#include "quakedef.h"
#include "snd_ogg.h"
+#include "snd_wav.h"
/*
qboolean OGG_OpenLibrary (void)
{
const char* dllname;
- const dllfunction_t *func;
// Already loaded?
if (vf_dll)
dllname = "libvorbisfile.so";
#endif
- // Initializations
- for (func = oggvorbisfuncs; func && func->name != NULL; func++)
- *func->funcvariable = NULL;
-
// Load the DLL
- if (! (vf_dll = Sys_LoadLibrary (dllname)))
+ if (! Sys_LoadLibrary (dllname, &vf_dll, oggvorbisfuncs))
{
- Con_DPrintf("Can't find %s. Ogg Vorbis support disabled\n", dllname);
+ Con_Printf ("Ogg Vorbis support disabled\n");
return false;
}
- // Get the function adresses
- for (func = oggvorbisfuncs; func && func->name != NULL; func++)
- if (!(*func->funcvariable = (void *) Sys_GetProcAddress (vf_dll, func->name)))
- {
- Con_Printf("missing function \"%s\" - broken Ogg Vorbis library!\n", func->name);
- OGG_CloseLibrary ();
- return false;
- }
-
- Con_DPrintf("%s loaded. Ogg Vorbis support enabled\n", dllname);
+ Con_Printf ("Ogg Vorbis support enabled\n");
return true;
}
*/
void OGG_CloseLibrary (void)
{
- if (!vf_dll)
- return;
-
- Sys_UnloadLibrary (vf_dll);
- vf_dll = NULL;
+ Sys_UnloadLibrary (&vf_dll);
}
=================================================================
*/
-#define STREAM_BUFFER_SIZE (128 * 1024)
+#define STREAM_BUFFER_DURATION 1.5f // 1.5 sec
-// Note: it must be able to contain enough samples at 48 KHz (max speed)
-// to fill STREAM_BUFFER_SIZE bytes of samples at 8 KHz (min speed)
-// TODO: dynamically allocate this buffer depending on the shm and min sound speeds
-static qbyte resampling_buffer [STREAM_BUFFER_SIZE * (48000 / 8000)];
+// We work with 1 sec sequences, so this buffer must be able to contain
+// 1 sec of sound of the highest quality (48 KHz, 16 bit samples, stereo)
+static qbyte resampling_buffer [48000 * 2 * 2];
// Per-sfx data structure
{
ogg_stream_perchannel_t* per_ch;
sfxbuffer_t* sb;
+ sfx_t* sfx;
int newlength, done, ret, bigendian;
unsigned int factor;
+ size_t buff_len;
per_ch = ch->fetcher_data;
+ sfx = ch->sfx;
+ buff_len = ceil (STREAM_BUFFER_DURATION * (sfx->format.speed * sfx->format.width * sfx->format.channels));
// If there's no fetcher structure attached to the channel yet
if (per_ch == NULL)
{
- sfx_t* sfx;
vorbis_info *vi;
ogg_stream_persfx_t* per_sfx;
- sfx = ch->sfx;
- per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + STREAM_BUFFER_SIZE);
+ per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len);
per_sfx = sfx->fetcher_data;
// Open it with the VorbisFile API
per_ch->format.speed = vi->rate;
per_ch->format.width = sfx->format.width;
per_ch->format.channels = sfx->format.channels;
-
+
per_ch->sb.offset = 0;
per_ch->sb.length = 0;
per_ch->bs = 0;
}
sb = &per_ch->sb;
+ factor = per_ch->format.width * per_ch->format.channels;
+
+ // If the stream buffer can't contain that much samples anyway
+ if (nbsamples * factor > buff_len)
+ {
+ Con_Printf ("OGG_FetchSound: stream buffer too small (%u bytes required)\n", nbsamples * factor);
+ return NULL;
+ }
// If the data we need has already been decompressed in the sfxbuffer, just return it
if (sb->offset <= start && sb->offset + sb->length >= start + nbsamples)
return sb;
newlength = sb->offset + sb->length - start;
- factor = per_ch->format.width * per_ch->format.channels;
// If we need to skip some data before decompressing the rest, or if the stream has looped
if (newlength < 0 || sb->offset > start)
sb->length = newlength;
}
- // How many free bytes do we have in the sfxbuffer now?
- newlength = STREAM_BUFFER_SIZE - (newlength * factor);
+ // We add exactly "per_ch->format.speed" samples per channel to the buffer (i.e. 1 sec of sound):
+ // 1- to ensure we won't lose any sample during the resampling process
+ // 2- to force one call to OGG_FetchSound per second to regulate the workload
+ newlength = per_ch->format.speed * factor;
+ if (newlength + sb->length * factor > buff_len)
+ {
+ Con_Printf ("OGG_FetchSound: stream buffer overflow (%u bytes / %u)\n",
+ newlength + sb->length * factor, buff_len);
+ return NULL;
+ }
- // Decompress in the resampling_buffer to get STREAM_BUFFER_SIZE samples after resampling
+ // Decompress in the resampling_buffer
#if BYTE_ORDER == LITTLE_ENDIAN
bigendian = 0;
#else
done += ret;
// Resample in the sfxbuffer
- newlength = ResampleSfx (resampling_buffer, (size_t)done / factor, &per_ch->format, sb->data + sb->length * factor, ch->sfx->name);
+ newlength = ResampleSfx (resampling_buffer, (size_t)done / factor, &per_ch->format, sb->data + sb->length * factor, sfx->name);
sb->length += newlength;
return sb;
}
static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd };
-extern snd_fetcher_t wav_fetcher;
/*
ov_decode_t ov_decode;
OggVorbis_File vf;
vorbis_info *vi;
- ogg_int64_t len;
+ ogg_int64_t len, buff_len;
if (!vf_dll)
return false;
len = qov_pcm_total (&vf, -1) * vi->channels * 2; // 16 bits => "* 2"
// Decide if we go for a stream or a simple PCM cache
- if (snd_streaming.integer && len > fs_filesize + 3 * STREAM_BUFFER_SIZE)
+ buff_len = ceil (STREAM_BUFFER_DURATION * (shm->format.speed * 2 * vi->channels));
+ if (snd_streaming.integer && len > fs_filesize + 3 * buff_len)
{
ogg_stream_persfx_t* per_sfx;
s->format.width = 2; // We always work with 16 bits samples
s->format.channels = vi->channels;
s->loopstart = -1;
- s->total_length = (size_t)len / (vi->channels * 2) * (float)(shm->format.speed / vi->rate);
+ s->flags |= SFXFLAG_STREAMED;
+ s->total_length = (size_t)len / (vi->channels * 2) * ((float)shm->format.speed / vi->rate);
}
else
{
s->format.width = 2; // We always work with 16 bits samples
s->format.channels = vi->channels;
s->loopstart = -1;
+ s->flags &= ~SFXFLAG_STREAMED;
sb->length = ResampleSfx (buff, (size_t)done / (vi->channels * 2), &s->format, sb->data, s->name);
s->format.speed = shm->format.speed;
projects / xonotic / darkplaces.git / blobdiff