Improved sound resampling. It can handle sounds up to 96KHz stereo at a constant...
[xonotic/darkplaces.git] / snd_mem.c
index be03af6..f61a5a9 100644 (file)
--- 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 ; i<outcount ; i++)
-                               ((signed char *)sc->data)[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<outcount ; i+=2)
-                       {
-                               srcsample = samplefrac >> 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<outcount ; i++)
+                       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++)
                        {
-                               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;
 }
-