X-Git-Url: http://de.git.xonotic.org/?a=blobdiff_plain;f=snd_mem.c;h=07c32eb1865d4c1f780dd89e2e73cea311e64455;hb=17fdc60aff5ea50886d4fd180ff00af11f7272d9;hp=2cb01bee3adae3d371761c105cad2975153f9c53;hpb=f3e79d752a76a9d6329759a83ec9800a5e4cc92b;p=xonotic%2Fdarkplaces.git diff --git a/snd_mem.c b/snd_mem.c index 2cb01bee..07c32eb1 100644 --- a/snd_mem.c +++ b/snd_mem.c @@ -17,10 +17,11 @@ 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" +#include "snd_main.h" #include "snd_ogg.h" #include "snd_wav.h" @@ -32,173 +33,111 @@ ResampleSfx */ size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname) { - int samplefrac, fracstep; - size_t i, srcsample, srclength, outcount; - - // this is usually 0.5 (128), 1 (256), or 2 (512) - fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0; + size_t srclength, outcount, i; srclength = in_length * in_format->channels; + outcount = (double)in_length * shm->format.speed / in_format->speed; - outcount = (double) in_length * (double) shm->format.speed / (double) in_format->speed; - 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); - -// resample / decimate to the current source rate + //Con_DPrintf("ResampleSfx(%s): %d samples @ %dHz -> %d samples @ %dHz\n", + // sfxname, in_length, in_format->speed, outcount, shm->format.speed); - if (fracstep == 256) + // Trivial case (direct transfer) + if (in_format->speed == shm->format.speed) { - // fast case for direct transfer - if (in_format->width == 1) // 8bit - for (i = 0;i < srclength;i++) - ((signed char *)out_data)[i] = ((unsigned char *)in_data)[i] - 128; - else //if (sb->width == 2) // 16bit - for (i = 0;i < srclength;i++) - ((short *)out_data)[i] = ((short *)in_data)[i]; + if (in_format->width == 1) + { + for (i = 0; i < srclength; i++) + ((signed char*)out_data)[i] = in_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 - samplefrac = 0; - if ((fracstep & 255) == 0) // skipping points on perfect multiple + 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) { - srcsample = 0; - fracstep >>= 8; - if (in_format->width == 2) - { - short *out = (short*)out_data; - const short *in = (const short*)in_data; - if (in_format->channels == 2) // LordHavoc: stereo sound support - { - fracstep <<= 1; - for (i=0 ; ichannels == 2) - { - fracstep <<= 1; - for (i=0 ; iwidth == 2) - { - short *out = (short*)out_data; - const short *in = (const short*)in_data; - if (in_format->channels == 2) - { - for (i=0 ; i> 8) << 1; - a = in[srcsample ]; - if (srcsample+2 >= srclength) - b = 0; - else - b = in[srcsample+2]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - a = in[srcsample+1]; - if (srcsample+2 >= srclength) - b = 0; - else - b = in[srcsample+3]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - samplefrac += fracstep; - } - } - else - { - for (i=0 ; i> 8; - a = in[srcsample ]; - if (srcsample+1 >= srclength) - b = 0; - else - b = in[srcsample+1]; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (short) sample; - samplefrac += fracstep; - } - } - } + size_t tmpcount; + + samplefrac = 0; + + // If more than 1 sec of sound remains to be converted + if (outcount - total_out > 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++) { - signed char *out = out_data; - const unsigned char *in = in_data; - if (in_format->channels == 2) + 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 (i=0 ; ichannels; j++, srcsample++) { - srcsample = (samplefrac >> 8) << 1; - a = (int) in[srcsample ] - 128; - if (srcsample+2 >= srclength) - b = 0; - else - b = (int) in[srcsample+2] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - a = (int) in[srcsample+1] - 128; - if (srcsample+2 >= srclength) - b = 0; + // 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 - b = (int) in[srcsample+3] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - samplefrac += fracstep; + *((short*)out_ptr) = ((const short*)in_ptr)[srcsample]; + + out_ptr += sizeof (short); } } - else + // 8 bit samples + else // if (in_format->width == 1) { - for (i=0 ; ichannels; j++, srcsample++) { - srcsample = samplefrac >> 8; - a = (int) in[srcsample ] - 128; - if (srcsample+1 >= srclength) - b = 0; + // 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 - b = (int) in[srcsample+1] - 128; - sample = (((b - a) * (samplefrac & 255)) >> 8) + a; - *out++ = (signed char) sample; - samplefrac += fracstep; + *((signed char*)out_ptr) = ((const qbyte*)in_ptr)[srcsample] - 128; + + out_ptr += sizeof (signed char); } } + + samplefrac += fracstep; } + + // 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; } } @@ -214,13 +153,17 @@ S_LoadSound */ qboolean S_LoadSound (sfx_t *s, qboolean complain) { - char namebuffer[MAX_QPATH]; + char namebuffer[MAX_QPATH + 16]; size_t len; - qboolean modified_name = false; - // see if still in memory if (!shm || !shm->format.speed) return false; + + // If we weren't able to load it previously, no need to retry + if (s->flags & SFXFLAG_FILEMISSING) + return false; + + // See if in memory if (s->fetcher != NULL) { if (s->format.speed != shm->format.speed) @@ -228,51 +171,59 @@ qboolean S_LoadSound (sfx_t *s, qboolean complain) return true; } - len = strlcpy (namebuffer, s->name, sizeof (namebuffer)); - if (len >= sizeof (namebuffer)) - return false; + // LordHavoc: if the sound filename does not begin with sound/, try adding it + if (strncasecmp(s->name, "sound/", 6)) + { + len = dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", s->name); + if (len < 0) + { + // name too long + Con_Printf("S_LoadSound: name \"%s\" is too long\n", s->name); + return false; + } + if (S_LoadWavFile (namebuffer, s)) + return true; + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav")) + strcpy (namebuffer + len - 3, "ogg"); + if (OGG_LoadVorbisFile (namebuffer, s)) + return true; + } - // Try to load it as a WAV file + // LordHavoc: then try without the added sound/ as wav and ogg + len = dpsnprintf (namebuffer, sizeof(namebuffer), "%s", s->name); + if (len < 0) + { + // name too long + Con_Printf("S_LoadSound: name \"%s\" is too long\n", s->name); + return false; + } if (S_LoadWavFile (namebuffer, s)) return true; - - // Else, try to load it as an Ogg Vorbis file - if (!strcasecmp (namebuffer + len - 4, ".wav")) - { + if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav")) strcpy (namebuffer + len - 3, "ogg"); - modified_name = true; - } if (OGG_LoadVorbisFile (namebuffer, s)) return true; // Can't load the sound! - if (!complain) - s->flags |= SFXFLAG_SILENTLYMISSING; - else - s->flags &= ~SFXFLAG_SILENTLYMISSING; + s->flags |= SFXFLAG_FILEMISSING; if (complain) - { - if (modified_name) - strcpy (namebuffer + len - 3, "wav"); - Con_Printf("Couldn't load %s\n", namebuffer); - } + Con_Printf("S_LoadSound: Couldn't load \"%s\"\n", s->name); return false; } -void S_UnloadSound(sfx_t *s) +void S_UnloadSound (sfx_t *s) { if (s->fetcher != NULL) { unsigned int i; + // Stop all channels that use this sound + for (i = 0; i < total_channels ; i++) + if (channels[i].sfx == s) + S_StopChannel (i); + s->fetcher = NULL; s->fetcher_data = NULL; Mem_FreePool(&s->mempool); - - // 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; } }