/pr_edict.o \
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source/pr_exec.o \
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source/snd_sdl.o \
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source/snd_dma.o \
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source/snd_mem.o \
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source/snd_mix.o \
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@ -108,14 +108,13 @@ HARDWARE_VIDEO_ONLY_FLAGS = -DPSP_HARDWARE_VIDEO
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OBJS = $(COMMON_OBJS) $(HARDWARE_VIDEO_ONLY_OBJS)
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SDL_LIBS = -lSDL2 -lSDL2main -lGL
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GU_LIBS = -lpspgum_vfpu -lpspvfpu -lpspgu -lpspvram
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AUDIO_LIBS = -lpspaudio -lpspmp3 source/psp/m33libs/libpspaudiocodec.a source/psp/m33libs/libpspkubridge.a
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AUDIO_LIBS = -lpspaudiolib -lpspaudio -lpspaudiocodec -lpspmp3 source/psp/m33libs/libpspkubridge.a
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MISC_LIBS = -lpsppower -lpspmath -lpsphprm -ljpeg -lpng -lz
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NET_LIBS = -lpspwlan -lpspnet_adhoc -lpspnet_adhocctl
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STD_LIBS = -lstdc++ -lm -lc
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LIBS = $(GPROF_LIBS) $(SDL_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
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LIBS = $(GPROF_LIBS) $(GU_LIBS) $(AUDIO_LIBS) $(MISC_LIBS) $(STD_LIBS) $(NET_LIBS)
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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
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CXXFLAGS = -fno-rtti -Wcast-qual -Wno-write-strings -Wno-sign-compare -Wno-strict-aliasing
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@ -36,9 +36,6 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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#include <pspge.h>
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#include <pspsysevent.h>
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#include <SDL2/SDL.h>
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#include <SDL2/SDL_mixer.h>
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extern "C"
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{
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#include "../quakedef.h"
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@ -534,13 +531,6 @@ int user_main(SceSize argc, void* argp)
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// Initialize the Common module.
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InitExtModules ();
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// Initialize SDL
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if (SDL_Init(SDL_INIT_AUDIO) < 0)
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{
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Sys_Error("SDL2: Could not initialize!\n");
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return 0;
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}
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ramClockSpeed = cpuClockSpeed = scePowerGetCpuClockFrequencyInt();
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busClockSpeed = scePowerGetBusClockFrequencyInt();
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@ -23,4 +23,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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#include "module.h"
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// Set up the module info.
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// Set up the module info.
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PSP_MODULE_INFO("nzportable", 0, 1, 1);
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PSP_MAIN_THREAD_ATTR(PSP_THREAD_ATTR_USER | PSP_THREAD_ATTR_VFPU);
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PSP_HEAP_SIZE_KB(-1024);
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@ -0,0 +1,183 @@
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/*
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Copyright (C) 1996-1997 Id Software, Inc.
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Copyright (C) 2007 Peter Mackay and Chris Swindle.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License
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as published by the Free Software Foundation; either version 2
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of the License, or (at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include <pspaudiolib.h>
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#include <pspdebug.h>
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#include <pspkernel.h>
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extern "C"
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{
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#include "../quakedef.h"
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}
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namespace quake
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{
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namespace sound
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{
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struct Sample
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{
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short left;
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short right;
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};
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static const unsigned int channelCount = 2;
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static const unsigned int inputBufferSize = 16384;
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#if 1 //def NORMAL_MODE
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static const unsigned int inputFrequency = 11025;
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#else
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static const unsigned int inputFrequency = 22050;
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#endif
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static const unsigned int outputFrequency = 44100;
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static const unsigned int inputSamplesPerOutputSample = outputFrequency / inputFrequency;
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static Sample inputBuffer[inputBufferSize];
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static volatile unsigned int samplesRead;
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static inline void copySamples(const Sample* first, const Sample* last, Sample* destination)
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{
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switch (inputSamplesPerOutputSample)
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{
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case 1:
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memcpy(destination, first, (last - first) * sizeof(Sample));
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break;
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case 2:
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for (const Sample* source = first; source != last; ++source)
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{
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const Sample sample = *source;
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*destination++ = sample;
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*destination++ = sample;
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}
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break;
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case 4:
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for (const Sample* source = first; source != last; ++source)
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{
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const Sample sample = *source;
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*destination++ = sample;
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*destination++ = sample;
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*destination++ = sample;
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*destination++ = sample;
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}
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break;
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default:
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break;
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}
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}
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static void fillOutputBuffer(void* buffer, unsigned int samplesToWrite, void* userData)
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{
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// Where are we writing to?
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Sample* const destination = static_cast<Sample*> (buffer);
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// Where are we reading from?
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const Sample* const firstSampleToRead = &inputBuffer[samplesRead];
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// How many samples to read?
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const unsigned int samplesToRead = samplesToWrite / inputSamplesPerOutputSample;
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// Going to wrap past the end of the input buffer?
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const unsigned int samplesBeforeEndOfInput = inputBufferSize - samplesRead;
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if (samplesToRead > samplesBeforeEndOfInput)
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{
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// Yes, so write the first chunk from the end of the input buffer.
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copySamples(
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firstSampleToRead,
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firstSampleToRead + samplesBeforeEndOfInput,
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&destination[0]);
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// Write the second chunk from the start of the input buffer.
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const unsigned int samplesToReadFromBeginning = samplesToRead - samplesBeforeEndOfInput;
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copySamples(
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&inputBuffer[0],
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&inputBuffer[samplesToReadFromBeginning],
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&destination[samplesBeforeEndOfInput * inputSamplesPerOutputSample]);
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}
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else
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{
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// No wrapping, just copy.
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copySamples(
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firstSampleToRead,
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firstSampleToRead + samplesToRead,
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&destination[0]);
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}
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// Update the read offset.
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samplesRead = (samplesRead + samplesToRead) % inputBufferSize;
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}
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}
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}
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using namespace quake;
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using namespace quake::sound;
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qboolean SNDDMA_Init(void)
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{
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// Set up Quake's audio.
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shm = &sn;
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shm->channels = channelCount;
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shm->samplebits = 16;
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shm->speed = inputFrequency;
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shm->soundalive = qtrue;
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shm->splitbuffer = qfalse;
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shm->samples = inputBufferSize * channelCount;
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shm->samplepos = 0;
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shm->submission_chunk = 1;
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shm->buffer = (unsigned char *) inputBuffer;
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// Initialise the audio system. This initialises it for the CD audio module
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// too.
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pspAudioInit();
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// Set the channel callback.
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// Sound effects use channel 0, CD audio uses channel 1.
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pspAudioSetChannelCallback(0, fillOutputBuffer, 0);
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return qtrue;
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}
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void SNDDMA_Shutdown(void)
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{
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// Clear the mixing buffer so we don't get any noise during cleanup.
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memset(inputBuffer, 0, sizeof(inputBuffer));
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// Clear the channel callback.
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pspAudioSetChannelCallback(0, 0, 0);
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// Stop the audio system?
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pspAudioEndPre();
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// Insert a false delay so the thread can be cleaned up.
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sceKernelDelayThread(50 * 1000);
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// Shut down the audio system.
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pspAudioEnd();
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}
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int SNDDMA_GetDMAPos(void)
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{
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return samplesRead * channelCount;
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}
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void SNDDMA_Submit(void)
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{
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}
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148
source/snd_dma.c
148
source/snd_dma.c
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@ -21,12 +21,16 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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#include "quakedef.h"
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void S_Play(void);
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void S_PlayVol(void);
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void S_SoundList(void);
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void S_Play_f(void);
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void S_PlayVol_f(void);
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void S_SoundList_f(void);
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void S_Update_();
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void S_StopAllSounds(qboolean clear);
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void S_StopAllSoundsC(void);
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void S_StopAllSoundsC_f(void);
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void S_VolumeDown_f (void); // Baker 3.60 - from JoeQuake 0.15
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void S_VolumeUp_f (void); // Baker 3.60 - from JoeQuake 0.15
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// =======================================================================
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// Internal sound data & structures
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@ -47,49 +51,43 @@ vec3_t listener_origin;
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vec3_t listener_forward;
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vec3_t listener_right;
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vec3_t listener_up;
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vec_t sound_nominal_clip_dist=1000.0;
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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)
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int soundtime; // sample PAIRS
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int paintedtime; // sample PAIRS
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#define MAX_SFX 512
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sfx_t *known_sfx; // hunk allocated [MAX_SFX]
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int num_sfx;
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sfx_t *ambient_sfx[NUM_AMBIENTS];
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int desired_speed = 44100; //11025;
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int desired_speed = 11025;
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int desired_bits = 16;
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int sound_started=0;
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cvar_t bgmvolume = {"bgmvolume", "1", true};
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cvar_t bgmtype = {"bgmtype", "cd", true}; // cd or none
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cvar_t volume = {"volume", "0.7", true};
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cvar_t nosound = {"nosound", "0"};
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cvar_t precache = {"precache", "1"};
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cvar_t loadas8bit = {"loadas8bit", "0"};
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cvar_t bgmbuffer = {"bgmbuffer", "4096"};
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cvar_t ambient_level = {"ambient_level", "0.3"};
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cvar_t ambient_level = {"ambient_level", "0.3", true}; // Baker 3.60 - Save to config
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cvar_t ambient_fade = {"ambient_fade", "100"};
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cvar_t snd_noextraupdate = {"snd_noextraupdate", "0"};
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cvar_t snd_show = {"snd_show", "0"};
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cvar_t _snd_mixahead = {"_snd_mixahead", "0.1", true};
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// ====================================================================
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// User-setable variables
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// ====================================================================
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//
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// Fake dma is a synchronous faking of the DMA progress used for
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// isolating performance in the renderer. The fakedma_updates is
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// number of times S_Update() is called per second.
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//
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qboolean fakedma = false;
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int fakedma_updates = 15;
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@ -141,9 +139,7 @@ void S_Startup (void)
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if (!fakedma)
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{
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rc = SNDDMA_Init();
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if (!rc)
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if (!(rc = SNDDMA_Init()))
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{
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Con_Printf("S_Startup: SNDDMA_Init failed.\n");
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sound_started = 0;
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@ -160,22 +156,24 @@ void S_Startup (void)
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S_Init
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================
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*/
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void CDAudioSetVolume (void);
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void S_Init (void)
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{
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Con_Printf("\nSound Initialization\n");
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if (COM_CheckParm("-nosound"))
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return;
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Con_Printf("\nSound Initialization\n");
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if (COM_CheckParm("-simsound"))
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fakedma = true;
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Cmd_AddCommand("play", S_Play);
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Cmd_AddCommand("playvol", S_PlayVol);
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Cmd_AddCommand("stopsound", S_StopAllSoundsC);
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Cmd_AddCommand("soundlist", S_SoundList);
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Cmd_AddCommand("play", S_Play_f);
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Cmd_AddCommand("playvol", S_PlayVol_f);
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Cmd_AddCommand("stopsound", S_StopAllSoundsC_f);
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Cmd_AddCommand("soundlist", S_SoundList_f);
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Cmd_AddCommand("soundinfo", S_SoundInfo_f);
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Cmd_AddCommand ("volumedown", S_VolumeDown_f); // Baker 3.60 - from JoeQuake 0.15
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Cmd_AddCommand ("volumeup", S_VolumeUp_f); // Baker 3.60 - from JoeQuake 0.15
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Cvar_RegisterVariable(&nosound);
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Cvar_RegisterVariable(&volume);
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@ -196,7 +194,6 @@ void S_Init (void)
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}
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snd_initialized = true;
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S_Startup ();
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@ -223,9 +220,7 @@ void S_Init (void)
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shm->buffer = Hunk_AllocName(1<<16, "shmbuf");
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}
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if ( shm ) {
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Con_Printf ("Sound sampling rate: %i\n", shm->speed);
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}
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Con_Printf ("Sound sampling rate: %i Hz\n", shm->speed);
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// provides a tick sound until washed clean
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@ -242,7 +237,6 @@ void S_Init (void)
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void S_Shutdown(void)
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{
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if (!sound_started)
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return;
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@ -253,10 +247,8 @@ void S_Shutdown(void)
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sound_started = 0;
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if (!fakedma)
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{
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SNDDMA_Shutdown();
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}
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}
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// =======================================================================
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@ -277,15 +269,13 @@ sfx_t *S_FindName (char *name)
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if (!name)
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Sys_Error ("S_FindName: NULL\n");
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if (Q_strlen(name) >= MAX_QPATH)
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if (strlen(name) >= MAX_QPATH)
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Sys_Error ("Sound name too long: %s", name);
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// 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");
|
||||
|
|
@ -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;
|
||||
}
|
||||
|
|
@ -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;
|
||||
|
|
@ -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)
|
||||
if (!(sc = S_LoadSound (sfx)))
|
||||
{
|
||||
target_chan->sfx = NULL;
|
||||
return; // couldn't load the sound's data
|
||||
|
|
@ -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
|
||||
&& channels[i].entchannel == entchannel)
|
||||
if (channels[i].entnum == entnum && 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);
|
||||
}
|
||||
|
|
@ -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)
|
||||
if (!(sc = S_LoadSound (sfx)))
|
||||
return;
|
||||
|
||||
if (sc->loopstart == -1)
|
||||
|
|
@ -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;
|
||||
|
|
@ -769,13 +753,13 @@ void GetSoundtime(void)
|
|||
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
|
||||
|
|
@ -796,13 +780,12 @@ 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;
|
||||
|
||||
|
|
@ -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*/
|
||||
}
|
||||
|
||||
/*
|
||||
|
|
@ -839,7 +823,7 @@ console functions
|
|||
===============================================================================
|
||||
*/
|
||||
|
||||
void S_Play(void)
|
||||
void S_Play_f(void)
|
||||
{
|
||||
static int hash=345;
|
||||
int i;
|
||||
|
|
@ -849,20 +833,20 @@ void S_Play(void)
|
|||
i = 1;
|
||||
while (i<Cmd_Argc())
|
||||
{
|
||||
if (!Q_strrchr(Cmd_Argv(i), '.'))
|
||||
if (!strrchr(Cmd_Argv(i), '.'))
|
||||
{
|
||||
Q_strcpy(name, Cmd_Argv(i));
|
||||
Q_strcat(name, ".wav");
|
||||
strcpy(name, Cmd_Argv(i));
|
||||
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;
|
||||
|
|
@ -870,24 +854,30 @@ void S_PlayVol(void)
|
|||
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));
|
||||
Q_strcat(name, ".wav");
|
||||
strcpy(name, Cmd_Argv(i));
|
||||
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)
|
||||
if (!(sc = Cache_Check (&sfx->cache)))
|
||||
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)
|
||||
{
|
||||
|
|
@ -921,8 +929,7 @@ void S_LocalSound (char *sound)
|
|||
if (!sound_started)
|
||||
return;
|
||||
|
||||
sfx = S_PrecacheSound (sound);
|
||||
if (!sfx)
|
||||
if (!(sfx = S_PrecacheSound (sound)))
|
||||
{
|
||||
Con_Printf ("S_LocalSound: can't cache %s\n", sound);
|
||||
return;
|
||||
|
|
@ -944,3 +951,4 @@ void S_BeginPrecaching (void)
|
|||
void S_EndPrecaching (void)
|
||||
{
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
if ((sc = Cache_Check (&s->cache)))
|
||||
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)
|
||||
if (!(data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf))))
|
||||
{
|
||||
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;
|
||||
|
|
|
|||
|
|
@ -34,8 +34,7 @@ void Snd_WriteLinearBlastStereo16 (void);
|
|||
#if !id386
|
||||
void Snd_WriteLinearBlastStereo16 (void)
|
||||
{
|
||||
int i;
|
||||
int val;
|
||||
int i, 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 lpaintedtime;
|
||||
int lpos, 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 count;
|
||||
int out_mask;
|
||||
int *p;
|
||||
int step;
|
||||
int val;
|
||||
int snd_vol;
|
||||
int out_idx, count, out_mask, *p, step, val, 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 end;
|
||||
int i, end, ltime, count;
|
||||
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)
|
||||
if (!(sc = S_LoadSound (ch->sfx)))
|
||||
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 *lscale, *rscale;
|
||||
int i, data, *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 left, right;
|
||||
int leftvol, rightvol;
|
||||
int data, left, right, leftvol, rightvol;
|
||||
signed short *sfx;
|
||||
int i;
|
||||
|
||||
|
|
|
|||
102
source/snd_sdl.c
102
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;
|
||||
}
|
||||
}
|
||||
|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in New Issue