X-Git-Url: http://de.git.xonotic.org/?a=blobdiff_plain;f=snd_ogg.c;h=8a2682031d86fae844ec8c21601fc67c3ab5d9e5;hb=9a86055855339709591f493513e8192102db2998;hp=fd4e362a2d42269b4d7938e7657f15c8a71a78f5;hpb=ef1324d66e924550c8ef7c0c9950a3202f0a94ec;p=xonotic%2Fdarkplaces.git diff --git a/snd_ogg.c b/snd_ogg.c index fd4e362a..8a268203 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 @@ -221,7 +221,7 @@ static dllhandle_t vf_dll = NULL; typedef struct { - qbyte *buffer; + unsigned char *buffer; ogg_int64_t ind, buffsize; } ov_decode_t; @@ -294,7 +294,36 @@ Try to load the VorbisFile DLL */ qboolean OGG_OpenLibrary (void) { - const char *dllname_vo, *dllname_vf; + 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) @@ -304,22 +333,11 @@ qboolean OGG_OpenLibrary (void) if (COM_CheckParm("-novorbis")) return false; -#ifdef WIN32 - dllname_vo = "vorbis.dll"; - dllname_vf = "vorbisfile.dll"; -#elif defined(MACOSX) - dllname_vo = "libvorbis.dylib"; - dllname_vf = "libvorbisfile.dylib"; -#else - dllname_vo = "libvorbis.so.0"; - dllname_vf = "libvorbisfile.so.3"; -#endif - // 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 (dllname_vo, &vo_dll, NULL) || - ! Sys_LoadLibrary (dllname_vf, &vf_dll, oggvorbisfuncs)) + 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"); @@ -354,16 +372,17 @@ void OGG_CloseLibrary (void) */ #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; @@ -373,8 +392,9 @@ 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; @@ -385,28 +405,34 @@ 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 (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; - per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + buff_len); - per_sfx = sfx->fetcher_data; + 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); + sfx->memsize += memsize; // Open it with the VorbisFile API per_ch->ov_decode.buffer = per_sfx->file; @@ -418,73 +444,105 @@ static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, uns 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 > 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; + } - sb->offset = start; - sb->length = 0; - newlength = 0; + // 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; + + 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 // 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; } @@ -498,18 +556,57 @@ static void OGG_FetchEnd (channel_t* ch) { 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 }; /* @@ -519,9 +616,10 @@ 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; @@ -530,27 +628,25 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s) 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; } @@ -559,37 +655,36 @@ 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) { 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->loopstart = -1; + 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)); } else { @@ -597,39 +692,44 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *s) 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 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 = -1; + sfx->flags &= ~SFXFLAG_STREAMED; qov_clear (&vf); Mem_Free (data);