X-Git-Url: http://de.git.xonotic.org/?p=xonotic%2Fdarkplaces.git;a=blobdiff_plain;f=snd_ogg.c;h=ad4abadbd779f3eb61b031b8f2b363e6a55a8a28;hp=17fc7dff64105615a3ec4f65c20a6940d866d5a1;hb=d2ba4a8ed129ea666d17e21b30edbba56aef7c1c;hpb=a876a43c9f86c23d767a5301d8bf53e7a79dcd6b diff --git a/snd_ogg.c b/snd_ogg.c index 17fc7dff..ad4abadb 100644 --- a/snd_ogg.c +++ b/snd_ogg.c @@ -1,5 +1,5 @@ /* - 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 @@ -23,6 +23,7 @@ #include "quakedef.h" +#include "snd_main.h" #include "snd_ogg.h" #include "snd_wav.h" @@ -157,6 +158,14 @@ typedef struct void *internal; } vorbis_block; +typedef struct +{ + char **user_comments; + int *comment_lengths; + int comments; + char *vendor; +} vorbis_comment; + typedef struct { void *datasource; @@ -170,7 +179,7 @@ typedef struct 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; @@ -195,6 +204,8 @@ typedef struct // 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); @@ -203,23 +214,31 @@ 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); -static dllfunction_t oggvorbisfuncs[] = +static dllfunction_t vorbisfilefuncs[] = { - {"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}, + {"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} }; -// Handle for the Vorbisfile DLL +static dllfunction_t vorbisfuncs[] = +{ + {"vorbis_comment_query", (void **) &qvorbis_comment_query}, + {NULL, NULL} +}; + +// 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; @@ -292,27 +311,45 @@ Try to load the VorbisFile DLL */ qboolean OGG_OpenLibrary (void) { - const char* dllname; + const char* dllnames_vo [] = + { +#if 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(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 - - // Load the DLL - if (! Sys_LoadLibrary (dllname, &vf_dll, oggvorbisfuncs)) - { - Con_Printf ("Ogg Vorbis support disabled\n"); +// COMMANDLINEOPTION: Sound: -novorbis disables ogg vorbis sound support + if (COM_CheckParm("-novorbis")) return false; - } - Con_Printf ("Ogg Vorbis support enabled\n"); - return true; + // 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 + return Sys_LoadLibrary (dllnames_vo, &vo_dll, vorbisfuncs) && Sys_LoadLibrary (dllnames_vf, &vf_dll, vorbisfilefuncs); } @@ -326,6 +363,7 @@ Unload the VorbisFile DLL void OGG_CloseLibrary (void) { Sys_UnloadLibrary (&vf_dll); + Sys_UnloadLibrary (&vo_dll); } @@ -337,19 +375,14 @@ void OGG_CloseLibrary (void) ================================================================= */ -#define STREAM_BUFFER_DURATION 1.5f // 1.5 sec - -// 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; + 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 @@ -357,8 +390,9 @@ typedef struct { 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; @@ -369,28 +403,28 @@ static const ov_callbacks callbacks = {ovcb_read, ovcb_seek, ovcb_close, ovcb_te OGG_FetchSound ==================== */ -static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, unsigned int nbsamples) +static const snd_buffer_t* OGG_FetchSound (void *sfxfetcher, void **chfetcherpointer, unsigned int *start, unsigned int nbsampleframes) { - ogg_stream_perchannel_t* per_ch; - sfxbuffer_t* sb; - sfx_t* sfx; - ogg_stream_persfx_t* per_sfx; - int newlength, done, ret, bigendian; + 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; + unsigned int real_start; 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) { - 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); // Open it with the VorbisFile API per_ch->ov_decode.buffer = per_sfx->file; @@ -398,77 +432,107 @@ static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, uns 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); + Con_Printf("error while reading Ogg Vorbis stream \"%s\"\n", per_sfx->name); Mem_Free (per_ch); return NULL; } - - per_ch->sb.offset = 0; - per_ch->sb.length = 0; per_ch->bs = 0; - ch->fetcher_data = per_ch; + 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 (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)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) + { + 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 = (unsigned int) ((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; } - // 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) + per_ch->sb_offset = real_start; + + // We add more than one frame of sound to the buffer: + // 1- to ensure we won't lose many samples during the resampling process + // 2- to reduce calls to OGG_FetchSound to regulate workload + newlength = (int)(per_sfx->format.speed*STREAM_BUFFER_FILL); + if (newlength + 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 *= factor; // convert from sample frames to bytes + if(newlength > (int)sizeof(resampling_buffer)) + newlength = sizeof(resampling_buffer); // Decompress in the resampling_buffer -#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) + while ((ret = qov_read (&per_ch->vf, (char *)&resampling_buffer[done], (int)(newlength - done), mem_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; } @@ -478,23 +542,100 @@ static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, uns OGG_FetchEnd ==================== */ -static void OGG_FetchEnd (channel_t* ch) +static void OGG_FetchEnd (void *chfetcherdata) { - ogg_stream_perchannel_t* per_ch; + ogg_stream_perchannel_t* per_ch = (ogg_stream_perchannel_t *)chfetcherdata; - 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 }; +/* +==================== +OGG_FreeSfx +==================== +*/ +static void OGG_FreeSfx (void *sfxfetcherdata) +{ + ogg_stream_persfx_t* per_sfx = (ogg_stream_persfx_t *)sfxfetcherdata; + + // Free the Ogg Vorbis file + Mem_Free(per_sfx->file); + + // Free the stream 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_FetchSound, OGG_FetchEnd, OGG_FreeSfx, OGG_GetFormat }; + +static void OGG_DecodeTags(vorbis_comment *vc, unsigned int *start, unsigned int *length, double samplesfactor, 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) * samplesfactor, numsamples); + if(endcomment) + *length = (unsigned int) bound(0, atof(endcomment) * samplesfactor, numsamples); + else if(lengthcomment) + *length = (unsigned int) bound(0, *start + atof(lengthcomment) * samplesfactor, numsamples); + } +} /* ==================== @@ -503,38 +644,40 @@ OGG_LoadVorbisFile 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; + 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, gaindb; if (!vf_dll) return false; - Mem_FreePool (&s->mempool); - s->mempool = Mem_AllocPool (s->name); + // 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); + if (developer_loading.integer >= 2) + 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 = 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; } @@ -543,82 +686,96 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s) 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 || snd_streaming.integer >= 2)) { ogg_stream_persfx_t* per_sfx; - Con_DPrintf ("\"%s\" will be streamed\n", filename); - per_sfx = Mem_Alloc (s->mempool, sizeof (*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)); + 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)); + vc = qov_comment(&vf, -1); + OGG_DecodeTags(vc, &sfx->loopstart, &sfx->total_length, (double)snd_renderbuffer->format.speed / (double)per_sfx->format.speed, sfx->total_length, &peak, &gaindb); + per_sfx->total_length = sfx->total_length; + qov_clear (&vf); } else { char *buff; ogg_int64_t done; - int bs, bigendian; + int bs; long ret; - sfxbuffer_t *sb; + snd_buffer_t *sb; + snd_format_t ogg_format; - Con_DPrintf ("\"%s\" will be streamed\n", filename); + if (developer_loading.integer >= 2) + Con_Printf ("Ogg sound file \"%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 - bigendian = 1; -#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; - // Calculate resampled length - len = (double)done * (double)shm->format.speed / (double)vi->rate; + // 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; + } - // 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; + sfx->fetcher = &wav_fetcher; + sfx->fetcher_data = sb; - 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; + sfx->total_length = sb->nbframes; + sfx->memsize += sb->maxframes * sb->format.channels * sb->format.width + sizeof (*sb) - sizeof (sb->samples); + sfx->flags &= ~SFXFLAG_STREAMED; + vc = qov_comment(&vf, -1); + OGG_DecodeTags(vc, &sfx->loopstart, &sfx->total_length, (double)snd_renderbuffer->format.speed / (double)sb->format.speed, sfx->total_length, &peak, &gaindb); + sb->nbframes = sfx->total_length; qov_clear (&vf); Mem_Free (data); Mem_Free (buff); } + if(peak) + { + 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); + } + return true; }