Fix engine not starting on Windows if linked against SDL > 2.0.5
[xonotic/darkplaces.git] / snd_mem.c
index 2cb01be..f6f9c14 100644 (file)
--- a/snd_mem.c
+++ b/snd_mem.c
@@ -17,194 +17,283 @@ 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"
 
 
 /*
-================
-ResampleSfx
-================
+====================
+Snd_CreateRingBuffer
+
+If "buffer" is NULL, the function allocates one buffer of "sampleframes" sample frames itself
+(if "sampleframes" is 0, the function chooses the size).
+====================
 */
-size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t* in_format, qbyte *out_data, const char* sfxname)
+snd_ringbuffer_t *Snd_CreateRingBuffer (const snd_format_t* format, unsigned int sampleframes, void* buffer)
 {
-       int samplefrac, fracstep;
-       size_t i, srcsample, srclength, outcount;
+       snd_ringbuffer_t *ringbuffer;
+
+       // If the caller provides a buffer, it must give us its size
+       if (sampleframes == 0 && buffer != NULL)
+               return NULL;
+
+       ringbuffer = (snd_ringbuffer_t*)Mem_Alloc(snd_mempool, sizeof (*ringbuffer));
+       memset(ringbuffer, 0, sizeof(*ringbuffer));
+       memcpy(&ringbuffer->format, format, sizeof(ringbuffer->format));
+
+       // If we haven't been given a buffer
+       if (buffer == NULL)
+       {
+               unsigned int maxframes;
+               size_t memsize;
+
+               if (sampleframes == 0)
+                       maxframes = (format->speed + 1) / 2;  // Make the sound buffer large enough for containing 0.5 sec of sound
+               else
+                       maxframes = sampleframes;
+
+               memsize = maxframes * format->width * format->channels;
+               ringbuffer->ring = (unsigned char *) Mem_Alloc(snd_mempool, memsize);
+               ringbuffer->maxframes = maxframes;
+       }
+       else
+       {
+               ringbuffer->ring = (unsigned char *) buffer;
+               ringbuffer->maxframes = sampleframes;
+       }
+
+       return ringbuffer;
+}
 
-       // this is usually 0.5 (128), 1 (256), or 2 (512)
-       fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0;
 
-       srclength = in_length * in_format->channels;
+/*
+====================
+Snd_CreateSndBuffer
+====================
+*/
+snd_buffer_t *Snd_CreateSndBuffer (const unsigned char *samples, unsigned int sampleframes, const snd_format_t* in_format, unsigned int sb_speed)
+{
+       size_t newsampleframes, memsize;
+       snd_buffer_t* sb;
+
+       newsampleframes = (size_t) ceil((double)sampleframes * (double)sb_speed / (double)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);
+       memsize = newsampleframes * in_format->channels * in_format->width;
+       memsize += sizeof (*sb) - sizeof (sb->samples);
 
-// resample / decimate to the current source rate
+       sb = (snd_buffer_t*)Mem_Alloc (snd_mempool, memsize);
+       sb->format.channels = in_format->channels;
+       sb->format.width = in_format->width;
+       sb->format.speed = sb_speed;
+       sb->maxframes = (unsigned int)newsampleframes;
+       sb->nbframes = 0;
 
-       if (fracstep == 256)
+       if (!Snd_AppendToSndBuffer (sb, samples, sampleframes, in_format))
        {
-               // 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];
+               Mem_Free (sb);
+               return NULL;
        }
+
+       return sb;
+}
+
+
+/*
+====================
+Snd_AppendToSndBuffer
+====================
+*/
+qboolean Snd_AppendToSndBuffer (snd_buffer_t* sb, const unsigned char *samples, unsigned int sampleframes, const snd_format_t* format)
+{
+       size_t srclength, outcount;
+       unsigned char *out_data;
+
+       //Con_DPrintf("ResampleSfx: %d samples @ %dHz -> %d samples @ %dHz\n",
+       //                      sampleframes, format->speed, outcount, sb->format.speed);
+
+       // If the formats are incompatible
+       if (sb->format.channels != format->channels || sb->format.width != format->width)
+       {
+               Con_Print("AppendToSndBuffer: incompatible sound formats!\n");
+               return false;
+       }
+
+       outcount = (size_t) ((double)sampleframes * (double)sb->format.speed / (double)format->speed);
+
+       // If the sound buffer is too short
+       if (outcount > sb->maxframes - sb->nbframes)
+       {
+               Con_Print("AppendToSndBuffer: sound buffer too short!\n");
+               return false;
+       }
+
+       out_data = &sb->samples[sb->nbframes * sb->format.width * sb->format.channels];
+       srclength = sampleframes * format->channels;
+
+       // Trivial case (direct transfer)
+       if (format->speed == sb->format.speed)
+       {
+               if (format->width == 1)
+               {
+                       size_t i;
+
+                       for (i = 0; i < srclength; i++)
+                               ((signed char*)out_data)[i] = samples[i] - 128;
+               }
+               else  // if (format->width == 2)
+                       memcpy (out_data, samples, srclength * 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 = (unsigned int)((double)format->speed / sb->format.speed * (1 << FRACTIONAL_BITS));
+               size_t remain_in = srclength, total_out = 0;
+               unsigned int samplefrac;
+               const unsigned char *in_ptr = samples;
+               unsigned char *out_ptr = out_data;
+
+               // Check that we can handle one second of that sound
+               if (format->speed * format->channels > (1 << INTEGER_BITS))
                {
-                       srcsample = 0;
-                       fracstep >>= 8;
-                       if (in_format->width == 2)
+                       Con_Printf ("ResampleSfx: sound quality too high for resampling (%uHz, %u channel(s))\n",
+                                          format->speed, format->channels);
+                       return 0;
+               }
+
+               // 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)
+               {
+                       size_t tmpcount, interpolation_limit, i, j;
+                       unsigned int srcsample;
+
+                       samplefrac = 0;
+
+                       // If more than 1 sec of sound remains to be converted
+                       if (outcount - total_out > sb->format.speed)
                        {
-                               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 ; i<outcount ; i++)
-                                       {
-                                               *out++ = in[srcsample  ];
-                                               *out++ = in[srcsample+1];
-                                               srcsample += fracstep;
-                                       }
-                               }
-                               else
-                               {
-                                       for (i=0 ; i<outcount ; i++)
-                                       {
-                                               *out++ = in[srcsample];
-                                               srcsample += fracstep;
-                                       }
-                               }
+                               tmpcount = sb->format.speed;
+                               interpolation_limit = tmpcount;  // all samples can be interpolated
                        }
                        else
                        {
-                               signed char *out = out_data;
-                               const unsigned char *in = in_data;
-                               if (in_format->channels == 2)
-                               {
-                                       fracstep <<= 1;
-                                       for (i=0 ; i<outcount ; i++)
-                                       {
-                                               *out++ = in[srcsample  ] - 128;
-                                               *out++ = in[srcsample+1] - 128;
-                                               srcsample += fracstep;
-                                       }
-                               }
-                               else
-                               {
-                                       for (i=0 ; i<outcount ; i++)
-                                       {
-                                               *out++ = in[srcsample  ] - 128;
-                                               srcsample += fracstep;
-                                       }
-                               }
+                               tmpcount = outcount - total_out;
+                               interpolation_limit = (int)ceil((double)(((remain_in / format->channels) - 1) << FRACTIONAL_BITS) / fracstep);
+                               if (interpolation_limit > tmpcount)
+                                       interpolation_limit = tmpcount;
                        }
-               }
-               else
-               {
-                       int sample;
-                       int a, b;
-                       if (in_format->width == 2)
+
+                       // 16 bit samples
+                       if (format->width == 2)
                        {
-                               short *out = (short*)out_data;
-                               const short *in = (const short*)in_data;
-                               if (in_format->channels == 2)
+                               const short* in_ptr_short;
+
+                               // Interpolated part
+                               for (i = 0; i < interpolation_limit; i++)
                                {
-                                       for (i=0 ; i<outcount ; i++)
+                                       srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels;
+                                       in_ptr_short = &((const short*)in_ptr)[srcsample];
+
+                                       for (j = 0; j < format->channels; j++)
                                        {
-                                               srcsample = (samplefrac >> 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;
+                                               int a, b;
+
+                                               a = *in_ptr_short;
+                                               b = *(in_ptr_short + format->channels);
+                                               *((short*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+
+                                               in_ptr_short++;
+                                               out_ptr += sizeof (short);
                                        }
+
+                                       samplefrac += fracstep;
                                }
-                               else
+
+                               // Non-interpolated part
+                               for (/* nothing */; i < tmpcount; i++)
                                {
-                                       for (i=0 ; i<outcount ; i++)
+                                       srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels;
+                                       in_ptr_short = &((const short*)in_ptr)[srcsample];
+
+                                       for (j = 0; j < format->channels; j++)
                                        {
-                                               srcsample = samplefrac >> 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;
+                                               *((short*)out_ptr) = *in_ptr_short;
+
+                                               in_ptr_short++;
+                                               out_ptr += sizeof (short);
                                        }
+
+                                       samplefrac += fracstep;
                                }
                        }
-                       else
+                       // 8 bit samples
+                       else  // if (format->width == 1)
                        {
-                               signed char *out = out_data;
-                               const unsigned char *in = in_data;
-                               if (in_format->channels == 2)
+                               const unsigned char* in_ptr_byte;
+
+                               // Convert up to 1 sec of sound
+                               for (i = 0; i < interpolation_limit; i++)
                                {
-                                       for (i=0 ; i<outcount ; i++)
+                                       srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels;
+                                       in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample];
+
+                                       for (j = 0; j < format->channels; j++)
                                        {
-                                               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;
-                                               else
-                                                       b = (int) in[srcsample+3] - 128;
-                                               sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
-                                               *out++ = (signed char) sample;
-                                               samplefrac += fracstep;
+                                               int a, b;
+
+                                               a = *in_ptr_byte - 128;
+                                               b = *(in_ptr_byte + format->channels) - 128;
+                                               *((signed char*)out_ptr) = (((b - a) * (samplefrac & FRACTIONAL_MASK)) >> FRACTIONAL_BITS) + a;
+
+                                               in_ptr_byte++;
+                                               out_ptr += sizeof (signed char);
                                        }
+
+                                       samplefrac += fracstep;
                                }
-                               else
+
+                               // Non-interpolated part
+                               for (/* nothing */; i < tmpcount; i++)
                                {
-                                       for (i=0 ; i<outcount ; i++)
+                                       srcsample = (samplefrac >> FRACTIONAL_BITS) * format->channels;
+                                       in_ptr_byte = &((const unsigned char*)in_ptr)[srcsample];
+
+                                       for (j = 0; j < format->channels; j++)
                                        {
-                                               srcsample = samplefrac >> 8;
-                                               a = (int) in[srcsample  ] - 128;
-                                               if (srcsample+1 >= srclength)
-                                                       b = 0;
-                                               else
-                                                       b = (int) in[srcsample+1] - 128;
-                                               sample = (((b - a) * (samplefrac & 255)) >> 8) + a;
-                                               *out++ = (signed char) sample;
-                                               samplefrac += fracstep;
+                                               *((signed char*)out_ptr) = *in_ptr_byte - 128;
+
+                                               in_ptr_byte++;
+                                               out_ptr += sizeof (signed char);
                                        }
+
+                                       samplefrac += fracstep;
                                }
                        }
+
+                       // Update the counters and the buffer position
+                       remain_in -= format->speed * format->channels;
+                       in_ptr += format->speed * format->channels * format->width;
+                       total_out += tmpcount;
                }
        }
 
-       return outcount;
+       sb->nbframes += (unsigned int)outcount;
+       return true;
 }
 
+
 //=============================================================================
 
 /*
@@ -212,67 +301,77 @@ size_t ResampleSfx (const qbyte *in_data, size_t in_length, const snd_format_t*
 S_LoadSound
 ==============
 */
-qboolean S_LoadSound (sfx_t *s, qboolean complain)
+qboolean S_LoadSound (sfx_t *sfx, 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 (s->fetcher != NULL)
-       {
-               if (s->format.speed != shm->format.speed)
-                       Sys_Error ("S_LoadSound: sound %s hasn't been resampled (%uHz instead of %uHz)", s->name);
+       // See if already loaded
+       if (sfx->fetcher != NULL)
                return true;
-       }
 
-       len = strlcpy (namebuffer, s->name, sizeof (namebuffer));
-       if (len >= sizeof (namebuffer))
+       // If we weren't able to load it previously, no need to retry
+       // Note: S_PrecacheSound clears this flag to cause a retry
+       if (sfx->flags & SFXFLAG_FILEMISSING)
                return false;
 
-       // Try to load it as a WAV file
-       if (S_LoadWavFile (namebuffer, s))
-               return true;
+       // No sound?
+       if (snd_renderbuffer == NULL)
+               return false;
 
-       // Else, try to load it as an Ogg Vorbis file
-       if (!strcasecmp (namebuffer + len - 4, ".wav"))
+       // Initialize volume peak to 0; if ReplayGain is supported, the loader will change this away
+       sfx->volume_peak = 0.0;
+
+       if (developer_loading.integer)
+               Con_Printf("loading sound %s\n", sfx->name);
+
+       SCR_PushLoadingScreen(true, sfx->name, 1);
+
+       // LordHavoc: if the sound filename does not begin with sound/, try adding it
+       if (strncasecmp(sfx->name, "sound/", 6))
        {
-               strcpy (namebuffer + len - 3, "ogg");
-               modified_name = true;
+               dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", sfx->name);
+               len = strlen(namebuffer);
+               if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
+               {
+                       if (S_LoadWavFile (namebuffer, sfx))
+                               goto loaded;
+                       memcpy (namebuffer + len - 3, "ogg", 4);
+               }
+               if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".ogg"))
+               {
+                       if (OGG_LoadVorbisFile (namebuffer, sfx))
+                               goto loaded;
+               }
        }
-       if (OGG_LoadVorbisFile (namebuffer, s))
-               return true;
 
-       // Can't load the sound!
-       if (!complain)
-               s->flags |= SFXFLAG_SILENTLYMISSING;
-       else
-               s->flags &= ~SFXFLAG_SILENTLYMISSING;
-       if (complain)
+       // LordHavoc: then try without the added sound/ as wav and ogg
+       dpsnprintf (namebuffer, sizeof(namebuffer), "%s", sfx->name);
+       len = strlen(namebuffer);
+       // request foo.wav: tries foo.wav, then foo.ogg
+       // request foo.ogg: tries foo.ogg only
+       // request foo.mod: tries foo.mod only
+       if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
        {
-               if (modified_name)
-                       strcpy (namebuffer + len - 3, "wav");
-               Con_Printf("Couldn't load %s\n", namebuffer);
+               if (S_LoadWavFile (namebuffer, sfx))
+                       goto loaded;
+               memcpy (namebuffer + len - 3, "ogg", 4);
        }
-       return false;
-}
-
-void S_UnloadSound(sfx_t *s)
-{
-       if (s->fetcher != NULL)
+       if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".ogg"))
        {
-               unsigned int i;
+               if (OGG_LoadVorbisFile (namebuffer, sfx))
+                       goto loaded;
+       }
 
-               s->fetcher = NULL;
-               s->fetcher_data = NULL;
-               Mem_FreePool(&s->mempool);
+       // Can't load the sound!
+       sfx->flags |= SFXFLAG_FILEMISSING;
+       if (complain)
+               Con_DPrintf("failed to load sound \"%s\"\n", sfx->name);
 
-               // 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;
-       }
+       SCR_PopLoadingScreen(false);
+       return false;
+
+loaded:
+       SCR_PopLoadingScreen(false);
+       return true;
 }