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quakespasm
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Merge branch 'master' into autosave

This commit is contained in:
Eric Wasylishen 2011-01-28 18:27:28 -07:00
parent 93881c3a45 13a6d072f8
commit b139cf7d86
13 changed files with 499 additions and 300 deletions
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267
MacOSX/QuakeSpasm.xcodeproj/project.pbxproj
View File

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66A5563A12E6D46D00FFA7D5 /* bitrate.c */,
66A5563B12E6D46D00FFA7D5 /* bitrate.h */,
66A5563C12E6D46D00FFA7D5 /* bitwise.c */,
66A5563D12E6D46D00FFA7D5 /* block.c */,
66A5563E12E6D46D00FFA7D5 /* books */,
66A5564612E6D46D00FFA7D5 /* codebook.c */,
66A5564712E6D46D00FFA7D5 /* codebook.h */,
66A5564812E6D46D00FFA7D5 /* codec_internal.h */,
66A5564912E6D46D00FFA7D5 /* COPYING */,
66A5564A12E6D46D00FFA7D5 /* envelope.c */,
66A5564B12E6D46D00FFA7D5 /* envelope.h */,
66A5564C12E6D46D00FFA7D5 /* floor0.c */,
66A5564D12E6D46D00FFA7D5 /* floor1.c */,
66A5564E12E6D46D00FFA7D5 /* framing.c */,
66A5564F12E6D46D00FFA7D5 /* highlevel.h */,
66A5565012E6D46D00FFA7D5 /* info.c */,
66A5565112E6D46D00FFA7D5 /* lookup.c */,
66A5565212E6D46D00FFA7D5 /* lookup.h */,
66A5565312E6D46D00FFA7D5 /* lookup_data.h */,
66A5565412E6D46D00FFA7D5 /* lpc.c */,
66A5565512E6D46D00FFA7D5 /* lpc.h */,
66A5565612E6D46D00FFA7D5 /* lsp.c */,
66A5565712E6D46D00FFA7D5 /* lsp.h */,
66A5565812E6D46D00FFA7D5 /* mapping0.c */,
66A5565912E6D46D00FFA7D5 /* masking.h */,
66A5565A12E6D46D00FFA7D5 /* mdct.c */,
66A5565B12E6D46D00FFA7D5 /* mdct.h */,
66A5565C12E6D46D00FFA7D5 /* misc.h */,
66A5565D12E6D46D00FFA7D5 /* modes */,
66A5567112E6D46D00FFA7D5 /* ogg */,
66A5567412E6D46D00FFA7D5 /* os.h */,
66A5567512E6D46D00FFA7D5 /* psy.c */,
66A5567612E6D46D00FFA7D5 /* psy.h */,
66A5567712E6D46D00FFA7D5 /* README */,
66A5567812E6D46D00FFA7D5 /* registry.c */,
66A5567912E6D46D00FFA7D5 /* registry.h */,
66A5567A12E6D46D00FFA7D5 /* res0.c */,
66A5567B12E6D46D00FFA7D5 /* scales.h */,
66A5567C12E6D46D00FFA7D5 /* sharedbook.c */,
66A5567D12E6D46D00FFA7D5 /* smallft.c */,
66A5567E12E6D46D00FFA7D5 /* smallft.h */,
66A5567F12E6D46D00FFA7D5 /* synthesis.c */,
66A5568012E6D46D00FFA7D5 /* vorbis */,
66A5568412E6D46D00FFA7D5 /* vorbisfile.c */,
66A5568512E6D46D00FFA7D5 /* window.c */,
66A5568612E6D46D00FFA7D5 /* window.h */,
);
name = OggVorbis;
path = ../OggVorbis;
sourceTree = SOURCE_ROOT;
};
66A5563E12E6D46D00FFA7D5 /* books */ = {
isa = PBXGroup;
children = (
66A5563F12E6D46D00FFA7D5 /* coupled */,
66A5564212E6D46D00FFA7D5 /* floor */,
66A5564412E6D46D00FFA7D5 /* uncoupled */,
);
path = books;
sourceTree = "<group>";
};
66A5563F12E6D46D00FFA7D5 /* coupled */ = {
isa = PBXGroup;
children = (
66A5564012E6D46D00FFA7D5 /* res_books_51.h */,
66A5564112E6D46D00FFA7D5 /* res_books_stereo.h */,
);
path = coupled;
sourceTree = "<group>";
};
66A5564212E6D46D00FFA7D5 /* floor */ = {
isa = PBXGroup;
children = (
66A5564312E6D46D00FFA7D5 /* floor_books.h */,
);
path = floor;
sourceTree = "<group>";
};
66A5564412E6D46D00FFA7D5 /* uncoupled */ = {
isa = PBXGroup;
children = (
66A5564512E6D46D00FFA7D5 /* res_books_uncoupled.h */,
);
path = uncoupled;
sourceTree = "<group>";
};
66A5565D12E6D46D00FFA7D5 /* modes */ = {
isa = PBXGroup;
children = (
66A5565E12E6D46D00FFA7D5 /* floor_all.h */,
66A5565F12E6D46D00FFA7D5 /* psych_11.h */,
66A5566012E6D46D00FFA7D5 /* psych_16.h */,
66A5566112E6D46D00FFA7D5 /* psych_44.h */,
66A5566212E6D46D00FFA7D5 /* psych_8.h */,
66A5566312E6D46D00FFA7D5 /* residue_16.h */,
66A5566412E6D46D00FFA7D5 /* residue_44.h */,
66A5566512E6D46D00FFA7D5 /* residue_44p51.h */,
66A5566612E6D46D00FFA7D5 /* residue_44u.h */,
66A5566712E6D46D00FFA7D5 /* residue_8.h */,
66A5566812E6D46D00FFA7D5 /* setup_11.h */,
66A5566912E6D46D00FFA7D5 /* setup_16.h */,
66A5566A12E6D46D00FFA7D5 /* setup_22.h */,
66A5566B12E6D46D00FFA7D5 /* setup_32.h */,
66A5566C12E6D46D00FFA7D5 /* setup_44.h */,
66A5566D12E6D46D00FFA7D5 /* setup_44p51.h */,
66A5566E12E6D46D00FFA7D5 /* setup_44u.h */,
66A5566F12E6D46D00FFA7D5 /* setup_8.h */,
66A5567012E6D46D00FFA7D5 /* setup_X.h */,
);
path = modes;
sourceTree = "<group>";
};
66A5567112E6D46D00FFA7D5 /* ogg */ = {
isa = PBXGroup;
children = (
66A5567212E6D46D00FFA7D5 /* ogg.h */,
66A5567312E6D46D00FFA7D5 /* os_types.h */,
);
path = ogg;
sourceTree = "<group>";
};
66A5568012E6D46D00FFA7D5 /* vorbis */ = {
isa = PBXGroup;
children = (
66A5568112E6D46D00FFA7D5 /* codec.h */,
66A5568212E6D46D00FFA7D5 /* vorbisenc.h */,
66A5568312E6D46D00FFA7D5 /* vorbisfile.h */,
);
path = vorbis;
sourceTree = "<group>";
};
/* End PBXGroup section */
/* Begin PBXNativeTarget section */
@ -674,6 +900,8 @@
48B9E7880D340B1E0001CACF /* Launcher.nib in Resources */,
484AA4B40D3FF6C0005D917A /* QuakeSpasm.icns in Resources */,
48FA25B712437EAB00667668 /* dsa_pub.pem in Resources */,
66A5568C12E6D46D00FFA7D5 /* COPYING in Resources */,
66A5569812E6D46D00FFA7D5 /* README in Resources */,
);
runOnlyForDeploymentPostprocessing = 0;
};
@ -776,6 +1004,29 @@
66A5470F12E3CF8100FFA7D5 /* snd_codec_ogg.c in Sources */,
66A54A4F12E429B800FFA7D5 /* snd_resample.c in Sources */,
66A54ECE12E4ECA800FFA7D5 /* resample.c in Sources */,
66A5568712E6D46D00FFA7D5 /* analysis.c in Sources */,
66A5568812E6D46D00FFA7D5 /* bitrate.c in Sources */,
66A5568912E6D46D00FFA7D5 /* bitwise.c in Sources */,
66A5568A12E6D46D00FFA7D5 /* block.c in Sources */,
66A5568B12E6D46D00FFA7D5 /* codebook.c in Sources */,
66A5568D12E6D46D00FFA7D5 /* envelope.c in Sources */,
66A5568E12E6D46D00FFA7D5 /* floor0.c in Sources */,
66A5568F12E6D46D00FFA7D5 /* floor1.c in Sources */,
66A5569012E6D46D00FFA7D5 /* framing.c in Sources */,
66A5569112E6D46D00FFA7D5 /* info.c in Sources */,
66A5569212E6D46D00FFA7D5 /* lookup.c in Sources */,
66A5569312E6D46D00FFA7D5 /* lpc.c in Sources */,
66A5569412E6D46D00FFA7D5 /* lsp.c in Sources */,
66A5569512E6D46D00FFA7D5 /* mapping0.c in Sources */,
66A5569612E6D46D00FFA7D5 /* mdct.c in Sources */,
66A5569712E6D46D00FFA7D5 /* psy.c in Sources */,
66A5569912E6D46D00FFA7D5 /* registry.c in Sources */,
66A5569A12E6D46D00FFA7D5 /* res0.c in Sources */,
66A5569B12E6D46D00FFA7D5 /* sharedbook.c in Sources */,
66A5569C12E6D46D00FFA7D5 /* smallft.c in Sources */,
66A5569D12E6D46D00FFA7D5 /* synthesis.c in Sources */,
66A5569E12E6D46D00FFA7D5 /* vorbisfile.c in Sources */,
66A5569F12E6D46D00FFA7D5 /* window.c in Sources */,
);
runOnlyForDeploymentPostprocessing = 0;
};

4
Quake/Makefile
View File

@ -132,7 +132,8 @@ DEFAULT_TARGET := quakespasm
# objects
# ----------------------------------------------------------------------------
COMOBJ_SND := snd_dma.o snd_mix.o snd_mem.o
OBJ_OGGVORBIS := $(patsubst %.c,%.o,$(wildcard ../OggVorbis/*.c))
COMOBJ_SND := cd_shared.o snd_dma.o snd_mix.o snd_mem.o snd_resample.o resample.o snd_codec.o snd_codec_wav.o snd_codec_ogg.o
SYSOBJ_SND := snd_sdl.o
SYSOBJ_CDA := cd_sdl.o
SYSOBJ_INPUT := in_sdl.o
@ -168,6 +169,7 @@ GLOBJS = \
gl_model.o
OBJS := \
$(OBJ_OGGVORBIS) \
$(GLOBJS) \
$(SYSOBJ_INPUT) \
$(COMOBJ_SND) \

102
Quake/cd_shared.c
View File

@ -2,6 +2,7 @@
cd_shared.c
Copyright (C) 1996-1997 Id Software, Inc.
Copyright (C) 2011 Eric Wasylishen
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
@ -46,19 +47,54 @@ static qboolean CDAudio_IsNumberedTrack(const char *trackName)
return true;
}
void CDAudio_NewMap ()
{
// hack copied from gl_fog.c:Fog_ParseWorldspawn()
char key[128], value[4096];
const char *data;
data = COM_Parse(cl.worldmodel->entities);
if (!data)
return; // error
if (com_token[0] != '{')
return; // error
while (1)
{
data = COM_Parse(data);
if (!data)
return; // error
if (com_token[0] == '}')
break; // end of worldspawn
if (com_token[0] == '_')
strcpy(key, com_token + 1);
else
strcpy(key, com_token);
while (key[strlen(key)-1] == ' ') // remove trailing spaces
key[strlen(key)-1] = 0;
data = COM_Parse(data);
if (!data)
return; // error
strcpy(value, com_token);
if (!strcmp("sounds", key))
{
if (!CDAudio_IsNumberedTrack(value))
{
CDAudio_PlayNamed(value, true);
}
}
}
}
static void CDAudio_FinishedCallback(void *userdata)
{
CDAudio_Next();
}
static qboolean CDAudio_TryPlayNamed(const char *name, qboolean looping)
{
qboolean success = S_Base_StartBackgroundTrack(name, looping, CDAudio_FinishedCallback, NULL);
return success;
}
void CDAudio_PlayNamed(const char *name, qboolean looping)
{
char filename[MAX_QPATH];
if (!enabled)
return;
@ -80,35 +116,9 @@ void CDAudio_PlayNamed(const char *name, qboolean looping)
track = remap[track];
}
q_snprintf(playTrackName, sizeof(playTrackName), "%03d", track);
}
else
{
q_snprintf(playTrackName, sizeof(playTrackName), "%s", name);
}
// First try to play a music file (path code from Darkplaces)
char filename[MAX_QPATH];
if (track > 0) // We were given a numbered track
{
q_snprintf(filename, sizeof(filename), "sound/cdtracks/track%03u", track);
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "sound/cdtracks/track%02u", track);
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "music/cdtracks/track%03u", track);// added by motorsep
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "music/cdtracks/track%02u", track);// added by motorsep
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "music/track%03u", track);// added by motorsep
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "music/track%02u", track);// added by motorsep
if (CDAudio_TryPlayNamed(filename, looping)) return;
if (S_Base_StartBackgroundTrack(filename, looping, CDAudio_FinishedCallback, NULL)) return;
// No music file, so try using the hardware CD player
@ -117,27 +127,21 @@ void CDAudio_PlayNamed(const char *name, qboolean looping)
{
usingBackend = true;
}
else
{
Con_Printf( "WARNING: Unable to play music track %d\n", track );
}
return;
}
// We were given a named track
q_snprintf(filename, sizeof(filename), "%s", playTrackName);
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "sound/%s", playTrackName);
if (CDAudio_TryPlayNamed(filename, looping)) return;
else
{
q_snprintf(playTrackName, sizeof(playTrackName), "%s", name);
q_snprintf(filename, sizeof(filename), "sound/cdtracks/%s", playTrackName);
if (CDAudio_TryPlayNamed(filename, looping)) return;
if (S_Base_StartBackgroundTrack(filename, looping, CDAudio_FinishedCallback, NULL)) return;
q_snprintf(filename, sizeof(filename), "music/%s", playTrackName);
if (CDAudio_TryPlayNamed(filename, looping)) return;
q_snprintf(filename, sizeof(filename), "music/cdtracks/%s", playTrackName);
if (CDAudio_TryPlayNamed(filename, looping)) return;
Con_Printf("WARNING: Couldn't find music track \"%s\"\n", playTrackName);
Con_Printf("WARNING: Unable to play music track \"%s\"\n", filename);
}
}
void CDAudio_Play(byte track, qboolean looping)

1
Quake/cdaudio.h
View File

@ -22,6 +22,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#ifndef __CDAUDIO_H
#define __CDAUDIO_H
void CDAudio_NewMap ();
int CDAudio_Init(void);
void CDAudio_PlayNamed(const char *name, qboolean looping);
void CDAudio_Play(byte track, qboolean looping);

1
Quake/cl_parse.c
View File

@ -373,6 +373,7 @@ void CL_ParseServerInfo (void)
cl_entities[0].model = cl.worldmodel = cl.model_precache[1];
R_NewMap ();
CDAudio_NewMap ();
//johnfitz -- clear out string; we don't consider identical
//messages to be duplicates if the map has changed in between

4
Quake/snd_codec.c
View File

@ -110,9 +110,7 @@ void S_CodecInit()
{
codecs = NULL;
S_CodecRegister(&wav_codec);
#ifdef USE_CODEC_VORBIS
S_CodecRegister(&ogg_codec);
#endif
}
/*
@ -226,6 +224,8 @@ snd_stream_t *S_CodecUtilOpen(const char *filename, snd_codec_t *codec)
stream->codec = codec;
stream->file = hnd;
stream->length = length;
stream->startpos = Sys_FileTell(hnd);
return stream;
}

3
Quake/snd_codec.h
View File

@ -44,6 +44,7 @@ typedef struct snd_stream_s
snd_info_t info;
int length;
int pos;
int startpos;
void *ptr;
} snd_stream_t;
@ -85,12 +86,10 @@ void S_WAV_CodecCloseStream(snd_stream_t *stream);
int S_WAV_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
// Ogg Vorbis codec
#ifdef USE_CODEC_VORBIS
extern snd_codec_t ogg_codec;
void *S_OGG_CodecLoad(const char *filename, snd_info_t *info);
snd_stream_t *S_OGG_CodecOpenStream(const char *filename);
void S_OGG_CodecCloseStream(snd_stream_t *stream);
int S_OGG_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
#endif // USE_CODEC_VORBIS
#endif // !_SND_CODEC_H_

14
Quake/snd_codec_ogg.c
View File

@ -23,9 +23,6 @@ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
// OGG support is enabled by this define
#ifdef USE_CODEC_VORBIS
// includes for the Q3 sound system
#include "quakedef.h"
#include "snd_codec.h"
@ -127,7 +124,7 @@ int S_OGG_Callback_seek(void *datasource, ogg_int64_t offset, int whence)
{
// set the file position in the actual file with the Q3 function
errno = 0;
Sys_FileSeek(stream->file, (long) offset);
Sys_FileSeek(stream->file, (long) stream->startpos + (long) offset);
// something has gone wrong, so we return here
if(errno != 0)
@ -144,7 +141,7 @@ int S_OGG_Callback_seek(void *datasource, ogg_int64_t offset, int whence)
{
// set the file position in the actual file with the Q3 function
errno = 0;
Sys_FileSeek(stream->file, (long) stream->pos + (long) offset);
Sys_FileSeek(stream->file, (long) stream->startpos + (long) stream->pos + (long) offset);
// something has gone wrong, so we return here
if(errno != 0)
@ -164,7 +161,7 @@ int S_OGG_Callback_seek(void *datasource, ogg_int64_t offset, int whence)
// set the file position in the actual file with the Q3 function
errno = 0;
Sys_FileSeek(stream->file, (long) stream->length + (long) offset);
Sys_FileSeek(stream->file, (long) stream->startpos + (long) stream->length + (long) offset);
// something has gone wrong, so we return here
if(errno != 0)
@ -214,7 +211,7 @@ long S_OGG_Callback_tell(void *datasource)
// snd_stream_t in the generic pointer
stream = (snd_stream_t *) datasource;
return (long) Sys_FileTell(stream->file);
return (long) Sys_FileTell(stream->file) - (long) stream->startpos;
}
// the callback structure
@ -446,6 +443,7 @@ void *S_OGG_CodecLoad(const char *filename, snd_info_t *info)
info->samples = stream->info.samples;
info->size = stream->info.size;
info->dataofs = stream->info.dataofs;
info->loopstart = -1;
// allocate a buffer
// this buffer must be free-ed by the caller of this function
@ -473,5 +471,3 @@ void *S_OGG_CodecLoad(const char *filename, snd_info_t *info)
return buffer;
}
#endif // USE_CODEC_VORBIS

136
Quake/snd_dma.c
View File

@ -71,7 +71,7 @@ qboolean sound_started = false;
int s_rawend[MAX_RAW_STREAMS];
portable_samplepair_t s_rawsamples[MAX_RAW_STREAMS][MAX_RAW_SAMPLES];
static void *s_rawresampler[MAX_RAW_STREAMS];
cvar_t bgmvolume = {"bgmvolume", "1", true};
cvar_t sfxvolume = {"volume", "0.7", true};
@ -182,8 +182,6 @@ void S_Init (void)
Con_Printf ("loading all sounds as 8bit\n");
}
SND_InitScaletable ();
known_sfx = (sfx_t *) Hunk_AllocName (MAX_SFX*sizeof(sfx_t), "sfx_t");
num_sfx = 0;
@ -221,6 +219,13 @@ void S_Shutdown (void)
SNDDMA_Shutdown();
shm = NULL;
int i;
for (i = 0; i<MAX_RAW_STREAMS; i++)
{
if (s_rawresampler[i] != NULL)
Snd_ResamplerClose(s_rawresampler[i]);
}
}
@ -581,31 +586,6 @@ void S_StaticSound (sfx_t *sfx, vec3_t origin, float vol, float attenuation)
//=============================================================================
/*
=================
S_ByteSwapRawSamples
If raw data has been loaded in little endien binary form, this must be done.
If raw data was calculated, as with ADPCM, this should not be called.
=================
*/
void S_ByteSwapRawSamples( int samples, int width, int s_channels, const byte *data ) {
int i;
if ( width != 2 ) {
return;
}
if ( LittleShort( 256 ) == 256 ) {
return;
}
if ( s_channels == 2 ) {
samples <<= 1;
}
for ( i = 0 ; i < samples ; i++ ) {
((short *)data)[i] = LittleShort( ((short *)data)[i] );
}
}
/*
============
@ -616,10 +596,11 @@ void S_ByteSwapRawSamples( int samples, int width, int s_channels, const byte *d
*/
void S_Base_RawSamples( int stream, int samples, int rate, int width, int s_channels, const byte *data, float volume ) {
int i;
int src, dst;
float scale;
int dst;
int intVolume;
portable_samplepair_t *rawsamples;
void *resampler;
int resampledNumSamples;
if ( !sound_started ) {
return;
@ -630,6 +611,13 @@ void S_Base_RawSamples( int stream, int samples, int rate, int width, int s_chan
}
rawsamples = s_rawsamples[stream];
// Set up resampler
if (s_rawresampler[stream] == NULL)
{
s_rawresampler[stream] = Snd_ResamplerInit();
}
resampler = s_rawresampler[stream];
intVolume = 256 * volume;
if ( s_rawend[stream] < soundtime ) {
@ -637,78 +625,39 @@ void S_Base_RawSamples( int stream, int samples, int rate, int width, int s_chan
s_rawend[stream] = soundtime;
}
scale = (float)rate / shm->speed;
void *resampled = Snd_Resample(resampler,
rate, width, samples, s_channels, data,
shm->speed, 2, &resampledNumSamples);
// old:
//Con_Printf ("%i < %i < %i\n", soundtime, s_paintedtime, s_rawend[stream]);
if (s_channels == 2 && width == 2)
if (s_channels == 2)
{
if (scale == 1.0)
{ // optimized case
for (i=0 ; i<samples ; i++)
for (i=0 ; i<resampledNumSamples ; i++)
{
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = ((short *)data)[i*2] * intVolume;
rawsamples[dst].right = ((short *)data)[i*2+1] * intVolume;
rawsamples[dst].left = ((short *)resampled)[i*2] * intVolume;
rawsamples[dst].right = ((short *)resampled)[i*2+1] * intVolume;
}
}
else if (s_channels == 1)
{
for (i=0 ; i<resampledNumSamples; i++)
{
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = ((short *)resampled)[i] * intVolume;
rawsamples[dst].right = ((short *)resampled)[i] * intVolume;
}
}
else
{
for (i=0 ; ; i++)
{
src = i*scale;
if (src >= samples)
break;
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = ((short *)data)[src*2] * intVolume;
rawsamples[dst].right = ((short *)data)[src*2+1] * intVolume;
Con_Printf( "S_Base_RawSamples: unsupported number of channels %d\n", s_channels );
}
}
}
else if (s_channels == 1 && width == 2)
{
for (i=0 ; ; i++)
{
src = i*scale;
if (src >= samples)
break;
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = ((short *)data)[src] * intVolume;
rawsamples[dst].right = ((short *)data)[src] * intVolume;
}
}
else if (s_channels == 2 && width == 1)
{
intVolume *= 256;
for (i=0 ; ; i++)
{
src = i*scale;
if (src >= samples)
break;
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = ((char *)data)[src*2] * intVolume;
rawsamples[dst].right = ((char *)data)[src*2+1] * intVolume;
}
}
else if (s_channels == 1 && width == 1)
{
intVolume *= 256;
for (i=0 ; ; i++)
{
src = i*scale;
if (src >= samples)
break;
dst = s_rawend[stream]&(MAX_RAW_SAMPLES-1);
s_rawend[stream]++;
rawsamples[dst].left = (((byte *)data)[src]-128) * intVolume;
rawsamples[dst].right = (((byte *)data)[src]-128) * intVolume;
}
}
if ( s_rawend[stream] > soundtime + MAX_RAW_SAMPLES ) {
Con_DPrintf( "S_Base_RawSamples: overflowed %i > %i\n", s_rawend[stream], soundtime );
@ -956,6 +905,7 @@ void S_Update_(void)
===============================================================================
background music functions
from ioquake3
===============================================================================
*/
@ -1198,12 +1148,6 @@ void S_Play (void)
i = 1;
while (i < Cmd_Argc())
{
if (!Q_strrchr(Cmd_Argv(i), '.'))
{
Q_strcpy(name, Cmd_Argv(i));
Q_strcat(name, ".wav");
}
else
Q_strcpy(name, Cmd_Argv(i));
sfx = S_PrecacheSound(name);
S_StartSound(hash++, 0, sfx, listener_origin, 1.0, 1.0);

22
Quake/snd_mem.c
View File

@ -23,6 +23,8 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#include "quakedef.h"
#include "snd_codec.h"
static void *resampler = NULL;
/*
==============
S_LoadSound
@ -62,17 +64,25 @@ sfxcache_t *S_LoadSound (sfx_t *s)
return NULL;
}
int resampledNumSamples;
void *resampled = Snd_Resample(info.rate, info.width, info.samples, info.channels, data, shm->speed, shm->samplebits/8, &resampledNumSamples);
if (info.rate == shm->speed && info.width == shm->samplebits/8)
// set up the resampler
if (resampler == NULL)
{
if (resampledNumSamples != info.samples) exit(5);
if (0 != memcmp(data, resampled, (info.width * info.samples * info.channels))) exit(6);
resampler = Snd_ResamplerInit();
}
else
{
Snd_ResamplerReset(resampler);
}
int resampledNumSamples;
void *resampled = Snd_Resample(resampler,
info.rate, info.width, info.samples, info.channels, data,
shm->speed, shm->samplebits/8, &resampledNumSamples);
len = resampledNumSamples * (shm->samplebits/8) * info.channels;
// set up the sample struct
sc = (sfxcache_t *) Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
if (!sc)
return NULL;

39
Quake/snd_mix.c
View File

@ -237,29 +237,10 @@ void S_PaintChannels (int endtime)
}
}
void SND_InitScaletable (void)
{
int i, j;
for (i = 0; i < 32; i++)
{
for (j = 0; j < 256; j++)
/* When compiling with gcc-4.1.0 at optimisations O1 and
higher, the tricky signed char type conversion is not
guaranteed. Therefore we explicity calculate the signed
value from the index as required. From Kevin Shanahan.
See: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=26719
*/
// snd_scaletable[i][j] = ((signed char)j) * i * 8;
snd_scaletable[i][j] = ((j < 128) ? j : j - 0xff) * i * 8;
}
}
void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count)
{
int data;
int *lscale, *rscale;
unsigned char *sfx;
int i;
@ -268,15 +249,13 @@ void SND_PaintChannelFrom8 (channel_t *ch, sfxcache_t *sc, int count)
if (ch->rightvol > 255)
ch->rightvol = 255;
lscale = snd_scaletable[ch->leftvol >> 3];
rscale = snd_scaletable[ch->rightvol >> 3];
sfx = (unsigned char *) ((signed char *)sc->data + ch->pos);
sfx = (unsigned char *) ((unsigned char *)sc->data + ch->pos);
for (i = 0; i < count; i++)
{
data = sfx[i];
paintbuffer[i].left += lscale[data] * snd_vol;
paintbuffer[i].right += rscale[data] * snd_vol;
data = (sfx[i] - 128) * snd_vol;
paintbuffer[i].left += data * ch->leftvol;
paintbuffer[i].right += data * ch->rightvol;
}
ch->pos += count;
@ -301,21 +280,11 @@ void SND_PaintChannelFrom16 (channel_t *ch, sfxcache_t *sc, int count)
for (i = 0; i < count; i++)
{
if ((ch->pos + i) >= sc->length)
{
Con_Printf("Overran Sample!\n");
}
data = sfx[i];
left = (data * leftvol) >> 8;
right = (data * rightvol) >> 8;
paintbuffer[i].left += left;
paintbuffer[i].right += right;
if (paintbuffer[i].left > (1 << 24))
{
Con_Printf("Clipping!\n");
}
}
ch->pos += count;

159
Quake/snd_resample.c
View File

@ -1,16 +1,72 @@
/*
Copyright (C) 2011 Eric Wasylishen
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 "quakedef.h"
#include "sound.h"
#include "speex_resampler.h"
void *Snd_Resample(int inrate, int inwidth, int innumsamples, int channels, const void *indata,
#define Snd_ResamplerQuality 0
struct resampler {
int channels;
SpeexResamplerState *st;
};
void *Snd_ResamplerInit()
{
struct resampler *data = malloc(sizeof(struct resampler));
data->channels = 1;
data->st = speex_resampler_init(1, 44100, 44100, Snd_ResamplerQuality, NULL);
return data;
}
void Snd_ResamplerClose(void *handle)
{
struct resampler *data = (struct resampler *)handle;
speex_resampler_destroy(data->st);
free(data);
}
void Snd_ResamplerReset(void *handle)
{
struct resampler *data = (struct resampler *)handle;
speex_resampler_reset_mem(data->st);
}
void *Snd_Resample(void *handle,
int inrate, int inwidth, int innumsamples, int channels, const void *indata,
int outrate, int outwidth, int *outnumsamples)
{
// check params
if ( !(inwidth == 1 || inwidth == 2) || !(outwidth == 1 || outwidth == 2) )
{
Sys_Error("Snd_ResampleStream only supports 1 or 2 bytes per sample");
}
const float frac = ((float)inrate) / ((float)outrate);
const int maxsamples = (innumsamples / frac) + 10;
short *outdata = malloc(maxsamples * channels * outwidth);
short *in16bit;
short *out16bit = malloc(maxsamples * channels * 2);
// Convert input to 16-bit if necessary
short *in16bit;
if (inwidth == 2)
{
in16bit = (short*)indata;
@ -22,104 +78,75 @@ void *Snd_Resample(int inrate, int inwidth, int innumsamples, int channels, cons
for (i=0; i<innumsamples; i++)
{
unsigned char sample = ((unsigned char *)indata)[i];
if (sample == 255)
{
//Con_Printf("8-bit clipping\n");
}
in16bit[i] = (((short)sample) - 128) << 8;
}
}
else
{
exit(5);
}
// See if we need to resample
// Call the resampler
if (inrate == outrate)
{
memcpy(outdata, in16bit, innumsamples * channels * 2);
memcpy(out16bit, in16bit, innumsamples * channels * 2);
*outnumsamples = innumsamples;
}
else
{
struct resampler *data = (struct resampler *)handle;
// Call the resampler
static SpeexResamplerState *st = NULL;
if (st == NULL)
unsigned int old_inrate, old_outrate;
speex_resampler_get_rate(data->st, &old_inrate, &old_outrate);
if (data->channels != channels)
{
st = speex_resampler_init(channels, inrate, outrate, 0, NULL);
speex_resampler_destroy(data->st);
data->st = speex_resampler_init(channels, inrate, outrate, Snd_ResamplerQuality, NULL);
data->channels = channels;
}
else
{
speex_resampler_reset_mem(st);
speex_resampler_set_rate(data->st, inrate, outrate);
}
speex_resampler_set_rate(st, inrate, outrate);
*outnumsamples = 0;
unsigned int consumedtotal = 0;
unsigned int outputtotal = 0;
unsigned int loops = 0;
unsigned int consumed, output;
while (consumedtotal < innumsamples)
{
int roomToConsume, roomToOutput;
consumed = innumsamples - consumedtotal;
output = maxsamples - outputtotal;
roomToConsume = consumed;
roomToOutput = output;
speex_resampler_process_interleaved_int(st, in16bit + consumedtotal, &consumed, outdata + outputtotal, &output);
unsigned int consumed = innumsamples - consumedtotal;
unsigned int output = maxsamples - outputtotal;
speex_resampler_process_interleaved_int(data->st, in16bit + consumedtotal, &consumed, out16bit + outputtotal, &output);
consumedtotal += consumed;
outputtotal += output;
loops++;
if (loops > 100)
{
Con_Printf("Infinite loop\n");
}
}
*outnumsamples = outputtotal;
if (*outnumsamples != (innumsamples / frac))
{
Con_Printf("Output %d, predicted %d\n", *outnumsamples, (innumsamples / frac));
}
//speex_resampler_destroy(resampler);
}
// Check for clipping.
{
int i;
for (i=0; i<*outnumsamples; i++)
{
short sample = outdata[i];
if (sample == 32767)
{
//Con_Printf("16-bit clipping\n");
}
}
}
// Prepare to return
if (in16bit != indata)
{
free(in16bit);
}
if(outwidth != 2) exit(5);
void *outdata;
if (outwidth == 2)
{
outdata = out16bit;
}
else // (outputwidth == 1)
{
int i;
int len = (*outnumsamples) * channels;
outdata = malloc(len);
for (i = 0; i<len; i++)
{
int s16sample = out16bit[i];
unsigned char u8sample = ((s16sample + 32768) >> 8);
((unsigned char *)outdata)[i] = u8sample;
}
free(out16bit);
}
return outdata;
}
void *Snd_ResamplerInit(int inrate, int inwidth, int outrate, int outwidth, int channels) { return NULL; }
void Snd_ResamplerClose(void *resampler) {}
void Snd_ResampleStream(void *resampler,
int *innumsamples, void *indata,
int *outnumsamples, void *outdata) {}

25
Quake/sound.h
View File

@ -151,23 +151,21 @@ qboolean S_BackgroundTrackIsPlaying( void );
qboolean S_BackgroundTrackIsPaused( void );
qboolean S_BackgroundTrackIsLooping( void );
/* resamples a whole file. return value must be freed with free() */
void *Snd_Resample(int inrate, int inwidth, int innumsamples, int channels, const void *indata,
int outrate, int outwidth, int *outnumsamples);
/* creates a new stream resampler for the specified rates and number of channels. returns a handle */
void *Snd_ResamplerInit(int inrate, int inwidth, int outrate, int outwidth, int channels);
/* creates a new stream resampler. returns a handle */
void *Snd_ResamplerInit();
/* closes a resampler handle */
void Snd_ResamplerClose(void *resampler);
/* performs resampling on samples in the given channel. numinsamples takes
the number of samples to process, and returns the number actually processed.
*/
void Snd_ResampleStream(void *resampler,
int *innumsamples, void *indata,
int *outnumsamples, void *outdata);
/* prepare resampler for processing an unrelated sound */
void Snd_ResamplerReset(void *resampler);
/* performs resampling on samples in the given channel.
return value must be freed with free()
*/
void *Snd_Resample(void *resampler,
int inrate, int inwidth, int innumsamples, int channels, const void *indata,
int outrate, int outwidth, int *outnumsamples);
// ====================================================================
// User-setable variables
@ -217,9 +215,6 @@ extern int snd_blocked;
void S_LocalSound (const char *s);
sfxcache_t *S_LoadSound (sfx_t *s);
wavinfo_t GetWavinfo (const char *name, byte *wav, int wavlength);
void SND_InitScaletable (void);
void S_AmbientOff (void);
void S_AmbientOn (void);