#include "quakedef.h"
+#include "snd_main.h"
#include "snd_ogg.h"
#include "snd_wav.h"
{NULL, NULL}
};
-// Handle for the Vorbisfile DLL
+// Handles for the Vorbis and Vorbisfile DLLs
+static dllhandle_t vo_dll = NULL;
static dllhandle_t vf_dll = NULL;
typedef struct
*/
qboolean OGG_OpenLibrary (void)
{
- const char* dllname;
+ const char* dllnames_vo [] =
+ {
+#ifdef WIN32
+ "vorbis.dll",
+#elif defined(MACOSX)
+ "libvorbis.dylib",
+#else
+ "libvorbis.so.0",
+ "libvorbis.so",
+#endif
+ NULL
+ };
+ const char* dllnames_vf [] =
+ {
+#ifdef WIN32
+ "vorbisfile.dll",
+#elif defined(MACOSX)
+ "libvorbisfile.dylib",
+#else
+ "libvorbisfile.so.3",
+ "libvorbisfile.so",
+#endif
+ NULL
+ };
// Already loaded?
if (vf_dll)
return true;
-#ifdef WIN32
- dllname = "vorbisfile.dll";
-#else
- dllname = "libvorbisfile.so";
-#endif
+// COMMANDLINEOPTION: Sound: -novorbis disables ogg vorbis sound support
+ if (COM_CheckParm("-novorbis"))
+ return false;
- // Load the DLL
- if (! Sys_LoadLibrary (dllname, &vf_dll, oggvorbisfuncs))
+ // Load the DLLs
+ // We need to load both by hand because some OSes seem to not load
+ // the vorbis DLL automatically when loading the VorbisFile DLL
+ if (! Sys_LoadLibrary (dllnames_vo, &vo_dll, NULL) ||
+ ! Sys_LoadLibrary (dllnames_vf, &vf_dll, oggvorbisfuncs))
{
+ Sys_UnloadLibrary (&vo_dll);
Con_Printf ("Ogg Vorbis support disabled\n");
return false;
}
void OGG_CloseLibrary (void)
{
Sys_UnloadLibrary (&vf_dll);
+ Sys_UnloadLibrary (&vo_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
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)
{
- 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
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_sfx->format.width * per_sfx->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;
+ newlength = (int)(sb->offset + sb->length) - 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 (qov_pcm_seek (&per_ch->vf, (ogg_int64_t)start) != 0)
return NULL;
-
- sb->offset = start;
sb->length = 0;
- newlength = 0;
}
// Else, move forward the samples we need to keep in the sfxbuffer
else
{
memmove (sb->data, sb->data + (start - sb->offset) * factor, newlength * factor);
- sb->offset = start;
sb->length = newlength;
}
- // How many free bytes do we have in the sfxbuffer now?
- newlength = STREAM_BUFFER_SIZE - (newlength * factor);
+ sb->offset = 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)
+ {
+ Con_Printf ("OGG_FetchSound: stream buffer overflow (%u bytes / %u)\n",
+ (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 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);
- sb->length += newlength;
+ newlength = ResampleSfx (resampling_buffer, (size_t)done / (size_t)factor, &per_sfx->format, sb->data + sb->length * (size_t)factor, sfx->name);
+ sb->length += newlength;
return sb;
}
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;
Mem_FreePool (&s->mempool);
- s->mempool = Mem_AllocPool (s->name);
+ s->mempool = Mem_AllocPool (s->name, 0, NULL);
// Load the file
data = FS_LoadFile (filename, s->mempool, 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;
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->total_length = (size_t)len / (vi->channels * 2) * (float)(shm->format.speed / vi->rate);
+ s->flags |= SFXFLAG_STREAMED;
+ s->total_length = (size_t)len / per_sfx->format.channels / 2 * ((float)s->format.speed / per_sfx->format.speed);
}
else
{
long ret;
sfxbuffer_t *sb;
- Con_DPrintf ("\"%s\" will be streamed\n", filename);
+ Con_DPrintf ("\"%s\" will be cached\n", filename);
// Decode it
buff = Mem_Alloc (s->mempool, (int)len);
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