// Per-sfx data structure
typedef struct
{
- qbyte *file;
- size_t filesize;
+ qbyte *file;
+ size_t filesize;
+ snd_format_t format;
} ogg_stream_persfx_t;
// Per-channel data structure
OggVorbis_File vf;
ov_decode_t ov_decode;
int bs;
- snd_format_t format;
sfxbuffer_t sb; // must be at the end due to its dynamically allocated size
} ogg_stream_perchannel_t;
ogg_stream_perchannel_t* per_ch;
sfxbuffer_t* sb;
sfx_t* sfx;
+ ogg_stream_persfx_t* per_sfx;
int newlength, done, ret, bigendian;
unsigned int factor;
size_t buff_len;
per_ch = ch->fetcher_data;
sfx = ch->sfx;
+ per_sfx = sfx->fetcher_data;
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)
{
- vorbis_info *vi;
ogg_stream_persfx_t* per_sfx;
per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len);
return NULL;
}
- // Get the stream information
- vi = qov_info (&per_ch->vf, -1);
- 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;
+ factor = per_sfx->format.width * per_sfx->format.channels;
// If the stream buffer can't contain that much samples anyway
if (nbsamples * factor > buff_len)
sb->length = newlength;
}
- // We add exactly "per_ch->format.speed" samples per channel to the buffer (i.e. 1 sec of sound):
+ // We add exactly 1 sec of sound to the buffer:
// 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)
+ if ((sfx->format.speed + sb->length) * factor > buff_len)
{
Con_Printf ("OGG_FetchSound: stream buffer overflow (%u bytes / %u)\n",
- newlength + sb->length * factor, buff_len);
+ (sfx->format.speed + sb->length) * factor, buff_len);
return NULL;
}
+ newlength = per_sfx->format.speed * factor; // 1 sec of sound before resampling
// Decompress in the resampling_buffer
#if BYTE_ORDER == LITTLE_ENDIAN
done += ret;
// Resample in the sfxbuffer
- newlength = ResampleSfx (resampling_buffer, (size_t)done / factor, &per_ch->format, sb->data + sb->length * factor, sfx->name);
+ newlength = ResampleSfx (resampling_buffer, (size_t)done / factor, &per_sfx->format, sb->data + sb->length * factor, sfx->name);
sb->length += newlength;
return sb;
per_sfx = Mem_Alloc (s->mempool, sizeof (*per_sfx));
per_sfx->file = data;
per_sfx->filesize = fs_filesize;
+
+ per_sfx->format.speed = vi->rate;
+ per_sfx->format.width = 2; // We always work with 16 bits samples
+ per_sfx->format.channels = vi->channels;
+ s->format.speed = shm->format.speed;
+ s->format.width = per_sfx->format.width;
+ s->format.channels = per_sfx->format.channels;
+
s->fetcher_data = per_sfx;
s->fetcher = &ogg_fetcher;
- s->format.speed = shm->format.speed;
- s->format.width = 2; // We always work with 16 bits samples
- s->format.channels = vi->channels;
s->loopstart = -1;
s->flags |= SFXFLAG_STREAMED;
- s->total_length = (size_t)len / (vi->channels * 2) * ((float)shm->format.speed / vi->rate);
+ s->total_length = (size_t)len / per_sfx->format.channels / 2 * ((float)s->format.speed / per_sfx->format.speed);
}
else
{