/*
- Copyright (C) 2003 Mathieu Olivier
+ Copyright (C) 2003-2004 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 "snd_ogg.h"
-extern void ResampleSfx (sfxcache_t *sc, qbyte *data, char *name);
-
-
/*
=================================================================
static int (*qov_open_callbacks) (void *datasource, OggVorbis_File *vf,
char *initial, long ibytes,
ov_callbacks callbacks);
+static int (*qov_pcm_seek) (OggVorbis_File *vf,ogg_int64_t pos);
static ogg_int64_t (*qov_pcm_total) (OggVorbis_File *vf,int i);
static long (*qov_read) (OggVorbis_File *vf,char *buffer,int length,
int bigendianp,int word,int sgned,int *bitstream);
{"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},
{NULL, NULL}
=================================================================
*/
+#define STREAM_BUFFER_SIZE (128 * 1024)
+
+// 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)];
+
+
+// Per-sfx data structure
+typedef struct
+{
+ qbyte *file;
+ size_t filesize;
+} ogg_stream_persfx_t;
+
+// Per-channel data structure
+typedef struct
+{
+ 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;
+
+
+static const ov_callbacks callbacks = {ovcb_read, ovcb_seek, ovcb_close, ovcb_tell};
+
+/*
+====================
+OGG_FetchSound
+====================
+*/
+static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples)
+{
+ ogg_stream_perchannel_t* per_ch;
+ sfxbuffer_t* sb;
+ int newlength, done, ret, bigendian;
+ unsigned int factor;
+
+ per_ch = ch->fetcher_data;
+
+ // 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_sfx = sfx->fetcher_data;
+
+ // Open it with the VorbisFile API
+ per_ch->ov_decode.buffer = per_sfx->file;
+ per_ch->ov_decode.ind = 0;
+ per_ch->ov_decode.buffsize = per_sfx->filesize;
+ if (qov_open_callbacks (&per_ch->ov_decode, &per_ch->vf, NULL, 0, callbacks) < 0)
+ {
+ Con_Printf("error while reading Ogg Vorbis stream \"%s\"\n", sfx->name);
+ Mem_Free (per_ch);
+ 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;
+
+ ch->fetcher_data = per_ch;
+ }
+
+ sb = &per_ch->sb;
+
+ // 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;
+
+ // 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);
+
+ // Decompress in the resampling_buffer to get STREAM_BUFFER_SIZE samples after resampling
+#if BYTE_ORDER == LITTLE_ENDIAN
+ bigendian = 0;
+#else
+ bigendian = 1;
+#endif
+ done = 0;
+ while ((ret = qov_read (&per_ch->vf, &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;
+
+ return sb;
+}
+
+
+/*
+====================
+OGG_FetchEnd
+====================
+*/
+static void OGG_FetchEnd (channel_t* ch)
+{
+ ogg_stream_perchannel_t* per_ch;
+
+ per_ch = ch->fetcher_data;
+ if (per_ch != NULL)
+ {
+ // Free the ogg vorbis decoder
+ qov_clear (&per_ch->vf);
+
+ Mem_Free (per_ch);
+ ch->fetcher_data = NULL;
+ }
+}
+
+static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd };
+extern snd_fetcher_t wav_fetcher;
+
+
/*
====================
OGG_LoadVorbisFile
-Load an Ogg Vorbis file into a sfxcache_t
+Load an Ogg Vorbis file into memory
====================
*/
-sfxcache_t *OGG_LoadVorbisFile (const char *filename, sfx_t *s)
+qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s)
{
qbyte *data;
ov_decode_t ov_decode;
OggVorbis_File vf;
- ov_callbacks callbacks = {ovcb_read, ovcb_seek, ovcb_close, ovcb_tell};
vorbis_info *vi;
ogg_int64_t len;
- char *buff;
- ogg_int64_t done;
- int bs, bigendian;
- long ret;
- sfxcache_t *sc;
if (!vf_dll)
- return NULL;
+ return false;
+
+ Mem_FreePool (&s->mempool);
+ s->mempool = Mem_AllocPool (s->name);
// Load the file
- data = FS_LoadFile (filename, tempmempool, false);
+ data = FS_LoadFile (filename, s->mempool, false);
if (data == NULL)
- return 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.buffsize = fs_filesize;
if (qov_open_callbacks (&ov_decode, &vf, NULL, 0, callbacks) < 0)
{
- Con_Printf("error while opening Ogg Vorbis file \"%s\"\n", filename);
- Mem_Free (data);
- return NULL;
+ Con_Printf ("error while opening Ogg Vorbis file \"%s\"\n", filename);
+ Mem_FreePool (&s->mempool);
+ return false;
}
// Get the stream information
Con_Printf("%s has an unsupported number of channels (%i)\n",
s->name, vi->channels);
qov_clear (&vf);
- Mem_Free (data);
- return NULL;
+ Mem_FreePool (&s->mempool);
+ return false;
}
- // Decode it
len = qov_pcm_total (&vf, -1) * vi->channels * 2; // 16 bits => "* 2"
- buff = Mem_Alloc (tempmempool, (int)len);
- done = 0;
- bs = 0;
-#if BYTE_ORDER == LITTLE_ENDIAN
- bigendian = 0;
-#else
- bigendian = 1;
-#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->speed / (double)vi->rate;
- // Resample it
- Mem_FreePool (&s->mempool);
- s->mempool = Mem_AllocPool (s->name);
- sc = s->sfxcache = Mem_Alloc (s->mempool, (int)len + sizeof (sfxcache_t));
- if (sc != NULL)
+ // Decide if we go for a stream or a simple PCM cache
+ if (snd_streaming.integer && len > fs_filesize + 3 * STREAM_BUFFER_SIZE)
{
- sc->length = (int)done / (vi->channels * 2);
- sc->loopstart = -1;
- sc->speed = vi->rate;
- sc->width = 2; // We always work with 16 bits samples
- sc->stereo = (vi->channels == 2);
-
- ResampleSfx (sc, buff, s->name);
+ 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->file = data;
+ per_sfx->filesize = fs_filesize;
+ 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);
}
else
{
- Con_Printf("failed to allocate memory for sound \"%s\"\n", s->name);
- Mem_FreePool (&s->mempool);
- }
+ char *buff;
+ ogg_int64_t done;
+ int bs, bigendian;
+ long ret;
+ sfxbuffer_t *sb;
+
+ Con_DPrintf ("\"%s\" will be streamed\n", filename);
+
+ // Decode it
+ buff = Mem_Alloc (s->mempool, (int)len);
+ done = 0;
+ bs = 0;
+#if BYTE_ORDER == LITTLE_ENDIAN
+ bigendian = 0;
+#else
+ bigendian = 1;
+#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;
+
+ 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;
- qov_clear (&vf);
- Mem_Free (buff);
- Mem_Free (data);
+ qov_clear (&vf);
+ Mem_Free (data);
+ Mem_Free (buff);
+ }
- return sc;
+ return true;
}
projects / xonotic / darkplaces.git / blobdiff