X-Git-Url: http://de.git.xonotic.org/?p=xonotic%2Fdarkplaces.git;a=blobdiff_plain;f=snd_mem.c;h=f61a5a98cf5c16e5ce578033708bb1bba9e69032;hp=be03af6ae5a14fb4a961ea86001b50d8dd137271;hb=137e29f228f1309a5cfb2c709a362ec159efd581;hpb=cc63b89849022ef37ef113a7dc9489c2e846bd1b diff --git a/snd_mem.c b/snd_mem.c index be03af6a..f61a5a98 100644 --- a/snd_mem.c +++ b/snd_mem.c @@ -8,7 +8,7 @@ of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of -MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. @@ -17,128 +17,130 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ -// snd_mem.c: sound caching + #include "quakedef.h" -int cache_full_cycle; +#include "snd_ogg.h" +#include "snd_wav.h" -byte *S_Alloc (int size); /* ================ ResampleSfx ================ */ -void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data) +size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname) { - int outcount; - int srcsample; - float stepscale; - int i; - int sample, samplefrac, fracstep; - sfxcache_t *sc; - - sc = Cache_Check (&sfx->cache); - if (!sc) - return; - - stepscale = (float)inrate / shm->speed; // this is usually 0.5, 1, or 2 - - outcount = sc->length / stepscale; - sc->length = outcount; - if (sc->loopstart != -1) - sc->loopstart = sc->loopstart / stepscale; - - sc->speed = shm->speed; - if (loadas8bit.value) - sc->width = 1; - else - sc->width = inwidth; -// sc->stereo = 0; + size_t srclength, outcount, i; + + srclength = in_length * in_format->channels; + outcount = (double)in_length * shm->format.speed / in_format->speed; -// resample / decimate to the current source rate + Con_DPrintf("ResampleSfx: resampling sound \"%s\" from %dHz to %dHz (%d samples to %d samples)\n", + sfxname, in_format->speed, shm->format.speed, in_length, outcount); - if (stepscale == 1 && inwidth == 1 && sc->width == 1) + // Trivial case (direct transfer) + if (in_format->speed == shm->format.speed) { -// fast special case - // LordHavoc: I do not serve the readability gods... - int *indata, *outdata; - int count4, count1; - count1 = outcount << sc->stereo; - count4 = count1 >> 2; - indata = (void *)data; - outdata = (void *)sc->data; - while (count4--) - *outdata++ = *indata++ ^ 0x80808080; - if (count1 & 2) - ((short*)outdata)[0] = ((short*)indata)[0] ^ 0x8080; - if (count1 & 1) - ((char*)outdata)[2] = ((char*)indata)[2] ^ 0x80; - /* - if (sc->stereo) // LordHavoc: stereo sound support + if (in_format->width == 1) { - for (i=0 ; i<(outcount<<1) ; i++) - ((signed char *)sc->data)[i] = (int)( (unsigned char)(data[i]) - 128); + for (i = 0; i < srclength; i++) + ((signed char*)out_data)[i] = in_data[i] - 128; } - else - { - for (i=0 ; idata)[i] = (int)( (unsigned char)(data[i]) - 128); - } - */ + else // if (in_format->width == 2) + memcpy (out_data, in_data, srclength * in_format->width); } + + // General case (linear interpolation with a fixed-point fractional + // step, 18-bit integer part and 14-bit fractional part) + // Can handle up to 2^18 (262144) samples per second (> 96KHz stereo) + #define FRACTIONAL_BITS 14 + #define FRACTIONAL_MASK ((1 << FRACTIONAL_BITS) - 1) + #define INTEGER_BITS (sizeof(samplefrac)*8 - FRACTIONAL_BITS) else { -// general case - Con_DPrintf("ResampleSfx: resampling sound %s\n", sfx->name); - samplefrac = 0; - fracstep = stepscale*256; - if (sc->stereo) // LordHavoc: stereo sound support - { - for (i=0 ; i> 8; - samplefrac += fracstep; - srcsample <<= 1; - // left - if (inwidth == 2) - sample = LittleShort ( ((short *)data)[srcsample] ); - else - sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8; - if (sc->width == 2) - ((short *)sc->data)[i] = sample; - else - ((signed char *)sc->data)[i] = sample >> 8; - // right - srcsample++; - if (inwidth == 2) - sample = LittleShort ( ((short *)data)[srcsample] ); - else - sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8; - if (sc->width == 2) - ((short *)sc->data)[i+1] = sample; - else - ((signed char *)sc->data)[i+1] = sample >> 8; - } - } - else + const unsigned int fracstep = (double)in_format->speed / shm->format.speed * (1 << FRACTIONAL_BITS); + size_t remain_in = srclength, total_out = 0; + unsigned int samplefrac; + const qbyte *in_ptr = in_data; + qbyte *out_ptr = out_data; + + // Check that we can handle one second of that sound + if (in_format->speed * in_format->channels > (1 << INTEGER_BITS)) + Sys_Error ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))", + in_format->speed, in_format->channels); + + // We work 1 sec at a time to make sure we don't accumulate any + // significant error when adding "fracstep" over several seconds, and + // also to be able to handle very long sounds. + while (total_out < outcount) { - for (i=0 ; i shm->format.speed) + tmpcount = shm->format.speed; + else + tmpcount = outcount - total_out; + + // Convert up to 1 sec of sound + for (i = 0; i < tmpcount; i++) { - srcsample = samplefrac >> 8; + unsigned int j = 0; + unsigned int srcsample = (samplefrac >> FRACTIONAL_BITS) * in_format->channels; + int a, b; + + // 16 bit samples + if (in_format->width == 2) + { + for (j = 0; j < in_format->channels; j++, srcsample++) + { + // No value to interpolate with? + if (srcsample + in_format->channels < remain_in) + { + a = ((const short*)in_ptr)[srcsample]; + b = ((const short*)in_ptr)[srcsample + in_format->channels]; + *((short*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a; + } + else + *((short*)out_ptr) = ((const short*)in_ptr)[srcsample]; + + out_ptr += sizeof (short); + } + } + // 8 bit samples + else // if (in_format->width == 1) + { + for (j = 0; j < in_format->channels; j++, srcsample++) + { + // No more value to interpolate with? + if (srcsample + in_format->channels < remain_in) + { + a = ((const qbyte*)in_ptr)[srcsample] - 128; + b = ((const qbyte*)in_ptr)[srcsample + in_format->channels] - 128; + *((signed char*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a; + } + else + *((signed char*)out_ptr) = ((const qbyte*)in_ptr)[srcsample] - 128; + + out_ptr += sizeof (signed char); + } + } + samplefrac += fracstep; - if (inwidth == 2) - sample = LittleShort ( ((short *)data)[srcsample] ); - else - sample = (int)( (unsigned char)(data[srcsample]) - 128) << 8; - if (sc->width == 2) - ((short *)sc->data)[i] = sample; - else - ((signed char *)sc->data)[i] = sample >> 8; } + + // Update the counters and the buffer position + remain_in -= in_format->speed * in_format->channels; + in_ptr += in_format->speed * in_format->channels * in_format->width; + total_out += tmpcount; } } + + return outcount; } //============================================================================= @@ -148,256 +150,67 @@ void ResampleSfx (sfx_t *sfx, int inrate, int inwidth, byte *data) S_LoadSound ============== */ -sfxcache_t *S_LoadSound (sfx_t *s) +qboolean S_LoadSound (sfx_t *s, qboolean complain) { - char namebuffer[256]; - byte *data; - wavinfo_t info; - int len; - float stepscale; - sfxcache_t *sc; - byte stackbuf[1*1024]; // avoid dirtying the cache heap - -// see if still in memory - sc = Cache_Check (&s->cache); - if (sc) - return sc; - -//Con_Printf ("S_LoadSound: %x\n", (int)stackbuf); -// load it in - strcpy(namebuffer, "sound/"); - strcat(namebuffer, s->name); - -// Con_Printf ("loading %s\n",namebuffer); - - data = COM_LoadStackFile(namebuffer, stackbuf, sizeof(stackbuf), false); - - if (!data) + char namebuffer[MAX_QPATH]; + size_t len; + qboolean modified_name = false; + + // see if still in memory + if (!shm || !shm->format.speed) + return false; + if (s->fetcher != NULL) { - Con_Printf ("Couldn't load %s\n", namebuffer); - return NULL; + if (s->format.speed != shm->format.speed) + Sys_Error ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name); + return true; } - info = GetWavinfo (s->name, data, com_filesize); - // LordHavoc: stereo sounds are now allowed (intended for music) - if (info.channels < 1 || info.channels > 2) - { - Con_Printf ("%s has an unsupported number of channels (%i)\n",s->name, info.channels); - return NULL; - } - /* - if (info.channels != 1) - { - Con_Printf ("%s is a stereo sample\n",s->name); - return NULL; - } - */ - - stepscale = (float)info.rate / shm->speed; - len = info.samples / stepscale; - - len = len * info.width * info.channels; - - sc = Cache_Alloc ( &s->cache, len + sizeof(sfxcache_t), s->name); - if (!sc) - return NULL; - - sc->length = info.samples; - sc->loopstart = info.loopstart; - sc->speed = info.rate; - sc->width = info.width; - sc->stereo = info.channels == 2; - - ResampleSfx (s, sc->speed, sc->width, data + info.dataofs); - - return sc; -} - + len = strlcpy (namebuffer, s->name, sizeof (namebuffer)); + if (len >= sizeof (namebuffer)) + return false; + // Try to load it as a WAV file + if (S_LoadWavFile (namebuffer, s)) + return true; -/* -=============================================================================== - -WAV loading - -=============================================================================== -*/ - - -byte *data_p; -byte *iff_end; -byte *last_chunk; -byte *iff_data; -int iff_chunk_len; - - -short GetLittleShort(void) -{ - short val = 0; - val = *data_p; - val = val + (*(data_p+1)<<8); - data_p += 2; - return val; -} - -int GetLittleLong(void) -{ - int val = 0; - val = *data_p; - val = val + (*(data_p+1)<<8); - val = val + (*(data_p+2)<<16); - val = val + (*(data_p+3)<<24); - data_p += 4; - return val; -} - -void FindNextChunk(char *name) -{ - while (1) + // Else, try to load it as an Ogg Vorbis file + if (!strcasecmp (namebuffer + len - 4, ".wav")) { - data_p=last_chunk; - - if (data_p >= iff_end) - { // didn't find the chunk - data_p = NULL; - return; - } - - data_p += 4; - iff_chunk_len = GetLittleLong(); - if (iff_chunk_len < 0) - { - data_p = NULL; - return; - } -// if (iff_chunk_len > 1024*1024) -// 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 (!strncmp(data_p, name, 4)) - return; + strcpy (namebuffer + len - 3, "ogg"); + modified_name = true; } -} + if (OGG_LoadVorbisFile (namebuffer, s)) + return true; -void FindChunk(char *name) -{ - last_chunk = iff_data; - FindNextChunk (name); -} - - -void DumpChunks(void) -{ - char str[5]; - - str[4] = 0; - data_p=iff_data; - do + // Can't load the sound! + if (!complain) + s->flags |= SFXFLAG_SILENTLYMISSING; + else + s->flags &= ~SFXFLAG_SILENTLYMISSING; + if (complain) { - memcpy (str, data_p, 4); - data_p += 4; - iff_chunk_len = GetLittleLong(); - Con_Printf ("0x%x : %s (%d)\n", (int)(data_p - 4), str, iff_chunk_len); - data_p += (iff_chunk_len + 1) & ~1; - } while (data_p < iff_end); + if (modified_name) + strcpy (namebuffer + len - 3, "wav"); + Con_Printf("Couldn't load %s\n", namebuffer); + } + return false; } -/* -============ -GetWavinfo -============ -*/ -wavinfo_t GetWavinfo (char *name, byte *wav, int wavlength) +void S_UnloadSound(sfx_t *s) { - wavinfo_t info; - int i; - int format; - int samples; - - memset (&info, 0, sizeof(info)); - - if (!wav) - return info; - - iff_data = wav; - iff_end = wav + wavlength; - -// find "RIFF" chunk - FindChunk("RIFF"); - if (!(data_p && !strncmp(data_p+8, "WAVE", 4))) - { - Con_Printf("Missing RIFF/WAVE chunks\n"); - return info; - } - -// get "fmt " chunk - iff_data = data_p + 12; -// DumpChunks (); - - FindChunk("fmt "); - if (!data_p) - { - Con_Printf("Missing fmt chunk\n"); - return info; - } - data_p += 8; - format = GetLittleShort(); - if (format != 1) + if (s->fetcher != NULL) { - Con_Printf("Microsoft PCM format only\n"); - return info; - } + unsigned int i; - info.channels = GetLittleShort(); - info.rate = GetLittleLong(); - data_p += 4+2; - info.width = GetLittleShort() / 8; + s->fetcher = NULL; + s->fetcher_data = NULL; + Mem_FreePool(&s->mempool); -// get cue chunk - FindChunk("cue "); - if (data_p) - { - data_p += 32; - info.loopstart = GetLittleLong(); -// Con_Printf("loopstart=%d\n", sfx->loopstart); - - // if the next chunk is a LIST chunk, look for a cue length marker - FindNextChunk ("LIST"); - if (data_p) - { - if (!strncmp (data_p + 28, "mark", 4)) - { // this is not a proper parse, but it works with cooledit... - data_p += 24; - i = GetLittleLong (); // samples in loop - info.samples = info.loopstart + i; -// Con_Printf("looped length: %i\n", i); - } - } + // At this point, some per-channel data pointers may point to freed zones. + // Practically, it shouldn't be a problem; but it's wrong, so we fix that + for (i = 0; i < total_channels ; i++) + if (channels[i].sfx == s) + channels[i].fetcher_data = NULL; } - else - info.loopstart = -1; - -// find data chunk - FindChunk("data"); - if (!data_p) - { - Con_Printf("Missing data chunk\n"); - return info; - } - - data_p += 4; - samples = GetLittleLong () / info.width; - - if (info.samples) - { - if (samples < info.samples) - Host_Error ("Sound %s has a bad loop length", name); - } - else - info.samples = samples; - - info.dataofs = data_p - wav; - - return info; } -