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

[xonotic/darkplaces.git] / snd_ogg.c

diff --git a/snd_ogg.c b/snd_ogg.c
index 5b7bbcba10d8570a42fcadb38f690e0ed3be6780..ad4abadbd779f3eb61b031b8f2b363e6a55a8a28 100644 (file)
--- a/snd_ogg.c
+++ b/snd_ogg.c
@@ -313,9 +313,7 @@ qboolean OGG_OpenLibrary (void)
 {
        const char* dllnames_vo [] =
        {
-#if defined(WIN64)
-               "libvorbis64.dll",
-#elif defined(WIN32)
+#if defined(WIN32)
                "libvorbis.dll",
                "vorbis.dll",
 #elif defined(MACOSX)
@@ -328,9 +326,7 @@ qboolean OGG_OpenLibrary (void)
        };
        const char* dllnames_vf [] =
        {
-#if defined(WIN64)
-               "libvorbisfile64.dll",
-#elif defined(WIN32)
+#if defined(WIN32)
                "libvorbisfile.dll",
                "vorbisfile.dll",
 #elif defined(MACOSX)
@@ -412,7 +408,7 @@ static const snd_buffer_t* OGG_FetchSound (void *sfxfetcher, void **chfetcherpoi
        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, bigendian;
+       int newlength, done, ret;
        unsigned int real_start;
        unsigned int factor;
 
@@ -515,27 +511,23 @@ 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);
 
        // Decompress in the resampling_buffer
-#if BYTE_ORDER == BIG_ENDIAN
-       bigendian = 1;
-#else
-       bigendian = 0;
-#endif
        done = 0;
-       while ((ret = qov_read (&per_ch->vf, (char *)&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;
 
        Snd_AppendToSndBuffer (sb, resampling_buffer, (size_t)done / (size_t)factor, &per_sfx->format);
@@ -704,7 +696,7 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
 
        // 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)
+       if (snd_streaming.integer && (len > (ogg_int64_t)filesize + 3 * buff_len || snd_streaming.integer >= 2))
        {
                ogg_stream_persfx_t* per_sfx;
 
@@ -734,7 +726,7 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
        {
                char *buff;
                ogg_int64_t done;
-               int bs, bigendian;
+               int bs;
                long ret;
                snd_buffer_t *sb;
                snd_format_t ogg_format;
@@ -746,12 +738,7 @@ qboolean OGG_LoadVorbisFile (const char *filename, sfx_t *sfx)
                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)
+               while ((ret = qov_read (&vf, &buff[done], (int)(len - done), mem_bigendian, 2, 1, &bs)) > 0)
                        done += ret;
 
                // Build the sound buffer