From a0593ee6b57c492e4ec9c4f1e62d0dc0114f119c Mon Sep 17 00:00:00 2001 From: havoc Date: Sat, 14 Nov 2009 14:06:50 +0000 Subject: [PATCH] always stream decode ogg, this reduces Nexuiz sound memory usage even more git-svn-id: svn://svn.icculus.org/twilight/trunk/darkplaces@9488 d7cf8633-e32d-0410-b094-e92efae38249 --- snd_3dras.c | 1 - snd_main.c | 2 - snd_main.h | 1 - snd_modplug.c | 11 ++--- snd_ogg.c | 115 +++++++++++++------------------------------------- 5 files changed, 35 insertions(+), 95 deletions(-) diff --git a/snd_3dras.c b/snd_3dras.c index 0f54d174..c984a3b4 100644 --- a/snd_3dras.c +++ b/snd_3dras.c @@ -9,7 +9,6 @@ cvar_t volume = {CVAR_SAVE, "volume", "0.7", "volume of sound effects"}; cvar_t snd_staticvolume = {CVAR_SAVE, "snd_staticvolume", "1", "volume of ambient sound effects (such as swampy sounds at the start of e1m2)"}; cvar_t snd_initialized = { CVAR_READONLY, "snd_initialized", "0", "indicates the sound subsystem is active"}; cvar_t snd_mutewhenidle = {CVAR_SAVE, "snd_mutewhenidle", "1", "whether to disable sound output when game window is inactive"}; -cvar_t snd_streaming = { CVAR_SAVE, "snd_streaming", "1", "enables keeping compressed ogg sound files compressed, decompressing them only as needed, otherwise they will be decompressed completely at load (may use a lot of memory)"}; static cvar_t snd_precache = {0, "snd_precache", "1", "loads sounds before they are used"}; static dllhandle_t ras_dll = NULL; diff --git a/snd_main.c b/snd_main.c index 2ade24b4..5f60089a 100644 --- a/snd_main.c +++ b/snd_main.c @@ -170,7 +170,6 @@ cvar_t snd_spatialization_occlusion = {CVAR_SAVE, "snd_spatialization_occlusion" // Cvars declared in snd_main.h (shared with other snd_*.c files) cvar_t _snd_mixahead = {CVAR_SAVE, "_snd_mixahead", "0.11", "how much sound to mix ahead of time"}; -cvar_t snd_streaming = { CVAR_SAVE, "snd_streaming", "1", "enables keeping compressed ogg sound files compressed, decompressing them only as needed, otherwise they will be decompressed completely at load (may use a lot of memory)"}; cvar_t snd_swapstereo = {CVAR_SAVE, "snd_swapstereo", "0", "swaps left/right speakers for old ISA soundblaster cards"}; extern cvar_t v_flipped; cvar_t snd_channellayout = {0, "snd_channellayout", "0", "channel layout. Can be 0 (auto - snd_restart needed), 1 (standard layout), or 2 (ALSA layout)"}; @@ -844,7 +843,6 @@ void S_Init(void) Cvar_RegisterVariable(&nosound); Cvar_RegisterVariable(&snd_precache); Cvar_RegisterVariable(&snd_initialized); - Cvar_RegisterVariable(&snd_streaming); Cvar_RegisterVariable(&ambient_level); Cvar_RegisterVariable(&ambient_fade); Cvar_RegisterVariable(&snd_noextraupdate); diff --git a/snd_main.h b/snd_main.h index 62c50342..f16dbb26 100644 --- a/snd_main.h +++ b/snd_main.h @@ -119,7 +119,6 @@ extern qboolean snd_usethreadedmixing; // if true, the main thread does not mix extern cvar_t _snd_mixahead; extern cvar_t snd_swapstereo; -extern cvar_t snd_streaming; #define SND_CHANNELLAYOUT_AUTO 0 #define SND_CHANNELLAYOUT_STANDARD 1 diff --git a/snd_modplug.c b/snd_modplug.c index 748d9e45..9cb48693 100644 --- a/snd_modplug.c +++ b/snd_modplug.c @@ -329,16 +329,17 @@ static const snd_buffer_t* ModPlug_FetchSound (void *sfxfetcher, void **chfetche 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 ModPlug_FetchSound per second to regulate the workload - if ((int)(sb->format.speed * STREAM_BUFFER_FILL) + sb->nbframes > sb->maxframes) + // 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 ModPlug_FetchSound to regulate workload + newlength = (int)(per_sfx->format.speed*STREAM_BUFFER_FILL); + if (newlength + sb->nbframes > sb->maxframes) { Con_Printf ("ModPlug_FetchSound: stream buffer overflow (%u sample frames / %u)\n", sb->format.speed + sb->nbframes, sb->maxframes); return NULL; } - newlength = (int)(per_sfx->format.speed*STREAM_BUFFER_FILL) * 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); diff --git a/snd_ogg.c b/snd_ogg.c index 5b7bbcba..fb21b93a 100644 --- a/snd_ogg.c +++ b/snd_ogg.c @@ -25,7 +25,6 @@ #include "quakedef.h" #include "snd_main.h" #include "snd_ogg.h" -#include "snd_wav.h" /* @@ -515,16 +514,17 @@ static const snd_buffer_t* OGG_FetchSound (void *sfxfetcher, void **chfetcherpoi 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 ((int)(sb->format.speed * STREAM_BUFFER_FILL) + sb->nbframes > sb->maxframes) + // 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 sample frames / %u)\n", sb->format.speed + sb->nbframes, sb->maxframes); return NULL; } - newlength = (int)(per_sfx->format.speed*STREAM_BUFFER_FILL) * 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); @@ -658,9 +658,10 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx) fs_offset_t filesize; ov_decode_t ov_decode; OggVorbis_File vf; + ogg_stream_persfx_t* per_sfx; vorbis_info *vi; vorbis_comment *vc; - ogg_int64_t len, buff_len; + ogg_int64_t len; double peak, gaindb; if (!vf_dll) @@ -702,85 +703,27 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx) 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 = (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; - - 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 = 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; - 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; - 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); - 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) - 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->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 (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 = 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; + 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); if(peak) { -- 2.39.2