/*
- 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"
void *internal;
} vorbis_block;
+typedef struct
+{
+ char **user_comments;
+ int *comment_lengths;
+ int comments;
+ char *vendor;
+} vorbis_comment;
+
typedef struct
{
void *datasource;
long *serialnos;
ogg_int64_t *pcmlengths;
vorbis_info *vi;
- void *vc; // VOIDED POINTER
+ vorbis_comment *vc;
ogg_int64_t pcm_offset;
int ready_state;
long current_serialno;
// Functions exported from the vorbisfile library
static int (*qov_clear) (OggVorbis_File *vf);
static vorbis_info* (*qov_info) (OggVorbis_File *vf,int link);
+static vorbis_comment* (*qov_comment) (OggVorbis_File *vf,int link);
+static char * (*qvorbis_comment_query) (vorbis_comment *vc, char *tag, int count);
static int (*qov_open_callbacks) (void *datasource, OggVorbis_File *vf,
char *initial, long ibytes,
ov_callbacks callbacks);
static long (*qov_read) (OggVorbis_File *vf,char *buffer,int length,
int bigendianp,int word,int sgned,int *bitstream);
-static dllfunction_t oggvorbisfuncs[] =
+static dllfunction_t vorbisfilefuncs[] =
+{
+ {"ov_clear", (void **) &qov_clear},
+ {"ov_info", (void **) &qov_info},
+ {"ov_comment", (void **) &qov_comment},
+ {"ov_open_callbacks", (void **) &qov_open_callbacks},
+ {"ov_pcm_seek", (void **) &qov_pcm_seek},
+ {"ov_pcm_total", (void **) &qov_pcm_total},
+ {"ov_read", (void **) &qov_read},
+ {NULL, NULL}
+};
+
+static dllfunction_t vorbisfuncs[] =
{
- {"ov_clear", (void **) &qov_clear},
- {"ov_info", (void **) &qov_info},
- {"ov_open_callbacks", (void **) &qov_open_callbacks},
- {"ov_pcm_seek", (void **) &qov_pcm_seek},
- {"ov_pcm_total", (void **) &qov_pcm_total},
- {"ov_read", (void **) &qov_read},
+ {"vorbis_comment_query", (void **) &qvorbis_comment_query},
{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;
+// COMMANDLINEOPTION: Sound: -novorbis disables ogg vorbis sound support
if (COM_CheckParm("-novorbis"))
return false;
-#ifdef WIN32
- dllname = "vorbisfile.dll";
-#else
- dllname = "libvorbisfile.so";
-#endif
-
- // 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, vorbisfuncs) ||
+ ! Sys_LoadLibrary (dllnames_vf, &vf_dll, vorbisfilefuncs))
{
+ 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_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))))
// 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];
+static unsigned char resampling_buffer [48000 * 2 * 2];
// Per-sfx data structure
typedef struct
{
- qbyte *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;
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 = ch->fetcher_data;
+ per_ch = (ogg_stream_perchannel_t *)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));
+ per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
// If there's no fetcher structure attached to the channel yet
if (per_ch == NULL)
{
- 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;
- per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len);
- per_sfx = sfx->fetcher_data;
+ 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;
// 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 = 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 > (unsigned int)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;
+ }
+
+ // 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;
- sb->offset = start;
- sb->length = 0;
- newlength = 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->offset = start;
- sb->length = newlength;
+ memmove (sb->samples, sb->samples + (real_start - per_ch->sb_offset) * factor, newlength * factor);
+ sb->nbframes = newlength;
}
+ 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
+ newlength = per_sfx->format.speed * factor; // -> 1 sec of sound before resampling
+ if(newlength > sizeof(resampling_buffer))
+ newlength = sizeof(resampling_buffer);
// Decompress in the resampling_buffer
-#if BYTE_ORDER == LITTLE_ENDIAN
- bigendian = 0;
-#else
+#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_sfx->format, sb->data + sb->length * 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;
}
{
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;
+
// 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;
}
}
-static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd };
+
+/*
+====================
+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;
+}
+
+
+/*
+====================
+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)
{
- qbyte *data;
+ unsigned char *data;
+ const char *loopcomment;
+ fs_offset_t filesize;
ov_decode_t ov_decode;
OggVorbis_File vf;
vorbis_info *vi;
+ vorbis_comment *vc;
ogg_int64_t len, buff_len;
if (!vf_dll)
return false;
- Mem_FreePool (&s->mempool);
- s->mempool = Mem_AllocPool (s->name, 0, NULL);
+ // Already loaded?
+ if (sfx->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;
}
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_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
- buff_len = ceil (STREAM_BUFFER_DURATION * (shm->format.speed * 2 * vi->channels));
- if (snd_streaming.integer && len > fs_filesize + 3 * buff_len)
+ 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 = Mem_Alloc (s->mempool, sizeof (*per_sfx));
+ per_sfx = (ogg_stream_persfx_t *)Mem_Alloc (snd_mempool, sizeof (*per_sfx));
+ sfx->memsize += sizeof (*per_sfx);
per_sfx->file = data;
- per_sfx->filesize = fs_filesize;
+ per_sfx->filesize = 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->flags |= SFXFLAG_STREAMED;
+ sfx->total_length = (int)((size_t)len / (per_sfx->format.channels * 2) * ((double)snd_renderbuffer->format.speed / per_sfx->format.speed));
+ sfx->loopstart = sfx->total_length;
+ vc = qov_comment(&vf, -1);
+ if(vc)
+ {
+ loopcomment = qvorbis_comment_query(vc, "LOOP_START", 0);
+ if(loopcomment)
+ sfx->loopstart = bound(0, (unsigned int) (atof(loopcomment) * (double)snd_renderbuffer->format.speed / (double)per_sfx->format.speed), sfx->total_length);
+ }
}
else
{
ogg_int64_t done;
int bs, bigendian;
long ret;
- sfxbuffer_t *sb;
+ snd_buffer_t *sb;
+ snd_format_t ogg_format;
- 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
- 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
- sb = Mem_Alloc (s->mempool, (size_t)len + sizeof (*sb) - sizeof (sb->data));
- 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);
- 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 = sfx->total_length;
+ sfx->flags &= ~SFXFLAG_STREAMED;
+ vc = qov_comment(&vf, -1);
+ if(vc)
+ {
+ loopcomment = qvorbis_comment_query(vc, "LOOP_START", 0);
+ if(loopcomment)
+ sfx->loopstart = bound(0, (unsigned int) (atoi(loopcomment) * (double)snd_renderbuffer->format.speed / (double)sb->format.speed), sfx->total_length);
+ }
qov_clear (&vf);
Mem_Free (data);
projects / xonotic / darkplaces.git / blobdiff