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

diff --git a/snd_3dras.c b/snd_3dras.c
index 0f54d1742fe2fecf3dd0266359ed4309b099ba6c..c984a3b465a582c9fa0bf0fcf6d3df2b08d426d6 100644 (file)
--- 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 2ade24b49f5ecd1f6628111e8b219994b2749fee..5f60089af00a4af2db292a7a17675c4800483c43 100644 (file)
--- 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 62c5034297058b802394ed65658e2bf905767ac1..f16dbb26539151b8b9e6b9a58abddf5d6111db41 100644 (file)
--- 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 748d9e45ee009c30f15ed057cf918ba19aa45a61..9cb486931fc402316a41464a8970a7502516d79c 100644 (file)
--- 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 5b7bbcba10d8570a42fcadb38f690e0ed3be6780..fb21b93a16e48eb5397aa3f9bb60cb75ec8410b6 100644 (file)
--- 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)
        {