*/
#define STREAM_BUFFER_DURATION 1.5f // 1.5 sec
-#define STREAM_BUFFER_SIZE(format_ptr) (ceil (STREAM_BUFFER_DURATION * ((format_ptr)->speed * (format_ptr)->width * (format_ptr)->channels)))
+#define STREAM_BUFFER_SIZE(format_ptr) ((int)(ceil (STREAM_BUFFER_DURATION * ((format_ptr)->speed * (format_ptr)->width * (format_ptr)->channels))))
// 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)
// Per-sfx data structure
typedef struct
{
- unsigned char *file;
+ unsigned char *file;
size_t filesize;
snd_format_t format;
} ogg_stream_persfx_t;
{
OggVorbis_File vf;
ov_decode_t ov_decode;
+ unsigned int sb_offset;
int bs;
- sfxbuffer_t sb; // must be at the end due to its dynamically allocated size
+ snd_buffer_t sb; // must be at the end due to its dynamically allocated size
} ogg_stream_perchannel_t;
OGG_FetchSound
====================
*/
-static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples)
+static const snd_buffer_t* OGG_FetchSound (channel_t* ch, unsigned int* start, unsigned int nbsampleframes)
{
ogg_stream_perchannel_t* per_ch;
- sfxbuffer_t* sb;
sfx_t* sfx;
- snd_format_t* format;
ogg_stream_persfx_t* per_sfx;
+ snd_buffer_t* sb;
int newlength, done, ret, bigendian;
+ unsigned int real_start;
unsigned int factor;
- size_t buff_len;
per_ch = (ogg_stream_perchannel_t *)ch->fetcher_data;
sfx = ch->sfx;
per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
- format = &sfx->format;
- buff_len = STREAM_BUFFER_SIZE(format);
// If there's no fetcher structure attached to the channel yet
if (per_ch == NULL)
{
- size_t memsize;
- ogg_stream_persfx_t* per_sfx;
+ size_t buff_len, memsize;
+ snd_format_t sb_format;
+
+ sb_format.speed = snd_renderbuffer->format.speed;
+ sb_format.width = per_sfx->format.width;
+ sb_format.channels = per_sfx->format.channels;
- memsize = sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len;
+ buff_len = STREAM_BUFFER_SIZE(&sb_format);
+ memsize = sizeof (*per_ch) - sizeof (per_ch->sb.samples) + buff_len;
per_ch = (ogg_stream_perchannel_t *)Mem_Alloc (snd_mempool, memsize);
sfx->memsize += memsize;
- per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
// Open it with the VorbisFile API
per_ch->ov_decode.buffer = per_sfx->file;
Mem_Free (per_ch);
return NULL;
}
-
- per_ch->sb.offset = 0;
- per_ch->sb.length = 0;
per_ch->bs = 0;
+ per_ch->sb_offset = 0;
+ per_ch->sb.format = sb_format;
+ per_ch->sb.nbframes = 0;
+ per_ch->sb.maxframes = buff_len / (per_ch->sb.format.channels * per_ch->sb.format.width);
+
ch->fetcher_data = per_ch;
}
+
+ real_start = *start;
sb = &per_ch->sb;
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)
+ if (nbsampleframes > sb->maxframes)
{
- Con_Printf ("OGG_FetchSound: stream buffer too small (%u bytes required)\n", nbsamples * factor);
+ Con_Printf ("OGG_FetchSound: stream buffer too small (%u sample frames required)\n", nbsampleframes);
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)
+ if (per_ch->sb_offset <= real_start && per_ch->sb_offset + sb->nbframes >= real_start + nbsampleframes)
+ {
+ *start = per_ch->sb_offset;
return sb;
+ }
- newlength = (int)(sb->offset + sb->length) - start;
+ newlength = (int)(per_ch->sb_offset + sb->nbframes) - real_start;
// If we need to skip some data before decompressing the rest, or if the stream has looped
- if (newlength < 0 || sb->offset > start)
+ if (newlength < 0 || per_ch->sb_offset > real_start)
{
- if (qov_pcm_seek (&per_ch->vf, (ogg_int64_t)start) != 0)
+ unsigned int time_start;
+ ogg_int64_t ogg_start;
+ int err;
+
+ if (real_start > sfx->total_length)
+ {
+ Con_Printf ("OGG_FetchSound: asked for a start position after the end of the sfx! (%u > %u)\n",
+ real_start, sfx->total_length);
return NULL;
- sb->length = 0;
+ }
+
+ // We work with 200ms (1/5 sec) steps to avoid rounding errors
+ time_start = real_start * 5 / snd_renderbuffer->format.speed;
+ ogg_start = time_start * (per_sfx->format.speed / 5);
+ err = qov_pcm_seek (&per_ch->vf, ogg_start);
+ if (err != 0)
+ {
+ Con_Printf ("OGG_FetchSound: qov_pcm_seek(..., %d) returned %d\n",
+ real_start, err);
+ return NULL;
+ }
+ sb->nbframes = 0;
+
+ real_start = (float)ogg_start / per_sfx->format.speed * snd_renderbuffer->format.speed;
+ if (*start - real_start + nbsampleframes > sb->maxframes)
+ {
+ Con_Printf ("OGG_FetchSound: stream buffer too small after seek (%u sample frames required)\n",
+ *start - real_start + nbsampleframes);
+ per_ch->sb_offset = real_start;
+ return NULL;
+ }
}
- // Else, move forward the samples we need to keep in the sfxbuffer
+ // Else, move forward the samples we need to keep in the sound buffer
else
{
- memmove (sb->data, sb->data + (start - sb->offset) * factor, newlength * factor);
- sb->length = newlength;
+ memmove (sb->samples, sb->samples + (real_start - per_ch->sb_offset) * factor, newlength * factor);
+ sb->nbframes = newlength;
}
- sb->offset = start;
+ per_ch->sb_offset = real_start;
// 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
- if ((sfx->format.speed + sb->length) * factor > buff_len)
+ if (sb->format.speed + sb->nbframes > sb->maxframes)
{
- Con_Printf ("OGG_FetchSound: stream buffer overflow (%u bytes / %u)\n",
- (sfx->format.speed + sb->length) * factor, buff_len);
+ Con_Printf ("OGG_FetchSound: stream buffer overflow (%u sample frames / %u)\n",
+ sb->format.speed + sb->nbframes, sb->maxframes);
return NULL;
}
newlength = per_sfx->format.speed * factor; // -> 1 sec of sound before resampling
while ((ret = qov_read (&per_ch->vf, (char *)&resampling_buffer[done], (int)(newlength - done), bigendian, 2, 1, &per_ch->bs)) > 0)
done += ret;
- // Resample in the sfxbuffer
- newlength = (int)ResampleSfx (resampling_buffer, (size_t)done / (size_t)factor, &per_sfx->format, sb->data + sb->length * (size_t)factor, sfx->name);
- sb->length += newlength;
+ Snd_AppendToSndBuffer (sb, resampling_buffer, (size_t)done / (size_t)factor, &per_sfx->format);
+ *start = per_ch->sb_offset;
return sb;
}
if (per_ch != NULL)
{
size_t buff_len;
- snd_format_t* format;
// Free the ogg vorbis decoder
qov_clear (&per_ch->vf);
+ buff_len = per_ch->sb.maxframes * per_ch->sb.format.channels * per_ch->sb.format.width;
+ ch->sfx->memsize -= sizeof (*per_ch) - sizeof (per_ch->sb.samples) + buff_len;
+
Mem_Free (per_ch);
ch->fetcher_data = NULL;
-
- format = &ch->sfx->format;
- buff_len = STREAM_BUFFER_SIZE(format);
- ch->sfx->memsize -= sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len;
}
}
sfx->fetcher = NULL;
}
-static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd, OGG_FreeSfx };
+
+/*
+====================
+OGG_GetFormat
+====================
+*/
+static const snd_format_t* OGG_GetFormat (sfx_t* sfx)
+{
+ ogg_stream_persfx_t* per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
+ return &per_sfx->format;
+}
+
+static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd, OGG_FreeSfx, OGG_GetFormat };
/*
Load an Ogg Vorbis file into memory
====================
*/
-qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s)
+qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
{
unsigned char *data;
fs_offset_t filesize;
return false;
// Already loaded?
- if (s->fetcher != NULL)
+ if (sfx->fetcher != NULL)
return true;
// Load the file
if (vi->channels < 1 || vi->channels > 2)
{
Con_Printf("%s has an unsupported number of channels (%i)\n",
- s->name, vi->channels);
+ sfx->name, vi->channels);
qov_clear (&vf);
Mem_Free(data);
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
- buff_len = ceil (STREAM_BUFFER_DURATION * (shm->format.speed * 2 * vi->channels));
+ buff_len = (int)ceil (STREAM_BUFFER_DURATION * (snd_renderbuffer->format.speed * 2 * vi->channels));
if (snd_streaming.integer && len > (ogg_int64_t)filesize + 3 * buff_len)
{
ogg_stream_persfx_t* per_sfx;
Con_DPrintf ("\"%s\" will be streamed\n", filename);
per_sfx = (ogg_stream_persfx_t *)Mem_Alloc (snd_mempool, sizeof (*per_sfx));
- s->memsize += sizeof (*per_sfx);
+ sfx->memsize += sizeof (*per_sfx);
per_sfx->file = data;
per_sfx->filesize = filesize;
- s->memsize += filesize;
+ sfx->memsize += 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->loopstart = -1;
- s->flags |= SFXFLAG_STREAMED;
- s->total_length = (size_t)len / per_sfx->format.channels / 2 * ((float)s->format.speed / per_sfx->format.speed);
+
+ sfx->fetcher_data = per_sfx;
+ sfx->fetcher = &ogg_fetcher;
+ sfx->loopstart = -1;
+ sfx->flags |= SFXFLAG_STREAMED;
+ sfx->total_length = (int)((size_t)len / (per_sfx->format.channels * 2) * ((double)snd_renderbuffer->format.speed / per_sfx->format.speed));
}
else
{
ogg_int64_t done;
int bs, bigendian;
long ret;
- sfxbuffer_t *sb;
- size_t memsize;
+ snd_buffer_t *sb;
+ snd_format_t ogg_format;
Con_DPrintf ("\"%s\" will be cached\n", filename);
buff = (char *)Mem_Alloc (snd_mempool, (int)len);
done = 0;
bs = 0;
-#if BYTE_ORDER == LITTLE_ENDIAN
- bigendian = 0;
-#else
+#if BYTE_ORDER == BIG_ENDIAN
bigendian = 1;
+#else
+ bigendian = 0;
#endif
while ((ret = qov_read (&vf, &buff[done], (int)(len - done), bigendian, 2, 1, &bs)) > 0)
done += ret;
- // Calculate resampled length
- len = (double)done * (double)shm->format.speed / (double)vi->rate;
-
- // Resample it
- memsize = (size_t)len + sizeof (*sb) - sizeof (sb->data);
- sb = (sfxbuffer_t *)Mem_Alloc (snd_mempool, memsize);
- s->memsize += memsize;
- s->fetcher_data = sb;
- s->fetcher = &wav_fetcher;
- s->format.speed = vi->rate;
- 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 = (unsigned int)ResampleSfx ((unsigned char *)buff, (size_t)done / (vi->channels * 2), &s->format, sb->data, s->name);
- s->format.speed = shm->format.speed;
- s->total_length = sb->length;
- sb->offset = 0;
+ // Build the sound buffer
+ ogg_format.speed = vi->rate;
+ ogg_format.channels = vi->channels;
+ ogg_format.width = 2; // We always work with 16 bits samples
+ sb = Snd_CreateSndBuffer ((unsigned char *)buff, (size_t)done / (vi->channels * 2), &ogg_format, snd_renderbuffer->format.speed);
+ if (sb == NULL)
+ {
+ qov_clear (&vf);
+ Mem_Free (data);
+ Mem_Free (buff);
+ return false;
+ }
+
+ sfx->fetcher = &wav_fetcher;
+ sfx->fetcher_data = sb;
+
+ sfx->total_length = sb->nbframes;
+ sfx->memsize += sb->maxframes * sb->format.channels * sb->format.width + sizeof (*sb) - sizeof (sb->samples);
+
+ sfx->loopstart = -1;
+ sfx->flags &= ~SFXFLAG_STREAMED;
qov_clear (&vf);
Mem_Free (data);
projects / xonotic / darkplaces.git / blobdiff