mirror of
https://github.com/Shpoike/Quakespasm.git
synced 2025-02-15 08:31:03 +00:00
Refactor resampler code. Use Quake 3 wav codec
and delete most of snd_mem.c.
This commit is contained in:
Eric Wasylishen
parent
e10670b8bc
commit
91ff92b33c
6 changed files with 113 additions and 273 deletions
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@ -108,6 +108,7 @@
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66A5470F12E3CF8100FFA7D5 /* snd_codec_ogg.c in Sources */ = {isa = PBXBuildFile; fileRef = 66A5470C12E3CF8100FFA7D5 /* snd_codec_ogg.c */; };
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66A5487A12E3EA6900FFA7D5 /* Ogg.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 66A5487912E3EA6900FFA7D5 /* Ogg.framework */; };
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66A5487C12E3EA6E00FFA7D5 /* Vorbis.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 66A5487B12E3EA6E00FFA7D5 /* Vorbis.framework */; };
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66A54A4F12E429B800FFA7D5 /* snd_resample.c in Sources */ = {isa = PBXBuildFile; fileRef = 66A54A4E12E429B800FFA7D5 /* snd_resample.c */; };
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8D11072B0486CEB800E47090 /* InfoPlist.strings in Resources */ = {isa = PBXBuildFile; fileRef = 089C165CFE840E0CC02AAC07 /* InfoPlist.strings */; };
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8D11072F0486CEB800E47090 /* Cocoa.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 1058C7A1FEA54F0111CA2CBB /* Cocoa.framework */; };
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/* End PBXBuildFile section */
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@ -288,6 +289,7 @@
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66A5470C12E3CF8100FFA7D5 /* snd_codec_ogg.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = snd_codec_ogg.c; path = ../Quake/snd_codec_ogg.c; sourceTree = SOURCE_ROOT; };
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66A5487912E3EA6900FFA7D5 /* Ogg.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Ogg.framework; path = Library/Frameworks/Ogg.framework; sourceTree = SDKROOT; };
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66A5487B12E3EA6E00FFA7D5 /* Vorbis.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = Vorbis.framework; path = Library/Frameworks/Vorbis.framework; sourceTree = SDKROOT; };
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66A54A4E12E429B800FFA7D5 /* snd_resample.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = snd_resample.c; path = ../Quake/snd_resample.c; sourceTree = SOURCE_ROOT; };
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8D1107310486CEB800E47090 /* Info.plist */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.plist; path = Info.plist; sourceTree = "<group>"; };
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8D1107320486CEB800E47090 /* QuakeSpasm.app */ = {isa = PBXFileReference; explicitFileType = wrapper.application; includeInIndex = 0; path = QuakeSpasm.app; sourceTree = BUILT_PRODUCTS_DIR; };
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/* End PBXFileReference section */
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@ -495,6 +497,7 @@
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66A5470A12E3CF8100FFA7D5 /* snd_codec.c */,
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66A5470B12E3CF8100FFA7D5 /* snd_codec_wav.c */,
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66A5470C12E3CF8100FFA7D5 /* snd_codec_ogg.c */,
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66A54A4E12E429B800FFA7D5 /* snd_resample.c */,
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);
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name = Sound;
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sourceTree = "<group>";
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@ -770,6 +773,7 @@
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66A5470D12E3CF8100FFA7D5 /* snd_codec.c in Sources */,
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66A5470E12E3CF8100FFA7D5 /* snd_codec_wav.c in Sources */,
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66A5470F12E3CF8100FFA7D5 /* snd_codec_ogg.c in Sources */,
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66A54A4F12E429B800FFA7D5 /* snd_resample.c in Sources */,
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);
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runOnlyForDeploymentPostprocessing = 0;
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};
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@ -32,6 +32,7 @@ typedef struct snd_info_s
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int samples;
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int size;
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int dataofs;
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int loopstart; // -1 if not looped
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} snd_info_t;
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typedef struct snd_codec_s snd_codec_t;
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@ -171,6 +171,31 @@ static qboolean S_ReadRIFFHeader( int file, snd_info_t *info)
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Sys_FileSeekRelative( file, fmtlen );
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}
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// get cue chunk
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// FIXME: port code
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/*
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FindChunk("cue ");
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if (data_p)
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{
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data_p += 32;
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info.loopstart = GetLittleLong();
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// if the next chunk is a LIST chunk, look for a cue length marker
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FindNextChunk ("LIST");
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if (data_p)
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{
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if (!strncmp (data_p + 28, "mark", 4))
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{ // this is not a proper parse, but it works with cooledit...
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data_p += 24;
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i = GetLittleLong (); // samples in loop
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info.samples = info.loopstart + i;
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// Con_Printf("looped length: %i\n", i);
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}
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}
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}
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else*/
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info->loopstart = -1;
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// Scan for the data chunk
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if( (info->size = S_FindRIFFChunk(file, "data")) < 0)
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{
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293
Quake/snd_mem.c
293
Quake/snd_mem.c
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@ -21,74 +21,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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// snd_mem.c: sound caching
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#include "quakedef.h"
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int cache_full_cycle;
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byte *S_Alloc (int size);
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/*
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================
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ResampleSfx
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================
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*/
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void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data)
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{
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int outcount;
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int srcsample;
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float stepscale;
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int i;
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int sample, samplefrac, fracstep;
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sfxcache_t *sc;
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sc = (sfxcache_t *) Cache_Check (&sfx->cache);
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if (!sc)
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return;
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stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2
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outcount = sc->length / stepscale;
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sc->length = outcount;
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if (sc->loopstart != -1)
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sc->loopstart = sc->loopstart / stepscale;
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sc->speed = shm->speed;
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if (loadas8bit.value)
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sc->width = 1;
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else
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sc->width = inwidth;
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sc->stereo = 0;
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// resample / decimate to the current source rate
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if (stepscale == 1 && inwidth == 1 && sc->width == 1)
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{
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// fast special case
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for (i=0 ; i<outcount ; i++)
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((signed char *)sc->data)[i]
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= (int)( (unsigned char)(data[i]) - 128);
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}
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else
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{
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// general case
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samplefrac = 0;
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fracstep = stepscale*256;
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for (i=0 ; i<outcount ; i++)
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{
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srcsample = samplefrac >> 8;
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samplefrac += fracstep;
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if (inwidth == 2)
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sample = LittleShort ( ((short *)data)[srcsample] );
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else
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sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8;
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if (sc->width == 2)
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((short *)sc->data)[i] = sample;
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else
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((signed char *)sc->data)[i] = sample >> 8;
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}
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}
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}
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//=============================================================================
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#include "snd_codec.h"
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/*
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==============
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@ -97,13 +30,11 @@ S_LoadSound
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*/
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sfxcache_t *S_LoadSound (sfx_t *s)
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{
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snd_info_t info;
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char namebuffer[256];
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byte *data;
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wavinfo_t info;
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int len;
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float stepscale;
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sfxcache_t *sc;
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byte stackbuf[1*1024]; // avoid dirtying the cache heap
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// see if still in memory
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sc = (sfxcache_t *) Cache_Check (&s->cache);
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@ -117,225 +48,41 @@ sfxcache_t *S_LoadSound (sfx_t *s)
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// Con_Printf ("loading %s\n",namebuffer);
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data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf));
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// load it in
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data = S_CodecLoad(namebuffer, &info);
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if (!data)
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{
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Con_Printf ("Couldn't load %s\n", namebuffer);
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return NULL;
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}
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info = GetWavinfo (s->name, data, com_filesize);
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if (info.channels != 1)
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{
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Con_Printf ("%s is a stereo sample\n",s->name);
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return NULL;
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}
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stepscale = (float)info.rate / shm->speed;
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len = info.samples / stepscale;
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len = len * info.width * info.channels;
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int resampledNumSamples;
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void *resampled = Snd_Resample(info.rate, info.width, info.samples, info.channels, data, shm->speed, shm->samplebits/8, &resampledNumSamples);
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len = resampledNumSamples * (shm->samplebits/8) * info.channels;
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sc = (sfxcache_t *) Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name);
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if (!sc)
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return NULL;
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sc->length = info.samples;
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sc->loopstart = info.loopstart;
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sc->speed = info.rate;
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sc->width = info.width;
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float ratio = info.rate / shm->speed;
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sc->length = resampledNumSamples;
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sc->loopstart = (info.loopstart == -1 ? -1 : info.loopstart / ratio); // reposition loop marker to take resampling into account
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sc->speed = shm->speed;
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sc->width = (shm->samplebits/8);
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sc->stereo = info.channels;
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ResampleSfx (s, sc->speed, sc->width, data + info.dataofs);
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memcpy(sc->data, resampled, len);
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free(resampled);
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Z_Free(data);
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return sc;
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}
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/*
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===============================================================================
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WAV loading
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===============================================================================
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*/
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byte *data_p;
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byte *iff_end;
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byte *last_chunk;
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byte *iff_data;
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int iff_chunk_len;
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short GetLittleShort(void)
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{
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short val = 0;
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val = *data_p;
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val = val + (*(data_p+1)<<8);
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data_p += 2;
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return val;
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}
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int GetLittleLong(void)
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{
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int val = 0;
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val = *data_p;
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val = val + (*(data_p+1)<<8);
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val = val + (*(data_p+2)<<16);
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val = val + (*(data_p+3)<<24);
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data_p += 4;
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return val;
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}
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void FindNextChunk(const char *name)
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{
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while (1)
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{
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// Need at least 8 bytes for a chunk
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if (last_chunk + 8 >= iff_end)
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{
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data_p = NULL;
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return;
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}
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data_p = last_chunk + 4;
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iff_chunk_len = GetLittleLong();
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if (iff_chunk_len < 0 || iff_chunk_len > iff_end - data_p)
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{
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data_p = NULL;
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Con_DPrintf("Bad \"%s\" chunk length (%d) in wav file\n", name, iff_chunk_len);
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return;
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}
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// if (iff_chunk_len > 1024*1024)
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// Sys_Error ("FindNextChunk: %i length is past the 1 meg sanity limit", iff_chunk_len);
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last_chunk = data_p + ((iff_chunk_len + 1) & ~1);
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data_p -= 8;
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if (!Q_strncmp((char *)data_p, name, 4))
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return;
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}
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}
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void FindChunk(const char *name)
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{
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last_chunk = iff_data;
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FindNextChunk (name);
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}
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void DumpChunks(void)
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{
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char str[5];
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str[4] = 0;
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data_p=iff_data;
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do
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{
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memcpy (str, data_p, 4);
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data_p += 4;
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iff_chunk_len = GetLittleLong();
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Con_Printf ("%p : %s (%d)\n", (data_p - 4), str, iff_chunk_len);
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data_p += (iff_chunk_len + 1) & ~1;
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} while (data_p < iff_end);
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}
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/*
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============
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GetWavinfo
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============
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*/
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wavinfo_t GetWavinfo (const char *name, byte *wav, int wavlength)
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{
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wavinfo_t info;
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int i;
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int format;
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int samples;
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memset (&info, 0, sizeof(info));
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if (!wav)
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return info;
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iff_data = wav;
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iff_end = wav + wavlength;
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// find "RIFF" chunk
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FindChunk("RIFF");
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if (!(data_p && !Q_strncmp((char *)data_p + 8, "WAVE", 4)))
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{
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Con_Printf("Missing RIFF/WAVE chunks\n");
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return info;
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}
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// get "fmt " chunk
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iff_data = data_p + 12;
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// DumpChunks ();
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FindChunk("fmt ");
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if (!data_p)
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{
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Con_Printf("Missing fmt chunk\n");
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return info;
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}
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data_p += 8;
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format = GetLittleShort();
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if (format != 1)
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{
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Con_Printf("Microsoft PCM format only\n");
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return info;
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}
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info.channels = GetLittleShort();
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info.rate = GetLittleLong();
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data_p += 4+2;
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info.width = GetLittleShort() / 8;
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// get cue chunk
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FindChunk("cue ");
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if (data_p)
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{
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data_p += 32;
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info.loopstart = GetLittleLong();
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// if the next chunk is a LIST chunk, look for a cue length marker
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FindNextChunk ("LIST");
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if (data_p)
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{
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if (!strncmp((char *)data_p + 28, "mark", 4))
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{ // this is not a proper parse, but it works with cooledit...
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data_p += 24;
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i = GetLittleLong (); // samples in loop
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info.samples = info.loopstart + i;
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// Con_Printf("looped length: %i\n", i);
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}
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}
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}
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else
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info.loopstart = -1;
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// find data chunk
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FindChunk("data");
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if (!data_p)
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{
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Con_Printf("Missing data chunk\n");
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return info;
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}
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data_p += 4;
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samples = GetLittleLong () / info.width;
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if (info.samples)
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{
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if (samples < info.samples)
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Sys_Error ("Sound %s has a bad loop length", name);
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}
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else
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info.samples = samples;
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info.dataofs = data_p - wav;
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return info;
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}
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|
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45
Quake/snd_resample.c
Normal file
45
Quake/snd_resample.c
Normal file
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@ -0,0 +1,45 @@
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#include "quakedef.h"
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#include "sound.h"
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// FIXME: call a real resampler :-)
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void *Snd_Resample(int inrate, int inwidth, int innumsamples, int channels, const void *indata,
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int outrate, int outwidth, int *outnumsamples)
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{
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char *outdata;
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int i;
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int sample, samplefrac, fracstep;
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float stepscale = ((float)inrate) / ((float)outrate);
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*outnumsamples = innumsamples / stepscale;
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outdata = malloc((*outnumsamples) * channels * outwidth);
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// resample / decimate to the current source rate
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samplefrac = 0;
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fracstep = stepscale*256;
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for (i=0 ; i<(*outnumsamples) ; i++)
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{
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int srcsample = samplefrac >> 8;
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samplefrac += fracstep;
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if (inwidth == 2)
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sample = LittleShort ( ((short *)indata)[srcsample] );
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else
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sample = (int)( (((unsigned char *)indata)[srcsample]) - 128) << 8;
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if (outwidth == 2)
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((short *)outdata)[i] = sample;
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else
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((signed char *)outdata)[i] = sample >> 8;
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}
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return outdata;
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}
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void *Snd_ResamplerInit(int inrate, int inwidth, int outrate, int outwidth, int channels) { return NULL; }
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void Snd_ResamplerClose(void *resampler) {}
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void Snd_ResampleStream(void *resampler,
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int *innumsamples, void *indata,
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int *outnumsamples, void *outdata) {}
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|
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@ -149,6 +149,24 @@ qboolean S_Base_StartBackgroundTrack( const char *intro, qboolean loop, S_Backgr
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void S_UpdateBackgroundTrack( void );
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qboolean S_BackgroundTrackIsPlaying( void );
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/* 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);
|
||||
|
||||
/* 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);
|
||||
|
||||
|
||||
// ====================================================================
|
||||
// User-setable variables
|
||||
// ====================================================================
|
||||
|
|
|
|||
Loading…
Reference in a new issue