#include "snd_ogg.h"
#include "snd_wav.h"
+#ifdef LINK_TO_LIBVORBIS
+#define OV_EXCLUDE_STATIC_CALLBACKS
+#include <ogg/ogg.h>
+#include <vorbis/vorbisfile.h>
+
+#define qov_clear ov_clear
+#define qov_info ov_info
+#define qov_comment ov_comment
+#define qov_open_callbacks ov_open_callbacks
+#define qov_pcm_seek ov_pcm_seek
+#define qov_pcm_total ov_pcm_total
+#define qov_read ov_read
+#define qvorbis_comment_query vorbis_comment_query
+
+qboolean OGG_OpenLibrary (void) {return true;}
+void OGG_CloseLibrary (void) {}
+#else
/*
=================================================================
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 char * (*qvorbis_comment_query) (vorbis_comment *vc, const char *tag, int count);
static int (*qov_open_callbacks) (void *datasource, OggVorbis_File *vf,
char *initial, long ibytes,
ov_callbacks callbacks);
static dllhandle_t vo_dll = NULL;
static dllhandle_t vf_dll = NULL;
-typedef struct
-{
- unsigned char *buffer;
- ogg_int64_t ind, buffsize;
-} ov_decode_t;
-
-
-static size_t ovcb_read (void *ptr, size_t size, size_t nb, void *datasource)
-{
- ov_decode_t *ov_decode = (ov_decode_t*)datasource;
- size_t remain, len;
-
- remain = ov_decode->buffsize - ov_decode->ind;
- len = size * nb;
- if (remain < len)
- len = remain - remain % size;
-
- memcpy (ptr, ov_decode->buffer + ov_decode->ind, len);
- ov_decode->ind += len;
-
- return len / size;
-}
-
-static int ovcb_seek (void *datasource, ogg_int64_t offset, int whence)
-{
- ov_decode_t *ov_decode = (ov_decode_t*)datasource;
-
- switch (whence)
- {
- case SEEK_SET:
- break;
- case SEEK_CUR:
- offset += ov_decode->ind;
- break;
- case SEEK_END:
- offset += ov_decode->buffsize;
- break;
- default:
- return -1;
- }
- if (offset < 0 || offset > ov_decode->buffsize)
- return -1;
-
- ov_decode->ind = offset;
- return 0;
-}
-
-static int ovcb_close (void *ov_decode)
-{
- return 0;
-}
-
-static long ovcb_tell (void *ov_decode)
-{
- return ((ov_decode_t*)ov_decode)->ind;
-}
-
/*
=================================================================
{
const char* dllnames_vo [] =
{
-#if defined(WIN64)
- "libvorbis64.dll",
-#elif defined(WIN32)
+#if defined(WIN32)
+ "libvorbis-0.dll",
"libvorbis.dll",
"vorbis.dll",
#elif defined(MACOSX)
};
const char* dllnames_vf [] =
{
-#if defined(WIN64)
- "libvorbisfile64.dll",
-#elif defined(WIN32)
+#if defined(WIN32)
+ "libvorbisfile-3.dll",
"libvorbisfile.dll",
"vorbisfile.dll",
#elif defined(MACOSX)
Sys_UnloadLibrary (&vo_dll);
}
+#endif
/*
=================================================================
=================================================================
*/
-#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))))
+typedef struct
+{
+ unsigned char *buffer;
+ ogg_int64_t ind, buffsize;
+} ov_decode_t;
+
+static size_t ovcb_read (void *ptr, size_t size, size_t nb, void *datasource)
+{
+ ov_decode_t *ov_decode = (ov_decode_t*)datasource;
+ size_t remain, len;
+
+ remain = ov_decode->buffsize - ov_decode->ind;
+ len = size * nb;
+ if (remain < len)
+ len = remain - remain % size;
+
+ memcpy (ptr, ov_decode->buffer + ov_decode->ind, len);
+ ov_decode->ind += len;
-// 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];
+ return len / size;
+}
+static int ovcb_seek (void *datasource, ogg_int64_t offset, int whence)
+{
+ ov_decode_t *ov_decode = (ov_decode_t*)datasource;
+
+ switch (whence)
+ {
+ case SEEK_SET:
+ break;
+ case SEEK_CUR:
+ offset += ov_decode->ind;
+ break;
+ case SEEK_END:
+ offset += ov_decode->buffsize;
+ break;
+ default:
+ return -1;
+ }
+ if (offset < 0 || offset > ov_decode->buffsize)
+ return -1;
+
+ ov_decode->ind = offset;
+ return 0;
+}
+
+static int ovcb_close (void *ov_decode)
+{
+ return 0;
+}
+
+static long ovcb_tell (void *ov_decode)
+{
+ return ((ov_decode_t*)ov_decode)->ind;
+}
// Per-sfx data structure
typedef struct
{
unsigned char *file;
size_t filesize;
- snd_format_t format;
- unsigned int total_length;
- char name[128];
} ogg_stream_persfx_t;
// Per-channel data structure
{
OggVorbis_File vf;
ov_decode_t ov_decode;
- unsigned int sb_offset;
int bs;
- snd_buffer_t sb; // must be at the end due to its dynamically allocated size
+ int buffer_firstframe;
+ int buffer_numframes;
+ unsigned char buffer[STREAM_BUFFERSIZE*4];
} ogg_stream_perchannel_t;
/*
====================
-OGG_FetchSound
+OGG_GetSamplesFloat
====================
*/
-static const snd_buffer_t* OGG_FetchSound (void *sfxfetcher, void **chfetcherpointer, unsigned int *start, unsigned int nbsampleframes)
+static void OGG_GetSamplesFloat (channel_t *ch, sfx_t *sfx, int firstsampleframe, int numsampleframes, float *outsamplesfloat)
{
- ogg_stream_perchannel_t* per_ch = (ogg_stream_perchannel_t *)*chfetcherpointer;
- ogg_stream_persfx_t* per_sfx = (ogg_stream_persfx_t *)sfxfetcher;
- snd_buffer_t* sb;
- int newlength, done, ret, bigendian;
- unsigned int real_start;
- unsigned int factor;
-
- // If there's no fetcher structure attached to the channel yet
+ ogg_stream_perchannel_t *per_ch = (ogg_stream_perchannel_t *)ch->fetcher_data;
+ ogg_stream_persfx_t *per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
+ int f = sfx->format.width * sfx->format.channels; // bytes per frame in the buffer
+ short *buf;
+ int i, len;
+ int newlength, done, ret;
+
+ // if this channel does not yet have a channel fetcher, make one
if (per_ch == NULL)
{
- 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;
-
- 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);
-
- // Open it with the VorbisFile API
+ // allocate a struct to keep track of our file position and buffer
+ per_ch = (ogg_stream_perchannel_t *)Mem_Alloc(snd_mempool, sizeof(*per_ch));
+ // begin decoding the file
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)
+ 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", per_sfx->name);
- Mem_Free (per_ch);
- return NULL;
+ // this never happens - this function succeeded earlier on the same data
+ Mem_Free(per_ch);
+ return;
}
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);
-
- *chfetcherpointer = 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 (nbsampleframes > sb->maxframes)
- {
- Con_Printf ("OGG_FetchSound: stream buffer too small (%u sample frames required)\n", nbsampleframes);
- return NULL;
+ per_ch->buffer_firstframe = 0;
+ per_ch->buffer_numframes = 0;
+ // attach the struct to our channel
+ ch->fetcher_data = (void *)per_ch;
}
- // If the data we need has already been decompressed in the sfxbuffer, just return it
- if (per_ch->sb_offset <= real_start && per_ch->sb_offset + sb->nbframes >= real_start + nbsampleframes)
+ // if the request is too large for our buffer, loop...
+ while (numsampleframes * f > (int)sizeof(per_ch->buffer))
{
- *start = per_ch->sb_offset;
- return sb;
+ done = sizeof(per_ch->buffer) / f;
+ OGG_GetSamplesFloat(ch, sfx, firstsampleframe, done, outsamplesfloat);
+ firstsampleframe += done;
+ numsampleframes -= done;
+ outsamplesfloat += done * sfx->format.channels;
}
- 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 || per_ch->sb_offset > real_start)
+ // seek if the request is before the current buffer (loop back)
+ // seek if the request starts beyond the current buffer by at least one frame (channel was zero volume for a while)
+ // do not seek if the request overlaps the buffer end at all (expected behavior)
+ if (per_ch->buffer_firstframe > firstsampleframe || per_ch->buffer_firstframe + per_ch->buffer_numframes < firstsampleframe)
{
- unsigned int time_start;
- ogg_int64_t ogg_start;
- int err;
-
- if (real_start > (unsigned int)per_sfx->total_length)
- {
- Con_Printf ("OGG_FetchSound: asked for a start position after the end of the sfx! (%u > %u)\n",
- real_start, per_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)
+ // we expect to decode forward from here so this will be our new buffer start
+ per_ch->buffer_firstframe = firstsampleframe;
+ per_ch->buffer_numframes = 0;
+ ret = qov_pcm_seek(&per_ch->vf, (ogg_int64_t)firstsampleframe);
+ if (ret != 0)
{
- Con_Printf ("OGG_FetchSound: qov_pcm_seek(..., %d) returned %d\n",
- real_start, err);
- return NULL;
+ // LordHavoc: we can't Con_Printf here, not thread safe...
+ //Con_Printf("OGG_FetchSound: qov_pcm_seek(..., %d) returned %d\n", firstsampleframe, ret);
+ return;
}
- 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 sound buffer
- else
- {
- 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 (sb->format.speed + sb->nbframes > sb->maxframes)
+ // decompress the file as needed
+ if (firstsampleframe + numsampleframes > per_ch->buffer_firstframe + per_ch->buffer_numframes)
{
- Con_Printf ("OGG_FetchSound: stream buffer overflow (%u sample frames / %u)\n",
- sb->format.speed + sb->nbframes, sb->maxframes);
- return NULL;
+ // first slide the buffer back, discarding any data preceding the range we care about
+ int offset = firstsampleframe - per_ch->buffer_firstframe;
+ int keeplength = per_ch->buffer_numframes - offset;
+ if (keeplength > 0)
+ memmove(per_ch->buffer, per_ch->buffer + offset * sfx->format.width * sfx->format.channels, keeplength * sfx->format.width * sfx->format.channels);
+ per_ch->buffer_firstframe = firstsampleframe;
+ per_ch->buffer_numframes -= offset;
+ // decompress as much as we can fit in the buffer
+ newlength = sizeof(per_ch->buffer) - per_ch->buffer_numframes * f;
+ done = 0;
+ while (newlength > done && (ret = qov_read(&per_ch->vf, (char *)per_ch->buffer + per_ch->buffer_numframes * f + done, (int)(newlength - done), mem_bigendian, 2, 1, &per_ch->bs)) > 0)
+ done += ret;
+ // clear the missing space if any
+ if (done < newlength)
+ memset(per_ch->buffer + done, 0, newlength - done);
+ // we now have more data in the buffer
+ per_ch->buffer_numframes += done / f;
}
- newlength = per_sfx->format.speed * factor; // -> 1 sec of sound before resampling
- if(newlength > (int)sizeof(resampling_buffer))
- newlength = sizeof(resampling_buffer);
-
- // 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, (char *)&resampling_buffer[done], (int)(newlength - done), bigendian, 2, 1, &per_ch->bs)) > 0)
- done += ret;
- Snd_AppendToSndBuffer (sb, resampling_buffer, (size_t)done / (size_t)factor, &per_sfx->format);
-
- *start = per_ch->sb_offset;
- return sb;
+ // convert the sample format for the caller
+ buf = (short *)((char *)per_ch->buffer + (firstsampleframe - per_ch->buffer_firstframe) * f);
+ len = numsampleframes * sfx->format.channels;
+ for (i = 0;i < len;i++)
+ outsamplesfloat[i] = buf[i] * (1.0f / 32768.0f);
}
/*
====================
-OGG_FetchEnd
+OGG_StopChannel
====================
*/
-static void OGG_FetchEnd (void *chfetcherdata)
+static void OGG_StopChannel(channel_t *ch)
{
- ogg_stream_perchannel_t* per_ch = (ogg_stream_perchannel_t *)chfetcherdata;
-
+ ogg_stream_perchannel_t *per_ch = (ogg_stream_perchannel_t *)ch->fetcher_data;
if (per_ch != NULL)
{
- // Free the ogg vorbis decoder
- qov_clear (&per_ch->vf);
-
- Mem_Free (per_ch);
+ // release the vorbis decompressor
+ qov_clear(&per_ch->vf);
+ Mem_Free(per_ch);
}
}
OGG_FreeSfx
====================
*/
-static void OGG_FreeSfx (void *sfxfetcherdata)
+static void OGG_FreeSfx(sfx_t *sfx)
{
- ogg_stream_persfx_t* per_sfx = (ogg_stream_persfx_t *)sfxfetcherdata;
-
- // Free the Ogg Vorbis file
+ ogg_stream_persfx_t *per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
+ // free the complete file we were keeping around
Mem_Free(per_sfx->file);
-
- // Free the stream structure
+ // free the file information structure
Mem_Free(per_sfx);
}
-/*
-====================
-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_GetSamplesFloat, OGG_StopChannel, OGG_FreeSfx};
-static const snd_fetcher_t ogg_fetcher = { OGG_FetchSound, OGG_FetchEnd, OGG_FreeSfx, OGG_GetFormat };
+static void OGG_DecodeTags(vorbis_comment *vc, unsigned int *start, unsigned int *length, unsigned int numsamples, double *peak, double *gaindb)
+{
+ const char *startcomment = NULL, *lengthcomment = NULL, *endcomment = NULL, *thiscomment = NULL;
+
+ *start = numsamples;
+ *length = numsamples;
+ *peak = 0.0;
+ *gaindb = 0.0;
+
+ if(!vc)
+ return;
+
+ thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_PEAK", 0);
+ if(thiscomment)
+ *peak = atof(thiscomment);
+ thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_GAIN", 0);
+ if(thiscomment)
+ *gaindb = atof(thiscomment);
+
+ startcomment = qvorbis_comment_query(vc, "LOOP_START", 0); // DarkPlaces, and some Japanese app
+ if(startcomment)
+ {
+ endcomment = qvorbis_comment_query(vc, "LOOP_END", 0);
+ if(!endcomment)
+ lengthcomment = qvorbis_comment_query(vc, "LOOP_LENGTH", 0);
+ }
+ else
+ {
+ startcomment = qvorbis_comment_query(vc, "LOOPSTART", 0); // RPG Maker VX
+ if(startcomment)
+ {
+ lengthcomment = qvorbis_comment_query(vc, "LOOPLENGTH", 0);
+ if(!lengthcomment)
+ endcomment = qvorbis_comment_query(vc, "LOOPEND", 0);
+ }
+ else
+ {
+ startcomment = qvorbis_comment_query(vc, "LOOPPOINT", 0); // Sonic Robo Blast 2
+ }
+ }
+ if(startcomment)
+ {
+ *start = (unsigned int) bound(0, atof(startcomment), numsamples);
+ if(endcomment)
+ *length = (unsigned int) bound(0, atof(endcomment), numsamples);
+ else if(lengthcomment)
+ *length = (unsigned int) bound(0, *start + atof(lengthcomment), numsamples);
+ }
+}
/*
====================
Load an Ogg Vorbis file into memory
====================
*/
-qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
+qboolean OGG_LoadVorbisFile(const char *filename, sfx_t *sfx)
{
unsigned char *data;
- const char *thiscomment;
fs_offset_t filesize;
ov_decode_t ov_decode;
OggVorbis_File vf;
vorbis_info *vi;
vorbis_comment *vc;
- ogg_int64_t len, buff_len;
- double peak = 0.0;
- double gaindb = 0.0;
+ double peak, gaindb;
+#ifndef LINK_TO_LIBVORBIS
if (!vf_dll)
return false;
+#endif
- // Already loaded?
+ // Return if already loaded
if (sfx->fetcher != NULL)
return true;
- // Load the file
- data = FS_LoadFile (filename, snd_mempool, false, &filesize);
+ // Load the file completely
+ data = FS_LoadFile(filename, snd_mempool, false, &filesize);
if (data == NULL)
return false;
if (developer_loading.integer >= 2)
- Con_Printf ("Loading Ogg Vorbis file \"%s\"\n", filename);
+ Con_Printf("Loading Ogg Vorbis file \"%s\"\n", filename);
// Open it with the VorbisFile API
ov_decode.buffer = data;
ov_decode.ind = 0;
ov_decode.buffsize = filesize;
- if (qov_open_callbacks (&ov_decode, &vf, NULL, 0, callbacks) < 0)
+ if (qov_open_callbacks(&ov_decode, &vf, NULL, 0, callbacks) < 0)
{
- Con_Printf ("error while opening Ogg Vorbis file \"%s\"\n", filename);
+ Con_Printf("error while opening Ogg Vorbis file \"%s\"\n", filename);
Mem_Free(data);
return false;
}
// Get the stream information
- vi = qov_info (&vf, -1);
+ vi = qov_info(&vf, -1);
if (vi->channels < 1 || vi->channels > 2)
{
Con_Printf("%s has an unsupported number of channels (%i)\n",
return false;
}
- len = qov_pcm_total (&vf, -1) * vi->channels * 2; // 16 bits => "* 2"
+ sfx->format.speed = vi->rate;
+ sfx->format.channels = vi->channels;
+ sfx->format.width = 2; // We always work with 16 bits samples
+
+ sfx->total_length = qov_pcm_total(&vf, -1);
- // Decide if we go for a stream or a simple PCM cache
- 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)
+ if (snd_streaming.integer && (snd_streaming.integer >= 2 || sfx->total_length > max(sizeof(ogg_stream_perchannel_t), snd_streaming_length.value * sfx->format.speed)))
{
+ // large sounds use the OGG fetcher to decode the file on demand (but the entire file is held in memory)
ogg_stream_persfx_t* per_sfx;
-
if (developer_loading.integer >= 2)
- Con_Printf ("Ogg sound file \"%s\" will be streamed\n", filename);
- per_sfx = (ogg_stream_persfx_t *)Mem_Alloc (snd_mempool, sizeof (*per_sfx));
- strlcpy(per_sfx->name, sfx->name, sizeof(per_sfx->name));
+ Con_Printf("Ogg sound file \"%s\" will be streamed\n", filename);
+ 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 = 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;
-
sfx->fetcher_data = per_sfx;
sfx->fetcher = &ogg_fetcher;
sfx->flags |= SFXFLAG_STREAMED;
- per_sfx->total_length = 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)
- {
- thiscomment = qvorbis_comment_query(vc, "LOOP_START", 0);
- if(thiscomment)
- sfx->loopstart = bound(0, (unsigned int) (atof(thiscomment) * (double)snd_renderbuffer->format.speed / (double)per_sfx->format.speed), sfx->total_length);
- thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_PEAK", 0);
- if(thiscomment)
- peak = atof(thiscomment);
- thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_GAIN", 0);
- if(thiscomment)
- gaindb = atof(thiscomment);
- }
+ OGG_DecodeTags(vc, &sfx->loopstart, &sfx->total_length, sfx->total_length, &peak, &gaindb);
+ qov_clear(&vf);
}
else
{
+ // small sounds are entirely loaded and use the PCM fetcher
char *buff;
+ ogg_int64_t len;
ogg_int64_t done;
- int bs, bigendian;
+ int bs;
long ret;
- snd_buffer_t *sb;
- snd_format_t ogg_format;
-
if (developer_loading.integer >= 2)
Con_Printf ("Ogg sound file \"%s\" will be cached\n", filename);
-
- // Decode it
- buff = (char *)Mem_Alloc (snd_mempool, (int)len);
+ len = sfx->total_length * sfx->format.channels * sfx->format.width;
+ sfx->flags &= ~SFXFLAG_STREAMED;
+ sfx->memsize += len;
+ sfx->fetcher = &wav_fetcher;
+ sfx->fetcher_data = Mem_Alloc(snd_mempool, (size_t)len);
+ buff = (char *)sfx->fetcher_data;
done = 0;
bs = 0;
-#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)
+ while ((ret = qov_read(&vf, &buff[done], (int)(len - done), mem_bigendian, 2, 1, &bs)) > 0)
done += ret;
-
- // 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)
- {
- thiscomment = qvorbis_comment_query(vc, "LOOP_START", 0);
- if(thiscomment)
- sfx->loopstart = bound(0, (unsigned int) (atoi(thiscomment) * (double)snd_renderbuffer->format.speed / (double)sb->format.speed), sfx->total_length);
- thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_PEAK", 0);
- if(thiscomment)
- peak = atof(thiscomment);
- thiscomment = qvorbis_comment_query(vc, "REPLAYGAIN_TRACK_GAIN", 0);
- if(thiscomment)
- gaindb = atof(thiscomment);
- }
-
- qov_clear (&vf);
- Mem_Free (data);
- Mem_Free (buff);
+ OGG_DecodeTags(vc, &sfx->loopstart, &sfx->total_length, sfx->total_length, &peak, &gaindb);
+ qov_clear(&vf);
+ Mem_Free(data);
}
if(peak)
{
- sfx->volume_mult = min(1 / peak, exp(gaindb * 0.05 * log(10)));
+ sfx->volume_mult = min(1.0f / peak, exp(gaindb * 0.05f * log(10.0f)));
sfx->volume_peak = peak;
if (developer_loading.integer >= 2)
Con_Printf ("Ogg sound file \"%s\" uses ReplayGain (gain %f, peak %f)\n", filename, sfx->volume_mult, sfx->volume_peak);
}
+ else if(gaindb != 0)
+ {
+ sfx->volume_mult = min(1.0f / peak, exp(gaindb * 0.05f * log(10.0f)));
+ sfx->volume_peak = 1.0; // if peak is not defined, we won't trust it
+ if (developer_loading.integer >= 2)
+ Con_Printf ("Ogg sound file \"%s\" uses ReplayGain (gain %f, peak not defined and assumed to be %f)\n", filename, sfx->volume_mult, sfx->volume_peak);
+ }
return true;
}
projects / xonotic / darkplaces.git / blobdiff