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;
- // this is usually 0.5 (128), 1 (256), or 2 (512)
- fracstep = ((double) in_format->speed / (double) shm->format.speed) * 256.0;
+ // If the caller provides a buffer, it must give us its size
+ if (sampleframes == 0 && buffer != NULL)
+ return NULL;
- srclength = in_length * in_format->channels;
+ ringbuffer = (snd_ringbuffer_t*)Mem_Alloc(snd_mempool, sizeof (*ringbuffer));
+ memset(ringbuffer, 0, sizeof(*ringbuffer));
+ memcpy(&ringbuffer->format, format, sizeof(ringbuffer->format));
- 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);
+ // If we haven't been given a buffer
+ if (buffer == NULL)
+ {
+ unsigned int maxframes;
+ size_t memsize;
-// resample / decimate to the current source rate
+ if (sampleframes == 0)
+ maxframes = (format->speed + 1) / 2; // Make the sound buffer large enough for containing 0.5 sec of sound
+ else
+ maxframes = sampleframes;
- if (fracstep == 256)
+ memsize = maxframes * format->width * format->channels;
+ ringbuffer->ring = Mem_Alloc(snd_mempool, memsize);
+ ringbuffer->maxframes = maxframes;
+ }
+ else
{
- // 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];
+ ringbuffer->ring = buffer;
+ ringbuffer->maxframes = sampleframes;
}
+
+ return ringbuffer;
+}
+
+
+/*
+====================
+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 = (double)sampleframes * (double)sb_speed / (double)in_format->speed;
+
+ memsize = newsampleframes * in_format->channels * in_format->width;
+ memsize += sizeof (*sb) - sizeof (sb->samples);
+
+ 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 = newsampleframes;
+ sb->nbframes = 0;
+
+ if (!Snd_AppendToSndBuffer (sb, samples, sampleframes, in_format))
+ {
+ 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 = (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 += outcount;
+ return true;
}
+
//=============================================================================
/*
S_LoadSound
==============
*/
-qboolean S_LoadSound (sfx_t *s, int 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 = snprintf (namebuffer, sizeof (namebuffer), "sound/%s", s->name);
- 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;
+
+ // LordHavoc: if the sound filename does not begin with sound/, try adding it
+ if (strncasecmp(sfx->name, "sound/", 6))
+ {
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "sound/%s", sfx->name);
+ if (len < 0)
+ {
+ // name too long
+ Con_DPrintf("S_LoadSound: name \"%s\" is too long\n", sfx->name);
+ return false;
+ }
+ if (S_LoadWavFile (namebuffer, sfx))
+ return true;
+ if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
+ memcpy (namebuffer + len - 3, "ogg", 4);
+ if (OGG_LoadVorbisFile (namebuffer, sfx))
+ return true;
+ }
- // Else, try to load it as an Ogg Vorbis file
- if (!strcasecmp (namebuffer + len - 4, ".wav"))
+ // LordHavoc: then try without the added sound/ as wav and ogg
+ len = dpsnprintf (namebuffer, sizeof(namebuffer), "%s", sfx->name);
+ if (len < 0)
{
- strcpy (namebuffer + len - 3, "ogg");
- modified_name = true;
+ // name too long
+ Con_DPrintf("S_LoadSound: name \"%s\" is too long\n", sfx->name);
+ return false;
}
- if (OGG_LoadVorbisFile (namebuffer, s))
+ if (S_LoadWavFile (namebuffer, sfx))
+ return true;
+ if (len >= 4 && !strcasecmp (namebuffer + len - 4, ".wav"))
+ memcpy (namebuffer + len - 3, "ogg", 4);
+ if (OGG_LoadVorbisFile (namebuffer, sfx))
return true;
// Can't load the sound!
- if (!complain)
- s->flags |= SFXFLAG_SILENTLYMISSING;
- else
- s->flags &= ~SFXFLAG_SILENTLYMISSING;
+ sfx->flags |= SFXFLAG_FILEMISSING;
if (complain)
- {
- if (modified_name)
- strcpy (namebuffer + len - 3, "wav");
- Con_Printf("Couldn't load %s\n", namebuffer);
- }
+ Con_DPrintf("S_LoadSound: Couldn't load \"%s\"\n", sfx->name);
return false;
}
-
-void S_UnloadSound(sfx_t *s)
-{
- if (s->fetcher != NULL)
- {
- unsigned int 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;
- }
-}
projects / xonotic / darkplaces.git / blobdiff