due to performance issues with streaming ogg decoding of all sounds, the

cvar snd_streaming has been brought back from r9487 and defaults to 1
(decode small sounds at load), a new mode of 2 has been added
snd_streaming 0 = always load entire sound, 1 = decode small sounds, 2 =
never decode at load (least memory, slow)

git-svn-id: svn://svn.icculus.org/twilight/trunk/darkplaces@9757 d7cf8633-e32d-0410-b094-e92efae38249

diff --git a/snd_main.c b/snd_main.c
index f5351c9bebfac1f71a0c5805399a72745858c323..acfad510299b8c38f12cf341bd6a325a59c594f2 100644 (file)
--- a/snd_main.c
+++ b/snd_main.c
@@ -171,6 +171,7 @@ 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", "0.15", "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)"};
@@ -844,6 +845,7 @@ 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 f16dbb26539151b8b9e6b9a58abddf5d6111db41..62c5034297058b802394ed65658e2bf905767ac1 100644 (file)
--- a/snd_main.h
+++ b/snd_main.h
@@ -119,6 +119,7 @@ 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_ogg.c b/snd_ogg.c
index a5dba1da6d97fe90178b10ca8d99c20cc86b8045..ad4abadbd779f3eb61b031b8f2b363e6a55a8a28 100644 (file)
--- a/snd_ogg.c
+++ b/snd_ogg.c
@@ -25,6 +25,7 @@
 #include "quakedef.h"
 #include "snd_main.h"
 #include "snd_ogg.h"
+#include "snd_wav.h"
 
 
 /*
@@ -649,10 +650,9 @@ 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;
+       ogg_int64_t len, buff_len;
        double peak, gaindb;
 
        if (!vf_dll)
@@ -694,27 +694,80 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
 
        len = qov_pcm_total (&vf, -1) * vi->channels * 2;  // 16 bits => "* 2"
 
-       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);
+       // 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 || snd_streaming.integer >= 2))
+       {
+               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;
+               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;
+               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->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)
        {