From dd6679b0b84abd3868249ff23902a0d044e124c9 Mon Sep 17 00:00:00 2001
From: janwas <jan@wildfiregames.com>
Date: Thu, 6 Jul 2006 15:52:59 +0000
Subject: [PATCH] # fixes to string_s selftest snd_mgr: fix cppdoc comments
 (got shredded by IDE autofmt) string_s: disable WARN_IF_PTR_LEN (too many
 false positives) test_string_s: add missing debug_skip_next_err (fixes
 complaints on VC2003)

refs #130

This was SVN commit r4066.
---
 source/lib/res/sound/snd_mgr.cpp | 1014 +++++++++++++++---------------
 source/lib/string_s.cpp          |    8 +-
 source/lib/tests/test_string_s.h |   23 +-
 3 files changed, 535 insertions(+), 510 deletions(-)

diff --git a/source/lib/res/sound/snd_mgr.cpp b/source/lib/res/sound/snd_mgr.cpp
index 8b0ea2f635..981ba83a5b 100644
--- a/source/lib/res/sound/snd_mgr.cpp
+++ b/source/lib/res/sound/snd_mgr.cpp
@@ -7,7 +7,7 @@
  *
  * @author Jan.Wassenberg@stud.uni-karlsruhe.de
  * =========================================================================
- */
+  */
 
 /*
  * Copyright (c) 2004-2005 Jan Wassenberg
@@ -19,7 +19,7 @@
  * This program is distributed in the hope that it will be useful, but
  * WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
- */
+  */
 
 #include "precompiled.h"
 
@@ -121,12 +121,12 @@ static void hsd_list_free_all();
 
 
 /**
-* check if OpenAL indicates an error has occurred. it can only report
-* 1 error at a time, so this is called after every OpenAL request.
-*
-* @param caller Name of calling function (typically passed via __func__)
-*/
-static void al_check(const char* caller = "(unknown)")
+ * check if OpenAL indicates an error has occurred. it can only report
+ * 1 error at a time, so this is called after every OpenAL request.
+ *
+ * @param caller Name of calling function (typically passed via __func__)
+ */
+static void al_check(const char * caller = "(unknown)")
 {
 	debug_assert(al_initialized);
 
@@ -134,7 +134,7 @@ static void al_check(const char* caller = "(unknown)")
 	if(err == AL_NO_ERROR)
 		return;
 
-	const char* str = (const char*)alGetString(err);
+	const char * str = (const char*)alGetString(err);
 	debug_printf("OpenAL error: %s; called from %s\n", str, caller);
 	debug_warn("OpenAL error");
 }
@@ -150,32 +150,32 @@ static void al_check(const char* caller = "(unknown)")
 //
 ///////////////////////////////////////////////////////////////////////////////
 
-static const char* alc_dev_name = 0;
+static const char * alc_dev_name = 0;
 	// default (0): use OpenAL default device.
 	// note: that's why this needs to be a pointer instead of an array.
 
 
 /**
-* tell OpenAL to use the specified device in future.
-*
-* @param alc_new_dev_name Device name.
-* @return LibError
-*
-* name = 0 reverts to OpenAL's default choice, which will also
-* be used if this routine is never called.
-*
-* the device name is typically taken from a config file at init-time;
-* the snd_dev* enumeration routines below are used to present a list
-* of choices to the user in the options screen.
-*
-* if OpenAL hasn't yet been initialized (i.e. no sounds have been opened),
-*   this just stores the device name for use when init does occur.
-*   note: we can't check now if it's invalid (if so, init will fail).
-* otherwise, we shut OpenAL down (thereby stopping all sounds) and
-* re-initialize with the new device. that's fairly time-consuming,
-* so preferably call this routine before sounds are loaded.
-*/
-LibError snd_dev_set(const char* alc_new_dev_name)
+ * tell OpenAL to use the specified device in future.
+ *
+ * @param alc_new_dev_name Device name.
+ * @return LibError
+ *
+ * name = 0 reverts to OpenAL's default choice, which will also
+ * be used if this routine is never called.
+ *
+ * the device name is typically taken from a config file at init-time;
+ * the snd_dev * enumeration routines below are used to present a list
+ * of choices to the user in the options screen.
+ *
+ * if OpenAL hasn't yet been initialized (i.e. no sounds have been opened),
+ *   this just stores the device name for use when init does occur.
+ *   note: we can't check now if it's invalid (if so, init will fail).
+ * otherwise, we shut OpenAL down (thereby stopping all sounds) and
+ * re-initialize with the new device. that's fairly time-consuming,
+ * so preferably call this routine before sounds are loaded.
+ */
+LibError snd_dev_set(const char * alc_new_dev_name)
 {
 	// requesting a specific device
 	if(alc_new_dev_name)
@@ -205,13 +205,13 @@ LibError snd_dev_set(const char* alc_new_dev_name)
 }
 
 
-static ALCcontext* alc_ctx = 0;
-static ALCdevice* alc_dev = 0;
+static ALCcontext * alc_ctx = 0;
+static ALCdevice * alc_dev = 0;
 
 
 /**
-* free the OpenAL context and device.
-*/
+ * free the OpenAL context and device.
+ */
 static void alc_shutdown()
 {
 	if(alc_ctx)
@@ -226,10 +226,10 @@ static void alc_shutdown()
 
 
 /**
-* Ready OpenAL for use by setting up a device and context.
-*
-* @return LibError
-*/
+ * Ready OpenAL for use by setting up a device and context.
+ *
+ * @return LibError
+ */
 static LibError alc_init()
 {
 	LibError ret = INFO_OK;
@@ -279,7 +279,7 @@ static LibError alc_init()
 
 	// make note of which sound device is actually being used
 	// (e.g. DS3D, native, MMSYSTEM) - needed when reporting OpenAL bugs.
-	const char* dev_name = (const char*)alcGetString(alc_dev, ALC_DEVICE_SPECIFIER);
+	const char * dev_name = (const char*)alcGetString(alc_dev, ALC_DEVICE_SPECIFIER);
 	wchar_t buf[200];
 	swprintf(buf, ARRAY_SIZE(buf), L"SND| alc_init: success, using %hs\n", dev_name);
 	ah_log(buf);
@@ -311,10 +311,10 @@ static float al_listener_orientation[6] = {0, 0, -1, 0, 1, 0};
 
 
 /**
-* send the current listener properties to OpenAL.
-*
-* also called from al_init.
-*/
+ * send the current listener properties to OpenAL.
+ *
+ * also called from al_init.
+ */
 static void al_listener_latch()
 {
 	if(al_initialized)
@@ -328,13 +328,13 @@ static void al_listener_latch()
 
 
 /**
-* set amplitude modifier, which is effectively applied to all sounds.
-* in layman's terms, this is the global "volume".
-*
-* @param gain Modifier: must be non-negative;
-* 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
-* @return LibError
-*/
+ * set amplitude modifier, which is effectively applied to all sounds.
+ * in layman's terms, this is the global "volume".
+ *
+ * @param gain Modifier: must be non-negative;
+ * 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
+ * @return LibError
+ */
 LibError snd_set_master_gain(float gain)
 {
 	if(gain < 0)
@@ -351,13 +351,13 @@ LibError snd_set_master_gain(float gain)
 
 
 /**
-* set position of the listener (corresponds to camera in graphics).
-* coordinates are in world space; the system doesn't matter.
-*
-* @param pos position support vector
-* @param dir view direction
-* @param up up vector
-*/
+ * set position of the listener (corresponds to camera in graphics).
+ * coordinates are in world space; the system doesn't matter.
+ *
+ * @param pos position support vector
+ * @param dir view direction
+ * @param up up vector
+ */
 static void al_listener_set_pos(const float pos[3], const float dir[3], const float up[3])
 {
 	int i;
@@ -373,12 +373,12 @@ static void al_listener_set_pos(const float pos[3], const float dir[3], const fl
 
 
 /**
-* get distance between listener and point.
-* this is used to determine sound priority.
-*
-* @param point position support vector
-* @return float euclidean distance squared
-*/
+ * get distance between listener and point.
+ * this is used to determine sound priority.
+ *
+ * @param point position support vector
+ * @return float euclidean distance squared
+ */
 static float al_listener_dist_2(const float point[3])
 {
 	const float dx = al_listener_pos[0] - point[0];
@@ -400,15 +400,15 @@ static int al_bufs_outstanding;
 
 
 /**
-* allocate a new buffer, and fill it with the specified data.
-*
-* @param data raw sound data buffer
-* @param size size of buffer in bytes
-* @param al_fmt AL_FORMAT_* describing the sound data
-* @param al_freq sampling frequency (typically 22050 Hz)
-* @return ALuint buffer name
-*/
-static ALuint al_buf_alloc(ALvoid* data, ALsizei size, ALenum al_fmt, ALsizei al_freq)
+ * allocate a new buffer, and fill it with the specified data.
+ *
+ * @param data raw sound data buffer
+ * @param size size of buffer in bytes
+ * @param al_fmt AL_FORMAT_ * describing the sound data
+ * @param al_freq sampling frequency (typically 22050 Hz)
+ * @return ALuint buffer name
+ */
+static ALuint al_buf_alloc(ALvoid * data, ALsizei size, ALenum al_fmt, ALsizei al_freq)
 {
 	ALuint al_buf;
 	alGenBuffers(1, &al_buf);
@@ -421,10 +421,10 @@ static ALuint al_buf_alloc(ALvoid* data, ALsizei size, ALenum al_fmt, ALsizei al
 
 
 /**
-* free the buffer and its contained sound data.
-*
-* @param al_buf buffer name
-*/
+ * free the buffer and its contained sound data.
+ *
+ * @param al_buf buffer name
+ */
 static void al_buf_free(ALuint al_buf)
 {
 	// no-op if 0 (needed in case SndData_reload fails -
@@ -441,9 +441,9 @@ static void al_buf_free(ALuint al_buf)
 
 
 /**
-* make sure all buffers have been returned to us via al_buf_free.
-* called from al_shutdown.
-*/
+ * make sure all buffers have been returned to us via al_buf_free.
+ * called from al_shutdown.
+ */
 static void al_buf_shutdown()
 {
 	if(al_bufs_outstanding != 0)
@@ -474,9 +474,9 @@ static uint al_src_cap = AL_SRC_MAX;
 
 
 /**
-* grab as many sources as possible up to the limit.
-* called from al_init.
-*/
+ * grab as many sources as possible up to the limit.
+ * called from al_init.
+ */
 static void al_src_init()
 {
 	// grab as many sources as possible and count how many we get.
@@ -501,10 +501,10 @@ static void al_src_init()
 
 
 /**
-* release all sources on freelist (currently stack).
-* all sources must have been returned to us via al_src_free.
-* called from al_shutdown.
-*/
+ * release all sources on freelist (currently stack).
+ * all sources must have been returned to us via al_src_free.
+ * called from al_shutdown.
+ */
 static void al_src_shutdown()
 {
 	debug_assert(al_src_used == 0);
@@ -515,10 +515,10 @@ static void al_src_shutdown()
 
 
 /**
-* try to allocate a source.
-*
-* @return ALuint source name, or 0 if none available
-*/
+ * try to allocate a source.
+ *
+ * @return ALuint source name, or 0 if none available
+ */
 static ALuint al_src_alloc()
 {
 	// no more to give
@@ -530,10 +530,10 @@ static ALuint al_src_alloc()
 
 
 /**
-* mark a source as free and available for reuse.
-*
-* @param al_src source name
-*/
+ * mark a source as free and available for reuse.
+ *
+ * @param al_src source name
+ */
 static void al_src_free(ALuint al_src)
 {
 	debug_assert(alIsSource(al_src));
@@ -545,14 +545,14 @@ static void al_src_free(ALuint al_src)
 
 
 /**
-* set maximum number of voices to play simultaneously,
-* to reduce mixing cost on low-end systems.
-* this limit may be ignored if e.g. there's a stricter
-* implementation- defined ceiling anyway.
-*
-* @param limit max. number of sources
-* @return LibError
-*/
+ * set maximum number of voices to play simultaneously,
+ * to reduce mixing cost on low-end systems.
+ * this limit may be ignored if e.g. there's a stricter
+ * implementation- defined ceiling anyway.
+ *
+ * @param limit max. number of sources
+ * @return LibError
+ */
 LibError snd_set_max_voices(uint limit)
 {
 	// valid if cap is legit (less than what we allocated in al_src_init),
@@ -582,11 +582,11 @@ LibError snd_set_max_voices(uint limit)
 ///////////////////////////////////////////////////////////////////////////////
 
 /**
-* master OpenAL init; makes sure all subsystems are ready for use.
-* called from each snd_open; no harm if called more than once.
-*
-* @return LibError
-*/
+ * master OpenAL init; makes sure all subsystems are ready for use.
+ * called from each snd_open; no harm if called more than once.
+ *
+ * @return LibError
+ */
 static LibError al_init()
 {
 	// only take action on first call, OR when re-initializing.
@@ -606,8 +606,8 @@ static LibError al_init()
 
 
 /**
-* shut down all module subsystems.
-*/
+ * shut down all module subsystems.
+ */
 static void al_shutdown()
 {
 	// was never initialized - nothing to do.
@@ -636,10 +636,10 @@ static void al_shutdown()
 
 
 /**
-* re-initialize OpenAL. currently only required for changing devices.
-*
-* @return LibError
-*/
+ * re-initialize OpenAL. currently only required for changing devices.
+ *
+ * @return LibError
+ */
 static LibError al_reinit()
 {
 	// not yet initialized. settings have been saved, and will be
@@ -663,19 +663,19 @@ static LibError al_reinit()
 // set by snd_dev_prepare_enum; used by snd_dev_next.
 // consists of back-to-back C strings, terminated by an extra '\0'.
 // (this is taken straight from OpenAL; dox say this format may change).
-static const char* devs;
+static const char * devs;
 
 /**
-* Prepare to enumerate all device names (this resets the list returned by
-* snd_dev_next).
-*
-* may be called each time the device list is needed.
-*
-* @return LibError; always successful unless the requisite device
-* enumeration extension isn't available. in the latter case, 
-* a "cannot enum device" message should be presented to the user,
-* and snd_dev_set need not be called; OpenAL will use its default device.
-*/
+ * Prepare to enumerate all device names (this resets the list returned by
+ * snd_dev_next).
+ *
+ * may be called each time the device list is needed.
+ *
+ * @return LibError; always successful unless the requisite device
+ * enumeration extension isn't available. in the latter case, 
+ * a "cannot enum device" message should be presented to the user,
+ * and snd_dev_set need not be called; OpenAL will use its default device.
+ */
 LibError snd_dev_prepare_enum()
 {
 	if(alcIsExtensionPresent(0, (ALubyte*)"ALC_ENUMERATION_EXT") != AL_TRUE)
@@ -687,18 +687,18 @@ LibError snd_dev_prepare_enum()
 
 
 /**
-* Get next device name.
-*
-* do not call unless snd_dev_prepare_enum succeeded!
-* not thread-safe! (static data from snd_dev_prepare_enum is used)
-*
-* @return const char* device name, or 0 if all have been returned
-*/
-const char* snd_dev_next()
+ * Get next device name.
+ *
+ * do not call unless snd_dev_prepare_enum succeeded!
+ * not thread-safe! (static data from snd_dev_prepare_enum is used)
+ *
+ * @return const char * device name, or 0 if all have been returned
+ */
+const char * snd_dev_next()
 {
 	if(!*devs)
 		return 0;
-	const char* dev = devs;
+	const char * dev = devs;
 	devs += strlen(dev)+1;
 	return dev;
 }
@@ -731,18 +731,18 @@ static const int TOTAL_IOS = MAX_STREAMS * MAX_IOS;
 static const size_t TOTAL_BUF_SIZE = TOTAL_IOS * STREAM_BUF_SIZE;
 
 // one large allocation for all buffers
-static void* io_bufs;
+static void * io_bufs;
 	
 // list of free buffers. start of buffer holds pointer to next in list.
-static void* io_buf_freelist;
+static void * io_buf_freelist;
 
 
 /**
-* Free an IO buffer.
-*
-* @param void* IO buffer
-*/
-static void io_buf_free(void* p)
+ * Free an IO buffer.
+ *
+ * @param void * IO buffer
+ */
+static void io_buf_free(void * p)
 {
 	debug_assert(io_bufs <= p && p <= (char*)io_bufs+TOTAL_BUF_SIZE);
 	*(void**)p = io_buf_freelist;
@@ -751,9 +751,9 @@ static void io_buf_free(void* p)
 
 
 /**
-* Allocate a memory pool for all IO buffers.
-* Called from first io_buf_alloc.
-*/
+ * Allocate a memory pool for all IO buffers.
+ * Called from first io_buf_alloc.
+ */
 static void io_buf_init()
 {
 	// allocate 1 big aligned block for all buffers.
@@ -763,7 +763,7 @@ static void io_buf_init()
 		return;
 
 	// build freelist.
-	char* p = (char*)io_bufs;
+	char * p = (char*)io_bufs;
 	for(int i = 0; i < TOTAL_IOS; i++)
 	{
 		io_buf_free(p);
@@ -773,15 +773,15 @@ static void io_buf_init()
 
 
 /**
-* Allocate a fixed-size IO buffer.
-*
-* @return void* buffer, or 0 (and warning) if not enough memory.
-*/
-static void* io_buf_alloc()
+ * Allocate a fixed-size IO buffer.
+ *
+ * @return void * buffer, or 0 (and warning) if not enough memory.
+ */
+static void * io_buf_alloc()
 {
 	ONCE(io_buf_init());
 
-	void* buf = io_buf_freelist;
+	void * buf = io_buf_freelist;
 	// note: we have to bail now; can't update io_buf_freelist.
 	if(!buf)
 	{
@@ -802,10 +802,10 @@ static void* io_buf_alloc()
 
 
 /**
-* Free memory pool holding all IO buffers.
-* no-op if io_buf_alloc was never called.
-* called by snd_shutdown.
-*/
+ * Free memory pool holding all IO buffers.
+ * no-op if io_buf_alloc was never called.
+ * called by snd_shutdown.
+ */
 static void io_buf_shutdown()
 {
 	mem_free(io_bufs);
@@ -822,32 +822,32 @@ static void io_buf_shutdown()
 // because we'll only have <= 2 streams active at a time.
 
 /**
-* Information for streaming sounds from file
-* (i.e. loading pieces of it in the background)
-*/
+ * Information for streaming sounds from file
+ * (i.e. loading pieces of it in the background)
+ */
 struct Stream
 {
 	Handle hf;
 	Handle ios[MAX_IOS];
 	uint active_ios;
 	/// set by stream_buf_get, used by stream_buf_discard to free buf.
-	void* last_buf;
+	void * last_buf;
 		
 };
 
 /**
-* Begin reading the next segment (asynchronously).
-* Called from SndData_reload and snd_data_buf_get.
-*
-* @param Stream*
-* @return LibError
-*/
-static LibError stream_issue(Stream* s)
+ * Begin reading the next segment (asynchronously).
+ * Called from SndData_reload and snd_data_buf_get.
+ *
+ * @param Stream*
+ * @return LibError
+ */
+static LibError stream_issue(Stream * s)
 {
 	if(s->active_ios >= MAX_IOS)
 		return INFO_OK;
 
-	void* buf = io_buf_alloc();
+	void * buf = io_buf_alloc();
 	if(!buf)
 		WARN_RETURN(ERR_NO_MEM);
 
@@ -859,15 +859,15 @@ static LibError stream_issue(Stream* s)
 
 
 /**
-* Access the data most recently streamed in.
-*
-* @param Stream*
-* @param data pointer to buffer
-* @param size [bytes]
-* @return LibError; if the first pending IO hasn't completed,
-* ERR_AGAIN (not an error).
-*/
-static LibError stream_buf_get(Stream* s, void*& data, size_t& size)
+ * Access the data most recently streamed in.
+ *
+ * @param Stream*
+ * @param data pointer to buffer
+ * @param size [bytes]
+ * @return LibError; if the first pending IO hasn't completed,
+ * ERR_AGAIN (not an error).
+ */
+static LibError stream_buf_get(Stream * s, void*& data, size_t& size)
 {
 	if(s->active_ios == 0)
 		WARN_RETURN(ERR_EOF);
@@ -890,16 +890,16 @@ static LibError stream_buf_get(Stream* s, void*& data, size_t& size)
 
 
 /**
-* Free the buffer that was last returned by stream_buf_get,
-* and remove its IO slot from our queue.
-*
-* Must be called exactly once after every successful stream_buf_get;
-* call before calling any other stream_* functions!
-*
-* @param Stream*
-* @return LibError
-*/
-static LibError stream_buf_discard(Stream* s)
+ * Free the buffer that was last returned by stream_buf_get,
+ * and remove its IO slot from our queue.
+ *
+ * Must be called exactly once after every successful stream_buf_get;
+ * call before calling any other stream_ * functions!
+ *
+ * @param Stream*
+ * @return LibError
+ */
+static LibError stream_buf_discard(Stream * s)
 {
 	Handle hio = s->ios[0];
 
@@ -922,13 +922,13 @@ static uint active_streams;
 
 
 /**
-* open a stream and begin reading from disk.
-*
-* @param Stream*
-* @param fn VFS filename.
-* @return LibError
-*/
-static LibError stream_open(Stream* s, const char* fn)
+ * open a stream and begin reading from disk.
+ *
+ * @param Stream*
+ * @param fn VFS filename.
+ * @return LibError
+ */
+static LibError stream_open(Stream * s, const char * fn)
 {
 	// bail because we wouldn't have enough IO buffers for all
 	if(active_streams >= MAX_STREAMS)
@@ -946,13 +946,13 @@ static LibError stream_open(Stream* s, const char* fn)
 
 
 /**
-* close a stream, which may currently be active.
-*
-* @param Stream*
-* @return LibError - the first error that occurred while waiting for
-* IOs / closing file.
-*/
-static LibError stream_close(Stream* s)
+ * close a stream, which may currently be active.
+ *
+ * @param Stream*
+ * @return LibError - the first error that occurred while waiting for
+ * IOs / closing file.
+ */
+static LibError stream_close(Stream * s)
 {
 	LibError ret = INFO_OK;
 	LibError err;
@@ -961,7 +961,7 @@ static LibError stream_close(Stream* s)
 	for(uint i = 0; i < s->active_ios; i++)
 	{
 		// .. wait until complete,
-		void* data; size_t size;	// unused
+		void * data; size_t size;	// unused
 		do
 			err = stream_buf_get(s, data, size);
 		while(err == ERR_AGAIN);
@@ -985,12 +985,12 @@ static LibError stream_close(Stream* s)
 
 
 /**
-* Make sure the given Stream is valid/self-consistent.
-*
-* @param const Stream*
-* @return LibError
-*/
-static LibError stream_validate(const Stream* s)
+ * Make sure the given Stream is valid/self-consistent.
+ *
+ * @param const Stream*
+ * @return LibError
+ */
+static LibError stream_validate(const Stream * s)
 {
 	if(s->active_ios > MAX_IOS)
 		WARN_RETURN(ERR_1);
@@ -1023,8 +1023,8 @@ static LibError stream_validate(const Stream* s)
 // IOs are passed to OpenAL and then discarded immediately.
 
 /**
-* Holder for sound data - either a clip, or stream.
-*/
+ * Holder for sound data - either a clip, or stream.
+ */
 struct SndData
 {
 	// stream
@@ -1040,19 +1040,19 @@ struct SndData
 
 #ifdef OGG_HACK
 // pointer to Ogg instance
-void* o;
+void * o;
 #endif
 };
 
 H_TYPE_DEFINE(SndData);
 
-static void SndData_init(SndData* sd, va_list args)
+static void SndData_init(SndData * sd, va_list args)
 {
 	// olsner: pass as int instead of bool for GCC compat.
 	sd->is_stream = va_arg(args, int) != 0;
 }
 
-static void SndData_dtor(SndData* sd)
+static void SndData_dtor(SndData * sd)
 {
 	// in case reload failed and the following haven't been initialized yet
 	// (freeing them would fail in that case, so bail)
@@ -1071,7 +1071,7 @@ if(sd->o) ogg_release(sd->o);
 
 static void hsd_list_add(Handle hsd);
 
-static LibError SndData_reload(SndData* sd, const char* fn, Handle hsd)
+static LibError SndData_reload(SndData * sd, const char * fn, Handle hsd)
 {
 	hsd_list_add(hsd);
 
@@ -1086,7 +1086,7 @@ static LibError SndData_reload(SndData* sd, const char* fn, Handle hsd)
 	}
 	file_type;
 
-	const char* ext = path_extension(fn);
+	const char * ext = path_extension(fn);
 	// .. OGG (data will be passed directly to OpenAL)
 	if(!stricmp(ext, "ogg"))
 	{
@@ -1132,12 +1132,12 @@ static LibError SndData_reload(SndData* sd, const char* fn, Handle hsd)
 	size_t file_size;
 	RETURN_ERR(vfs_load(fn, file, file_size));
 
-	ALvoid* al_data = (ALvoid*)file;
+	ALvoid * al_data = (ALvoid*)file;
 	ALsizei al_size = (ALsizei)file_size;
 
 	if(file_type == FT_WAV)
 	{
-		ALbyte* memory = (ALbyte*)file;
+		ALbyte * memory = (ALbyte*)file;
 		ALboolean al_loop;	// unused
 		alutLoadWAVMemory(memory, &sd->al_fmt, &al_data, &al_size, &sd->al_freq, &al_loop);
 	}
@@ -1178,7 +1178,7 @@ else
 	return INFO_OK;
 }
 
-static LibError SndData_validate(const SndData* sd)
+static LibError SndData_validate(const SndData * sd)
 {
 	if(sd->al_fmt == 0)
 		WARN_RETURN(ERR_11);
@@ -1193,25 +1193,25 @@ static LibError SndData_validate(const SndData* sd)
 	return INFO_OK;
 }
 
-static LibError SndData_to_string(const SndData* sd, char* buf)
+static LibError SndData_to_string(const SndData * sd, char * buf)
 {
-	const char* type = sd->is_stream? "stream" : "clip";
+	const char * type = sd->is_stream? "stream" : "clip";
 	snprintf(buf, H_STRING_LEN, "%s; al_buf=%d", type, sd->al_buf);
 	return INFO_OK;
 }
 
 
 /**
-* open and return a handle to a sound file's data.
-*
-* @param fn VFS filename
-* @param is_stream (default false) indicates whether this file should be
-* streamed in (opening is faster, it won't be kept in memory, but
-* only one instance can be open at a time; makes sense for large music files)
-* or loaded immediately.
-* @return Handle or LibError on failure
-*/
-static Handle snd_data_load(const char* fn, bool is_stream)
+ * open and return a handle to a sound file's data.
+ *
+ * @param fn VFS filename
+ * @param is_stream (default false) indicates whether this file should be
+ * streamed in (opening is faster, it won't be kept in memory, but
+ * only one instance can be open at a time; makes sense for large music files)
+ * or loaded immediately.
+ * @return Handle or LibError on failure
+ */
+static Handle snd_data_load(const char * fn, bool is_stream)
 {
 	// don't allow reloading stream objects
 	// (both references would read from the same file handle).
@@ -1224,11 +1224,11 @@ static Handle snd_data_load(const char* fn, bool is_stream)
 
 
 /**
-* Free the sound.
-*
-* @param hsd Handle to SndData; set to 0 afterwards.
-* @return LibError
-*/
+ * Free the sound.
+ *
+ * @param hsd Handle to SndData; set to 0 afterwards.
+ * @return LibError
+ */
 static LibError snd_data_free(Handle& hsd)
 {
 	return h_free(hsd, H_SndData);
@@ -1260,10 +1260,10 @@ typedef std::vector<Handle> Handles;
 static Handles hsd_list;
 
 /**
-* Add hsd to the list.
-* called from SndData_reload; will later be removed via hsd_list_free_all.
-* @param hsd Handle to SndData
-*/
+ * Add hsd to the list.
+ * called from SndData_reload; will later be removed via hsd_list_free_all.
+ * @param hsd Handle to SndData
+ */
 static void hsd_list_add(Handle hsd)
 {
 	hsd_list.push_back(hsd);
@@ -1271,9 +1271,9 @@ static void hsd_list_add(Handle hsd)
 
 
 /**
-* Free all sounds on list.
-* called by al_shutdown (at exit, or when reinitializing OpenAL).
-*/
+ * Free all sounds on list.
+ * called by al_shutdown (at exit, or when reinitializing OpenAL).
+ */
 static void hsd_list_free_all()
 {
 	for(Handles::iterator it = hsd_list.begin(); it != hsd_list.end(); ++it)
@@ -1297,17 +1297,17 @@ static void hsd_list_free_all()
 ///////////////////////////////////////////////////////////////////////////////
 
 /**
-* Get the sound's AL buffer (typically to play it)
-*
-* @param hsd Handle to SndData.
-* @param al_buf buffer name.
-* @return LibError, most commonly:
-* INFO_OK    = buffer has been returned; more are expected to be available.
-* ERR_EOF   = buffer has been returned but is the last one
-*             (end of file reached)
-* ERR_AGAIN = no buffer returned yet; still streaming in ATM.
-*             call back later.
-*/
+ * Get the sound's AL buffer (typically to play it)
+ *
+ * @param hsd Handle to SndData.
+ * @param al_buf buffer name.
+ * @return LibError, most commonly:
+ * INFO_OK    = buffer has been returned; more are expected to be available.
+ * ERR_EOF   = buffer has been returned but is the last one
+ *             (end of file reached)
+ * ERR_AGAIN = no buffer returned yet; still streaming in ATM.
+ *             call back later.
+ */
 static LibError snd_data_buf_get(Handle hsd, ALuint& al_buf)
 {
 	LibError err = INFO_OK;
@@ -1327,7 +1327,7 @@ static LibError snd_data_buf_get(Handle hsd, ALuint& al_buf)
 	// stream:
 
 	// .. check if IO finished.
-	void* data;
+	void * data;
 	size_t size;
 	err = stream_buf_get(&sd->s, data, size);
 	if(err == ERR_AGAIN)
@@ -1351,12 +1351,12 @@ static LibError snd_data_buf_get(Handle hsd, ALuint& al_buf)
 
 
 /**
-* Indicate the sound's buffer is no longer needed.
-*
-* @param hsd Handle to SndData.
-* @param al_buf buffer name
-* @return LibError
-*/
+ * Indicate the sound's buffer is no longer needed.
+ *
+ * @param hsd Handle to SndData.
+ * @param al_buf buffer name
+ * @return LibError
+ */
 static LibError snd_data_buf_free(Handle hsd, ALuint al_buf)
 {
 	H_DEREF(hsd, SndData, sd);
@@ -1377,8 +1377,8 @@ static LibError snd_data_buf_free(Handle hsd, ALuint al_buf)
 //-----------------------------------------------------------------------------
 
 /**
-* control block for a fade operation.
-*/
+ * control block for a fade operation.
+ */
 struct FadeInfo
 {
 	double start_time;
@@ -1409,8 +1409,8 @@ static float fade_factor_s_curve(float t)
 
 
 /**
-* fade() return value; indicates if the fade operation is complete.
-*/
+ * fade() return value; indicates if the fade operation is complete.
+ */
 enum FadeRet
 {
 	FADE_NO_CHANGE,
@@ -1419,18 +1419,18 @@ enum FadeRet
 };
 
 /**
-* Carry out the requested fade operation.
-*
-* This is called for each active VSrc; if they have no fade operation
-* active, nothing happens. Note: as an optimization, we could make a
-* list of VSrc with fade active and only call this for those;
-* not yet necessary, though.
-*
-* @param fi Describes the fade operation
-* @param cur_time typically returned via get_time()
-* @param out_val Output gain value, i.e. the current result of the fade.
-* @return FadeRet
-*/
+ * Carry out the requested fade operation.
+ *
+ * This is called for each active VSrc; if they have no fade operation
+ * active, nothing happens. Note: as an optimization, we could make a
+ * list of VSrc with fade active and only call this for those;
+ * not yet necessary, though.
+ *
+ * @param fi Describes the fade operation
+ * @param cur_time typically returned via get_time()
+ * @param out_val Output gain value, i.e. the current result of the fade.
+ * @return FadeRet
+ */
 static FadeRet fade(FadeInfo& fi, double cur_time, float& out_val)
 {
 	// no fade in progress - abort immediately. this check is necessary to
@@ -1488,11 +1488,11 @@ static FadeRet fade(FadeInfo& fi, double cur_time, float& out_val)
 
 
 /**
-* Is the fade operation currently active?
-*
-* @param FadeInfo
-* @return bool
-*/
+ * Is the fade operation currently active?
+ *
+ * @param FadeInfo
+ * @return bool
+ */
 static bool fade_is_active(FadeInfo& fi)
 {
 	return (fi.type != FT_NONE);
@@ -1528,10 +1528,10 @@ enum VSrcFlags
 };
 
 /**
-* control block for a virtual source, which represents a sound that the
-* application wants played. it may or may not be played, depending on
-* priority and whether an actual OpenAL source is available.
-*/
+ * control block for a virtual source, which represents a sound that the
+ * application wants played. it may or may not be played, depending on
+ * priority and whether an actual OpenAL source is available.
+ */
 struct VSrc
 {
 	/// handle to this VSrc, so that it can close itself.
@@ -1561,16 +1561,16 @@ struct VSrc
 
 H_TYPE_DEFINE(VSrc);
 
-static void VSrc_init(VSrc* vs, va_list args)
+static void VSrc_init(VSrc * vs, va_list args)
 {
 	vs->flags = va_arg(args, uint);
 	vs->fade.type = FT_NONE;
 }
 
-static void list_remove(VSrc* vs);
-static LibError vsrc_reclaim(VSrc* vs);
+static void list_remove(VSrc * vs);
+static LibError vsrc_reclaim(VSrc * vs);
 
-static void VSrc_dtor(VSrc* vs)
+static void VSrc_dtor(VSrc * vs)
 {
 	// only remove if added (not the case if load failed)
 	if(vs->flags & VS_IN_LIST)
@@ -1584,7 +1584,7 @@ static void VSrc_dtor(VSrc* vs)
 	(void)snd_data_free(vs->hsd);
 }
 
-static LibError VSrc_reload(VSrc* vs, const char* fn, Handle hvs)
+static LibError VSrc_reload(VSrc * vs, const char * fn, Handle hvs)
 {
 	// cannot wait till play(), need to init here:
 	// must load OpenAL so that snd_data_load can check for OGG extension.
@@ -1601,12 +1601,12 @@ static LibError VSrc_reload(VSrc* vs, const char* fn, Handle hvs)
 	// and assume default gain (1.0).
 	//
 
-	const char* snd_fn;		// actual sound file name
+	const char * snd_fn;		// actual sound file name
 	std::string snd_fn_s;
 	// extracted from stringstream;
 	// declare here so that it doesn't go out of scope below.
 
-	const char* ext = path_extension(fn);
+	const char * ext = path_extension(fn);
 	if(!stricmp(ext, "txt"))
 	{
 		FileIOBuf buf; size_t size;
@@ -1641,7 +1641,7 @@ static LibError VSrc_reload(VSrc* vs, const char* fn, Handle hvs)
 	return INFO_OK;
 }
 
-static LibError VSrc_validate(const VSrc* vs)
+static LibError VSrc_validate(const VSrc * vs)
 {
 	// al_src can legitimately be 0 (if vs is low-pri)
 	if(vs->flags & ~VS_ALL_FLAGS)
@@ -1657,7 +1657,7 @@ static LibError VSrc_validate(const VSrc* vs)
 	return INFO_OK;
 }
 
-static LibError VSrc_to_string(const VSrc* vs, char* buf)
+static LibError VSrc_to_string(const VSrc * vs, char * buf)
 {
 	snprintf(buf, H_STRING_LEN, "al_src = %d", vs->al_src);
 	return INFO_OK;
@@ -1665,20 +1665,20 @@ static LibError VSrc_to_string(const VSrc* vs, char* buf)
 
 
 /**
-* open and return a handle to a sound instance.
-*
-* @param snd_fn VFS filename. if a text file (extension ".txt"),
-* it is assumed to be a definition file containing the
-* sound file name and its gain (0.0 .. 1.0).
-* otherwise, it is taken to be the sound file name and
-* gain is set to the default of 1.0 (no attenuation).
-* @param is_stream (default false) indicates whether this file should be
-* streamed in (opening is faster, it won't be kept in memory, but
-* only one instance can be open at a time; makes sense for large music files)
-* or loaded immediately.
-* @return Handle or LibError on failure
-*/
-Handle snd_open(const char* snd_fn, bool is_stream)
+ * open and return a handle to a sound instance.
+ *
+ * @param snd_fn VFS filename. if a text file (extension ".txt"),
+ * it is assumed to be a definition file containing the
+ * sound file name and its gain (0.0 .. 1.0).
+ * otherwise, it is taken to be the sound file name and
+ * gain is set to the default of 1.0 (no attenuation).
+ * @param is_stream (default false) indicates whether this file should be
+ * streamed in (opening is faster, it won't be kept in memory, but
+ * only one instance can be open at a time; makes sense for large music files)
+ * or loaded immediately.
+ * @return Handle or LibError on failure
+ */
+Handle snd_open(const char * snd_fn, bool is_stream)
 {
 	uint flags = 0;
 	if(is_stream)
@@ -1690,13 +1690,13 @@ Handle snd_open(const char* snd_fn, bool is_stream)
 
 
 /**
-* Free the sound; if it was playing, it will be stopped.
-* Note: sounds are closed automatically when done playing;
-* this is provided for completeness only.
-*
-* @param hvs Handle to VSrc. will be set to 0 afterwards.
-* @return LibError
-*/
+ * Free the sound; if it was playing, it will be stopped.
+ * Note: sounds are closed automatically when done playing;
+ * this is provided for completeness only.
+ *
+ * @param hvs Handle to VSrc. will be set to 0 afterwards.
+ * @return LibError
+ */
 LibError snd_free(Handle& hvs)
 {
 	return h_free(hvs, H_VSrc);
@@ -1721,19 +1721,19 @@ typedef VSrcs::iterator It;
 static VSrcs vsrcs;
 
 // don't need to sort now - caller will list_sort() during update.
-static void list_add(VSrc* vs)
+static void list_add(VSrc * vs)
 {
 	vsrcs.push_back(vs);
 }
 
 
 /**
-* call back for each VSrc entry in the list.
-*
-* @param cb Callback function
-* @param skip number of entries to skip (default 0)
-* @param end_idx if not the default value of 0, stop before that entry.
-*/
+ * call back for each VSrc entry in the list.
+ *
+ * @param cb Callback function
+ * @param skip number of entries to skip (default 0)
+ * @param end_idx if not the default value of 0, stop before that entry.
+ */
 static void list_foreach(void (*cb)(VSrc*), uint skip = 0, uint end_idx = 0)
 {
 	It begin = vsrcs.begin() + skip;
@@ -1745,14 +1745,14 @@ static void list_foreach(void (*cb)(VSrc*), uint skip = 0, uint end_idx = 0)
 	// (i.e. set to 0) since last update.
 	for(It it = begin; it != end; ++it)
 	{
-		VSrc* vs = *it;
+		VSrc * vs = *it;
 		if(vs)
 			cb(vs);
 	}
 }
 
 
-static bool is_greater(const VSrc* vs1, const VSrc* vs2)
+static bool is_greater(const VSrc * vs1, const VSrc * vs2)
 {
 	return vs1->cur_pri > vs2->cur_pri;
 }
@@ -1765,13 +1765,13 @@ static void list_sort()
 
 
 /**
-* scan list and remove the given VSrc (by setting it to 0; list will be
-* pruned later (see rationale below).
-* O(N)!
-*
-* @param vs VSrc to remove.
-*/
-static void list_remove(VSrc* vs)
+ * scan list and remove the given VSrc (by setting it to 0; list will be
+ * pruned later (see rationale below).
+ * O(N)!
+ *
+ * @param vs VSrc to remove.
+ */
+static void list_remove(VSrc * vs)
 {
 	for(size_t i = 0; i < vsrcs.size(); i++)
 	{
@@ -1792,12 +1792,12 @@ static void list_remove(VSrc* vs)
 }
 
 
-static bool is_null(VSrc* vs) { return (vs == 0); }
+static bool is_null(VSrc * vs) { return (vs == 0); }
 
 /**
-* remove entries that were set to 0 by list_remove, so that
-* code below can grant the first al_src_cap entries a soure.
-*/
+ * remove entries that were set to 0 by list_remove, so that
+ * code below can grant the first al_src_cap entries a soure.
+ */
 static void list_prune_removed()
 {
 	It new_end = remove_if(vsrcs.begin(), vsrcs.end(), is_null);
@@ -1805,7 +1805,7 @@ static void list_prune_removed()
 }
 
 
-static void vsrc_free(VSrc* vs) { snd_free(vs->hvs); }
+static void vsrc_free(VSrc * vs) { snd_free(vs->hvs); }
 
 static LibError list_free_all()
 {
@@ -1817,12 +1817,12 @@ static LibError list_free_all()
 ///////////////////////////////////////////////////////////////////////////////
 
 /**
-* Send the VSrc properties to OpenAL (when we actually have a source).
-* called by snd_set* and vsrc_grant.
-*
-* @param VSrc*
-*/
-static void vsrc_latch(VSrc* vs)
+ * Send the VSrc properties to OpenAL (when we actually have a source).
+ * called by snd_set * and vsrc_grant.
+ *
+ * @param VSrc*
+ */
+static void vsrc_latch(VSrc * vs)
 {
 	if(!vs->al_src)
 		return;
@@ -1848,12 +1848,12 @@ static void vsrc_latch(VSrc* vs)
 
 
 /**
-* Dequeue any of the VSrc's sound buffers that are finished playing.
-*
-* @param VSrc*
-* @return int number of entries that were removed.
-*/
-static int vsrc_deque_finished_bufs(VSrc* vs)
+ * Dequeue any of the VSrc's sound buffers that are finished playing.
+ *
+ * @param VSrc*
+ * @return int number of entries that were removed.
+ */
+static int vsrc_deque_finished_bufs(VSrc * vs)
 {
 	int num_processed;
 	alGetSourcei(vs->al_src, AL_BUFFERS_PROCESSED, &num_processed);
@@ -1877,13 +1877,13 @@ static int vsrc_deque_finished_bufs(VSrc* vs)
 static double snd_update_time;
 
 /**
-* Update the VSrc - perform fade (if active), queue/unqueue buffers.
-* Called once a frame.
-*
-* @param VSrc*
-* @return LibError
-*/
-static LibError vsrc_update(VSrc* vs)
+ * Update the VSrc - perform fade (if active), queue/unqueue buffers.
+ * Called once a frame.
+ *
+ * @param VSrc*
+ * @return LibError
+ */
+static LibError vsrc_update(VSrc * vs)
 {
 	if(!vs->al_src)
 		return INFO_OK;
@@ -1948,13 +1948,13 @@ static LibError vsrc_update(VSrc* vs)
 
 
 /**
-* Try to give the VSrc an AL source so that it can (re)start playing.
-* called by snd_play and voice management.
-*
-* @param VSrc*
-* @return LibError (ERR_FAIL if no AL source is available)
-*/
-static LibError vsrc_grant(VSrc* vs)
+ * Try to give the VSrc an AL source so that it can (re)start playing.
+ * called by snd_play and voice management.
+ *
+ * @param VSrc*
+ * @return LibError (ERR_FAIL if no AL source is available)
+ */
+static LibError vsrc_grant(VSrc * vs)
 {
 	// already playing - bail
 	if(vs->al_src)
@@ -1988,14 +1988,14 @@ static LibError vsrc_grant(VSrc* vs)
 
 
 /**
-* stop playback, and reclaim the OpenAL source.
-* called when closing the VSrc, or when voice management decides
-* this VSrc must yield to others of higher priority.
-*
-* @param VSrc*
-* @return LibError
-*/
-static LibError vsrc_reclaim(VSrc* vs)
+ * stop playback, and reclaim the OpenAL source.
+ * called when closing the VSrc, or when voice management decides
+ * this VSrc must yield to others of higher priority.
+ *
+ * @param VSrc*
+ * @return LibError
+ */
+static LibError vsrc_reclaim(VSrc * vs)
 {
 	// don't own a source - bail.
 	if(!vs->al_src)
@@ -2016,19 +2016,19 @@ static LibError vsrc_reclaim(VSrc* vs)
 ///////////////////////////////////////////////////////////////////////////////
 
 /**
-* Request the sound be played.
-*
-* Once done playing, the sound is automatically closed (allows
-* fire-and-forget play code).
-* if no hardware voice is available, this sound may not be played at all,
-* or in the case of looped sounds, start later.
-*
-* @param hvs Handle to VSrc
-* @param static_pri (min 0 .. max 1, default 0) indicates which sounds are
-* considered more important; this is attenuated by distance to the
-* listener (see snd_update).
-* @return LibError
-*/
+ * Request the sound be played.
+ *
+ * Once done playing, the sound is automatically closed (allows
+ * fire-and-forget play code).
+ * if no hardware voice is available, this sound may not be played at all,
+ * or in the case of looped sounds, start later.
+ *
+ * @param hvs Handle to VSrc
+ * @param static_pri (min 0 .. max 1, default 0) indicates which sounds are
+ * considered more important; this is attenuated by distance to the
+ * listener (see snd_update).
+ * @return LibError
+ */
 LibError snd_play(Handle hvs, float static_pri)
 {
 	H_DEREF(hvs, VSrc, vs);
@@ -2049,17 +2049,17 @@ LibError snd_play(Handle hvs, float static_pri)
 
 
 /**
-* Change 3d position of the sound source.
-*
-* May be called at any time; fails with invalid handle return if
-* the sound has already been closed (e.g. it never played).
-*
-* @param hvs Handle to VSrc
-* @param x,y,z coordinates (interpretation: see below)
-* @param relative if true, (x,y,z) is treated as relative to the listener;
-* otherwise, it is the position in world coordinates (default).
-* @return LibError
-*/
+ * Change 3d position of the sound source.
+ *
+ * May be called at any time; fails with invalid handle return if
+ * the sound has already been closed (e.g. it never played).
+ *
+ * @param hvs Handle to VSrc
+ * @param x,y,z coordinates (interpretation: see below)
+ * @param relative if true, (x,y,z) is treated as relative to the listener;
+ * otherwise, it is the position in world coordinates (default).
+ * @return LibError
+ */
 LibError snd_set_pos(Handle hvs, float x, float y, float z, bool relative)
 {
 	H_DEREF(hvs, VSrc, vs);
@@ -2073,17 +2073,17 @@ LibError snd_set_pos(Handle hvs, float x, float y, float z, bool relative)
 
 
 /**
-* Change gain (amplitude modifier) of the sound source.
-*
-* should not be called during a fade (see note in implementation);
-* fails with invalid handle return if the sound has already been
-* closed (e.g. it never played).
-*
-* @param hvs Handle to VSrc
-* @param gain modifier; must be non-negative;
-* 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
-* @return LibError
-*/
+ * Change gain (amplitude modifier) of the sound source.
+ *
+ * should not be called during a fade (see note in implementation);
+ * fails with invalid handle return if the sound has already been
+ * closed (e.g. it never played).
+ *
+ * @param hvs Handle to VSrc
+ * @param gain modifier; must be non-negative;
+ * 1 -> unattenuated, 0.5 -> -6 dB, 0 -> silence.
+ * @return LibError
+ */
 LibError snd_set_gain(Handle hvs, float gain)
 {
 	H_DEREF(hvs, VSrc, vs);
@@ -2105,16 +2105,16 @@ LibError snd_set_gain(Handle hvs, float gain)
 
 
 /**
-* Change pitch shift of the sound source.
-*
-* may be called at any time; fails with invalid handle return if
-* the sound has already been closed (e.g. it never played).
-*
-* @param hvs Handle to VSrc
-* @param pitch shift: 1.0 means no change; each reduction by 50% equals a
-* pitch shift of -12 semitones (one octave). zero is invalid.
-* @return LibError
-*/
+ * Change pitch shift of the sound source.
+ *
+ * may be called at any time; fails with invalid handle return if
+ * the sound has already been closed (e.g. it never played).
+ *
+ * @param hvs Handle to VSrc
+ * @param pitch shift: 1.0 means no change; each reduction by 50% equals a
+ * pitch shift of -12 semitones (one octave). zero is invalid.
+ * @return LibError
+ */
 LibError snd_set_pitch(Handle hvs, float pitch)
 {
 	H_DEREF(hvs, VSrc, vs);
@@ -2130,22 +2130,22 @@ LibError snd_set_pitch(Handle hvs, float pitch)
 
 
 /**
-* Enable/disable looping on the sound source.
-* used to implement variable-length sounds (e.g. while building).
-*
-* may be called at any time; fails with invalid handle return if
-* the sound has already been closed (e.g. it never played).
-*
-* notes:
-* - looping sounds are not discarded if they cannot be played for lack of
-*   a hardware voice at the moment play was requested.
-* - once looping is again disabled and the sound has reached its end,
-*   the sound instance is freed automatically (as if never looped).
-*
-* @param hvs Handle to VSrc
-* @param bool loop
-* @return LibError
-*/
+ * Enable/disable looping on the sound source.
+ * used to implement variable-length sounds (e.g. while building).
+ *
+ * may be called at any time; fails with invalid handle return if
+ * the sound has already been closed (e.g. it never played).
+ *
+ * notes:
+ * - looping sounds are not discarded if they cannot be played for lack of
+ *   a hardware voice at the moment play was requested.
+ * - once looping is again disabled and the sound has reached its end,
+ *   the sound instance is freed automatically (as if never looped).
+ *
+ * @param hvs Handle to VSrc
+ * @param bool loop
+ * @return LibError
+ */
 LibError snd_set_loop(Handle hvs, bool loop)
 {
 	H_DEREF(hvs, VSrc, vs);
@@ -2158,34 +2158,34 @@ LibError snd_set_loop(Handle hvs, bool loop)
 
 
 /**
-* Fade the sound source in or out over time.
-* Its gain starts at <initial_gain> immediately and is moved toward
-* <final_gain> over <length> seconds.
-*
-* may be called at any time; fails with invalid handle return if
-* the sound has already been closed (e.g. it never played).
-*
-* note that this function doesn't busy-wait until the fade is complete;
-* any number of fades may be active at a time (allows cross-fading).
-* each snd_update calculates a new gain value for all pending fades.
-* it is safe to start another fade on the same sound source while
-* one is currently in progress; the old one is dropped.
-*
-* @param hvs Handle to VSrc
-* @param initial_gain gain. if < 0 (an otherwise illegal value), the sound's
-* current gain is used as the start value (useful for fading out).
-* @param final_gain gain. if 0, the sound is freed when the fade completes or
-* is aborted, thus allowing fire-and-forget fadeouts. no cases are
-* foreseen where this is undesirable, and it is easier to implement
-* than an extra set-free-after-fade-flag function.
-* @param length duration of fade [s]
-* @param type determines the fade curve: linear, exponential or S-curve.
-* for guidance on which to use, see
-* http://www.transom.org/tools/editing_mixing/200309.stupidfadetricks.html
-* you can also pass FT_ABORT to stop fading (if in progress) and
-* set gain to the final_gain parameter passed here.
-* @return LibError
-*/
+ * Fade the sound source in or out over time.
+ * Its gain starts at <initial_gain> immediately and is moved toward
+ * <final_gain> over <length> seconds.
+ *
+ * may be called at any time; fails with invalid handle return if
+ * the sound has already been closed (e.g. it never played).
+ *
+ * note that this function doesn't busy-wait until the fade is complete;
+ * any number of fades may be active at a time (allows cross-fading).
+ * each snd_update calculates a new gain value for all pending fades.
+ * it is safe to start another fade on the same sound source while
+ * one is currently in progress; the old one is dropped.
+ *
+ * @param hvs Handle to VSrc
+ * @param initial_gain gain. if < 0 (an otherwise illegal value), the sound's
+ * current gain is used as the start value (useful for fading out).
+ * @param final_gain gain. if 0, the sound is freed when the fade completes or
+ * is aborted, thus allowing fire-and-forget fadeouts. no cases are
+ * foreseen where this is undesirable, and it is easier to implement
+ * than an extra set-free-after-fade-flag function.
+ * @param length duration of fade [s]
+ * @param type determines the fade curve: linear, exponential or S-curve.
+ * for guidance on which to use, see
+ * http://www.transom.org/tools/editing_mixing/200309.stupidfadetricks.html
+ * you can also pass FT_ABORT to stop fading (if in progress) and
+ * set gain to the final_gain parameter passed here.
+ * @return LibError
+ */
 LibError snd_fade(Handle hvs, float initial_gain, float final_gain,
 	float length, FadeType type)
 {
@@ -2231,13 +2231,13 @@ static float magnitude_2(const float v[3])
 
 
 /**
-* determine new priority of the VSrc based on distance to listener and
-* static priority.
-* called via list_foreach.
-*
-* @param VSrc*
-*/
-static void calc_cur_pri(VSrc* vs)
+ * determine new priority of the VSrc based on distance to listener and
+ * static priority.
+ * called via list_foreach.
+ *
+ * @param VSrc*
+ */
+static void calc_cur_pri(VSrc * vs)
 {
 	const float MAX_DIST_2 = 1000.0f;
 	const float falloff = 10.0f;
@@ -2260,10 +2260,10 @@ static void calc_cur_pri(VSrc* vs)
 
 
 /**
-* convenience function that strips all unimportant VSrc of their AL source.
-* called via list_foreach; also immediately frees discarded clips.
-*/
-static void reclaim(VSrc* vs)
+ * convenience function that strips all unimportant VSrc of their AL source.
+ * called via list_foreach; also immediately frees discarded clips.
+ */
+static void reclaim(VSrc * vs)
 {
 	vsrc_reclaim(vs);
 
@@ -2272,18 +2272,18 @@ static void reclaim(VSrc* vs)
 }
 
 /// thunk that allows calling via list_foreach (return types differ)
-static void grant(VSrc* vs)
+static void grant(VSrc * vs)
 {
 	vsrc_grant(vs);
 }
 
 
 /**
-* update voice management, i.e. recalculate priority and assign AL sources.
-* no-op if OpenAL not yet initialized.
-*
-* @return LibError
-*/
+ * update voice management, i.e. recalculate priority and assign AL sources.
+ * no-op if OpenAL not yet initialized.
+ *
+ * @return LibError
+ */
 static LibError vm_update()
 {
 	list_prune_removed();
@@ -2308,16 +2308,16 @@ static LibError vm_update()
 ///////////////////////////////////////////////////////////////////////////////
 
 /**
-* perform housekeeping (e.g. streaming); call once a frame.
-*
-* @param pos position support vector. if NULL, all parameters are
-* ignored and listener position unchanged; this is useful in case the
-* world isn't initialized yet.
-* @param dir view direction
-* @param up up vector
-* @return LibError
-*/
-LibError snd_update(const float* pos, const float* dir, const float* up)
+ * perform housekeeping (e.g. streaming); call once a frame.
+ *
+ * @param pos position support vector. if NULL, all parameters are
+ * ignored and listener position unchanged; this is useful in case the
+ * world isn't initialized yet.
+ * @param dir view direction
+ * @param up up vector
+ * @return LibError
+ */
+LibError snd_update(const float * pos, const float * dir, const float * up)
 {
 	// there's no sense in updating anything if we weren't initialized
 	// yet (most notably, if sound is disabled). we check for this to
@@ -2344,11 +2344,11 @@ LibError snd_update(const float* pos, const float* dir, const float* up)
 static bool snd_disabled = false;
 
 /**
-* extra layer on top of al_init that allows 'disabling' sound.
-* called from each snd_open.
-*
-* @return LibError from al_init, or ERR_AGAIN if sound disabled
-*/
+ * extra layer on top of al_init that allows 'disabling' sound.
+ * called from each snd_open.
+ *
+ * @return LibError from al_init, or ERR_AGAIN if sound disabled
+ */
 static inline LibError snd_init()
 {
 	// (note: each VSrc_reload and therefore snd_open will fail)
@@ -2360,21 +2360,21 @@ static inline LibError snd_init()
 
 
 /**
-* (temporarily) disable all sound output. because it causes future snd_open
-* calls to immediately abort before they demand-initialize OpenAL,
-* startup is sped up considerably (500..1000ms). therefore, this must be
-* called before the first snd_open to have any effect; otherwise, the
-* cat will already be out of the bag and we debug_warn of it.
-*
-* rationale: this is a quick'n dirty way of speeding up startup during
-* development without having to change the game's sound code.
-*
-* can later be called to reactivate sound; all settings ever changed
-* will be applied and subsequent sound load / play requests will work.
-*
-* @param bool disabled
-* @return LibError
-*/
+ * (temporarily) disable all sound output. because it causes future snd_open
+ * calls to immediately abort before they demand-initialize OpenAL,
+ * startup is sped up considerably (500..1000ms). therefore, this must be
+ * called before the first snd_open to have any effect; otherwise, the
+ * cat will already be out of the bag and we debug_warn of it.
+ *
+ * rationale: this is a quick'n dirty way of speeding up startup during
+ * development without having to change the game's sound code.
+ *
+ * can later be called to reactivate sound; all settings ever changed
+ * will be applied and subsequent sound load / play requests will work.
+ *
+ * @param bool disabled
+ * @return LibError
+ */
 LibError snd_disable(bool disabled)
 {
 	snd_disabled = disabled;
@@ -2392,9 +2392,9 @@ LibError snd_disable(bool disabled)
 
 
 /**
-* free all resources and shut down the sound system.
-* call before h_mgr_shutdown.
-*/
+ * free all resources and shut down the sound system.
+ * call before h_mgr_shutdown.
+ */
 void snd_shutdown()
 {
 	io_buf_shutdown();
diff --git a/source/lib/string_s.cpp b/source/lib/string_s.cpp
index 662b7458d8..b72a298394 100644
--- a/source/lib/string_s.cpp
+++ b/source/lib/string_s.cpp
@@ -70,10 +70,14 @@
 // raise a debug warning if <len> is the size of a pointer.
 // catches bugs such as: tchar* s = ..; tcpy_s(s, sizeof(s), T(".."));
 // if warnings get annoying, replace with debug_printf. usable as a statement.
-#define WARN_IF_PTR_LEN(len) STMT(                            \
+//
+// currently disabled due to high risk of false positives.
+#define WARN_IF_PTR_LEN(len)\
+/*
+STMT(                                                         \
 	if(len == sizeof(char*))                                  \
 		debug_warn("make sure string buffer size is correct");\
-)
+)*/
 
 
 // skip our implementation if already available, but not the
diff --git a/source/lib/tests/test_string_s.h b/source/lib/tests/test_string_s.h
index b08c2501e0..162d0b8db2 100644
--- a/source/lib/tests/test_string_s.h
+++ b/source/lib/tests/test_string_s.h
@@ -119,31 +119,53 @@ public:
 		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CPY(d1,0,s1, ERANGE,"");	// max_dst_chars = 0
 
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CPY2(d1,1, s1, ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CPY2(d1,1, s5, ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CPY2(d5,5, s5, ERANGE,"");
 
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCPY(d1,1 ,s1,1, ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCPY(d1,1 ,s5,1, ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCPY(d5,5 ,s5,5, ERANGE,"");
 
+		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CAT(0 ,0,0 , EINVAL,"");	// all invalid
+		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CAT(0 ,0,s1, EINVAL,"");	// dst = 0, max = 0
+		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CAT(0 ,1,s1, EINVAL,"");	// dst = 0, max > 0
+		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CAT(d1,1,0 , EINVAL,"");	// src = 0
+		debug_skip_next_err(ERR_INVALID_PARAM);
 		TEST_CAT(d1,0,s1, ERANGE,"");	// max_dst_chars = 0
+		debug_skip_next_err(ERR_STRING_NOT_TERMINATED);
 		TEST_CAT(no_null,5,s1, ERANGE,"");	// dst not terminated
 
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CAT2(d1,1, s1, "",ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CAT2(d1,1, s5, "",ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CAT2(d10,10, s10, "",ERANGE,"");		// empty, total overflow
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CAT2(d10,10, s5, "12345",ERANGE,"");	// not empty, overflow
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_CAT2(d10,10, s10, "12345",ERANGE,"");	// not empty, total overflow
 
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCAT(d1,1, s1,1, "",ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCAT(d1,1, s5,5, "",ERANGE,"");
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCAT(d10,10, s10,10, "",ERANGE,"");		// empty, total overflow
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCAT(d10,10, s5,5, "12345",ERANGE,"");		// not empty, overflow
+		debug_skip_next_err(ERR_BUF_SIZE);
 		TEST_NCAT(d10,10, s10,10, "12345",ERANGE,"");	// not empty, total overflow
 	#endif
 	}
@@ -179,7 +201,6 @@ public:
 		strcpy(d5, "----");
 		TEST_NCPY(d5,5, s5,0 , 0,"");	// specified behavior! see 3.6.2.1.1 #4
 		TEST_NCPY(d5,5, s5,1 , 0,"a");
-		TEST_NCPY(d5,5, s5,4 , 0,"abcd");
 		TEST_NCPY(d6,6, s5,5 , 0,"abcde");
 		TEST_NCPY(d6,6, s5,10, 0,"abcde");
 	}