/*
- Copyright (C) 2003-2004 Mathieu Olivier
+ Copyright (C) 2003-2005 Mathieu Olivier
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
#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
{
- qbyte *buffer;
+ unsigned char *buffer;
ogg_int64_t ind, buffsize;
} ov_decode_t;
*/
qboolean OGG_OpenLibrary (void)
{
- const char* dllname;
+ const char* dllnames_vo [] =
+ {
+#if defined(WIN64)
+ "libvorbis64.dll",
+#elif defined(WIN32)
+ "libvorbis.dll",
+ "vorbis.dll",
+#elif defined(MACOSX)
+ "libvorbis.dylib",
+#else
+ "libvorbis.so.0",
+ "libvorbis.so",
+#endif
+ NULL
+ };
+ const char* dllnames_vf [] =
+ {
+#if defined(WIN64)
+ "libvorbisfile64.dll",
+#elif defined(WIN32)
+ "libvorbisfile.dll",
+ "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
+#define STREAM_BUFFER_SIZE(format_ptr) ((int)(ceil (STREAM_BUFFER_DURATION * ((format_ptr)->speed * (format_ptr)->width * (format_ptr)->channels))))
-// 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 unsigned char resampling_buffer [48000 * 2 * 2];
// Per-sfx data structure
typedef struct
{
- qbyte *file;
- size_t filesize;
+ unsigned char *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;
+ snd_format_t* format;
+ ogg_stream_persfx_t* per_sfx;
int newlength, done, ret, bigendian;
unsigned int factor;
+ size_t buff_len;
- per_ch = ch->fetcher_data;
+ 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)
{
- sfx_t* sfx;
- vorbis_info *vi;
+ size_t memsize;
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_sfx = sfx->fetcher_data;
+ memsize = sizeof (*per_ch) - sizeof (per_ch->sb.data) + 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;
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;
- // Decompress in the resampling_buffer to get STREAM_BUFFER_SIZE samples after resampling
-#if BYTE_ORDER == LITTLE_ENDIAN
- bigendian = 0;
-#else
+ // 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
+#if BYTE_ORDER == BIG_ENDIAN
bigendian = 1;
+#else
+ bigendian = 0;
#endif
done = 0;
- while ((ret = qov_read (&per_ch->vf, &resampling_buffer[done], (int)(newlength - done), bigendian, 2, 1, &per_ch->bs)) > 0)
+ 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 = ResampleSfx (resampling_buffer, (size_t)done / factor, &per_ch->format, sb->data + sb->length * factor, ch->sfx->name);
- sb->length += newlength;
+ 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;
return sb;
}
{
ogg_stream_perchannel_t* per_ch;
- per_ch = ch->fetcher_data;
+ per_ch = (ogg_stream_perchannel_t *)ch->fetcher_data;
if (per_ch != NULL)
{
+ size_t buff_len;
+ snd_format_t* format;
+
// Free the ogg vorbis decoder
qov_clear (&per_ch->vf);
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;
}
}
-static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd };
-extern snd_fetcher_t wav_fetcher;
+
+/*
+====================
+OGG_FreeSfx
+====================
+*/
+static void OGG_FreeSfx (sfx_t* sfx)
+{
+ ogg_stream_persfx_t* per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
+
+ // Free the Ogg Vorbis file
+ Mem_Free(per_sfx->file);
+ sfx->memsize -= per_sfx->filesize;
+
+ // Free the stream structure
+ Mem_Free(per_sfx);
+ sfx->memsize -= sizeof (*per_sfx);
+
+ sfx->fetcher_data = NULL;
+ sfx->fetcher = NULL;
+}
+
+static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd, OGG_FreeSfx };
/*
*/
qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s)
{
- qbyte *data;
+ unsigned char *data;
+ fs_offset_t filesize;
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);
+ // Already loaded?
+ if (s->fetcher != NULL)
+ return true;
// Load the file
- data = FS_LoadFile (filename, s->mempool, false);
+ data = FS_LoadFile (filename, snd_mempool, false, &filesize);
if (data == NULL)
- {
- Mem_FreePool (&s->mempool);
return false;
- }
Con_DPrintf ("Loading Ogg Vorbis file \"%s\"\n", filename);
// Open it with the VorbisFile API
ov_decode.buffer = data;
ov_decode.ind = 0;
- ov_decode.buffsize = fs_filesize;
+ ov_decode.buffsize = filesize;
if (qov_open_callbacks (&ov_decode, &vf, NULL, 0, callbacks) < 0)
{
Con_Printf ("error while opening Ogg Vorbis file \"%s\"\n", filename);
- Mem_FreePool (&s->mempool);
+ Mem_Free(data);
return false;
}
Con_Printf("%s has an unsupported number of channels (%i)\n",
s->name, vi->channels);
qov_clear (&vf);
- Mem_FreePool (&s->mempool);
+ 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
- if (snd_streaming.integer && len > fs_filesize + 3 * STREAM_BUFFER_SIZE)
+ buff_len = (int)ceil (STREAM_BUFFER_DURATION * (shm->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 = Mem_Alloc (s->mempool, sizeof (*per_sfx));
+ per_sfx = (ogg_stream_persfx_t *)Mem_Alloc (snd_mempool, sizeof (*per_sfx));
+ s->memsize += sizeof (*per_sfx);
per_sfx->file = data;
- per_sfx->filesize = fs_filesize;
+ per_sfx->filesize = filesize;
+ s->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->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 = (int)((size_t)len / per_sfx->format.channels / 2 * ((float)s->format.speed / per_sfx->format.speed));
}
else
{
int bs, bigendian;
long ret;
sfxbuffer_t *sb;
+ size_t memsize;
- 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);
+ buff = (char *)Mem_Alloc (snd_mempool, (int)len);
done = 0;
bs = 0;
#if BYTE_ORDER == LITTLE_ENDIAN
done += ret;
// Calculate resampled length
- len = (double)done * (double)shm->format.speed / (double)vi->rate;
+ // FIXME: is this using the correct rounding direction? ceil may be better
+ len = (int)((double)done * (double)shm->format.speed / (double)vi->rate);
// Resample it
- sb = Mem_Alloc (s->mempool, (size_t)len + sizeof (*sb) - sizeof (sb->data));
+ 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 = ResampleSfx (buff, (size_t)done / (vi->channels * 2), &s->format, sb->data, s->name);
+ 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;