X-Git-Url: http://de.git.xonotic.org/?a=blobdiff_plain;f=snd_ogg.c;h=8a2682031d86fae844ec8c21601fc67c3ab5d9e5;hb=14bec1a4c2a0623c08d1e746a023971e311fda36;hp=6c78a6768f8594d25fa9858d3fadbe6a6fed0ec6;hpb=c0cf058bf3c8d866dd2279b2ae418550803360cd;p=xonotic%2Fdarkplaces.git

diff --git a/snd_ogg.c b/snd_ogg.c
index 6c78a676..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
@@ -23,6 +23,7 @@
 
 
 #include "quakedef.h"
+#include "snd_main.h"
 #include "snd_ogg.h"
 #include "snd_wav.h"
 
@@ -214,12 +215,13 @@ static dllfunction_t oggvorbisfuncs[] =
 	{NULL, NULL}
 };
 
-// Handle for the Vorbisfile DLL
+// Handles for the Vorbis and Vorbisfile DLLs
+static dllhandle_t vo_dll = NULL;
 static dllhandle_t vf_dll = NULL;
 
 typedef struct
 {
-	qbyte *buffer;
+	unsigned char *buffer;
 	ogg_int64_t ind, buffsize;
 } ov_decode_t;
 
@@ -292,21 +294,52 @@ Try to load the VorbisFile DLL
 */
 qboolean OGG_OpenLibrary (void)
 {
-	const char* dllname;
+	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)
 		return true;
 
-#ifdef WIN32
-	dllname = "vorbisfile.dll";
-#else
-	dllname = "libvorbisfile.so";
-#endif
+// COMMANDLINEOPTION: Sound: -novorbis disables ogg vorbis sound support
+	if (COM_CheckParm("-novorbis"))
+		return false;
 
-	// Load the DLL
-	if (! Sys_LoadLibrary (dllname, &vf_dll, oggvorbisfuncs))
+	// 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 (dllnames_vo, &vo_dll, NULL) ||
+		! Sys_LoadLibrary (dllnames_vf, &vf_dll, oggvorbisfuncs))
 	{
+		Sys_UnloadLibrary (&vo_dll);
 		Con_Printf ("Ogg Vorbis support disabled\n");
 		return false;
 	}
@@ -326,6 +359,7 @@ Unload the VorbisFile DLL
 void OGG_CloseLibrary (void)
 {
 	Sys_UnloadLibrary (&vf_dll);
+	Sys_UnloadLibrary (&vo_dll);
 }
 
 
@@ -337,19 +371,20 @@ void OGG_CloseLibrary (void)
 =================================================================
 */
 
-#define STREAM_BUFFER_SIZE (128 * 1024)
+#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))))
 
-// Note: it must be able to contain enough samples at 48 KHz (max speed)
-//       to fill STREAM_BUFFER_SIZE bytes of samples at 8 KHz (min speed)
-// TODO: dynamically allocate this buffer depending on the shm and min sound speeds
-static qbyte resampling_buffer [STREAM_BUFFER_SIZE * (48000 / 8000)];
+// 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 unsigned char resampling_buffer [48000 * 2 * 2];
 
 
 // Per-sfx data structure
 typedef struct
 {
-	qbyte	*file;
-	size_t	filesize;
+	unsigned char	*file;
+	size_t			filesize;
+	snd_format_t	format;
 } ogg_stream_persfx_t;
 
 // Per-channel data structure
@@ -357,9 +392,9 @@ typedef struct
 {
 	OggVorbis_File	vf;
 	ov_decode_t		ov_decode;
+	unsigned int	sb_offset;
 	int				bs;
-	snd_format_t	format;
-	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;
 
 
@@ -370,25 +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;
 
-	per_ch = ch->fetcher_data;
+	per_ch = (ogg_stream_perchannel_t *)ch->fetcher_data;
+	sfx = ch->sfx;
+	per_sfx = (ogg_stream_persfx_t *)sfx->fetcher_data;
 
 	// If there's no fetcher structure attached to the channel yet
 	if (per_ch == NULL)
 	{
-		sfx_t* sfx;
-		vorbis_info *vi;
-		ogg_stream_persfx_t* per_sfx;
+		size_t buff_len, memsize;
+		snd_format_t sb_format;
+
+		sb_format.speed = snd_renderbuffer->format.speed;
+		sb_format.width = per_sfx->format.width;
+		sb_format.channels = per_sfx->format.channels;
 
-		sfx = ch->sfx;
-		per_ch = Mem_Alloc (sfx->mempool, sizeof (*per_ch) - sizeof (per_ch->sb.data) + STREAM_BUFFER_SIZE);
-		per_sfx = sfx->fetcher_data;
+		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;
@@ -400,64 +444,105 @@ static const sfxbuffer_t* OGG_FetchSound (channel_t* ch, unsigned int start, uns
 			Mem_Free (per_ch);
 			return NULL;
 		}
-
-		// Get the stream information
-		vi = qov_info (&per_ch->vf, -1);
-		per_ch->format.speed = vi->rate;
-		per_ch->format.width = sfx->format.width;
-		per_ch->format.channels = sfx->format.channels;
-		
-		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 (nbsampleframes > sb->maxframes)
+	{
+		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;
-	factor = per_ch->format.width * per_ch->format.channels;
+	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;
+		}
+
+		// 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;
 
-		sb->offset = start;
-		sb->length = 0;
-		newlength = 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;
 	}
 
-	// How many free bytes do we have in the sfxbuffer now?
-	newlength = STREAM_BUFFER_SIZE - (newlength * factor);
+	per_ch->sb_offset = real_start;
 
-	// Decompress in the resampling_buffer to get STREAM_BUFFER_SIZE samples after resampling
-#if BYTE_ORDER == LITTLE_ENDIAN
-	bigendian = 0;
-#else
+	// 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 (sb->format.speed + 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 = per_sfx->format.speed * factor;  // -> 1 sec of sound before resampling
+
+	// 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, &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_ch->format, sb->data + sb->length * factor, ch->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;
 }
 
@@ -471,18 +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 };
 
 
 /*
@@ -492,38 +616,37 @@ 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;
-	ogg_int64_t len;
+	ogg_int64_t len, buff_len;
 
 	if (!vf_dll)
 		return false;
 
-	Mem_FreePool (&s->mempool);
-	s->mempool = Mem_AllocPool (s->name);
+	// 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;
 	}
 
@@ -532,31 +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
-	if (snd_streaming.integer && len > fs_filesize + 3 * STREAM_BUFFER_SIZE)
+	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;
-		s->fetcher_data = per_sfx;
-		s->fetcher = &ogg_fetcher;
-		s->format.speed = shm->format.speed;
-		s->format.width = 2;  // We always work with 16 bits samples
-		s->format.channels = vi->channels;
-		s->loopstart = -1;
-		s->flags |= SFXFLAG_STREAMED;
-		s->total_length = (size_t)len / (vi->channels * 2) * (float)(shm->format.speed / vi->rate);
+		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->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
 	{
@@ -564,39 +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 streamed\n", filename);
+		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);