WIP new AudioServer, with buses, effects, etc.

55 files changed, 9483 insertions, 246 deletions

diff --git a/core/math/audio_frame.h b/core/math/audio_frame.h
index dbababf762..acd74903bb 100644
--- a/core/math/audio_frame.h
+++ b/core/math/audio_frame.h
@@ -3,8 +3,25 @@
 
 #include "typedefs.h"
 
+
+static inline float undenormalise(volatile float f)
+{
+	union {
+		uint32_t i;
+		float f;
+	} v;
+
+	v.f = f;
+
+	// original: return (v.i & 0x7f800000) == 0 ? 0.0f : f;
+	// version from Tim Blechmann:
+	return (v.i & 0x7f800000) < 0x08000000 ? 0.0f : f;
+}
+
+
 struct AudioFrame {
 
+	//left and right samples
 	float l,r;
 
 	_ALWAYS_INLINE_ const float& operator[](int idx) const { return idx==0?l:r; }
@@ -15,14 +32,30 @@ struct AudioFrame {
 	_ALWAYS_INLINE_ AudioFrame operator*(const AudioFrame& p_frame) const { return AudioFrame(l*p_frame.l,r*p_frame.r); }
 	_ALWAYS_INLINE_ AudioFrame operator/(const AudioFrame& p_frame) const { return AudioFrame(l/p_frame.l,r/p_frame.r); }
 
+	_ALWAYS_INLINE_ AudioFrame operator+(float p_sample) const { return AudioFrame(l+p_sample,r+p_sample); }
+	_ALWAYS_INLINE_ AudioFrame operator-(float p_sample) const { return AudioFrame(l-p_sample,r-p_sample); }
+	_ALWAYS_INLINE_ AudioFrame operator*(float p_sample) const { return AudioFrame(l*p_sample,r*p_sample); }
+	_ALWAYS_INLINE_ AudioFrame operator/(float p_sample) const { return AudioFrame(l/p_sample,r/p_sample); }
+
 	_ALWAYS_INLINE_ void operator+=(const AudioFrame& p_frame) { l+=p_frame.l; r+=p_frame.r; }
 	_ALWAYS_INLINE_ void operator-=(const AudioFrame& p_frame) { l-=p_frame.l; r-=p_frame.r; }
 	_ALWAYS_INLINE_ void operator*=(const AudioFrame& p_frame) { l*=p_frame.l; r*=p_frame.r; }
 	_ALWAYS_INLINE_ void operator/=(const AudioFrame& p_frame) { l/=p_frame.l; r/=p_frame.r; }
 
+	_ALWAYS_INLINE_ void operator+=(float p_sample) { l+=p_sample; r+=p_sample; }
+	_ALWAYS_INLINE_ void operator-=(float p_sample) { l-=p_sample; r-=p_sample; }
+	_ALWAYS_INLINE_ void operator*=(float p_sample) { l*=p_sample; r*=p_sample; }
+	_ALWAYS_INLINE_ void operator/=(float p_sample) { l/=p_sample; r/=p_sample; }
+
+	_ALWAYS_INLINE_ void undenormalise() {
+		l = ::undenormalise(l);
+		r = ::undenormalise(r);
+	}
+
 	_ALWAYS_INLINE_ AudioFrame(float p_l, float p_r) {l=p_l; r=p_r;}
 	_ALWAYS_INLINE_ AudioFrame(const AudioFrame& p_frame) {l=p_frame.l; r=p_frame.r;}
 
+	_ALWAYS_INLINE_ AudioFrame() {}
 };
 
 #endif
diff --git a/core/resource.cpp b/core/resource.cpp
index 4b09a506ff..6e8d9a8b17 100644
--- a/core/resource.cpp
+++ b/core/resource.cpp
@@ -486,6 +486,7 @@ Resource::Resource() {
 #endif
 
 	subindex=0;
+	local_to_scene=false;
 	local_scene=NULL;
 }
 
diff --git a/drivers/pulseaudio/audio_driver_pulseaudio.cpp b/drivers/pulseaudio/audio_driver_pulseaudio.cpp
index 14b1b229b2..5f0e647545 100644
--- a/drivers/pulseaudio/audio_driver_pulseaudio.cpp
+++ b/drivers/pulseaudio/audio_driver_pulseaudio.cpp
@@ -65,15 +65,15 @@ Error AudioDriverPulseAudio::init() {
 
 	int error_code;
 	pulse = pa_simple_new(	NULL,         // default server
-                          	"Godot",      // application name
-                          	PA_STREAM_PLAYBACK,
-                          	NULL,         // default device
-                          	"Sound",      // stream description
-                          	&spec,
-                          	NULL,         // use default channel map
-                          	&attr,         // use buffering attributes from above
-                          	&error_code
-                          	);
+				"Godot",      // application name
+				PA_STREAM_PLAYBACK,
+				NULL,         // default device
+				"Sound",      // stream description
+				&spec,
+				NULL,         // use default channel map
+				&attr,         // use buffering attributes from above
+				&error_code
+				);
 
 	if (pulse == NULL) {
 		fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
@@ -103,6 +103,7 @@ float AudioDriverPulseAudio::get_latency() {
 
 void AudioDriverPulseAudio::thread_func(void* p_udata) {
 
+	print_line("thread");
 	AudioDriverPulseAudio* ad = (AudioDriverPulseAudio*)p_udata;
 
 	while (!ad->exit_thread) {
@@ -121,9 +122,9 @@ void AudioDriverPulseAudio::thread_func(void* p_udata) {
 			for (unsigned int i=0; i < ad->buffer_size * ad->channels;i ++) {
 				ad->samples_out[i] = ad->samples_in[i] >> 16;
 			}
-        	}
+		}
 
-        	// pa_simple_write always consumes the entire buffer
+		// pa_simple_write always consumes the entire buffer
 
 		int error_code;
 		int byte_size = ad->buffer_size * sizeof(int16_t) * ad->channels;
@@ -134,7 +135,7 @@ void AudioDriverPulseAudio::thread_func(void* p_udata) {
 			ad->exit_thread = true;
 			break;
 		}
-    	}
+	}
 
 	ad->thread_exited = true;
 }
diff --git a/modules/stb_vorbis/SCsub b/modules/stb_vorbis/SCsub
new file mode 100644
index 0000000000..897d05961c
--- /dev/null
+++ b/modules/stb_vorbis/SCsub
@@ -0,0 +1,10 @@
+#!/usr/bin/env python
+
+Import('env')
+Import('env_modules')
+
+# Thirdparty source files
+
+env_stb_vorbis = env_modules.Clone()
+
+env_stb_vorbis.add_source_files(env.modules_sources, "*.cpp")
diff --git a/modules/stb_vorbis/audio_stream_ogg_vorbis.cpp b/modules/stb_vorbis/audio_stream_ogg_vorbis.cpp
new file mode 100644
index 0000000000..31bf5ac292
--- /dev/null
+++ b/modules/stb_vorbis/audio_stream_ogg_vorbis.cpp
@@ -0,0 +1,236 @@
+
+#include "audio_stream_ogg_vorbis.h"
+#include "thirdparty/stb_vorbis/stb_vorbis.c"
+#include "os/file_access.h"
+
+
+void AudioStreamPlaybackOGGVorbis::_mix_internal(AudioFrame* p_buffer,int p_frames) {
+
+	ERR_FAIL_COND(!active);
+
+	int todo=p_frames;
+
+	while(todo) {
+
+		int mixed = stb_vorbis_get_samples_float_interleaved(ogg_stream,2,(float*)p_buffer,todo*2);
+		todo-=mixed;
+
+		if (todo) {
+			//end of file!
+			if (false) {
+				//loop
+				seek_pos(0);
+				loops++;
+			} else {
+				for(int i=mixed;i<p_frames;i++) {
+					p_buffer[i]=AudioFrame(0,0);
+				}
+				active=false;
+			}
+		}
+	}
+
+
+}
+
+float AudioStreamPlaybackOGGVorbis::get_stream_sampling_rate() {
+
+	return vorbis_stream->sample_rate;
+}
+
+
+void AudioStreamPlaybackOGGVorbis::start(float p_from_pos) {
+
+	seek_pos(p_from_pos);
+	active=true;
+	loops=0;
+	_begin_resample();
+
+
+}
+
+void AudioStreamPlaybackOGGVorbis::stop() {
+
+	active=false;
+}
+bool AudioStreamPlaybackOGGVorbis::is_playing() const {
+
+	return active;
+}
+
+int AudioStreamPlaybackOGGVorbis::get_loop_count() const {
+
+	return loops;
+}
+
+float AudioStreamPlaybackOGGVorbis::get_pos() const {
+
+	return float(frames_mixed)/vorbis_stream->sample_rate;
+}
+void AudioStreamPlaybackOGGVorbis::seek_pos(float p_time) {
+
+	if (!active)
+		return;
+
+	stb_vorbis_seek(ogg_stream, uint32_t(p_time*vorbis_stream->sample_rate));
+}
+
+float AudioStreamPlaybackOGGVorbis::get_length() const {
+
+	return vorbis_stream->length;
+}
+
+AudioStreamPlaybackOGGVorbis::~AudioStreamPlaybackOGGVorbis() {
+	if (ogg_alloc.alloc_buffer) {
+		AudioServer::get_singleton()->audio_data_free(ogg_alloc.alloc_buffer);
+		stb_vorbis_close(ogg_stream);
+	}
+}
+
+Ref<AudioStreamPlayback> AudioStreamOGGVorbis::instance_playback() {
+
+
+
+	Ref<AudioStreamPlaybackOGGVorbis> ovs;
+	printf("instance at %p, data %p\n",this,data);
+
+	ERR_FAIL_COND_V(data==NULL,ovs);
+
+	ovs.instance();
+	ovs->vorbis_stream=Ref<AudioStreamOGGVorbis>(this);
+	ovs->ogg_alloc.alloc_buffer=(char*)AudioServer::get_singleton()->audio_data_alloc(decode_mem_size);
+	ovs->ogg_alloc.alloc_buffer_length_in_bytes=decode_mem_size;
+	ovs->frames_mixed=0;
+	ovs->active=false;
+	ovs->loops=0;
+	int error ;
+	ovs->ogg_stream = stb_vorbis_open_memory( (const unsigned char*)data, data_len, &error, &ovs->ogg_alloc );
+	if (!ovs->ogg_stream) {
+
+		AudioServer::get_singleton()->audio_data_free(ovs->ogg_alloc.alloc_buffer);
+		ovs->ogg_alloc.alloc_buffer=NULL;
+		ERR_FAIL_COND_V(!ovs->ogg_stream,Ref<AudioStreamPlaybackOGGVorbis>());
+	}
+
+	return ovs;
+}
+
+String AudioStreamOGGVorbis::get_stream_name() const {
+
+	return "";//return stream_name;
+}
+
+Error AudioStreamOGGVorbis::setup(const uint8_t *p_data,uint32_t p_data_len) {
+
+
+#define MAX_TEST_MEM (1<<20)
+
+	uint32_t alloc_try=1024;
+	PoolVector<char> alloc_mem;
+	PoolVector<char>::Write w;
+	stb_vorbis * ogg_stream=NULL;
+	stb_vorbis_alloc ogg_alloc;
+
+	while(alloc_try<MAX_TEST_MEM) {
+
+		alloc_mem.resize(alloc_try);
+		w = alloc_mem.write();
+
+		ogg_alloc.alloc_buffer=w.ptr();
+		ogg_alloc.alloc_buffer_length_in_bytes=alloc_try;
+
+		int error;
+		ogg_stream = stb_vorbis_open_memory( (const unsigned char*)p_data, p_data_len, &error, &ogg_alloc );
+
+		if (!ogg_stream && error==VORBIS_outofmem) {
+			w = PoolVector<char>::Write();
+			alloc_try*=2;
+		} else {
+			break;
+		}
+	}
+	ERR_FAIL_COND_V(alloc_try==MAX_TEST_MEM,ERR_OUT_OF_MEMORY);
+	ERR_FAIL_COND_V(ogg_stream==NULL,ERR_FILE_CORRUPT);
+
+	stb_vorbis_info info = stb_vorbis_get_info(ogg_stream);
+
+	channels = info.channels;
+	sample_rate = info.sample_rate;
+	decode_mem_size = alloc_try;
+	//does this work? (it's less mem..)
+	//decode_mem_size = ogg_alloc.alloc_buffer_length_in_bytes + info.setup_memory_required + info.temp_memory_required + info.max_frame_size;
+
+	//print_line("succeded "+itos(ogg_alloc.alloc_buffer_length_in_bytes)+" setup "+itos(info.setup_memory_required)+" setup temp "+itos(info.setup_temp_memory_required)+" temp "+itos(info.temp_memory_required)+" maxframe"+itos(info.max_frame_size));
+
+	length=stb_vorbis_stream_length_in_seconds(ogg_stream);
+	stb_vorbis_close(ogg_stream);
+
+	data = AudioServer::get_singleton()->audio_data_alloc(p_data_len,p_data);
+	data_len=p_data_len;
+
+	printf("create at %p, data %p\n",this,data);
+	return OK;
+}
+
+AudioStreamOGGVorbis::AudioStreamOGGVorbis() {
+
+
+	data=NULL;
+	length=0;
+	sample_rate=1;
+	channels=1;
+	decode_mem_size=0;
+}
+
+
+
+
+RES ResourceFormatLoaderAudioStreamOGGVorbis::load(const String &p_path, const String& p_original_path, Error *r_error) {
+	if (r_error)
+		*r_error=OK;
+
+	FileAccess *f = FileAccess::open(p_path,FileAccess::READ);
+	if (!f) {
+		*r_error=ERR_CANT_OPEN;
+		ERR_FAIL_COND_V(!f,RES());
+	}
+
+	size_t len = f->get_len();
+
+	PoolVector<uint8_t> data;
+	data.resize(len);
+	PoolVector<uint8_t>::Write w = data.write();
+
+	f->get_buffer(w.ptr(),len);
+
+	memdelete(f);
+
+	Ref<AudioStreamOGGVorbis> ogg_stream;
+	ogg_stream.instance();
+
+	Error err = ogg_stream->setup(w.ptr(),len);
+
+	if (err!=OK) {
+		*r_error=err;
+		ogg_stream.unref();
+		ERR_FAIL_V(RES());
+	}
+
+	return ogg_stream;
+}
+
+void ResourceFormatLoaderAudioStreamOGGVorbis::get_recognized_extensions(List<String> *p_extensions) const {
+
+	p_extensions->push_back("ogg");
+}
+String ResourceFormatLoaderAudioStreamOGGVorbis::get_resource_type(const String &p_path) const {
+
+	if (p_path.get_extension().to_lower()=="ogg")
+		return "AudioStreamOGGVorbis";
+	return "";
+}
+
+bool ResourceFormatLoaderAudioStreamOGGVorbis::handles_type(const String& p_type) const {
+	return (p_type=="AudioStream" || p_type=="AudioStreamOGG" || p_type=="AudioStreamOGGVorbis");
+}
+
diff --git a/modules/stb_vorbis/audio_stream_ogg_vorbis.h b/modules/stb_vorbis/audio_stream_ogg_vorbis.h
new file mode 100644
index 0000000000..4555423f85
--- /dev/null
+++ b/modules/stb_vorbis/audio_stream_ogg_vorbis.h
@@ -0,0 +1,84 @@
+#ifndef AUDIO_STREAM_STB_VORBIS_H
+#define AUDIO_STREAM_STB_VORBIS_H
+
+#include "servers/audio/audio_stream.h"
+#include "io/resource_loader.h"
+
+#define STB_VORBIS_HEADER_ONLY
+#include "thirdparty/stb_vorbis/stb_vorbis.c"
+#undef STB_VORBIS_HEADER_ONLY
+
+
+class AudioStreamOGGVorbis;
+
+class AudioStreamPlaybackOGGVorbis : public AudioStreamPlaybackResampled {
+
+	GDCLASS( AudioStreamPlaybackOGGVorbis, AudioStreamPlaybackResampled )
+
+	stb_vorbis * ogg_stream;
+	stb_vorbis_alloc ogg_alloc;
+	uint32_t frames_mixed;
+	bool active;
+	int loops;
+
+friend class AudioStreamOGGVorbis;
+
+	Ref<AudioStreamOGGVorbis> vorbis_stream;
+protected:
+
+	virtual void _mix_internal(AudioFrame* p_buffer, int p_frames);
+	virtual float get_stream_sampling_rate();
+
+public:
+	virtual void start(float p_from_pos=0.0);
+	virtual void stop();
+	virtual bool is_playing() const;
+
+	virtual int get_loop_count() const; //times it looped
+
+	virtual float get_pos() const;
+	virtual void seek_pos(float p_time);
+
+	virtual float get_length() const; //if supported, otherwise return 0
+
+	AudioStreamPlaybackOGGVorbis() {   }
+	~AudioStreamPlaybackOGGVorbis();
+};
+
+class AudioStreamOGGVorbis : public AudioStream {
+
+	GDCLASS( AudioStreamOGGVorbis, AudioStream )
+	OBJ_SAVE_TYPE( AudioStream ) //children are all saved as AudioStream, so they can be exchanged
+
+friend class AudioStreamPlaybackOGGVorbis;
+
+	void *data;
+	uint32_t data_len;
+
+	int decode_mem_size;
+	float sample_rate;
+	int channels;
+	float length;
+
+public:
+
+
+	virtual Ref<AudioStreamPlayback> instance_playback();
+	virtual String get_stream_name() const;
+
+	Error setup(const uint8_t *p_data, uint32_t p_data_len);
+
+	AudioStreamOGGVorbis();
+};
+
+class ResourceFormatLoaderAudioStreamOGGVorbis : public ResourceFormatLoader {
+public:
+	virtual RES load(const String &p_path,const String& p_original_path="",Error *r_error=NULL);
+	virtual void get_recognized_extensions(List<String> *p_extensions) const;
+	virtual bool handles_type(const String& p_type) const;
+	virtual String get_resource_type(const String &p_path) const;
+};
+
+
+
+#endif
diff --git a/modules/stb_vorbis/config.py b/modules/stb_vorbis/config.py
new file mode 100644
index 0000000000..fb920482f5
--- /dev/null
+++ b/modules/stb_vorbis/config.py
@@ -0,0 +1,7 @@
+
+def can_build(platform):
+    return True
+
+
+def configure(env):
+    pass
diff --git a/scene/resources/audio_stream.cpp b/modules/stb_vorbis/register_types.cpp
index 7c269de007..143ad6f47e 100644
--- a/scene/resources/audio_stream.cpp
+++ b/modules/stb_vorbis/register_types.cpp
@@ -1,5 +1,5 @@
 /*************************************************************************/
-/*  audio_stream.cpp                                                     */
+/*  register_types.cpp                                                   */
 /*************************************************************************/
 /*                       This file is part of:                           */
 /*                           GODOT ENGINE                                */
@@ -26,36 +26,19 @@
 /* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
 /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
 /*************************************************************************/
-#include "audio_stream.h"
+#include "register_types.h"
+#include "audio_stream_ogg_vorbis.h"
 
-//////////////////////////////
-
-
-void AudioStreamPlayback::_bind_methods() {
-
-	ClassDB::bind_method(_MD("play","from_pos_sec"),&AudioStreamPlayback::play,DEFVAL(0));
-	ClassDB::bind_method(_MD("stop"),&AudioStreamPlayback::stop);
-	ClassDB::bind_method(_MD("is_playing"),&AudioStreamPlayback::is_playing);
-
-	ClassDB::bind_method(_MD("set_loop","enabled"),&AudioStreamPlayback::set_loop);
-	ClassDB::bind_method(_MD("has_loop"),&AudioStreamPlayback::has_loop);
-
-	ClassDB::bind_method(_MD("get_loop_count"),&AudioStreamPlayback::get_loop_count);
-
-	ClassDB::bind_method(_MD("seek_pos","pos"),&AudioStreamPlayback::seek_pos);
-	ClassDB::bind_method(_MD("get_pos"),&AudioStreamPlayback::get_pos);
-
-	ClassDB::bind_method(_MD("get_length"),&AudioStreamPlayback::get_length);
-	ClassDB::bind_method(_MD("get_channels"),&AudioStreamPlayback::get_channels);
-	ClassDB::bind_method(_MD("get_mix_rate"),&AudioStreamPlayback::get_mix_rate);
-	ClassDB::bind_method(_MD("get_minimum_buffer_size"),&AudioStreamPlayback::get_minimum_buffer_size);
+static ResourceFormatLoaderAudioStreamOGGVorbis *vorbis_stream_loader = NULL;
 
+void register_stb_vorbis_types() {
 
+	vorbis_stream_loader = memnew( ResourceFormatLoaderAudioStreamOGGVorbis );
+	ResourceLoader::add_resource_format_loader(vorbis_stream_loader);
+	ClassDB::register_class<AudioStreamOGGVorbis>();
 }
 
+void unregister_stb_vorbis_types() {
 
-void AudioStream::_bind_methods() {
-
-
+	memdelete( vorbis_stream_loader );
 }
-
diff --git a/modules/stb_vorbis/register_types.h b/modules/stb_vorbis/register_types.h
new file mode 100644
index 0000000000..2824aa9f0c
--- /dev/null
+++ b/modules/stb_vorbis/register_types.h
@@ -0,0 +1,30 @@
+/*************************************************************************/
+/*  register_types.h                                                     */
+/*************************************************************************/
+/*                       This file is part of:                           */
+/*                           GODOT ENGINE                                */
+/*                    http://www.godotengine.org                         */
+/*************************************************************************/
+/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur.                 */
+/*                                                                       */
+/* Permission is hereby granted, free of charge, to any person obtaining */
+/* a copy of this software and associated documentation files (the       */
+/* "Software"), to deal in the Software without restriction, including   */
+/* without limitation the rights to use, copy, modify, merge, publish,   */
+/* distribute, sublicense, and/or sell copies of the Software, and to    */
+/* permit persons to whom the Software is furnished to do so, subject to */
+/* the following conditions:                                             */
+/*                                                                       */
+/* The above copyright notice and this permission notice shall be        */
+/* included in all copies or substantial portions of the Software.       */
+/*                                                                       */
+/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
+/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
+/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
+/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
+/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
+/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
+/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
+/*************************************************************************/
+void register_stb_vorbis_types();
+void unregister_stb_vorbis_types();
diff --git a/platform/x11/os_x11.cpp b/platform/x11/os_x11.cpp
index 41746c2431..8baeb8bc90 100644
--- a/platform/x11/os_x11.cpp
+++ b/platform/x11/os_x11.cpp
@@ -295,6 +295,7 @@ void OS_X11::initialize(const VideoMode& p_desired,int p_video_driver,int p_audi
 	}
 
 
+
 	ERR_FAIL_COND(!visual_server);
 	ERR_FAIL_COND(x11_window==0);
 
diff --git a/scene/audio/audio_player.cpp b/scene/audio/audio_player.cpp
new file mode 100644
index 0000000000..9fd005e6fb
--- /dev/null
+++ b/scene/audio/audio_player.cpp
@@ -0,0 +1,301 @@
+#include "audio_player.h"
+
+
+void AudioPlayer::_mix_audio() {
+
+	if (!stream_playback.is_valid()) {
+		return;
+	}
+
+	if (!active) {
+		return;
+	}
+
+	if (setseek>=0.0) {
+		stream_playback->start(setseek);
+		setseek=-1.0; //reset seek
+
+	}
+
+	int bus_index = AudioServer::get_singleton()->thread_find_bus_index(bus);
+
+	//get data
+	AudioFrame *buffer = mix_buffer.ptr();
+	int buffer_size = mix_buffer.size();
+
+	//mix
+	stream_playback->mix(buffer,1.0,buffer_size);
+
+	//multiply volume interpolating to avoid clicks if this changes
+	float vol = Math::db2linear(mix_volume_db);
+	float vol_inc = (Math::db2linear(volume_db) - vol)/float(buffer_size);
+
+	for(int i=0;i<buffer_size;i++) {
+		buffer[i]*=vol;
+		vol+=vol_inc;
+	}
+	//set volume for next mix
+	mix_volume_db = volume_db;
+
+	AudioFrame * targets[3]={NULL,NULL,NULL};
+
+	if (AudioServer::get_singleton()->get_speaker_mode()==AudioServer::SPEAKER_MODE_STEREO) {
+		targets[0] = AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,0);
+	} else {
+		switch(mix_target) {
+			case MIX_TARGET_STEREO: {
+				targets[0]=AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,1);
+			} break;
+			case MIX_TARGET_SURROUND: {
+				targets[0]=AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,1);
+				targets[1]=AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,2);
+				if (AudioServer::get_singleton()->get_speaker_mode()==AudioServer::SPEAKER_SURROUND_71) {
+					targets[2]=AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,3);
+				}
+			} break;
+			case MIX_TARGET_CENTER: {
+				targets[0]=AudioServer::get_singleton()->thread_get_channel_mix_buffer(bus_index,0);
+			} break;
+
+		}
+	}
+
+	for(int c=0;c<3;c++) {
+		if (!targets[c])
+			break;
+		for(int i=0;i<buffer_size;i++) {
+			targets[c][i]+=buffer[i];
+		}
+	}
+
+
+}
+
+void AudioPlayer::_notification(int p_what) {
+
+	if (p_what==NOTIFICATION_ENTER_TREE) {
+
+		AudioServer::get_singleton()->add_callback(_mix_audios,this);
+		if (autoplay && !get_tree()->is_editor_hint()) {
+			play();
+		}
+	}
+
+	if (p_what==NOTIFICATION_EXIT_TREE) {
+
+		AudioServer::get_singleton()->remove_callback(_mix_audios,this);
+
+	}
+}
+
+void AudioPlayer::set_stream(Ref<AudioStream> p_stream) {
+
+	AudioServer::get_singleton()->lock();
+
+	mix_buffer.resize(AudioServer::get_singleton()->thread_get_mix_buffer_size());
+
+	if (stream_playback.is_valid()) {
+		stream_playback.unref();
+		stream.unref();
+		active=false;
+		setseek=-1;
+	}
+
+	stream=p_stream;
+	stream_playback=p_stream->instance_playback();
+
+	if (stream_playback.is_null()) {
+		stream.unref();
+		ERR_FAIL_COND(stream_playback.is_null());
+	}
+
+	AudioServer::get_singleton()->unlock();
+
+}
+
+Ref<AudioStream> AudioPlayer::get_stream() const {
+
+	return stream;
+}
+
+void AudioPlayer::set_volume_db(float p_volume) {
+
+	volume_db=p_volume;
+}
+float AudioPlayer::get_volume_db() const {
+
+	return volume_db;
+}
+
+void AudioPlayer::play(float p_from_pos) {
+
+	if (stream_playback.is_valid()) {
+		mix_volume_db=volume_db; //reset volume ramp
+		setseek=p_from_pos;
+		active=true;
+	}
+}
+
+void AudioPlayer::seek(float p_seconds) {
+
+	if (stream_playback.is_valid()) {
+		setseek=p_seconds;
+	}
+}
+
+void AudioPlayer::stop()  {
+
+	if (stream_playback.is_valid()) {
+		active=false;
+	}
+
+
+}
+
+bool AudioPlayer::is_playing() const {
+
+	if (stream_playback.is_valid()) {
+		return active && stream_playback->is_playing();
+	}
+
+	return false;
+}
+
+float AudioPlayer::get_pos()  {
+
+	if (stream_playback.is_valid()) {
+		return stream_playback->get_pos();
+	}
+
+	return 0;
+}
+
+void AudioPlayer::set_bus(const StringName& p_bus) {
+
+	//if audio is active, must lock this
+	AudioServer::get_singleton()->lock();
+	bus=p_bus;
+	AudioServer::get_singleton()->unlock();
+
+}
+StringName AudioPlayer::get_bus() const {
+
+	for(int i=0;i<AudioServer::get_singleton()->get_bus_count();i++) {
+		if (AudioServer::get_singleton()->get_bus_name(i)==bus) {
+			return bus;
+		}
+	}
+	return "Master";
+}
+
+void AudioPlayer::set_autoplay(bool p_enable)  {
+
+	autoplay=p_enable;
+}
+bool AudioPlayer::is_autoplay_enabled()  {
+
+	return autoplay;
+}
+
+void AudioPlayer::set_mix_target(MixTarget p_target) {
+
+	mix_target=p_target;
+}
+
+AudioPlayer::MixTarget AudioPlayer::get_mix_target() const{
+
+	return mix_target;
+}
+
+void AudioPlayer::_set_playing(bool p_enable) {
+
+	if (p_enable)
+		play();
+	else
+		stop();
+}
+bool AudioPlayer::_is_active() const {
+
+	return active;
+}
+
+
+void AudioPlayer::_validate_property(PropertyInfo& property) const {
+
+	if (property.name=="bus") {
+
+		String options;
+		for(int i=0;i<AudioServer::get_singleton()->get_bus_count();i++) {
+			if (i>0)
+				options+=",";
+			String name = AudioServer::get_singleton()->get_bus_name(i);
+			options+=name;
+		}
+
+		property.hint_string=options;
+	}
+}
+
+void AudioPlayer::_bus_layout_changed() {
+
+	_change_notify();
+}
+
+void AudioPlayer::_bind_methods() {
+
+	ClassDB::bind_method(_MD("set_stream","stream:AudioStream"),&AudioPlayer::set_stream);
+	ClassDB::bind_method(_MD("get_stream"),&AudioPlayer::get_stream);
+
+	ClassDB::bind_method(_MD("set_volume_db","volume_db"),&AudioPlayer::set_volume_db);
+	ClassDB::bind_method(_MD("get_volume_db"),&AudioPlayer::get_volume_db);
+
+	ClassDB::bind_method(_MD("play","from_pos"),&AudioPlayer::play,DEFVAL(0.0));
+	ClassDB::bind_method(_MD("seek","to_pos"),&AudioPlayer::seek);
+	ClassDB::bind_method(_MD("stop"),&AudioPlayer::stop);
+
+	ClassDB::bind_method(_MD("is_playing"),&AudioPlayer::is_playing);
+	ClassDB::bind_method(_MD("get_pos"),&AudioPlayer::get_pos);
+
+	ClassDB::bind_method(_MD("set_bus","bus"),&AudioPlayer::set_bus);
+	ClassDB::bind_method(_MD("get_bus"),&AudioPlayer::get_bus);
+
+	ClassDB::bind_method(_MD("set_autoplay","enable"),&AudioPlayer::set_autoplay);
+	ClassDB::bind_method(_MD("is_autoplay_enabled"),&AudioPlayer::is_autoplay_enabled);
+
+	ClassDB::bind_method(_MD("set_mix_target","mix_target"),&AudioPlayer::set_mix_target);
+	ClassDB::bind_method(_MD("get_mix_target"),&AudioPlayer::get_mix_target);
+
+	ClassDB::bind_method(_MD("_set_playing","enable"),&AudioPlayer::_set_playing);
+	ClassDB::bind_method(_MD("_is_active"),&AudioPlayer::_is_active);
+
+	ClassDB::bind_method(_MD("_bus_layout_changed"),&AudioPlayer::_bus_layout_changed);
+
+
+	ADD_PROPERTY( PropertyInfo(Variant::OBJECT,"stream",PROPERTY_HINT_RESOURCE_TYPE,"AudioStream"),_SCS("set_stream"),_SCS("get_stream") );
+	ADD_PROPERTY( PropertyInfo(Variant::REAL,"volume_db",PROPERTY_HINT_RANGE,"-80,24"),_SCS("set_volume_db"),_SCS("get_volume_db") );
+	ADD_PROPERTY( PropertyInfo(Variant::BOOL,"playing",PROPERTY_HINT_NONE,"",PROPERTY_USAGE_EDITOR),_SCS("_set_playing"),_SCS("_is_active" ));
+	ADD_PROPERTY( PropertyInfo(Variant::BOOL,"autoplay"),_SCS("set_autoplay"),_SCS("is_autoplay_enabled") );
+	ADD_PROPERTY( PropertyInfo(Variant::INT,"mix_target",PROPERTY_HINT_ENUM,"Stereo,Surround,Center"),_SCS("set_mix_target"),_SCS("get_mix_target"));
+	ADD_PROPERTY( PropertyInfo(Variant::STRING,"bus",PROPERTY_HINT_ENUM,""),_SCS("set_bus"),_SCS("get_bus"));
+
+}
+
+AudioPlayer::AudioPlayer() {
+
+	mix_volume_db=0;
+	volume_db=0;
+	autoplay=false;
+	setseek=-1;
+	active=false;
+	mix_target=MIX_TARGET_STEREO;
+
+	AudioServer::get_singleton()->connect("bus_layout_changed",this,"_bus_layout_changed");
+}
+
+
+
+AudioPlayer::~AudioPlayer() {
+
+
+}
+
diff --git a/scene/audio/audio_player.h b/scene/audio/audio_player.h
new file mode 100644
index 0000000000..249e5d0381
--- /dev/null
+++ b/scene/audio/audio_player.h
@@ -0,0 +1,75 @@
+#ifndef AUDIOPLAYER_H
+#define AUDIOPLAYER_H
+
+#include "scene/main/node.h"
+#include "servers/audio/audio_stream.h"
+
+
+class AudioPlayer : public Node {
+
+	GDCLASS( AudioPlayer, Node )
+
+public:
+
+	enum MixTarget {
+		MIX_TARGET_STEREO,
+		MIX_TARGET_SURROUND,
+		MIX_TARGET_CENTER
+	};
+private:
+	Ref<AudioStreamPlayback> stream_playback;
+	Ref<AudioStream> stream;
+	Vector<AudioFrame> mix_buffer;
+
+	volatile float setseek;
+	volatile bool active;
+
+	float mix_volume_db;
+	float volume_db;
+	bool autoplay;
+	StringName bus;
+
+	MixTarget mix_target;
+
+	void _mix_audio();
+	static void _mix_audios(void *self) { reinterpret_cast<AudioPlayer*>(self)->_mix_audio(); }
+
+	void _set_playing(bool p_enable);
+	bool _is_active() const;
+
+	void _bus_layout_changed();
+
+protected:
+
+	void _validate_property(PropertyInfo& property) const;
+	void _notification(int p_what);
+	static void _bind_methods();
+public:
+
+	void set_stream(Ref<AudioStream> p_stream);
+	Ref<AudioStream> get_stream() const;
+
+	void set_volume_db(float p_volume);
+	float get_volume_db() const;
+
+	void play(float p_from_pos=0.0);
+	void seek(float p_seconds);
+	void stop();
+	bool is_playing() const;
+	float get_pos();
+
+	void set_bus(const StringName& p_bus);
+	StringName get_bus() const;
+
+	void set_autoplay(bool p_enable);
+	bool is_autoplay_enabled();
+
+	void set_mix_target(MixTarget p_target);
+	MixTarget get_mix_target() const;
+
+	AudioPlayer();
+	~AudioPlayer();
+};
+
+VARIANT_ENUM_CAST(AudioPlayer::MixTarget)
+#endif // AUDIOPLAYER_H
diff --git a/scene/gui/texture_progress.cpp b/scene/gui/texture_progress.cpp
index f6a33b5643..7d8373976b 100644
--- a/scene/gui/texture_progress.cpp
+++ b/scene/gui/texture_progress.cpp
@@ -46,7 +46,9 @@ void TextureProgress::set_over_texture(const Ref<Texture>& p_texture) {
 
 	over=p_texture;
 	update();
-	minimum_size_changed();
+	if (under.is_null()) {
+		minimum_size_changed();
+	}
 }
 
 Ref<Texture> TextureProgress::get_over_texture() const{
@@ -302,4 +304,5 @@ TextureProgress::TextureProgress()
 	rad_init_angle=0;
 	rad_center_off=Point2();
 	rad_max_degrees=360;
+	set_mouse_filter(MOUSE_FILTER_PASS);
 }
diff --git a/scene/gui/texture_rect.cpp b/scene/gui/texture_rect.cpp
index cbb077ef5d..6a4b59c5ec 100644
--- a/scene/gui/texture_rect.cpp
+++ b/scene/gui/texture_rect.cpp
@@ -160,7 +160,7 @@ TextureRect::TextureRect() {
 
 
 	expand=false;
-	set_mouse_filter(MOUSE_FILTER_IGNORE);
+	set_mouse_filter(MOUSE_FILTER_PASS);
 	stretch_mode=STRETCH_SCALE_ON_EXPAND;
 }
 
diff --git a/scene/gui/tree.cpp b/scene/gui/tree.cpp
index c52283ecde..1a7392f27e 100644
--- a/scene/gui/tree.cpp
+++ b/scene/gui/tree.cpp
@@ -1008,7 +1008,7 @@ int Tree::draw_item(const Point2i& p_pos,const Point2& p_draw_ofs, const Size2&
 	/* Draw label, if height fits */
 
 
-		bool skip=(p_item==root && hide_root);
+	bool skip=(p_item==root && hide_root);
 
 
 	if (!skip && (p_pos.y+label_h-cache.offset.y)>0) {
@@ -1711,8 +1711,15 @@ int Tree::propagate_mouse_event(const Point2i &p_pos,int x_ofs,int y_ofs,bool p_
 			case TreeItem::CELL_MODE_CHECK: {
 
 				bring_up_editor=false; //checkboxes are not edited with editor
-				p_item->set_checked(col, !c.checked);
-				item_edited(col, p_item);
+				if (force_edit_checkbox_only_on_checkbox) {
+					if (x < cache.checked->get_width()) {
+						p_item->set_checked(col, !c.checked);
+						item_edited(col, p_item);
+					}
+				} else {
+					p_item->set_checked(col, !c.checked);
+					item_edited(col, p_item);
+				}
 				click_handled = true;
 				//p_item->edited_signal.call(col);
 
@@ -3555,6 +3562,16 @@ bool Tree::get_single_select_cell_editing_only_when_already_selected() const {
 	return force_select_on_already_selected;
 }
 
+void Tree::set_edit_checkbox_cell_only_when_checkbox_is_pressed(bool p_enable) {
+
+	force_edit_checkbox_only_on_checkbox=p_enable;
+}
+
+bool Tree::get_edit_checkbox_cell_only_when_checkbox_is_pressed() const {
+
+	return force_edit_checkbox_only_on_checkbox;
+}
+
 
 void Tree::set_allow_rmb_select(bool p_allow) {
 
@@ -3733,6 +3750,7 @@ Tree::Tree() {
 	force_select_on_already_selected=false;
 
 	allow_rmb_select=false;
+	force_edit_checkbox_only_on_checkbox=false;
 
 	set_clip_contents(true);
 }
diff --git a/scene/gui/tree.h b/scene/gui/tree.h
index d715ff4772..351cc4cb50 100644
--- a/scene/gui/tree.h
+++ b/scene/gui/tree.h
@@ -452,6 +452,7 @@ friend class TreeItem;
 	bool scrolling;
 
 	bool force_select_on_already_selected;
+	bool force_edit_checkbox_only_on_checkbox;
 
 	bool hide_folding;
 
@@ -531,6 +532,10 @@ public:
 	void set_single_select_cell_editing_only_when_already_selected(bool p_enable);
 	bool get_single_select_cell_editing_only_when_already_selected() const;
 
+	void set_edit_checkbox_cell_only_when_checkbox_is_pressed(bool p_enable);
+	bool get_edit_checkbox_cell_only_when_checkbox_is_pressed() const;
+
+
 	void set_allow_rmb_select(bool p_allow);
 	bool get_allow_rmb_select() const;
 
diff --git a/scene/gui/video_player.cpp b/scene/gui/video_player.cpp
index 907b5a771f..46c0eeca65 100644
--- a/scene/gui/video_player.cpp
+++ b/scene/gui/video_player.cpp
@@ -28,6 +28,7 @@
 /*************************************************************************/
 #include "video_player.h"
 #include "os/os.h"
+#include "servers/audio_server.h"
 /*
 
 int VideoPlayer::InternalStream::get_channel_count() const {
diff --git a/scene/main/viewport.cpp b/scene/main/viewport.cpp
index 14acf9583d..63c86c8e9d 100644
--- a/scene/main/viewport.cpp
+++ b/scene/main/viewport.cpp
@@ -1901,9 +1901,28 @@ void Viewport::_gui_input_event(InputEvent p_event) {
 					}*/
 #endif
 
-				if (gui.mouse_focus->get_focus_mode()!=Control::FOCUS_NONE && gui.mouse_focus!=gui.key_focus && p_event.mouse_button.button_index==BUTTON_LEFT) {
-					// also get keyboard focus
-					gui.mouse_focus->grab_focus();
+				if (p_event.mouse_button.button_index==BUTTON_LEFT) { //assign focus
+					CanvasItem *ci=gui.mouse_focus;
+					while(ci) {
+
+						Control *control = ci->cast_to<Control>();
+						if (control) {
+							if (control->get_focus_mode()!=Control::FOCUS_NONE) {
+								if (control!=gui.key_focus) {
+									control->grab_focus();
+								}
+								break;
+							}
+
+							if (control->data.mouse_filter==Control::MOUSE_FILTER_STOP)
+								break;
+						}
+
+						if (ci->is_set_as_toplevel())
+							break;
+
+						ci=ci->get_parent_item();
+					}
 				}
 
 
diff --git a/scene/register_scene_types.cpp b/scene/register_scene_types.cpp
index 1932f9cbf6..7e8a033c40 100644
--- a/scene/register_scene_types.cpp
+++ b/scene/register_scene_types.cpp
@@ -139,7 +139,7 @@
 
 #include "scene/main/timer.h"
 
-//#include "scene/audio/stream_player.h"
+#include "scene/audio/audio_player.h"
 //#include "scene/audio/event_player.h"
 //#include "scene/audio/sound_room_params.h"
 #include "scene/resources/sphere_shape.h"
@@ -177,7 +177,7 @@
 #include "scene/resources/world_2d.h"
 
 //#include "scene/resources/sample_library.h"
-#include "scene/resources/audio_stream.h"
+//#include "scene/resources/audio_stream.h"
 #include "scene/resources/gibberish_stream.h"
 #include "scene/resources/bit_mask.h"
 #include "scene/resources/color_ramp.h"
@@ -592,10 +592,7 @@ void register_scene_types() {
 
 	OS::get_singleton()->yield(); //may take time to init
 
-	ClassDB::register_virtual_class<AudioStream>();
-	ClassDB::register_virtual_class<AudioStreamPlayback>();
-//TODO: Adapt to the new AudioStream API or drop (GH-3307)
-	//ClassDB::register_type<AudioStreamGibberish>();
+	ClassDB::register_class<AudioPlayer>();
 	ClassDB::register_virtual_class<VideoStream>();
 
 	OS::get_singleton()->yield(); //may take time to init
diff --git a/scene/resources/audio_stream_resampled.h b/scene/resources/audio_stream_resampled.h
index 761643b027..7ceb6cef84 100644
--- a/scene/resources/audio_stream_resampled.h
+++ b/scene/resources/audio_stream_resampled.h
@@ -29,7 +29,7 @@
 #ifndef AUDIO_STREAM_RESAMPLED_H
 #define AUDIO_STREAM_RESAMPLED_H
 
-#include "scene/resources/audio_stream.h"
+//#include "scene/resources/audio_stream.h"
 
 #if 0
 
diff --git a/scene/resources/default_theme/theme_data.h b/scene/resources/default_theme/theme_data.h
index 46fd770a27..5b5868ba14 100644
--- a/scene/resources/default_theme/theme_data.h
+++ b/scene/resources/default_theme/theme_data.h
@@ -45,7 +45,7 @@ static const unsigned char button_normal_png[]={
 
 
 static const unsigned char button_pressed_png[]={
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 };
 
 
@@ -540,12 +540,12 @@ static const unsigned char vslider_bg_png[]={
 
 
 static const unsigned char vslider_grabber_png[]={
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 };
 
 
 static const unsigned char vslider_grabber_hl_png[]={
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+0x89,0x50,0x4e,0x47,0xd,0xa,0x1a,0xa,0x0,0x0,0x0,0xd,0x49,0x48,0x44,0x52,0x0,0x0,0x0,0x10,0x0,0x0,0x0,0x10,0x8,0x3,0x0,0x0,0x0,0x28,0x2d,0xf,0x53,0x0,0x0,0x0,0xc3,0x50,0x4c,0x54,0x45,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x17,0x2a,0x29,0x3a,0x69,0x69,0x5b,0xa6,0xa5,0x61,0xb3,0xbc,0x63,0xb7,0xc8,0x65,0xbb,0xca,0x60,0xaf,0xb1,0x48,0x83,0x83,0x0,0x0,0x0,0x0,0x0,0x0,0x8,0xf,0xf,0x55,0x9b,0x9a,0x60,0xb2,0xbd,0x5e,0xb1,0xcd,0x61,0xb3,0xc2,0x0,0x0,0x0,0x27,0x48,0x47,0x62,0xb4,0xbd,0x51,0x93,0x92,0x68,0xc0,0xcf,0x0,0x0,0x0,0x56,0x9d,0x9c,0x68,0xc1,0xcf,0x2d,0x52,0x52,0x63,0xb7,0xbf,0x52,0x96,0x95,0x62,0xb3,0xbf,0x5e,0xb0,0xcd,0x0,0x0,0x0,0x3,0x5,0x5,0x36,0x63,0x63,0x63,0xb4,0xb6,0x60,0xb1,0xbc,0x63,0xb7,0xc7,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x0,0x55,0xa3,0xc8,0x4f,0x98,0xc4,0x4b,0x93,0xc2,0x4c,0x94,0xc2,0x54,0xa2,0xc8,0x5a,0xab,0xcb,0x4e,0x97,0xc4,0x49,0x8f,0xc0,0x47,0x8c,0xbf,0x48,0x8e,0xc0,0x52,0x9e,0xc6,0x51,0x9d,0xc6,0x5a,0xac,0xcc,0x53,0x9f,0xc7,0x4d,0x96,0xc3,0x4b,0x92,0xc2,0xff,0xff,0xff,0x76,0xbd,0x27,0x7a,0x0,0x0,0x0,0x1,0x74,0x52,0x4e,0x53,0x0,0x40,0xe6,0xd8,0x66,0x0,0x0,0x0,0x1,0x62,0x4b,0x47,0x44,0x0,0x88,0x5,0x1d,0x48,0x0,0x0,0x0,0x9,0x70,0x48,0x59,0x73,0x0,0x0,0xb,0x13,0x0,0x0,0xb,0x13,0x1,0x0,0x9a,0x9c,0x18,0x0,0x0,0x0,0x7,0x74,0x49,0x4d,0x45,0x7,0xe1,0x1,0x12,0x1,0x36,0x11,0x34,0xd2,0xf,0x93,0x0,0x0,0x0,0x19,0x74,0x45,0x58,0x74,0x43,0x6f,0x6d,0x6d,0x65,0x6e,0x74,0x0,0x43,0x72,0x65,0x61,0x74,0x65,0x64,0x20,0x77,0x69,0x74,0x68,0x20,0x47,0x49,0x4d,0x50,0x57,0x81,0xe,0x17,0x0,0x0,0x0,0x48,0x49,0x44,0x41,0x54,0x18,0xd3,0x63,0x60,0xa0,0x12,0x10,0x14,0xe0,0xe7,0xe3,0x45,0xe2,0x4b,0x9a,0x18,0x1b,0x19,0x1a,0x48,0x88,0x8b,0xc1,0xe4,0x4d,0x2c,0x2d,0x80,0xc0,0xdc,0xcc,0x54,0x6,0x22,0x20,0x60,0x6c,0x1,0x1,0xe6,0x56,0x72,0x68,0x2,0xd6,0x8a,0xa8,0x5a,0x6c,0x94,0x11,0x86,0xda,0xdb,0xd9,0xaa,0xa9,0xaa,0x20,0x59,0xab,0xa3,0xad,0xc5,0x40,0x3d,0x0,0x0,0xbf,0x8e,0xc,0xed,0xed,0xc7,0x67,0x72,0x0,0x0,0x0,0x0,0x49,0x45,0x4e,0x44,0xae,0x42,0x60,0x82
 };
 
 
diff --git a/scene/resources/default_theme/vslider_grabber.png b/scene/resources/default_theme/vslider_grabber.png
index a61a6a57c9..afc490be45 100644
--- a/scene/resources/default_theme/vslider_grabber.png
+++ b/scene/resources/default_theme/vslider_grabber.png
diff --git a/scene/resources/default_theme/vslider_grabber_hl.png b/scene/resources/default_theme/vslider_grabber_hl.png
index 548dbbbff8..548972e115 100644
--- a/scene/resources/default_theme/vslider_grabber_hl.png
+++ b/scene/resources/default_theme/vslider_grabber_hl.png
diff --git a/servers/audio/SCsub b/servers/audio/SCsub
index ccc76e823f..afaffcfe93 100644
--- a/servers/audio/SCsub
+++ b/servers/audio/SCsub
@@ -5,3 +5,5 @@ Import('env')
 env.add_source_files(env.servers_sources, "*.cpp")
 
 Export('env')
+
+SConscript("effects/SCsub")
diff --git a/servers/audio/audio_effect.h b/servers/audio/audio_effect.h
index 2fcd22251b..02eb258f99 100644
--- a/servers/audio/audio_effect.h
+++ b/servers/audio/audio_effect.h
@@ -10,7 +10,7 @@ class AudioEffectInstance : public Reference {
 
 public:
 
-	virtual void process(AudioFrame *p_frames,int p_frame_count)=0;
+	virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count)=0;
 
 };
 
diff --git a/servers/audio/audio_filter_sw.h b/servers/audio/audio_filter_sw.h
index 0f3e2410fd..b711944ca8 100644
--- a/servers/audio/audio_filter_sw.h
+++ b/servers/audio/audio_filter_sw.h
@@ -65,7 +65,7 @@ public:
 		void set_filter(AudioFilterSW * p_filter);
 		void process(float *p_samples,int p_amount, int p_stride=1);
 		void update_coeffs();
-		inline void process_one(float& p_sample);
+		_ALWAYS_INLINE_ void process_one(float& p_sample);
 
 		Processor();
 	};
diff --git a/servers/audio/audio_stream.cpp b/servers/audio/audio_stream.cpp
new file mode 100644
index 0000000000..f4214838a1
--- /dev/null
+++ b/servers/audio/audio_stream.cpp
@@ -0,0 +1,86 @@
+/*************************************************************************/
+/*  audio_stream.cpp                                                     */
+/*************************************************************************/
+/*                       This file is part of:                           */
+/*                           GODOT ENGINE                                */
+/*                    http://www.godotengine.org                         */
+/*************************************************************************/
+/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur.                 */
+/*                                                                       */
+/* Permission is hereby granted, free of charge, to any person obtaining */
+/* a copy of this software and associated documentation files (the       */
+/* "Software"), to deal in the Software without restriction, including   */
+/* without limitation the rights to use, copy, modify, merge, publish,   */
+/* distribute, sublicense, and/or sell copies of the Software, and to    */
+/* permit persons to whom the Software is furnished to do so, subject to */
+/* the following conditions:                                             */
+/*                                                                       */
+/* The above copyright notice and this permission notice shall be        */
+/* included in all copies or substantial portions of the Software.       */
+/*                                                                       */
+/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
+/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
+/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
+/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
+/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
+/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
+/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
+/*************************************************************************/
+#include "audio_stream.h"
+
+//////////////////////////////
+
+
+
+void AudioStreamPlaybackResampled::_begin_resample() {
+
+	//clear cubic interpolation history
+	internal_buffer[0]=AudioFrame(0.0,0.0);
+	internal_buffer[1]=AudioFrame(0.0,0.0);
+	internal_buffer[2]=AudioFrame(0.0,0.0);
+	internal_buffer[3]=AudioFrame(0.0,0.0);
+	//mix buffer
+	_mix_internal(internal_buffer+4,INTERNAL_BUFFER_LEN);
+	mix_offset=0;
+}
+
+void AudioStreamPlaybackResampled::mix(AudioFrame* p_buffer,float p_rate_scale,int p_frames) {
+
+	float target_rate = AudioServer::get_singleton()->get_mix_rate() * p_rate_scale;
+
+	uint64_t mix_increment = uint64_t((get_stream_sampling_rate() / double(target_rate)) * double( FP_LEN ));
+
+	for(int i=0;i<p_frames;i++) {
+
+
+
+		uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
+		//standard cubic interpolation (great quality/performance ratio)
+		//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
+		float mu = (mix_offset&FP_MASK)/float(FP_LEN);
+		AudioFrame y0 = internal_buffer[idx-3];
+		AudioFrame y1 = internal_buffer[idx-2];
+		AudioFrame y2 = internal_buffer[idx-1];
+		AudioFrame y3 = internal_buffer[idx-0];
+
+		float mu2 = mu*mu;
+		AudioFrame a0 = y3 - y2 - y0 + y1;
+		AudioFrame a1 = y0 - y1 - a0;
+		AudioFrame a2 = y2 - y0;
+		AudioFrame a3 = y1;
+
+		p_buffer[i] = (a0*mu*mu2 + a1*mu2 + a2*mu + a3);
+
+		mix_offset+=mix_increment;
+
+		while ( (mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN ) {
+
+			internal_buffer[0]=internal_buffer[INTERNAL_BUFFER_LEN+0];
+			internal_buffer[1]=internal_buffer[INTERNAL_BUFFER_LEN+1];
+			internal_buffer[2]=internal_buffer[INTERNAL_BUFFER_LEN+2];
+			internal_buffer[3]=internal_buffer[INTERNAL_BUFFER_LEN+3];
+			_mix_internal(internal_buffer+4,INTERNAL_BUFFER_LEN);
+			mix_offset-=(INTERNAL_BUFFER_LEN<<FP_BITS);
+		}
+	}
+}
diff --git a/scene/resources/audio_stream.h b/servers/audio/audio_stream.h
index b79707cd32..d08fedb084 100644
--- a/scene/resources/audio_stream.h
+++ b/servers/audio/audio_stream.h
@@ -34,47 +34,65 @@
 
 class AudioStreamPlayback : public Reference {
 
-	GDCLASS( AudioStreamPlayback, Reference );
-protected:
-	static void _bind_methods();
-public:
+	GDCLASS( AudioStreamPlayback, Reference )
 
+public:
 
-	virtual void play(float p_from_pos=0)=0;
+	virtual void start(float p_from_pos=0.0)=0;
 	virtual void stop()=0;
 	virtual bool is_playing() const=0;
 
-	virtual void set_loop(bool p_enable)=0;
-	virtual bool has_loop() const=0;
-
-	virtual void set_loop_restart_time(float p_time)=0;
-
-	virtual int get_loop_count() const=0;
+	virtual int get_loop_count() const=0; //times it looped
 
 	virtual float get_pos() const=0;
 	virtual void seek_pos(float p_time)=0;
 
-	virtual int mix(int16_t* p_bufer,int p_frames)=0;
+	virtual void mix(AudioFrame* p_bufer,float p_rate_scale,int p_frames)=0;
 
-	virtual float get_length() const=0;
-	virtual String get_stream_name() const=0;
+	virtual float get_length() const=0; //if supported, otherwise return 0
 
-	virtual int get_channels() const=0;
-	virtual int get_mix_rate() const=0;
-	virtual int get_minimum_buffer_size() const=0;
 
 };
 
-class AudioStream : public Resource {
+class AudioStreamPlaybackResampled : public AudioStreamPlayback {
+
+	GDCLASS( AudioStreamPlaybackResampled, AudioStreamPlayback )
+
+
+
+	enum {
+		FP_BITS=16, //fixed point used for resampling
+		FP_LEN=(1<<FP_BITS),
+		FP_MASK=FP_LEN-1,
+		INTERNAL_BUFFER_LEN=256,
+		CUBIC_INTERP_HISTORY=4
+	};
 
-	GDCLASS( AudioStream, Resource );
-	OBJ_SAVE_TYPE( AudioStream ); //children are all saved as AudioStream, so they can be exchanged
+	AudioFrame internal_buffer[INTERNAL_BUFFER_LEN+CUBIC_INTERP_HISTORY];
+	uint64_t mix_offset;
 
 protected:
-	static void _bind_methods();
+	void _begin_resample();
+	virtual void _mix_internal(AudioFrame* p_bufer,int p_frames)=0;
+	virtual float get_stream_sampling_rate()=0;
+
+public:
+
+	virtual void mix(AudioFrame* p_bufer,float p_rate_scale,int p_frames);
+
+	AudioStreamPlaybackResampled() { mix_offset=0;  }
+};
+
+class AudioStream : public Resource {
+
+	GDCLASS( AudioStream, Resource )
+	OBJ_SAVE_TYPE( AudioStream ) //children are all saved as AudioStream, so they can be exchanged
+
+
 public:
 
 	virtual Ref<AudioStreamPlayback> instance_playback()=0;
+	virtual String get_stream_name() const=0;
 
 
 };
diff --git a/servers/audio/effects/SCsub b/servers/audio/effects/SCsub
new file mode 100644
index 0000000000..ccc76e823f
--- /dev/null
+++ b/servers/audio/effects/SCsub
@@ -0,0 +1,7 @@
+#!/usr/bin/env python
+
+Import('env')
+
+env.add_source_files(env.servers_sources, "*.cpp")
+
+Export('env')
diff --git a/servers/audio/effects/audio_effect_amplify.cpp b/servers/audio/effects/audio_effect_amplify.cpp
new file mode 100644
index 0000000000..d723f8d2fe
--- /dev/null
+++ b/servers/audio/effects/audio_effect_amplify.cpp
@@ -0,0 +1,50 @@
+#include "audio_effect_amplify.h"
+
+
+void AudioEffectAmplifyInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
+
+
+	//multiply volume interpolating to avoid clicks if this changes
+	float volume_db = base->volume_db;
+	float vol = Math::db2linear(mix_volume_db);
+	float vol_inc = (Math::db2linear(volume_db) - vol)/float(p_frame_count);
+
+	for(int i=0;i<p_frame_count;i++) {
+		p_dst_frames[i]=p_src_frames[i]*vol;
+		vol+=vol_inc;
+	}
+	//set volume for next mix
+	mix_volume_db = volume_db;
+
+}
+
+
+Ref<AudioEffectInstance> AudioEffectAmplify::instance() {
+	Ref<AudioEffectAmplifyInstance> ins;
+	ins.instance();
+	ins->base=Ref<AudioEffectAmplify>(this);
+	ins->mix_volume_db=volume_db;
+	return ins;
+}
+
+void AudioEffectAmplify::set_volume_db(float p_volume) {
+	volume_db=p_volume;
+}
+
+float AudioEffectAmplify::get_volume_db() const {
+
+	return volume_db;
+}
+
+void AudioEffectAmplify::_bind_methods() {
+
+	ClassDB::bind_method(_MD("set_volume_db","volume"),&AudioEffectAmplify::set_volume_db);
+	ClassDB::bind_method(_MD("get_volume_db"),&AudioEffectAmplify::get_volume_db);
+
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"volume_db",PROPERTY_HINT_RANGE,"-80,24,0.01"),_SCS("set_volume_db"),_SCS("get_volume_db"));
+}
+
+AudioEffectAmplify::AudioEffectAmplify()
+{
+	volume_db=0;
+}
diff --git a/servers/audio/effects/audio_effect_amplify.h b/servers/audio/effects/audio_effect_amplify.h
new file mode 100644
index 0000000000..921054e2cd
--- /dev/null
+++ b/servers/audio/effects/audio_effect_amplify.h
@@ -0,0 +1,40 @@
+#ifndef AUDIOEFFECTAMPLIFY_H
+#define AUDIOEFFECTAMPLIFY_H
+
+#include "servers/audio/audio_effect.h"
+
+class AudioEffectAmplify;
+
+class AudioEffectAmplifyInstance : public AudioEffectInstance {
+	GDCLASS(AudioEffectAmplifyInstance,AudioEffectInstance)
+friend class AudioEffectAmplify;
+	Ref<AudioEffectAmplify> base;
+
+	float mix_volume_db;
+public:
+
+	virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+
+};
+
+
+class AudioEffectAmplify : public AudioEffect {
+	GDCLASS(AudioEffectAmplify,AudioEffect)
+
+friend class AudioEffectAmplifyInstance;
+	float volume_db;
+
+protected:
+
+	static void _bind_methods();
+public:
+
+
+	Ref<AudioEffectInstance> instance();
+	void set_volume_db(float p_volume);
+	float get_volume_db() const;
+
+	AudioEffectAmplify();
+};
+
+#endif // AUDIOEFFECTAMPLIFY_H
diff --git a/servers/audio/effects/audio_effect_eq.cpp b/servers/audio/effects/audio_effect_eq.cpp
new file mode 100644
index 0000000000..3c6a684224
--- /dev/null
+++ b/servers/audio/effects/audio_effect_eq.cpp
@@ -0,0 +1,122 @@
+#include "audio_effect_eq.h"
+#include "servers/audio_server.h"
+
+
+void AudioEffectEQInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
+
+	int band_count = bands[0].size();
+	EQ::BandProcess *proc_l = bands[0].ptr();
+	EQ::BandProcess *proc_r = bands[1].ptr();
+	float *bgain = gains.ptr();
+	for(int i=0;i<band_count;i++) {
+		bgain[i]=Math::db2linear(base->gain[i]);
+	}
+
+
+	for(int i=0;i<p_frame_count;i++) {
+
+		AudioFrame src = p_src_frames[i];
+		AudioFrame dst = AudioFrame(0,0);
+
+		for(int j=0;j<band_count;j++) {
+
+			float l = src.l;
+			float r = src.r;
+
+			proc_l[j].process_one(l);
+			proc_r[j].process_one(r);
+
+			dst.l+=l * bgain[j];
+			dst.r+=r * bgain[j];
+		}
+
+		p_dst_frames[i]=dst;
+	}
+
+}
+
+
+Ref<AudioEffectInstance> AudioEffectEQ::instance() {
+	Ref<AudioEffectEQInstance> ins;
+	ins.instance();
+	ins->base=Ref<AudioEffectEQ>(this);
+	ins->gains.resize(eq.get_band_count());
+	for(int i=0;i<2;i++) {
+		ins->bands[i].resize(eq.get_band_count());
+		for(int j=0;j<ins->bands[i].size();j++) {
+			ins->bands[i][j]=eq.get_band_processor(j);
+		}
+	}
+
+	return ins;
+}
+
+void AudioEffectEQ::set_band_gain_db(int p_band,float p_volume) {
+	ERR_FAIL_INDEX(p_band,gain.size());
+	gain[p_band]=p_volume;
+}
+
+float AudioEffectEQ::get_band_gain_db(int p_band) const {
+	ERR_FAIL_INDEX_V(p_band,gain.size(),0);
+
+	return gain[p_band];
+}
+int AudioEffectEQ::get_band_count() const {
+	return gain.size();
+}
+
+bool AudioEffectEQ::_set(const StringName& p_name, const Variant& p_value) {
+
+	const Map<StringName,int>::Element *E=prop_band_map.find(p_name);
+	if (E) {
+		set_band_gain_db(E->get(),p_value);
+		return true;
+	}
+
+	return false;
+}
+
+bool AudioEffectEQ::_get(const StringName& p_name,Variant &r_ret) const{
+
+	const Map<StringName,int>::Element *E=prop_band_map.find(p_name);
+	if (E) {
+		r_ret=get_band_gain_db(E->get());
+		return true;
+	}
+
+	return false;
+
+}
+
+void AudioEffectEQ::_get_property_list( List<PropertyInfo> *p_list) const{
+
+	for(int i=0;i<band_names.size();i++) {
+
+		p_list->push_back(PropertyInfo(Variant::REAL,band_names[i],PROPERTY_HINT_RANGE,"-60,24,0.1"));
+	}
+}
+
+
+
+void AudioEffectEQ::_bind_methods() {
+
+	ClassDB::bind_method(_MD("set_band_gain_db","band_idx","volume_db"),&AudioEffectEQ::set_band_gain_db);
+	ClassDB::bind_method(_MD("get_band_gain_db","band_idx"),&AudioEffectEQ::get_band_gain_db);
+	ClassDB::bind_method(_MD("get_band_count"),&AudioEffectEQ::get_band_count);
+
+}
+
+AudioEffectEQ::AudioEffectEQ(EQ::Preset p_preset)
+{
+
+
+	eq.set_mix_rate(AudioServer::get_singleton()->get_mix_rate());
+	eq.set_preset_band_mode(p_preset);
+	gain.resize(eq.get_band_count());
+	for(int i=0;i<gain.size();i++) {
+		gain[i]=0.0;
+		String name = "band_db/"+itos(eq.get_band_frequency(i))+"_hz";
+		prop_band_map[name]=i;
+		band_names.push_back(name);
+	}
+}
diff --git a/servers/audio/effects/audio_effect_eq.h b/servers/audio/effects/audio_effect_eq.h
new file mode 100644
index 0000000000..3fcc2c0056
--- /dev/null
+++ b/servers/audio/effects/audio_effect_eq.h
@@ -0,0 +1,72 @@
+#ifndef AUDIOEFFECTEQ_H
+#define AUDIOEFFECTEQ_H
+
+
+#include "servers/audio/audio_effect.h"
+#include "servers/audio/effects/eq.h"
+
+class AudioEffectEQ;
+
+class AudioEffectEQInstance : public AudioEffectInstance {
+	GDCLASS(AudioEffectEQInstance,AudioEffectInstance)
+friend class AudioEffectEQ;
+	Ref<AudioEffectEQ> base;
+
+	Vector<EQ::BandProcess> bands[2];
+	Vector<float> gains;
+public:
+
+	virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+
+};
+
+
+class AudioEffectEQ : public AudioEffect {
+	GDCLASS(AudioEffectEQ,AudioEffect)
+
+friend class AudioEffectEQInstance;
+
+	EQ eq;
+	Vector<float> gain;
+	Map<StringName,int> prop_band_map;
+	Vector<String> band_names;
+
+protected:
+	bool _set(const StringName& p_name, const Variant& p_value);
+	bool _get(const StringName& p_name,Variant &r_ret) const;
+	void _get_property_list( List<PropertyInfo> *p_list) const;
+
+
+
+	static void _bind_methods();
+public:
+
+
+	Ref<AudioEffectInstance> instance();
+	void set_band_gain_db(int p_band,float p_volume);
+	float get_band_gain_db(int p_band) const;
+	int get_band_count() const;
+
+	AudioEffectEQ(EQ::Preset p_preset=EQ::PRESET_6_BANDS);
+};
+
+
+class AudioEffectEQ6 : public AudioEffectEQ {
+	GDCLASS(AudioEffectEQ6,AudioEffectEQ)
+public:
+	AudioEffectEQ6() : AudioEffectEQ(EQ::PRESET_6_BANDS) {}
+};
+
+class AudioEffectEQ10 : public AudioEffectEQ {
+	GDCLASS(AudioEffectEQ10,AudioEffectEQ)
+public:
+	AudioEffectEQ10() : AudioEffectEQ(EQ::PRESET_10_BANDS) {}
+};
+
+class AudioEffectEQ21 : public AudioEffectEQ {
+	GDCLASS(AudioEffectEQ21,AudioEffectEQ)
+public:
+	AudioEffectEQ21() : AudioEffectEQ(EQ::PRESET_21_BANDS) {}
+};
+
+#endif // AUDIOEFFECTEQ_H
diff --git a/servers/audio/effects/audio_effect_filter.cpp b/servers/audio/effects/audio_effect_filter.cpp
new file mode 100644
index 0000000000..4e54ea1f3e
--- /dev/null
+++ b/servers/audio/effects/audio_effect_filter.cpp
@@ -0,0 +1,151 @@
+#include "audio_effect_filter.h"
+#include "servers/audio_server.h"
+
+template<int S>
+void AudioEffectFilterInstance::_process_filter(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
+
+	for(int i=0;i<p_frame_count;i++) {
+		float f = p_src_frames[i].l;
+		filter_process[0][0].process_one(f);
+		if (S>1)
+			filter_process[0][1].process_one(f);
+		if (S>2)
+			filter_process[0][2].process_one(f);
+		if (S>3)
+			filter_process[0][3].process_one(f);
+
+		p_dst_frames[i].l=f;
+	}
+
+	for(int i=0;i<p_frame_count;i++) {
+		float f = p_src_frames[i].r;
+		filter_process[1][0].process_one(f);
+		if (S>1)
+			filter_process[1][1].process_one(f);
+		if (S>2)
+			filter_process[1][2].process_one(f);
+		if (S>3)
+			filter_process[1][3].process_one(f);
+
+		p_dst_frames[i].r=f;
+	}
+
+}
+
+void AudioEffectFilterInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
+
+	filter.set_cutoff(base->cutoff);
+	filter.set_gain(base->gain);
+	filter.set_resonance(base->resonance);
+	filter.set_mode(base->mode);
+	int stages = int(base->db)+1;
+	filter.set_stages(stages);
+	filter.set_sampling_rate(AudioServer::get_singleton()->get_mix_rate());
+
+	for(int i=0;i<2;i++) {
+		for(int j=0;j<4;j++) {
+			filter_process[i][j].update_coeffs();
+		}
+	}
+
+
+	if (stages==1) {
+		_process_filter<1>(p_src_frames,p_dst_frames,p_frame_count);
+	} else if (stages==2) {
+		_process_filter<2>(p_src_frames,p_dst_frames,p_frame_count);
+	} else if (stages==3) {
+		_process_filter<3>(p_src_frames,p_dst_frames,p_frame_count);
+	} else if (stages==4) {
+		_process_filter<4>(p_src_frames,p_dst_frames,p_frame_count);
+	}
+
+}
+
+
+AudioEffectFilterInstance::AudioEffectFilterInstance() {
+
+	for(int i=0;i<2;i++) {
+		for(int j=0;j<4;j++) {
+			filter_process[i][j].set_filter(&filter);
+		}
+	}
+
+}
+
+
+Ref<AudioEffectInstance> AudioEffectFilter::instance() {
+	Ref<AudioEffectFilterInstance> ins;
+	ins.instance();
+	ins->base=Ref<AudioEffectFilter>(this);
+
+	return ins;
+}
+
+void AudioEffectFilter::set_cutoff(float p_freq) {
+
+	cutoff=p_freq;
+}
+
+float AudioEffectFilter::get_cutoff() const{
+
+	return cutoff;
+}
+
+void AudioEffectFilter::set_resonance(float p_amount){
+
+	resonance=p_amount;
+}
+float AudioEffectFilter::get_resonance() const{
+
+	return resonance;
+}
+
+void AudioEffectFilter::set_gain(float p_amount){
+
+	gain=p_amount;
+}
+float AudioEffectFilter::get_gain() const {
+
+	return gain;
+}
+
+
+
+void AudioEffectFilter::set_db(FilterDB p_db) {
+	db=p_db;
+}
+
+AudioEffectFilter::FilterDB AudioEffectFilter::get_db() const {
+
+	return db;
+}
+
+void AudioEffectFilter::_bind_methods() {
+
+	ClassDB::bind_method(_MD("set_cutoff","freq"),&AudioEffectFilter::set_cutoff);
+	ClassDB::bind_method(_MD("get_cutoff"),&AudioEffectFilter::get_cutoff);
+
+	ClassDB::bind_method(_MD("set_resonance","amount"),&AudioEffectFilter::set_resonance);
+	ClassDB::bind_method(_MD("get_resonance"),&AudioEffectFilter::get_resonance);
+
+	ClassDB::bind_method(_MD("set_gain","amount"),&AudioEffectFilter::set_gain);
+	ClassDB::bind_method(_MD("get_gain"),&AudioEffectFilter::get_gain);
+
+	ClassDB::bind_method(_MD("set_db","amount"),&AudioEffectFilter::set_db);
+	ClassDB::bind_method(_MD("get_db"),&AudioEffectFilter::get_db);
+
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"cutoff_hz",PROPERTY_HINT_RANGE,"1,40000,0.1"),_SCS("set_cutoff"),_SCS("get_cutoff"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"resonance",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_resonance"),_SCS("get_resonance"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"gain",PROPERTY_HINT_RANGE,"0,4,0.01"),_SCS("set_gain"),_SCS("get_gain"));
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"dB",PROPERTY_HINT_ENUM,"6db,12db,18db,24db"),_SCS("set_db"),_SCS("get_db"));
+}
+
+AudioEffectFilter::AudioEffectFilter(AudioFilterSW::Mode p_mode)
+{
+
+	mode=p_mode;
+	cutoff=2000;
+	resonance=0.5;
+	gain=1.0;
+	db=FILTER_6DB;
+}
diff --git a/servers/audio/effects/audio_effect_filter.h b/servers/audio/effects/audio_effect_filter.h
new file mode 100644
index 0000000000..7b5f1f1a1c
--- /dev/null
+++ b/servers/audio/effects/audio_effect_filter.h
@@ -0,0 +1,125 @@
+#ifndef AUDIOEFFECTFILTER_H
+#define AUDIOEFFECTFILTER_H
+
+#include "servers/audio/audio_effect.h"
+#include "servers/audio/audio_filter_sw.h"
+
+class AudioEffectFilter;
+
+class AudioEffectFilterInstance : public AudioEffectInstance {
+	GDCLASS(AudioEffectFilterInstance,AudioEffectInstance)
+friend class AudioEffectFilter;
+
+	Ref<AudioEffectFilter> base;
+
+	AudioFilterSW filter;
+	AudioFilterSW::Processor filter_process[2][4];
+
+	template<int S>
+	void _process_filter(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+public:
+
+	virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+
+	AudioEffectFilterInstance();
+};
+
+
+class AudioEffectFilter : public AudioEffect {
+	GDCLASS(AudioEffectFilter,AudioEffect)
+public:
+
+	enum FilterDB {
+	     FILTER_6DB,
+	     FILTER_12DB,
+	     FILTER_18DB,
+	     FILTER_24DB,
+	};
+	friend class AudioEffectFilterInstance;
+
+	AudioFilterSW::Mode mode;
+	float cutoff;
+	float resonance;
+	float gain;
+	FilterDB db;
+
+
+protected:
+
+
+	static void _bind_methods();
+public:
+
+	void set_cutoff(float p_freq);
+	float get_cutoff() const;
+
+	void set_resonance(float p_amount);
+	float get_resonance() const;
+
+	void set_gain(float p_amount);
+	float get_gain() const;
+
+	void set_db(FilterDB p_db);
+	FilterDB get_db() const;
+
+	Ref<AudioEffectInstance> instance();
+
+	AudioEffectFilter(AudioFilterSW::Mode p_mode=AudioFilterSW::LOWPASS);
+};
+
+VARIANT_ENUM_CAST(AudioEffectFilter::FilterDB)
+
+class AudioEffectLowPass : public AudioEffectFilter {
+	GDCLASS(AudioEffectLowPass,AudioEffectFilter)
+public:
+
+	AudioEffectLowPass() : AudioEffectFilter(AudioFilterSW::LOWPASS) {}
+};
+
+class AudioEffectHighPass : public AudioEffectFilter {
+	GDCLASS(AudioEffectHighPass,AudioEffectFilter)
+public:
+
+	AudioEffectHighPass() : AudioEffectFilter(AudioFilterSW::HIGHPASS) {}
+};
+
+class AudioEffectBandPass : public AudioEffectFilter {
+	GDCLASS(AudioEffectBandPass,AudioEffectFilter)
+public:
+
+	AudioEffectBandPass() : AudioEffectFilter(AudioFilterSW::BANDPASS) {}
+};
+
+class AudioEffectNotchPass : public AudioEffectFilter {
+	GDCLASS(AudioEffectNotchPass,AudioEffectFilter)
+public:
+
+	AudioEffectNotchPass() : AudioEffectFilter(AudioFilterSW::NOTCH) {}
+};
+
+class AudioEffectBandLimit : public AudioEffectFilter {
+	GDCLASS(AudioEffectBandLimit,AudioEffectFilter)
+public:
+
+	AudioEffectBandLimit() : AudioEffectFilter(AudioFilterSW::BANDLIMIT) {}
+};
+
+
+class AudioEffectLowShelf : public AudioEffectFilter {
+	GDCLASS(AudioEffectLowShelf,AudioEffectFilter)
+public:
+
+	AudioEffectLowShelf() : AudioEffectFilter(AudioFilterSW::LOWSHELF) {}
+};
+
+
+class AudioEffectHighShelf : public AudioEffectFilter {
+	GDCLASS(AudioEffectHighShelf,AudioEffectFilter)
+public:
+
+	AudioEffectHighShelf() : AudioEffectFilter(AudioFilterSW::HIGHSHELF) {}
+};
+
+
+
+#endif // AUDIOEFFECTFILTER_H
diff --git a/servers/audio/effects/audio_effect_reverb.cpp b/servers/audio/effects/audio_effect_reverb.cpp
new file mode 100644
index 0000000000..749814fd76
--- /dev/null
+++ b/servers/audio/effects/audio_effect_reverb.cpp
@@ -0,0 +1,182 @@
+#include "audio_effect_reverb.h"
+#include "servers/audio_server.h"
+void AudioEffectReverbInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
+
+	for(int i=0;i<2;i++) {
+		Reverb &r=reverb[i];
+
+		r.set_predelay( base->predelay);
+		r.set_predelay_feedback( base->predelay_fb );
+		r.set_highpass( base->hpf );
+		r.set_room_size( base->room_size );
+		r.set_damp( base->damping );
+		r.set_extra_spread( base->spread );
+		r.set_wet( base->wet );
+		r.set_dry( base->dry );
+	}
+
+	int todo = p_frame_count;
+	int offset=0;
+
+	while(todo) {
+
+		int to_mix = MIN(todo,Reverb::INPUT_BUFFER_MAX_SIZE);
+
+		for(int j=0;j<to_mix;j++) {
+			tmp_src[j]=p_src_frames[offset+j].l;
+		}
+
+		reverb[0].process(tmp_src,tmp_dst,to_mix);
+
+		for(int j=0;j<to_mix;j++) {
+			p_dst_frames[offset+j].l=tmp_dst[j];
+			tmp_src[j]=p_src_frames[offset+j].r;
+		}
+
+		reverb[1].process(tmp_src,tmp_dst,to_mix);
+
+		for(int j=0;j<to_mix;j++) {
+			p_dst_frames[offset+j].r=tmp_dst[j];
+		}
+
+		offset+=to_mix;
+		todo-=to_mix;
+	}
+}
+
+AudioEffectReverbInstance::AudioEffectReverbInstance() {
+
+	reverb[0].set_mix_rate( AudioServer::get_singleton()->get_mix_rate() );
+	reverb[0].set_extra_spread_base(0);
+	reverb[1].set_mix_rate( AudioServer::get_singleton()->get_mix_rate() );
+	reverb[1].set_extra_spread_base(0.000521); //for stereo effect
+
+}
+
+Ref<AudioEffectInstance> AudioEffectReverb::instance() {
+	Ref<AudioEffectReverbInstance> ins;
+	ins.instance();
+	ins->base=Ref<AudioEffectReverb>(this);
+	return ins;
+}
+
+void AudioEffectReverb::set_predelay_msec(float p_msec) {
+
+	predelay=p_msec;
+}
+
+void AudioEffectReverb::set_predelay_feedback(float p_feedback){
+
+	predelay_fb=p_feedback;
+}
+void AudioEffectReverb::set_room_size(float p_size){
+
+	room_size=p_size;
+}
+void AudioEffectReverb::set_damping(float p_damping){
+
+	damping=p_damping;
+}
+void AudioEffectReverb::set_spread(float p_spread){
+
+	spread=p_spread;
+}
+
+void AudioEffectReverb::set_dry(float p_dry){
+
+	dry=p_dry;
+}
+void AudioEffectReverb::set_wet(float p_wet){
+
+	wet=p_wet;
+}
+void AudioEffectReverb::set_hpf(float p_hpf) {
+
+	hpf=p_hpf;
+}
+
+float AudioEffectReverb::get_predelay_msec() const {
+
+	return predelay;
+}
+float AudioEffectReverb::get_predelay_feedback() const {
+
+	return predelay_fb;
+}
+float AudioEffectReverb::get_room_size() const {
+
+	return room_size;
+}
+float AudioEffectReverb::get_damping() const {
+
+	return damping;
+}
+float AudioEffectReverb::get_spread() const {
+
+	return spread;
+}
+float AudioEffectReverb::get_dry() const {
+
+	return dry;
+}
+float AudioEffectReverb::get_wet() const {
+
+	return wet;
+}
+float AudioEffectReverb::get_hpf() const {
+
+	return hpf;
+}
+
+
+void AudioEffectReverb::_bind_methods() {
+
+
+	ClassDB::bind_method(_MD("set_predelay_msec","msec"),&AudioEffectReverb::set_predelay_msec);
+	ClassDB::bind_method(_MD("get_predelay_msec"),&AudioEffectReverb::get_predelay_msec);
+
+	ClassDB::bind_method(_MD("set_predelay_feedback","feedback"),&AudioEffectReverb::set_predelay_feedback);
+	ClassDB::bind_method(_MD("get_predelay_feedback"),&AudioEffectReverb::get_predelay_feedback);
+
+	ClassDB::bind_method(_MD("set_room_size","size"),&AudioEffectReverb::set_room_size);
+	ClassDB::bind_method(_MD("get_room_size"),&AudioEffectReverb::get_room_size);
+
+	ClassDB::bind_method(_MD("set_damping","amount"),&AudioEffectReverb::set_damping);
+	ClassDB::bind_method(_MD("get_damping"),&AudioEffectReverb::get_damping);
+
+	ClassDB::bind_method(_MD("set_spread","amount"),&AudioEffectReverb::set_spread);
+	ClassDB::bind_method(_MD("get_spread"),&AudioEffectReverb::get_spread);
+
+	ClassDB::bind_method(_MD("set_dry","amount"),&AudioEffectReverb::set_dry);
+	ClassDB::bind_method(_MD("get_dry"),&AudioEffectReverb::get_dry);
+
+	ClassDB::bind_method(_MD("set_wet","amount"),&AudioEffectReverb::set_wet);
+	ClassDB::bind_method(_MD("get_wet"),&AudioEffectReverb::get_wet);
+
+	ClassDB::bind_method(_MD("set_hpf","amount"),&AudioEffectReverb::set_hpf);
+	ClassDB::bind_method(_MD("get_hpf"),&AudioEffectReverb::get_hpf);
+
+
+	ADD_GROUP("Predelay","predelay_");
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"predelay_msec",PROPERTY_HINT_RANGE,"20,500,1"),_SCS("set_predelay_msec"),_SCS("get_predelay_msec"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"predelay_feedback",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_predelay_msec"),_SCS("get_predelay_msec"));
+	ADD_GROUP("","");
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"room_size",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_room_size"),_SCS("get_room_size"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"damping",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_damping"),_SCS("get_damping"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"spread",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_spread"),_SCS("get_spread"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"hipass",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_hpf"),_SCS("get_hpf"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"dry",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_dry"),_SCS("get_dry"));
+	ADD_PROPERTY(PropertyInfo(Variant::REAL,"wet",PROPERTY_HINT_RANGE,"0,1,0.01"),_SCS("set_wet"),_SCS("get_wet"));
+}
+
+AudioEffectReverb::AudioEffectReverb() {
+	predelay=150;
+	predelay_fb=0.4;
+	hpf=0;
+	room_size=0.8;
+	damping=0.5;
+	spread=1.0;
+	dry=1.0;
+	wet=0.5;
+
+}
diff --git a/servers/audio/effects/audio_effect_reverb.h b/servers/audio/effects/audio_effect_reverb.h
new file mode 100644
index 0000000000..e05ffe422f
--- /dev/null
+++ b/servers/audio/effects/audio_effect_reverb.h
@@ -0,0 +1,76 @@
+#ifndef AUDIOEFFECTREVERB_H
+#define AUDIOEFFECTREVERB_H
+
+
+#include "servers/audio/audio_effect.h"
+#include "servers/audio/effects/reverb.h"
+
+class AudioEffectReverb;
+
+class AudioEffectReverbInstance : public AudioEffectInstance {
+	GDCLASS(AudioEffectReverbInstance,AudioEffectInstance)
+
+	Ref<AudioEffectReverb> base;
+
+	float tmp_src[Reverb::INPUT_BUFFER_MAX_SIZE];
+	float tmp_dst[Reverb::INPUT_BUFFER_MAX_SIZE];
+
+friend class AudioEffectReverb;
+
+	Reverb reverb[2];
+
+
+public:
+
+	virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+	AudioEffectReverbInstance();
+};
+
+
+class AudioEffectReverb : public AudioEffect {
+	GDCLASS(AudioEffectReverb,AudioEffect)
+
+friend class AudioEffectReverbInstance;
+
+	float predelay;
+	float predelay_fb;
+	float hpf;
+	float room_size;
+	float damping;
+	float spread;
+	float dry;
+	float wet;
+
+protected:
+
+	static void _bind_methods();
+public:
+
+
+	void set_predelay_msec(float p_msec);
+	void set_predelay_feedback(float p_feedback);
+	void set_room_size(float p_size);
+	void set_damping(float p_damping);
+	void set_spread(float p_spread);
+	void set_dry(float p_dry);
+	void set_wet(float p_wet);
+	void set_hpf(float p_hpf);
+
+	float get_predelay_msec() const;
+	float get_predelay_feedback() const;
+	float get_room_size() const;
+	float get_damping() const;
+	float get_spread() const;
+	float get_dry() const;
+	float get_wet() const;
+	float get_hpf() const;
+
+	Ref<AudioEffectInstance> instance();
+	void set_volume_db(float p_volume);
+	float get_volume_db() const;
+
+	AudioEffectReverb();
+};
+
+
+#endif // AUDIOEFFECTREVERB_H
diff --git a/servers/audio/effects/eq.cpp b/servers/audio/effects/eq.cpp
new file mode 100644
index 0000000000..14659585b8
--- /dev/null
+++ b/servers/audio/effects/eq.cpp
@@ -0,0 +1,218 @@
+//
+// C++ Interface: eq
+//
+// Description:
+//
+//
+// Author: reduzio@gmail.com (C) 2006
+//
+// Copyright: See COPYING file that comes with this distribution
+//
+//
+#include "eq.h"
+#include <math.h>
+#include "error_macros.h"
+
+#define POW2(v) ((v)*(v))
+
+/* Helper */
+ static int solve_quadratic(double a,double b,double c,double *r1, double *r2) {
+//solves quadractic and returns number of roots
+
+    double base=2*a;
+    if (base == 0.0f)
+	    return 0;
+
+    double squared=b*b-4*a*c;
+    if (squared<0.0)
+	    return 0;
+
+    squared=sqrt(squared);
+
+    *r1=(-b+squared)/base;
+    *r2=(-b-squared)/base;
+
+    if (*r1==*r2)
+	    return 1;
+    else
+	    return 2;
+ }
+
+EQ::BandProcess::BandProcess() {
+
+	c1=c2=c3=history.a1=history.a2=history.a3=0;
+	history.b1=history.b2=history.b3=0;
+}
+
+void EQ::recalculate_band_coefficients() {
+
+#define BAND_LOG( m_f ) ( log((m_f)) / log(2) )
+
+	for (int i=0;i<band.size();i++) {
+
+		double octave_size;
+
+		double frq=band[i].freq;
+
+		if (i==0) {
+
+			octave_size=BAND_LOG(band[1].freq)-BAND_LOG(frq);
+		} else if (i==(band.size()-1)) {
+
+			octave_size=BAND_LOG(frq)-BAND_LOG(band[i-1].freq);
+		} else {
+
+			double next=BAND_LOG(band[i+1].freq)-BAND_LOG(frq);
+			double prev=BAND_LOG(frq)-BAND_LOG(band[i-1].freq);
+			octave_size=(next+prev)/2.0;
+		}
+
+
+
+		double frq_l=round(frq/pow(2.0,octave_size/2.0));
+
+
+
+		double side_gain2=POW2(1.0/M_SQRT2);
+		double th=2.0*M_PI*frq/mix_rate;
+		double th_l=2.0*M_PI*frq_l/mix_rate;
+
+		double c2a=side_gain2 * POW2(cos(th))
+					- 2.0 * side_gain2 * cos(th_l) * cos(th)
+					+ side_gain2
+					- POW2(sin(th_l));
+
+		double c2b=2.0 * side_gain2 * POW2(cos(th_l))
+					+ side_gain2 * POW2(cos(th))
+					- 2.0 * side_gain2 * cos(th_l) * cos(th)
+					- side_gain2
+					+ POW2(sin(th_l));
+
+		double c2c=0.25 * side_gain2 * POW2(cos(th))
+					- 0.5 * side_gain2 * cos(th_l) * cos(th)
+					+ 0.25 * side_gain2
+					- 0.25 * POW2(sin(th_l));
+
+		//printf("band %i, precoefs = %f,%f,%f\n",i,c2a,c2b,c2c);
+
+		double r1,r2; //roots
+		int roots=solve_quadratic(c2a,c2b,c2c,&r1,&r2);
+
+		ERR_CONTINUE( roots==0 );
+
+		band[i].c1=2.0 * ((0.5-r1)/2.0);
+		band[i].c2=2.0 * r1;
+		band[i].c3=2.0 * (0.5+r1) * cos(th);
+		//printf("band %i, coefs = %f,%f,%f\n",i,(float)bands[i].c1,(float)bands[i].c2,(float)bands[i].c3);
+
+	}
+}
+
+void EQ::set_preset_band_mode(Preset p_preset) {
+
+
+	band.clear();
+
+#define PUSH_BANDS(m_bands) \
+	for (int i=0;i<m_bands;i++) { \
+		Band b; \
+		b.freq=bands[i];\
+		band.push_back(b);\
+	}
+
+	switch (p_preset) {
+
+		case PRESET_6_BANDS: {
+
+			static const double bands[] = {  32 ,  100 ,  320 ,  1e3,  3200,  10e3 };
+			PUSH_BANDS(6);
+
+		} break;
+
+		case PRESET_8_BANDS: {
+
+			static const double bands[] = { 32,72,192,512,1200,3000,7500,16e3 };
+
+			PUSH_BANDS(8);
+		} break;
+
+		case PRESET_10_BANDS: {
+			static const double bands[] = { 31.25, 62.5,  125 ,  250 ,  500 ,  1e3,  2e3,  4e3,  8e3,  16e3 };
+
+			PUSH_BANDS(10);
+
+		} break;
+
+		case PRESET_21_BANDS: {
+
+			static const double bands[] = {  22 ,  32 ,  44 ,  63 ,  90 ,  125 ,  175 ,  250 ,  350 ,  500 ,  700 ,  1e3,  1400 ,  2e3,  2800 ,  4e3,  5600 ,  8e3,  11e3,  16e3,  22e3 };
+			PUSH_BANDS(21);
+
+		} break;
+
+		case PRESET_31_BANDS: {
+
+			static const double bands[] = {  20,   25,   31.5,  40 ,  50 ,  63 ,  80 ,  100 ,  125 ,  160 ,  200 ,  250 ,  315 ,  400 ,  500 ,  630 ,  800 , 1e3 ,  1250  ,  1600 ,  2e3,  2500 ,  3150 ,  4e3,  5e3,  6300 ,  8e3,  10e3,  12500 ,  16e3,  20e3 };
+			PUSH_BANDS(31);
+		} break;
+
+	};
+
+	recalculate_band_coefficients();
+}
+
+int EQ::get_band_count() const {
+
+	return band.size();
+}
+float EQ::get_band_frequency(int p_band) {
+
+	ERR_FAIL_INDEX_V(p_band,band.size(),0);
+	return band[p_band].freq;
+}
+void EQ::set_bands(const Vector<float>& p_bands) {
+
+	band.resize(p_bands.size());
+	for (int i=0;i<p_bands.size();i++) {
+
+		band[i].freq=p_bands[i];
+	}
+
+	recalculate_band_coefficients();
+
+}
+
+void EQ::set_mix_rate(float p_mix_rate) {
+
+	mix_rate=p_mix_rate;
+	recalculate_band_coefficients();
+}
+
+EQ::BandProcess EQ::get_band_processor(int p_band) const {
+
+
+	EQ::BandProcess band_proc;
+
+	ERR_FAIL_INDEX_V(p_band,band.size(),band_proc);
+
+	band_proc.c1=band[p_band].c1;
+	band_proc.c2=band[p_band].c2;
+	band_proc.c3=band[p_band].c3;
+
+	return band_proc;
+
+
+}
+
+
+EQ::EQ()
+{
+	mix_rate=44100;
+}
+
+
+EQ::~EQ()
+{
+}
+
+
diff --git a/servers/audio/effects/eq.h b/servers/audio/effects/eq.h
new file mode 100644
index 0000000000..2c4668cd0b
--- /dev/null
+++ b/servers/audio/effects/eq.h
@@ -0,0 +1,106 @@
+//
+// C++ Interface: eq
+//
+// Description:
+//
+//
+// Author: reduzio@gmail.com (C) 2006
+//
+// Copyright: See COPYING file that comes with this distribution
+//
+//
+#ifndef EQ_FILTER_H
+#define EQ_FILTER_H
+
+
+#include "typedefs.h"
+#include "vector.h"
+
+
+/**
+@author Juan Linietsky
+*/
+
+class EQ {
+public:
+
+	enum Preset {
+
+		PRESET_6_BANDS,
+		PRESET_8_BANDS,
+		PRESET_10_BANDS,
+		PRESET_21_BANDS,
+		PRESET_31_BANDS
+	};
+
+
+
+	class BandProcess {
+
+	friend class EQ;
+		float c1,c2,c3;
+		struct History {
+			float a1,a2,a3;
+			float b1,b2,b3;
+
+		} history;
+
+	public:
+
+		inline void process_one(float & p_data);
+
+		BandProcess();
+	};
+
+private:
+	struct Band {
+
+		float freq;
+		float c1,c2,c3;
+	};
+
+	Vector<Band> band;
+
+	float mix_rate;
+
+	void recalculate_band_coefficients();
+
+public:
+
+
+	void set_mix_rate(float p_mix_rate);
+
+	int get_band_count() const;
+	void set_preset_band_mode(Preset p_preset);
+	void set_bands(const Vector<float>& p_bands);
+	BandProcess get_band_processor(int p_band) const;
+	float get_band_frequency(int p_band);
+
+	EQ();
+	~EQ();
+
+};
+
+
+/* Inline Function */
+
+inline void EQ::BandProcess::process_one(float & p_data) {
+
+
+	history.a1=p_data;
+
+	history.b1=	c1 * ( history.a1 - history.a3 )
+			+ c3 * history.b2
+			- c2 * history.b3;
+
+	p_data = history.b1;
+
+	history.a3=history.a2;
+	history.a2=history.a1;
+	history.b3=history.b2;
+	history.b2=history.b1;
+
+}
+
+
+#endif
diff --git a/servers/audio/effects/reverb.cpp b/servers/audio/effects/reverb.cpp
new file mode 100644
index 0000000000..2a4c728b3f
--- /dev/null
+++ b/servers/audio/effects/reverb.cpp
@@ -0,0 +1,363 @@
+//
+// C++ Interface: reverb
+//
+// Description:
+//
+//
+// Author: Juan Linietsky <reduzio@gmail.com>, (C) 2006
+//
+// Copyright: See COPYING file that comes with this distribution
+//
+//
+
+#include "reverb.h"
+#include <math.h>
+
+
+const float Reverb::comb_tunings[MAX_COMBS]={
+	//freeverb comb tunings
+	0.025306122448979593,
+	0.026938775510204082,
+	0.028956916099773241,
+	0.03074829931972789,
+	0.032244897959183672,
+	0.03380952380952381,
+	0.035306122448979592,
+	0.036666666666666667
+};
+
+const float Reverb::allpass_tunings[MAX_ALLPASS]={
+	//freeverb allpass tunings
+	0.0051020408163265302,
+	0.007732426303854875,
+	0.01,
+	0.012607709750566893
+};
+
+
+
+void Reverb::process(float *p_src,float *p_dst,int p_frames) {
+
+	if (p_frames>INPUT_BUFFER_MAX_SIZE)
+		p_frames=INPUT_BUFFER_MAX_SIZE;
+
+	int predelay_frames=lrint((params.predelay/1000.0)*params.mix_rate);
+	if (predelay_frames<10)
+		predelay_frames=10;
+	if (predelay_frames>=echo_buffer_size)
+		predelay_frames=echo_buffer_size-1;
+
+	for (int i=0;i<p_frames;i++) {
+
+		if (echo_buffer_pos>=echo_buffer_size)
+			echo_buffer_pos=0;
+
+		int read_pos=echo_buffer_pos-predelay_frames;
+		while (read_pos<0)
+			read_pos+=echo_buffer_size;
+
+		float in=undenormalise(echo_buffer[read_pos]*params.predelay_fb+p_src[i]);
+
+		echo_buffer[echo_buffer_pos]=in;
+
+		input_buffer[i]=in;
+
+		p_dst[i]=0; //take the chance and clear this
+
+		echo_buffer_pos++;
+	}
+
+	if (params.hpf>0) {
+		float hpaux=expf(-2.0*M_PI*params.hpf*6000/params.mix_rate);
+		float hp_a1=(1.0+hpaux)/2.0;
+		float hp_a2=-(1.0+hpaux)/2.0;
+		float hp_b1=hpaux;
+
+		for (int i=0;i<p_frames;i++) {
+
+			float in=input_buffer[i];
+			input_buffer[i]=in*hp_a1+hpf_h1*hp_a2+hpf_h2*hp_b1;
+			hpf_h2=input_buffer[i];
+			hpf_h1=in;
+		}
+	}
+
+	for (int i=0;i<MAX_COMBS;i++) {
+
+		Comb &c=comb[i];
+
+		int size_limit=c.size-lrintf((float)c.extra_spread_frames*(1.0-params.extra_spread));
+		for (int j=0;j<p_frames;j++) {
+
+			if (c.pos>=size_limit) //reset this now just in case
+				c.pos=0;
+
+			float out=undenormalise(c.buffer[c.pos]*c.feedback);
+			out=out*(1.0-c.damp)+c.damp_h*c.damp; //lowpass
+			c.damp_h=out;
+			c.buffer[c.pos]=input_buffer[j]+out;
+			p_dst[j]+=out;
+			c.pos++;
+		}
+
+	}
+
+
+	static const float allpass_feedback=0.7;
+	/* this one works, but the other version is just nicer....
+	int ap_size_limit[MAX_ALLPASS];
+
+	for (int i=0;i<MAX_ALLPASS;i++) {
+
+		AllPass &a=allpass[i];
+		ap_size_limit[i]=a.size-lrintf((float)a.extra_spread_frames*(1.0-params.extra_spread));
+	}
+
+	for (int i=0;i<p_frames;i++) {
+
+		float sample=p_dst[i];
+		float aux,in;
+		float AllPass*ap;
+
+#define PROCESS_ALLPASS(m_ap) 	\
+	ap=&allpass[m_ap];	\
+	if (ap->pos>=ap_size_limit[m_ap])	\
+		ap->pos=0;	\
+	aux=undenormalise(ap->buffer[ap->pos]);	\
+	in=sample;	\
+	sample=-in+aux;	\
+	ap->pos++;
+
+
+		PROCESS_ALLPASS(0);
+		PROCESS_ALLPASS(1);
+		PROCESS_ALLPASS(2);
+		PROCESS_ALLPASS(3);
+
+		p_dst[i]=sample;
+	}
+	*/
+
+	for (int i=0;i<MAX_ALLPASS;i++) {
+
+		AllPass &a=allpass[i];
+		int size_limit=a.size-lrintf((float)a.extra_spread_frames*(1.0-params.extra_spread));
+
+		for (int j=0;j<p_frames;j++) {
+
+			if (a.pos>=size_limit)
+				a.pos=0;
+
+			float aux=a.buffer[a.pos];
+			a.buffer[a.pos]=undenormalise(allpass_feedback*aux+p_dst[j]);
+			p_dst[j]=aux-allpass_feedback*a.buffer[a.pos];
+			a.pos++;
+
+		}
+	}
+
+	static const float wet_scale=0.6;
+
+	for (int i=0;i<p_frames;i++) {
+
+
+		p_dst[i]=p_dst[i]*params.wet*wet_scale+p_src[i]*params.dry;
+	}
+
+}
+
+
+void Reverb::set_room_size(float p_size) {
+
+	params.room_size=p_size;
+	update_parameters();
+
+}
+void Reverb::set_damp(float p_damp) {
+
+	params.damp=p_damp;
+	update_parameters();
+
+}
+void Reverb::set_wet(float p_wet) {
+
+	params.wet=p_wet;
+
+}
+
+void Reverb::set_dry(float p_dry) {
+
+	params.dry=p_dry;
+
+}
+
+void Reverb::set_predelay(float p_predelay) {
+
+	params.predelay=p_predelay;
+}
+void Reverb::set_predelay_feedback(float p_predelay_fb) {
+
+	params.predelay_fb=p_predelay_fb;
+
+}
+
+void Reverb::set_highpass(float p_frq) {
+
+	if (p_frq>1)
+		p_frq=1;
+	if (p_frq<0)
+		p_frq=0;
+	params.hpf=p_frq;
+}
+
+void Reverb::set_extra_spread(float p_spread) {
+
+	params.extra_spread=p_spread;
+
+}
+
+
+void Reverb::set_mix_rate(float p_mix_rate) {
+
+	params.mix_rate=p_mix_rate;
+	configure_buffers();
+}
+
+void Reverb::set_extra_spread_base(float p_sec) {
+
+	params.extra_spread_base=p_sec;
+	configure_buffers();
+}
+
+
+void Reverb::configure_buffers() {
+
+	clear_buffers(); //clear if necesary
+
+	for (int i=0;i<MAX_COMBS;i++) {
+
+		Comb &c=comb[i];
+
+
+		c.extra_spread_frames=lrint(params.extra_spread_base*params.mix_rate);
+
+		int len=lrint(comb_tunings[i]*params.mix_rate)+c.extra_spread_frames;
+		if (len<5)
+			len=5; //may this happen?
+
+		c.buffer = memnew_arr(float,len);
+		c.pos=0;
+		for (int j=0;j<len;j++)
+			c.buffer[j]=0;
+		c.size=len;
+
+	}
+
+	for (int i=0;i<MAX_ALLPASS;i++) {
+
+		AllPass &a=allpass[i];
+
+		a.extra_spread_frames=lrint(params.extra_spread_base*params.mix_rate);
+
+		int len=lrint(allpass_tunings[i]*params.mix_rate)+a.extra_spread_frames;
+		if (len<5)
+			len=5; //may this happen?
+
+		a.buffer = memnew_arr(float,len);
+		a.pos=0;
+		for (int j=0;j<len;j++)
+			a.buffer[j]=0;
+		a.size=len;
+	}
+
+	echo_buffer_size=(int)(((float)MAX_ECHO_MS/1000.0)*params.mix_rate+1.0);
+	echo_buffer = memnew_arr(float,echo_buffer_size);
+	for (int i=0;i<echo_buffer_size;i++) {
+
+		echo_buffer[i]=0;
+	}
+
+	echo_buffer_pos=0;
+}
+
+
+void Reverb::update_parameters() {
+
+	//more freeverb derived constants
+	static const float room_scale = 0.28f;
+	static const float room_offset = 0.7f;
+
+	for (int i=0;i<MAX_COMBS;i++) {
+
+		Comb &c=comb[i];
+		c.feedback=room_offset+params.room_size*room_scale;
+		if (c.feedback<room_offset)
+			c.feedback=room_offset;
+		else if (c.feedback>(room_offset+room_scale))
+			c.feedback=(room_offset+room_scale);
+
+		float auxdmp=params.damp/2.0+0.5; //only half the range (0.5 .. 1.0  is enough)
+		auxdmp*=auxdmp;
+
+		c.damp=expf(-2.0*M_PI*auxdmp*10000/params.mix_rate); // 0 .. 10khz
+	}
+
+}
+
+void Reverb::clear_buffers() {
+
+	if (echo_buffer)
+		memdelete_arr(echo_buffer);
+
+	for (int i=0;i<MAX_COMBS;i++) {
+
+		if (comb[i].buffer)
+			memdelete_arr(comb[i].buffer);
+
+		comb[i].buffer=0;
+
+	}
+
+	for (int i=0;i<MAX_ALLPASS;i++) {
+
+		if (allpass[i].buffer)
+			memdelete_arr(allpass[i].buffer);
+
+		allpass[i].buffer=0;
+	}
+
+}
+
+Reverb::Reverb() {
+
+	params.room_size=0.8;
+	params.damp=0.5;
+	params.dry=1.0;
+	params.wet=0.0;
+	params.mix_rate=44100;
+	params.extra_spread_base=0;
+	params.extra_spread=1.0;
+	params.predelay=150;
+	params.predelay_fb=0.4;
+	params.hpf=0;
+	hpf_h1=0;
+	hpf_h2=0;
+
+
+	input_buffer=memnew_arr(float,INPUT_BUFFER_MAX_SIZE);
+	echo_buffer=0;
+
+	configure_buffers();
+	update_parameters();
+
+
+}
+
+
+Reverb::~Reverb() {
+
+	memdelete_arr(input_buffer);
+	clear_buffers();
+}
+
+
diff --git a/servers/audio/effects/reverb.h b/servers/audio/effects/reverb.h
new file mode 100644
index 0000000000..2c82be9156
--- /dev/null
+++ b/servers/audio/effects/reverb.h
@@ -0,0 +1,111 @@
+//
+// C++ Interface: reverb
+//
+// Description:
+//
+//
+// Author: Juan Linietsky <reduzio@gmail.com>, (C) 2006
+//
+// Copyright: See COPYING file that comes with this distribution
+//
+//
+#ifndef REVERB_H
+#define REVERB_H
+
+#include "typedefs.h"
+#include "os/memory.h"
+#include "audio_frame.h"
+
+class Reverb {
+public:
+	enum {
+		INPUT_BUFFER_MAX_SIZE=1024,
+
+	};
+private:
+	enum {
+
+		MAX_COMBS=8,
+		MAX_ALLPASS=4,
+		MAX_ECHO_MS=500
+
+	};
+
+
+
+	static const float comb_tunings[MAX_COMBS];
+	static const float allpass_tunings[MAX_ALLPASS];
+
+	struct Comb {
+
+		int size;
+		float *buffer;
+		float feedback;
+		float damp; //lowpass
+		float damp_h; //history
+		int pos;
+		int extra_spread_frames;
+
+		Comb() { size=0; buffer=0; feedback=0; damp_h=0; pos=0; }
+	};
+
+	struct AllPass {
+
+		int size;
+		float *buffer;
+		int pos;
+		int extra_spread_frames;
+		AllPass() { size=0; buffer=0; pos=0; }
+	};
+
+	Comb comb[MAX_COMBS];
+	AllPass allpass[MAX_ALLPASS];
+	float *input_buffer;
+	float *echo_buffer;
+	int echo_buffer_size;
+	int echo_buffer_pos;
+
+	float hpf_h1,hpf_h2;
+
+
+	struct Parameters {
+
+		float room_size;
+		float damp;
+		float wet;
+		float dry;
+		float mix_rate;
+		float extra_spread_base;
+		float extra_spread;
+		float predelay;
+		float predelay_fb;
+		float hpf;
+	} params;
+
+	void configure_buffers();
+	void update_parameters();
+	void clear_buffers();
+public:
+
+	void set_room_size(float p_size);
+	void set_damp(float p_damp);
+	void set_wet(float p_wet);
+	void set_dry(float p_dry);
+	void set_predelay(float p_predelay); // in ms
+	void set_predelay_feedback(float p_predelay_fb); // in ms
+	void set_highpass(float p_frq);
+	void set_mix_rate(float p_mix_rate);
+	void set_extra_spread(float p_spread);
+	void set_extra_spread_base(float p_sec);
+
+	void process(float *p_src,float *p_dst,int p_frames);
+
+	Reverb();
+
+	~Reverb();
+
+};
+
+
+
+#endif
diff --git a/servers/audio_server.cpp b/servers/audio_server.cpp
index 9b938a7f86..90e35c7fce 100644
--- a/servers/audio_server.cpp
+++ b/servers/audio_server.cpp
@@ -42,12 +42,16 @@ void AudioDriver::set_singleton() {
 
 void AudioDriver::audio_server_process(int p_frames,int32_t *p_buffer,bool p_update_mix_time) {
 
-	AudioServer * audio_server = static_cast<AudioServer*>(AudioServer::get_singleton());
+
 	if (p_update_mix_time)
 		update_mix_time(p_frames);
-//	audio_server->driver_process(p_frames,p_buffer);
+
+	if (AudioServer::get_singleton())
+		AudioServer::get_singleton()->_driver_process(p_frames,p_buffer);
 }
 
+
+
 void AudioDriver::update_mix_time(int p_frames) {
 
 	_mix_amount+=p_frames;
@@ -74,7 +78,6 @@ AudioDriver *AudioDriverManager::drivers[MAX_DRIVERS];
 int AudioDriverManager::driver_count=0;
 
 
-
 void AudioDriverManager::add_driver(AudioDriver *p_driver) {
 
 	ERR_FAIL_COND(driver_count>=MAX_DRIVERS);
@@ -97,13 +100,286 @@ AudioDriver *AudioDriverManager::get_driver(int p_driver) {
 //////////////////////////////////////////////
 //////////////////////////////////////////////
 
+void AudioServer::_driver_process(int p_frames,int32_t* p_buffer) {
+
+	int todo=p_frames;
+
+	while(todo) {
+
+		if (to_mix==0) {
+			_mix_step();
+		}
+
+		int to_copy = MIN(to_mix,todo);
+
+		Bus *master = buses[0];
+
+		int from = buffer_size-to_mix;
+		int from_buf=p_frames-todo;
+
+		//master master, send to output
+		int cs = master->channels.size();
+		for(int k=0;k<cs;k++) {
+
+			if (master->channels[k].active) {
+
+				AudioFrame *buf = master->channels[k].buffer.ptr();
+
+				for(int j=0;j<to_copy;j++) {
+
+					float l = CLAMP(buf[from+j].l,-1.0,1.0);
+					int32_t vl = l*((1<<20)-1);
+					p_buffer[(from_buf+j)*(cs*2)+k*2+0]=vl<<11;
+
+					float r = CLAMP(buf[from+j].r,-1.0,1.0);
+					int32_t vr = r*((1<<20)-1);
+					p_buffer[(from_buf+j)*(cs*2)+k*2+1]=vr<<11;
+				}
+
+			} else {
+				for(int j=0;j<to_copy;j++) {
+
+					p_buffer[(from_buf+j)*(cs*2)+k*2+0]=0;
+					p_buffer[(from_buf+j)*(cs*2)+k*2+1]=0;
+				}
+			}
+
+		}
+
+		todo-=to_copy;
+		to_mix-=to_copy;
+
+	}
+
+}
+
+void AudioServer::_mix_step() {
+
+	for(int i=0;i<buses.size();i++) {
+		Bus *bus = buses[i];
+		bus->index_cache=i; //might be moved around by editor, so..
+		for(int k=0;k<bus->channels.size();k++) {
+
+			bus->channels[k].used=false;
+		}
+	}
+
+	//make callbacks for mixing the audio
+	for (Set<CallbackItem>::Element *E=callbacks.front();E;E=E->next()) {
+
+		E->get().callback(E->get().userdata);
+	}
+
+	for(int i=buses.size()-1;i>=0;i--) {
+		//go bus by bus
+		Bus *bus = buses[i];
+
+
+		for(int k=0;k<bus->channels.size();k++) {
+
+			if (bus->channels[k].active && !bus->channels[k].used) {
+				//buffer was not used, but it's still active, so it must be cleaned
+				AudioFrame *buf = bus->channels[k].buffer.ptr();
+
+				for(uint32_t j=0;j<buffer_size;j++) {
+
+					buf[j]=AudioFrame(0,0);
+				}
+			}
+
+		}
+
+
+		//process effects
+		for(int j=0;j<bus->effects.size();j++) {
+
+			if (!bus->effects[j].enabled)
+				continue;
+
+			for(int k=0;k<bus->channels.size();k++) {
+
+				if (!bus->channels[k].active)
+					continue;
+				bus->channels[k].effect_instances[j]->process(bus->channels[k].buffer.ptr(),temp_buffer[k].ptr(),buffer_size);
+			}
+
+			//swap buffers, so internal buffer always has the right data
+			for(int k=0;k<bus->channels.size();k++) {
+
+				if (!buses[i]->channels[k].active)
+					continue;
+				SWAP(bus->channels[k].buffer,temp_buffer[k]);
+			}
+		}
+
+		//process send
+
+		Bus *send=NULL;
+
+		if (i>0) {
+			//everything has a send save for master bus
+			if (!bus_map.has(bus->send)) {
+				send=buses[0];
+			} else {
+				send=bus_map[bus->send];
+				if (send->index_cache>=bus->index_cache) { //invalid, send to master
+					send=buses[0];
+				}
+			}
+		}
+
+
+		for(int k=0;k<bus->channels.size();k++) {
+
+			if (!bus->channels[k].active)
+				continue;
+
+			AudioFrame *buf = bus->channels[k].buffer.ptr();
+
+
+			AudioFrame peak = AudioFrame(0,0);
+			for(uint32_t j=0;j<buffer_size;j++) {
+				float l = ABS(buf[j].l);
+				if (l>peak.l) {
+					peak.l=l;
+				}
+				float r = ABS(buf[j].r);
+				if (r>peak.r) {
+					peak.r=r;
+				}
+			}
+
+			bus->channels[k].peak_volume=AudioFrame(Math::linear2db(peak.l+0.0000000001),Math::linear2db(peak.r+0.0000000001));
+
+
+			if (!bus->channels[k].used) {
+				//see if any audio is contained, because channel was not used
+
+
+				if (MAX(peak.r,peak.l) > Math::db2linear(channel_disable_treshold_db)) {
+					bus->channels[k].last_mix_with_audio=mix_frames;
+				} else if (mix_frames-bus->channels[k].last_mix_with_audio > channel_disable_frames ) {
+					bus->channels[k].active=false;
+					continue; //went inactive, dont mix.
+				}
+			}
+
+			if (send) {
+				//if not master bus, send
+				AudioFrame *target_buf = thread_get_channel_mix_buffer(send->index_cache,k);
+
+				for(uint32_t j=0;j<buffer_size;j++) {
+					target_buf[j]+=buf[j];
+				}
+			}
+
+		}
+
+	}
+
+
+	mix_frames+=buffer_size;
+	to_mix=buffer_size;
+
+}
+
+AudioFrame *AudioServer::thread_get_channel_mix_buffer(int p_bus,int p_buffer) {
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),NULL);
+	ERR_FAIL_INDEX_V(p_buffer,buses[p_bus]->channels.size(),NULL);
+
+	AudioFrame *data = buses[p_bus]->channels[p_buffer].buffer.ptr();
+
+
+	if (!buses[p_bus]->channels[p_buffer].used) {
+		buses[p_bus]->channels[p_buffer].used=true;
+		buses[p_bus]->channels[p_buffer].active=true;
+		buses[p_bus]->channels[p_buffer].last_mix_with_audio=mix_frames;
+		for(uint32_t i=0;i<buffer_size;i++) {
+			data[i]=AudioFrame(0,0);
+		}
+	}
+
+	return data;
+}
+
+int AudioServer::thread_get_mix_buffer_size() const {
+
+	return buffer_size;
+}
+
+int AudioServer::thread_find_bus_index(const StringName& p_name) {
+
+	if (bus_map.has(p_name)) {
+		return bus_map[p_name]->index_cache;
+	} else {
+		return 0;
+	}
+
+}
+
 void AudioServer::set_bus_count(int p_count) {
 
 	ERR_FAIL_COND(p_count<1);
 	ERR_FAIL_INDEX(p_count,256);
 	lock();
+	int cb = buses.size();
+
+	if (p_count<buses.size()) {
+		for(int i=p_count;i<buses.size();i++) {
+			bus_map.erase(buses[i]->name);
+			memdelete(buses[i]);
+		}
+	}
+
 	buses.resize(p_count);
+
+	for(int i=cb;i<buses.size();i++) {
+
+		String attempt="New Bus";
+		int attempts=1;
+		while(true) {
+
+			bool name_free=true;
+			for(int j=0;j<i;j++) {
+
+				if (buses[j]->name==attempt) {
+					name_free=false;
+					break;
+				}
+			}
+
+			if (!name_free) {
+				attempts++;
+				attempt="New Bus " +itos(attempts);
+			} else {
+				break;
+			}
+
+		}
+
+
+		buses[i]=memnew(Bus);
+		buses[i]->channels.resize(_get_channel_count());
+		for(int j=0;j<_get_channel_count();j++) {
+			buses[i]->channels[j].buffer.resize(buffer_size);
+		}
+		buses[i]->name=attempt;
+		buses[i]->solo=false;
+		buses[i]->mute=false;
+		buses[i]->bypass=false;
+		buses[i]->volume_db=0;
+		if (i>0) {
+			buses[i]->send="Master";
+		}
+
+		bus_map[attempt]=buses[i];
+
+	}
+
 	unlock();
+
+	emit_signal("bus_layout_changed");
 }
 
 int AudioServer::get_bus_count() const {
@@ -111,42 +387,138 @@ int AudioServer::get_bus_count() const {
 	return buses.size();
 }
 
-void AudioServer::set_bus_mode(int p_bus,BusMode p_mode) {
+
+void AudioServer::set_bus_name(int p_bus,const String& p_name) {
 
 	ERR_FAIL_INDEX(p_bus,buses.size());
+	if (p_bus==0 && p_name!="Master")
+		return; //bus 0 is always master
+	lock();
 
-}
-AudioServer::BusMode AudioServer::get_bus_mode(int p_bus) const {
+	if (buses[p_bus]->name==p_name) {
+		unlock();
+		return;
+	}
 
-	ERR_FAIL_INDEX_V(p_bus,buses.size(),BUS_MODE_STEREO);
+	String attempt=p_name;
+	int attempts=1;
 
-	return buses[p_bus].mode;
-}
+	while(true) {
 
-void AudioServer::set_bus_name(int p_bus,const String& p_name) {
+		bool name_free=true;
+		for(int i=0;i<buses.size();i++) {
 
-	ERR_FAIL_INDEX(p_bus,buses.size());
-	buses[p_bus].name=p_name;
+			if (buses[i]->name==attempt) {
+				name_free=false;
+				break;
+			}
+		}
+
+		if (name_free) {
+			break;
+		}
+
+		attempts++;
+		attempt=p_name+" "+itos(attempts);
+	}
+	bus_map.erase(buses[p_bus]->name);
+	buses[p_bus]->name=attempt;
+	bus_map[attempt]=buses[p_bus];
+	unlock();
+
+	emit_signal("bus_layout_changed");
 
 }
 String AudioServer::get_bus_name(int p_bus) const {
 
 	ERR_FAIL_INDEX_V(p_bus,buses.size(),String());
-	return buses[p_bus].name;
+	return buses[p_bus]->name;
 }
 
 void AudioServer::set_bus_volume_db(int p_bus,float p_volume_db) {
 
 	ERR_FAIL_INDEX(p_bus,buses.size());
-	buses[p_bus].volume_db=p_volume_db;
+	buses[p_bus]->volume_db=p_volume_db;
 
 }
 float AudioServer::get_bus_volume_db(int p_bus) const {
 
 	ERR_FAIL_INDEX_V(p_bus,buses.size(),0);
-	return buses[p_bus].volume_db;
+	return buses[p_bus]->volume_db;
+
+}
+
+void AudioServer::set_bus_send(int p_bus,const StringName& p_send) {
+
+	ERR_FAIL_INDEX(p_bus,buses.size());
+
+	buses[p_bus]->send=p_send;
+
+}
+
+StringName AudioServer::get_bus_send(int p_bus) const {
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),StringName());
+	return buses[p_bus]->send;
+
+}
+
+
+void AudioServer::set_bus_solo(int p_bus,bool p_enable) {
+
+	ERR_FAIL_INDEX(p_bus,buses.size());
+
+	buses[p_bus]->solo=p_enable;
+
+}
+
+bool AudioServer::is_bus_solo(int p_bus) const{
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),false);
+
+	return buses[p_bus]->solo;
+
+}
+
+void AudioServer::set_bus_mute(int p_bus,bool p_enable){
+
+	ERR_FAIL_INDEX(p_bus,buses.size());
+
+	buses[p_bus]->mute=p_enable;
+}
+bool AudioServer::is_bus_mute(int p_bus) const{
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),false);
+
+	return buses[p_bus]->mute;
+
+}
+
+void AudioServer::set_bus_bypass_effects(int p_bus,bool p_enable){
+
+	ERR_FAIL_INDEX(p_bus,buses.size());
 
+	buses[p_bus]->bypass=p_enable;
 }
+bool AudioServer::is_bus_bypassing_effects(int p_bus) const{
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),false);
+
+	return buses[p_bus]->bypass;
+
+}
+
+
+void AudioServer::_update_bus_effects(int p_bus) {
+
+	for(int i=0;i<buses[p_bus]->channels.size();i++) {
+		buses[p_bus]->channels[i].effect_instances.resize(buses[p_bus]->effects.size());
+		for(int j=0;j<buses[p_bus]->effects.size();j++) {
+			buses[p_bus]->channels[i].effect_instances[j]=buses[p_bus]->effects[j].effect->instance();
+		}
+	}
+}
+
 
 void AudioServer::add_bus_effect(int p_bus,const Ref<AudioEffect>& p_effect,int p_at_pos) {
 
@@ -160,12 +532,14 @@ void AudioServer::add_bus_effect(int p_bus,const Ref<AudioEffect>& p_effect,int
 	//fx.instance=p_effect->instance();
 	fx.enabled=true;
 
-	if (p_at_pos>=buses[p_bus].effects.size() || p_at_pos<0) {
-		buses[p_bus].effects.push_back(fx);
+	if (p_at_pos>=buses[p_bus]->effects.size() || p_at_pos<0) {
+		buses[p_bus]->effects.push_back(fx);
 	} else {
-		buses[p_bus].effects.insert(p_at_pos,fx);
+		buses[p_bus]->effects.insert(p_at_pos,fx);
 	}
 
+	_update_bus_effects(p_bus);
+
 	unlock();
 }
 
@@ -176,7 +550,8 @@ void AudioServer::remove_bus_effect(int p_bus,int p_effect) {
 
 	lock();
 
-	buses[p_bus].effects.remove(p_effect);
+	buses[p_bus]->effects.remove(p_effect);
+	_update_bus_effects(p_bus);
 
 	unlock();
 }
@@ -185,52 +560,117 @@ int AudioServer::get_bus_effect_count(int p_bus) {
 
 	ERR_FAIL_INDEX_V(p_bus,buses.size(),0);
 
-	return buses[p_bus].effects.size();
+	return buses[p_bus]->effects.size();
 
 }
 
 Ref<AudioEffect> AudioServer::get_bus_effect(int p_bus,int p_effect) {
 
 	ERR_FAIL_INDEX_V(p_bus,buses.size(),Ref<AudioEffect>());
-	ERR_FAIL_INDEX_V(p_effect,buses[p_bus].effects.size(),Ref<AudioEffect>());
+	ERR_FAIL_INDEX_V(p_effect,buses[p_bus]->effects.size(),Ref<AudioEffect>());
 
-	return buses[p_bus].effects[p_effect].effect;
+	return buses[p_bus]->effects[p_effect].effect;
 
 }
 
 void AudioServer::swap_bus_effects(int p_bus,int p_effect,int p_by_effect) {
 
 	ERR_FAIL_INDEX(p_bus,buses.size());
-	ERR_FAIL_INDEX(p_effect,buses[p_bus].effects.size());
-	ERR_FAIL_INDEX(p_by_effect,buses[p_bus].effects.size());
+	ERR_FAIL_INDEX(p_effect,buses[p_bus]->effects.size());
+	ERR_FAIL_INDEX(p_by_effect,buses[p_bus]->effects.size());
 
 	lock();
-	SWAP( buses[p_bus].effects[p_effect], buses[p_bus].effects[p_by_effect] );
+	SWAP( buses[p_bus]->effects[p_effect], buses[p_bus]->effects[p_by_effect] );
+	_update_bus_effects(p_bus);
 	unlock();
 }
 
 void AudioServer::set_bus_effect_enabled(int p_bus,int p_effect,bool p_enabled) {
 
 	ERR_FAIL_INDEX(p_bus,buses.size());
-	ERR_FAIL_INDEX(p_effect,buses[p_bus].effects.size());
-	buses[p_bus].effects[p_effect].enabled=p_enabled;
+	ERR_FAIL_INDEX(p_effect,buses[p_bus]->effects.size());
+	buses[p_bus]->effects[p_effect].enabled=p_enabled;
 
 }
 bool AudioServer::is_bus_effect_enabled(int p_bus,int p_effect) const {
 
 	ERR_FAIL_INDEX_V(p_bus,buses.size(),false);
-	ERR_FAIL_INDEX_V(p_effect,buses[p_bus].effects.size(),false);
-	return buses[p_bus].effects[p_effect].enabled;
+	ERR_FAIL_INDEX_V(p_effect,buses[p_bus]->effects.size(),false);
+	return buses[p_bus]->effects[p_effect].enabled;
+
+}
+
+void AudioServer::move_bus(int p_bus,int p_to_bus) {
+
+	ERR_FAIL_COND(p_bus<1 || p_bus>=buses.size());
+	ERR_FAIL_COND(p_bus<1 || p_to_bus>=buses.size());
+
+
+
+}
+
+float AudioServer::get_bus_peak_volume_left_db(int p_bus,int p_channel) const {
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),0);
+	ERR_FAIL_INDEX_V(p_channel,buses[p_bus]->channels.size(),0);
+
+	return buses[p_bus]->channels[p_channel].peak_volume.l;
+
+}
+float AudioServer::get_bus_peak_volume_right_db(int p_bus,int p_channel) const {
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),0);
+	ERR_FAIL_INDEX_V(p_channel,buses[p_bus]->channels.size(),0);
+
+	return buses[p_bus]->channels[p_channel].peak_volume.r;
+
+}
+
+bool  AudioServer::is_bus_channel_active(int p_bus,int p_channel) const {
+
+	ERR_FAIL_INDEX_V(p_bus,buses.size(),false);
+	ERR_FAIL_INDEX_V(p_channel,buses[p_bus]->channels.size(),false);
+
+	return buses[p_bus]->channels[p_channel].active;
 
 }
 
 void AudioServer::init() {
 
+	channel_disable_treshold_db=GLOBAL_DEF("audio/channel_disable_treshold_db",-60.0);
+	channel_disable_frames=float(GLOBAL_DEF("audio/channel_disable_time",2.0))*get_mix_rate();
+	buffer_size=1024; //harcoded for now
+	switch( get_speaker_mode() ) {
+		case SPEAKER_MODE_STEREO: {
+			temp_buffer.resize(1);
+		} break;
+		case SPEAKER_SURROUND_51: {
+			temp_buffer.resize(3);
+		} break;
+		case SPEAKER_SURROUND_71: {
+			temp_buffer.resize(4);
+		} break;
+	}
+
+	for(int i=0;i<temp_buffer.size();i++) {
+		temp_buffer[i].resize(buffer_size);
+	}
+
+	mix_count=0;
 	set_bus_count(1);;
 	set_bus_name(0,"Master");
+
+
+	if (AudioDriver::get_singleton())
+		AudioDriver::get_singleton()->start();
+
 }
 void AudioServer::finish() {
 
+	for(int i=0;i<buses.size();i++) {
+		memdelete(buses[i]);
+	}
+
 	buses.clear();
 }
 void AudioServer::update() {
@@ -282,18 +722,130 @@ double AudioServer::get_output_delay() const {
 AudioServer* AudioServer::singleton=NULL;
 
 
-void AudioServer::_bind_methods() {
 
+
+void* AudioServer::audio_data_alloc(uint32_t p_data_len,const uint8_t *p_from_data) {
+
+	void * ad = memalloc( p_data_len );
+	ERR_FAIL_COND_V(!ad,NULL);
+	if (p_from_data) {
+		copymem(ad,p_from_data,p_data_len);
+	}
+
+	audio_data_lock->lock();
+	audio_data[ad]=p_data_len;
+	audio_data_total_mem+=p_data_len;
+	audio_data_max_mem=MAX(audio_data_total_mem,audio_data_max_mem);
+	audio_data_lock->unlock();
+
+	return ad;
 }
 
+void AudioServer::audio_data_free(void* p_data) {
+
+	audio_data_lock->lock();
+	if (!audio_data.has(p_data)) {
+		audio_data_lock->unlock();
+		ERR_FAIL();
+	}
+
+	audio_data_total_mem-=audio_data[p_data];
+	audio_data.erase(p_data);
+	memfree(p_data);
+	audio_data_lock->unlock();
+
+
+}
+
+size_t AudioServer::audio_data_get_total_memory_usage() const{
+
+	return audio_data_total_mem;
+}
+size_t AudioServer::audio_data_get_max_memory_usage() const{
+
+	return audio_data_max_mem;
+
+}
+
+void AudioServer::add_callback(AudioCallback p_callback,void *p_userdata) {
+	lock();
+	CallbackItem ci;
+	ci.callback=p_callback;
+	ci.userdata=p_userdata;
+	callbacks.insert(ci);
+	unlock();
+}
+
+void AudioServer::remove_callback(AudioCallback p_callback,void *p_userdata) {
+
+	lock();
+	CallbackItem ci;
+	ci.callback=p_callback;
+	ci.userdata=p_userdata;
+	callbacks.erase(ci);
+	unlock();
+
+}
+
+void AudioServer::_bind_methods() {
+
+
+	ClassDB::bind_method(_MD("set_bus_count","amount"),&AudioServer::set_bus_count);
+	ClassDB::bind_method(_MD("get_bus_count"),&AudioServer::get_bus_count);
+
+	ClassDB::bind_method(_MD("set_bus_name","bus_idx","name"),&AudioServer::set_bus_name);
+	ClassDB::bind_method(_MD("get_bus_name","bus_idx"),&AudioServer::get_bus_name);
+
+	ClassDB::bind_method(_MD("set_bus_volume_db","bus_idx","volume_db"),&AudioServer::set_bus_volume_db);
+	ClassDB::bind_method(_MD("get_bus_volume_db","bus_idx"),&AudioServer::get_bus_volume_db);
+
+	ClassDB::bind_method(_MD("set_bus_send","bus_idx","send"),&AudioServer::set_bus_send);
+	ClassDB::bind_method(_MD("get_bus_send","bus_idx"),&AudioServer::get_bus_send);
+
+	ClassDB::bind_method(_MD("set_bus_solo","bus_idx","enable"),&AudioServer::set_bus_solo);
+	ClassDB::bind_method(_MD("is_bus_solo","bus_idx"),&AudioServer::is_bus_solo);
+
+	ClassDB::bind_method(_MD("set_bus_mute","bus_idx","enable"),&AudioServer::set_bus_mute);
+	ClassDB::bind_method(_MD("is_bus_mute","bus_idx"),&AudioServer::is_bus_mute);
+
+	ClassDB::bind_method(_MD("set_bus_bypass_effects","bus_idx","enable"),&AudioServer::set_bus_bypass_effects);
+	ClassDB::bind_method(_MD("is_bus_bypassing_effects","bus_idx"),&AudioServer::is_bus_bypassing_effects);
+
+	ClassDB::bind_method(_MD("add_bus_effect","bus_idx","effect:AudioEffect"),&AudioServer::add_bus_effect);
+	ClassDB::bind_method(_MD("remove_bus_effect","bus_idx","effect_idx"),&AudioServer::remove_bus_effect);
+
+	ClassDB::bind_method(_MD("get_bus_effect_count","bus_idx"),&AudioServer::add_bus_effect);
+	ClassDB::bind_method(_MD("get_bus_effect:AudioEffect","bus_idx","effect_idx"),&AudioServer::get_bus_effect);
+	ClassDB::bind_method(_MD("swap_bus_effects","bus_idx","effect_idx","by_effect_idx"),&AudioServer::swap_bus_effects);
+
+	ClassDB::bind_method(_MD("set_bus_effect_enabled","bus_idx","effect_idx","enabled"),&AudioServer::set_bus_effect_enabled);
+	ClassDB::bind_method(_MD("is_bus_effect_enabled","bus_idx","effect_idx"),&AudioServer::is_bus_effect_enabled);
+
+	ClassDB::bind_method(_MD("get_bus_peak_volume_left_db","bus_idx","channel"),&AudioServer::get_bus_peak_volume_left_db);
+	ClassDB::bind_method(_MD("get_bus_peak_volume_right_db","bus_idx","channel"),&AudioServer::get_bus_peak_volume_right_db);
+
+	ClassDB::bind_method(_MD("lock"),&AudioServer::lock);
+	ClassDB::bind_method(_MD("unlock"),&AudioServer::unlock);
+
+	ClassDB::bind_method(_MD("get_speaker_mode"),&AudioServer::get_speaker_mode);
+	ClassDB::bind_method(_MD("get_mix_rate"),&AudioServer::get_mix_rate);
+
+	ADD_SIGNAL(MethodInfo("bus_layout_changed") );
+}
 
 AudioServer::AudioServer() {
 
 	singleton=this;
+	audio_data_total_mem=0;
+	audio_data_max_mem=0;
+	audio_data_lock=Mutex::create();
+	mix_frames=0;
+	to_mix=0;
+
 }
 
 AudioServer::~AudioServer() {
 
-
+	memdelete(audio_data_lock);
 }
 
diff --git a/servers/audio_server.h b/servers/audio_server.h
index 77aca39760..8cc31752ba 100644
--- a/servers/audio_server.h
+++ b/servers/audio_server.h
@@ -100,63 +100,148 @@ public:
 };
 
 
+
+
 class AudioServer : public Object {
 
 	GDCLASS( AudioServer, Object )
 public:
-	enum BusMode {
-		BUS_MODE_STEREO,
-		BUS_MODE_SURROUND
-	};
-
 	//re-expose this her, as AudioDriver is not exposed to script
 	enum SpeakerMode {
 		SPEAKER_MODE_STEREO,
 		SPEAKER_SURROUND_51,
 		SPEAKER_SURROUND_71,
 	};
+
+	enum {
+		AUDIO_DATA_INVALID_ID=-1
+	};
+
+	typedef void (*AudioCallback)(void* p_userdata);
+
 private:
 	uint32_t buffer_size;
+	uint64_t mix_count;
+	uint64_t mix_frames;
+
+	float channel_disable_treshold_db;
+	uint32_t channel_disable_frames;
+
+	int to_mix;
 
 	struct Bus {
 
-		String name;
-		BusMode mode;
-		Vector<AudioFrame> buffer;
+		StringName name;
+		bool solo;
+		bool mute;
+		bool bypass;
+
+		//Each channel is a stereo pair.
+		struct Channel {
+			bool used;
+			bool active;
+			AudioFrame peak_volume;
+			Vector<AudioFrame> buffer;
+			Vector<Ref<AudioEffectInstance> > effect_instances;
+			uint64_t last_mix_with_audio;
+			Channel() { last_mix_with_audio=0; used=false; active=false; peak_volume=AudioFrame(0,0); }
+		};
+
+		Vector<Channel> channels;
+
 
 		struct Effect {
 			Ref<AudioEffect> effect;
-			Ref<AudioEffectInstance> instance;
 			bool enabled;
 		};
 
 		Vector<Effect> effects;
-
 		float volume_db;
+		StringName send;
+		int index_cache;
 	};
 
 
-	Vector<Bus> buses;
+	Vector< Vector<AudioFrame> >temp_buffer; //temp_buffer for each level
+	Vector<Bus*> buses;
+	Map<StringName,Bus*> bus_map;
 
+	_FORCE_INLINE_ int _get_channel_count() const {
+		switch (AudioDriver::get_singleton()->get_speaker_mode()) {
+			case AudioDriver::SPEAKER_MODE_STEREO: return 1;
+			case AudioDriver::SPEAKER_SURROUND_51: return 3;
+			case AudioDriver::SPEAKER_SURROUND_71: return 4;
+
+		}
+		ERR_FAIL_V(1);
+	}
+
+
+	void _update_bus_effects(int p_bus);
 
-	static void _bind_methods();
 
 	static AudioServer* singleton;
+
+	// TODO create an audiodata pool to optimize memory
+
+
+	Map<void*,uint32_t> audio_data;
+	size_t audio_data_total_mem;
+	size_t audio_data_max_mem;
+
+	Mutex *audio_data_lock;
+
+	void _mix_step();
+
+	struct CallbackItem {
+
+		AudioCallback callback;
+		void *userdata;
+
+		bool operator<(const CallbackItem& p_item) const {
+			return (callback==p_item.callback ? userdata < p_item.userdata : callback < p_item.callback);
+		}
+	};
+
+	Set<CallbackItem> callbacks;
+
+
+
+friend class AudioDriver;
+	void _driver_process(int p_frames, int32_t *p_buffer);
+protected:
+
+	static void _bind_methods();
 public:
 
+	//do not use from outside audio thread
+	AudioFrame *thread_get_channel_mix_buffer(int p_bus,int p_buffer);
+	int thread_get_mix_buffer_size() const;
+	int thread_find_bus_index(const StringName& p_name);
+
 
 	void set_bus_count(int p_count);
 	int get_bus_count() const;
 
-	void set_bus_mode(int p_bus,BusMode p_mode);
-	BusMode get_bus_mode(int p_bus) const;
-
 	void set_bus_name(int p_bus,const String& p_name);
 	String get_bus_name(int p_bus) const;
 
 	void set_bus_volume_db(int p_bus,float p_volume_db);
 	float get_bus_volume_db(int p_bus) const;
 
+
+	void set_bus_send(int p_bus,const StringName& p_send);
+	StringName get_bus_send(int p_bus) const;
+
+	void set_bus_solo(int p_bus,bool p_enable);
+	bool is_bus_solo(int p_bus) const;
+
+	void set_bus_mute(int p_bus,bool p_enable);
+	bool is_bus_mute(int p_bus) const;
+
+	void set_bus_bypass_effects(int p_bus,bool p_enable);
+	bool is_bus_bypassing_effects(int p_bus) const;
+
 	void add_bus_effect(int p_bus,const Ref<AudioEffect>& p_effect,int p_at_pos=-1);
 	void remove_bus_effect(int p_bus,int p_effect);
 
@@ -168,6 +253,13 @@ public:
 	void set_bus_effect_enabled(int p_bus,int p_effect,bool p_enabled);
 	bool is_bus_effect_enabled(int p_bus,int p_effect) const;
 
+	void move_bus(int p_bus,int p_to_bus);
+
+	float get_bus_peak_volume_left_db(int p_bus,int p_channel) const;
+	float get_bus_peak_volume_right_db(int p_bus,int p_channel) const;
+
+	bool is_bus_channel_active(int p_bus,int p_channel) const;
+
 	virtual void init();
 	virtual void finish();
 	virtual void update();
@@ -188,11 +280,21 @@ public:
 	virtual double get_mix_time() const; //useful for video -> audio sync
 	virtual double get_output_delay() const;
 
+	void* audio_data_alloc(uint32_t p_data_len, const uint8_t *p_from_data=NULL);
+	void audio_data_free(void* p_data);
+
+	size_t audio_data_get_total_memory_usage() const;
+	size_t audio_data_get_max_memory_usage() const;
+
+
+	void add_callback(AudioCallback p_callback,void *p_userdata);
+	void remove_callback(AudioCallback p_callback,void *p_userdata);
+
 	AudioServer();
 	virtual ~AudioServer();
 };
 
-VARIANT_ENUM_CAST( AudioServer::BusMode )
+
 VARIANT_ENUM_CAST( AudioServer::SpeakerMode )
 
 typedef AudioServer AS;
diff --git a/servers/register_server_types.cpp b/servers/register_server_types.cpp
index 8b831f4ff6..66f3bfd275 100644
--- a/servers/register_server_types.cpp
+++ b/servers/register_server_types.cpp
@@ -35,6 +35,12 @@
 #include "physics_2d_server.h"
 #include "script_debugger_remote.h"
 #include "visual/shader_types.h"
+#include "audio/audio_stream.h"
+#include "audio/audio_effect.h"
+#include "audio/effects/audio_effect_amplify.h"
+#include "audio/effects/audio_effect_reverb.h"
+#include "audio/effects/audio_effect_filter.h"
+#include "audio/effects/audio_effect_eq.h"
 
 static void _debugger_get_resource_usage(List<ScriptDebuggerRemote::ResourceUsage>* r_usage) {
 
@@ -67,6 +73,26 @@ void register_server_types() {
 
 	shader_types = memnew( ShaderTypes );
 
+	ClassDB::register_virtual_class<AudioStream>();
+	ClassDB::register_virtual_class<AudioStreamPlayback>();
+	ClassDB::register_virtual_class<AudioEffect>();
+
+	ClassDB::register_class<AudioEffectAmplify>();
+
+	ClassDB::register_class<AudioEffectReverb>();
+
+	ClassDB::register_class<AudioEffectLowPass>();
+	ClassDB::register_class<AudioEffectHighPass>();
+	ClassDB::register_class<AudioEffectBandPass>();
+	ClassDB::register_class<AudioEffectNotchPass>();
+	ClassDB::register_class<AudioEffectBandLimit>();
+	ClassDB::register_class<AudioEffectLowShelf>();
+	ClassDB::register_class<AudioEffectHighShelf>();
+
+	ClassDB::register_class<AudioEffectEQ6>();
+	ClassDB::register_class<AudioEffectEQ10>();
+	ClassDB::register_class<AudioEffectEQ21>();
+
 
 	ClassDB::register_virtual_class<Physics2DDirectBodyState>();
 	ClassDB::register_virtual_class<Physics2DDirectSpaceState>();
diff --git a/thirdparty/stb_vorbis/stb_vorbis.c b/thirdparty/stb_vorbis/stb_vorbis.c
new file mode 100644
index 0000000000..c4f24d5898
--- /dev/null
+++ b/thirdparty/stb_vorbis/stb_vorbis.c
@@ -0,0 +1,5399 @@
+// Ogg Vorbis audio decoder - v1.09 - public domain
+// http://nothings.org/stb_vorbis/
+//
+// Original version written by Sean Barrett in 2007.
+//
+// Originally sponsored by RAD Game Tools. Seeking sponsored
+// by Phillip Bennefall, Marc Andersen, Aaron Baker, Elias Software,
+// Aras Pranckevicius, and Sean Barrett.
+//
+// LICENSE
+//
+//   This software is dual-licensed to the public domain and under the following
+//   license: you are granted a perpetual, irrevocable license to copy, modify,
+//   publish, and distribute this file as you see fit.
+//
+// No warranty for any purpose is expressed or implied by the author (nor
+// by RAD Game Tools). Report bugs and send enhancements to the author.
+//
+// Limitations:
+//
+//   - floor 0 not supported (used in old ogg vorbis files pre-2004)
+//   - lossless sample-truncation at beginning ignored
+//   - cannot concatenate multiple vorbis streams
+//   - sample positions are 32-bit, limiting seekable 192Khz
+//       files to around 6 hours (Ogg supports 64-bit)
+//
+// Feature contributors:
+//    Dougall Johnson (sample-exact seeking)
+//
+// Bugfix/warning contributors:
+//    Terje Mathisen     Niklas Frykholm     Andy Hill
+//    Casey Muratori     John Bolton         Gargaj
+//    Laurent Gomila     Marc LeBlanc        Ronny Chevalier
+//    Bernhard Wodo      Evan Balster        alxprd@github
+//    Tom Beaumont       Ingo Leitgeb        Nicolas Guillemot
+//    Phillip Bennefall  Rohit               Thiago Goulart
+//    manxorist@github   saga musix
+//
+// Partial history:
+//    1.09    - 2016/04/04 - back out 'truncation of last frame' fix from previous version
+//    1.08    - 2016/04/02 - warnings; setup memory leaks; truncation of last frame
+//    1.07    - 2015/01/16 - fixes for crashes on invalid files; warning fixes; const
+//    1.06    - 2015/08/31 - full, correct support for seeking API (Dougall Johnson)
+//                           some crash fixes when out of memory or with corrupt files
+//                           fix some inappropriately signed shifts
+//    1.05    - 2015/04/19 - don't define __forceinline if it's redundant
+//    1.04    - 2014/08/27 - fix missing const-correct case in API
+//    1.03    - 2014/08/07 - warning fixes
+//    1.02    - 2014/07/09 - declare qsort comparison as explicitly _cdecl in Windows
+//    1.01    - 2014/06/18 - fix stb_vorbis_get_samples_float (interleaved was correct)
+//    1.0     - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in >2-channel;
+//                           (API change) report sample rate for decode-full-file funcs
+//
+// See end of file for full version history.
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+//  HEADER BEGINS HERE
+//
+
+#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define STB_VORBIS_INCLUDE_STB_VORBIS_H
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+#define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+///////////   THREAD SAFETY
+
+// Individual stb_vorbis* handles are not thread-safe; you cannot decode from
+// them from multiple threads at the same time. However, you can have multiple
+// stb_vorbis* handles and decode from them independently in multiple thrads.
+
+
+///////////   MEMORY ALLOCATION
+
+// normally stb_vorbis uses malloc() to allocate memory at startup,
+// and alloca() to allocate temporary memory during a frame on the
+// stack. (Memory consumption will depend on the amount of setup
+// data in the file and how you set the compile flags for speed
+// vs. size. In my test files the maximal-size usage is ~150KB.)
+//
+// You can modify the wrapper functions in the source (setup_malloc,
+// setup_temp_malloc, temp_malloc) to change this behavior, or you
+// can use a simpler allocation model: you pass in a buffer from
+// which stb_vorbis will allocate _all_ its memory (including the
+// temp memory). "open" may fail with a VORBIS_outofmem if you
+// do not pass in enough data; there is no way to determine how
+// much you do need except to succeed (at which point you can
+// query get_info to find the exact amount required. yes I know
+// this is lame).
+//
+// If you pass in a non-NULL buffer of the type below, allocation
+// will occur from it as described above. Otherwise just pass NULL
+// to use malloc()/alloca()
+
+typedef struct
+{
+   char *alloc_buffer;
+   int   alloc_buffer_length_in_bytes;
+} stb_vorbis_alloc;
+
+
+///////////   FUNCTIONS USEABLE WITH ALL INPUT MODES
+
+typedef struct stb_vorbis stb_vorbis;
+
+typedef struct
+{
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int setup_temp_memory_required;
+   unsigned int temp_memory_required;
+
+   int max_frame_size;
+} stb_vorbis_info;
+
+// get general information about the file
+extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f);
+
+// get the last error detected (clears it, too)
+extern int stb_vorbis_get_error(stb_vorbis *f);
+
+// close an ogg vorbis file and free all memory in use
+extern void stb_vorbis_close(stb_vorbis *f);
+
+// this function returns the offset (in samples) from the beginning of the
+// file that will be returned by the next decode, if it is known, or -1
+// otherwise. after a flush_pushdata() call, this may take a while before
+// it becomes valid again.
+// NOT WORKING YET after a seek with PULLDATA API
+extern int stb_vorbis_get_sample_offset(stb_vorbis *f);
+
+// returns the current seek point within the file, or offset from the beginning
+// of the memory buffer. In pushdata mode it returns 0.
+extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f);
+
+///////////   PUSHDATA API
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+// this API allows you to get blocks of data from any source and hand
+// them to stb_vorbis. you have to buffer them; stb_vorbis will tell
+// you how much it used, and you have to give it the rest next time;
+// and stb_vorbis may not have enough data to work with and you will
+// need to give it the same data again PLUS more. Note that the Vorbis
+// specification does not bound the size of an individual frame.
+
+extern stb_vorbis *stb_vorbis_open_pushdata(
+         const unsigned char * datablock, int datablock_length_in_bytes,
+         int *datablock_memory_consumed_in_bytes,
+         int *error,
+         const stb_vorbis_alloc *alloc_buffer);
+// create a vorbis decoder by passing in the initial data block containing
+//    the ogg&vorbis headers (you don't need to do parse them, just provide
+//    the first N bytes of the file--you're told if it's not enough, see below)
+// on success, returns an stb_vorbis *, does not set error, returns the amount of
+//    data parsed/consumed on this call in *datablock_memory_consumed_in_bytes;
+// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed
+// if returns NULL and *error is VORBIS_need_more_data, then the input block was
+//       incomplete and you need to pass in a larger block from the start of the file
+
+extern int stb_vorbis_decode_frame_pushdata(
+         stb_vorbis *f,
+         const unsigned char *datablock, int datablock_length_in_bytes,
+         int *channels,             // place to write number of float * buffers
+         float ***output,           // place to write float ** array of float * buffers
+         int *samples               // place to write number of output samples
+     );
+// decode a frame of audio sample data if possible from the passed-in data block
+//
+// return value: number of bytes we used from datablock
+//
+// possible cases:
+//     0 bytes used, 0 samples output (need more data)
+//     N bytes used, 0 samples output (resynching the stream, keep going)
+//     N bytes used, M samples output (one frame of data)
+// note that after opening a file, you will ALWAYS get one N-bytes,0-sample
+// frame, because Vorbis always "discards" the first frame.
+//
+// Note that on resynch, stb_vorbis will rarely consume all of the buffer,
+// instead only datablock_length_in_bytes-3 or less. This is because it wants
+// to avoid missing parts of a page header if they cross a datablock boundary,
+// without writing state-machiney code to record a partial detection.
+//
+// The number of channels returned are stored in *channels (which can be
+// NULL--it is always the same as the number of channels reported by
+// get_info). *output will contain an array of float* buffers, one per
+// channel. In other words, (*output)[0][0] contains the first sample from
+// the first channel, and (*output)[1][0] contains the first sample from
+// the second channel.
+
+extern void stb_vorbis_flush_pushdata(stb_vorbis *f);
+// inform stb_vorbis that your next datablock will not be contiguous with
+// previous ones (e.g. you've seeked in the data); future attempts to decode
+// frames will cause stb_vorbis to resynchronize (as noted above), and
+// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it
+// will begin decoding the _next_ frame.
+//
+// if you want to seek using pushdata, you need to seek in your file, then
+// call stb_vorbis_flush_pushdata(), then start calling decoding, then once
+// decoding is returning you data, call stb_vorbis_get_sample_offset, and
+// if you don't like the result, seek your file again and repeat.
+#endif
+
+
+//////////   PULLING INPUT API
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+// This API assumes stb_vorbis is allowed to pull data from a source--
+// either a block of memory containing the _entire_ vorbis stream, or a
+// FILE * that you or it create, or possibly some other reading mechanism
+// if you go modify the source to replace the FILE * case with some kind
+// of callback to your code. (But if you don't support seeking, you may
+// just want to go ahead and use pushdata.)
+
+#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output);
+#endif
+#if !defined(STB_VORBIS_NO_INTEGER_CONVERSION)
+extern int stb_vorbis_decode_memory(const unsigned char *mem, int len, int *channels, int *sample_rate, short **output);
+#endif
+// decode an entire file and output the data interleaved into a malloc()ed
+// buffer stored in *output. The return value is the number of samples
+// decoded, or -1 if the file could not be opened or was not an ogg vorbis file.
+// When you're done with it, just free() the pointer returned in *output.
+
+extern stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an ogg vorbis stream in memory (note
+// this must be the entire stream!). on failure, returns NULL and sets *error
+
+#ifndef STB_VORBIS_NO_STDIO
+extern stb_vorbis * stb_vorbis_open_filename(const char *filename,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from a filename via fopen(). on failure,
+// returns NULL and sets *error (possibly to VORBIS_file_open_failure).
+
+extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close,
+                                  int *error, const stb_vorbis_alloc *alloc_buffer);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell). on failure, returns NULL and sets *error.
+// note that stb_vorbis must "own" this stream; if you seek it in between
+// calls to stb_vorbis, it will become confused. Morever, if you attempt to
+// perform stb_vorbis_seek_*() operations on this file, it will assume it
+// owns the _entire_ rest of the file after the start point. Use the next
+// function, stb_vorbis_open_file_section(), to limit it.
+
+extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close,
+                int *error, const stb_vorbis_alloc *alloc_buffer, unsigned int len);
+// create an ogg vorbis decoder from an open FILE *, looking for a stream at
+// the _current_ seek point (ftell); the stream will be of length 'len' bytes.
+// on failure, returns NULL and sets *error. note that stb_vorbis must "own"
+// this stream; if you seek it in between calls to stb_vorbis, it will become
+// confused.
+#endif
+
+extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number);
+extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number);
+// these functions seek in the Vorbis file to (approximately) 'sample_number'.
+// after calling seek_frame(), the next call to get_frame_*() will include
+// the specified sample. after calling stb_vorbis_seek(), the next call to
+// stb_vorbis_get_samples_* will start with the specified sample. If you
+// do not need to seek to EXACTLY the target sample when using get_samples_*,
+// you can also use seek_frame().
+
+extern void stb_vorbis_seek_start(stb_vorbis *f);
+// this function is equivalent to stb_vorbis_seek(f,0)
+
+extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f);
+extern float        stb_vorbis_stream_length_in_seconds(stb_vorbis *f);
+// these functions return the total length of the vorbis stream
+
+extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output);
+// decode the next frame and return the number of samples. the number of
+// channels returned are stored in *channels (which can be NULL--it is always
+// the same as the number of channels reported by get_info). *output will
+// contain an array of float* buffers, one per channel. These outputs will
+// be overwritten on the next call to stb_vorbis_get_frame_*.
+//
+// You generally should not intermix calls to stb_vorbis_get_frame_*()
+// and stb_vorbis_get_samples_*(), since the latter calls the former.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts);
+extern int stb_vorbis_get_frame_short            (stb_vorbis *f, int num_c, short **buffer, int num_samples);
+#endif
+// decode the next frame and return the number of *samples* per channel.
+// Note that for interleaved data, you pass in the number of shorts (the
+// size of your array), but the return value is the number of samples per
+// channel, not the total number of samples.
+//
+// The data is coerced to the number of channels you request according to the
+// channel coercion rules (see below). You must pass in the size of your
+// buffer(s) so that stb_vorbis will not overwrite the end of the buffer.
+// The maximum buffer size needed can be gotten from get_info(); however,
+// the Vorbis I specification implies an absolute maximum of 4096 samples
+// per channel.
+
+// Channel coercion rules:
+//    Let M be the number of channels requested, and N the number of channels present,
+//    and Cn be the nth channel; let stereo L be the sum of all L and center channels,
+//    and stereo R be the sum of all R and center channels (channel assignment from the
+//    vorbis spec).
+//        M    N       output
+//        1    k      sum(Ck) for all k
+//        2    *      stereo L, stereo R
+//        k    l      k > l, the first l channels, then 0s
+//        k    l      k <= l, the first k channels
+//    Note that this is not _good_ surround etc. mixing at all! It's just so
+//    you get something useful.
+
+extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats);
+extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples);
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES.
+// Returns the number of samples stored per channel; it may be less than requested
+// at the end of the file. If there are no more samples in the file, returns 0.
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts);
+extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples);
+#endif
+// gets num_samples samples, not necessarily on a frame boundary--this requires
+// buffering so you have to supply the buffers. Applies the coercion rules above
+// to produce 'channels' channels. Returns the number of samples stored per channel;
+// it may be less than requested at the end of the file. If there are no more
+// samples in the file, returns 0.
+
+#endif
+
+////////   ERROR CODES
+
+enum STBVorbisError
+{
+   VORBIS__no_error,
+
+   VORBIS_need_more_data=1,             // not a real error
+
+   VORBIS_invalid_api_mixing,           // can't mix API modes
+   VORBIS_outofmem,                     // not enough memory
+   VORBIS_feature_not_supported,        // uses floor 0
+   VORBIS_too_many_channels,            // STB_VORBIS_MAX_CHANNELS is too small
+   VORBIS_file_open_failure,            // fopen() failed
+   VORBIS_seek_without_length,          // can't seek in unknown-length file
+
+   VORBIS_unexpected_eof=10,            // file is truncated?
+   VORBIS_seek_invalid,                 // seek past EOF
+
+   // decoding errors (corrupt/invalid stream) -- you probably
+   // don't care about the exact details of these
+
+   // vorbis errors:
+   VORBIS_invalid_setup=20,
+   VORBIS_invalid_stream,
+
+   // ogg errors:
+   VORBIS_missing_capture_pattern=30,
+   VORBIS_invalid_stream_structure_version,
+   VORBIS_continued_packet_flag_invalid,
+   VORBIS_incorrect_stream_serial_number,
+   VORBIS_invalid_first_page,
+   VORBIS_bad_packet_type,
+   VORBIS_cant_find_last_page,
+   VORBIS_seek_failed
+};
+
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H
+//
+//  HEADER ENDS HERE
+//
+//////////////////////////////////////////////////////////////////////////////
+
+#ifndef STB_VORBIS_HEADER_ONLY
+
+// global configuration settings (e.g. set these in the project/makefile),
+// or just set them in this file at the top (although ideally the first few
+// should be visible when the header file is compiled too, although it's not
+// crucial)
+
+// STB_VORBIS_NO_PUSHDATA_API
+//     does not compile the code for the various stb_vorbis_*_pushdata()
+//     functions
+// #define STB_VORBIS_NO_PUSHDATA_API
+
+// STB_VORBIS_NO_PULLDATA_API
+//     does not compile the code for the non-pushdata APIs
+// #define STB_VORBIS_NO_PULLDATA_API
+
+// STB_VORBIS_NO_STDIO
+//     does not compile the code for the APIs that use FILE *s internally
+//     or externally (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_STDIO
+
+// STB_VORBIS_NO_INTEGER_CONVERSION
+//     does not compile the code for converting audio sample data from
+//     float to integer (implied by STB_VORBIS_NO_PULLDATA_API)
+// #define STB_VORBIS_NO_INTEGER_CONVERSION
+
+// STB_VORBIS_NO_FAST_SCALED_FLOAT
+//      does not use a fast float-to-int trick to accelerate float-to-int on
+//      most platforms which requires endianness be defined correctly.
+//#define STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+
+// STB_VORBIS_MAX_CHANNELS [number]
+//     globally define this to the maximum number of channels you need.
+//     The spec does not put a restriction on channels except that
+//     the count is stored in a byte, so 255 is the hard limit.
+//     Reducing this saves about 16 bytes per value, so using 16 saves
+//     (255-16)*16 or around 4KB. Plus anything other memory usage
+//     I forgot to account for. Can probably go as low as 8 (7.1 audio),
+//     6 (5.1 audio), or 2 (stereo only).
+#ifndef STB_VORBIS_MAX_CHANNELS
+#define STB_VORBIS_MAX_CHANNELS    16  // enough for anyone?
+#endif
+
+// STB_VORBIS_PUSHDATA_CRC_COUNT [number]
+//     after a flush_pushdata(), stb_vorbis begins scanning for the
+//     next valid page, without backtracking. when it finds something
+//     that looks like a page, it streams through it and verifies its
+//     CRC32. Should that validation fail, it keeps scanning. But it's
+//     possible that _while_ streaming through to check the CRC32 of
+//     one candidate page, it sees another candidate page. This #define
+//     determines how many "overlapping" candidate pages it can search
+//     at once. Note that "real" pages are typically ~4KB to ~8KB, whereas
+//     garbage pages could be as big as 64KB, but probably average ~16KB.
+//     So don't hose ourselves by scanning an apparent 64KB page and
+//     missing a ton of real ones in the interim; so minimum of 2
+#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT
+#define STB_VORBIS_PUSHDATA_CRC_COUNT  4
+#endif
+
+// STB_VORBIS_FAST_HUFFMAN_LENGTH [number]
+//     sets the log size of the huffman-acceleration table.  Maximum
+//     supported value is 24. with larger numbers, more decodings are O(1),
+//     but the table size is larger so worse cache missing, so you'll have
+//     to probe (and try multiple ogg vorbis files) to find the sweet spot.
+#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH
+#define STB_VORBIS_FAST_HUFFMAN_LENGTH   10
+#endif
+
+// STB_VORBIS_FAST_BINARY_LENGTH [number]
+//     sets the log size of the binary-search acceleration table. this
+//     is used in similar fashion to the fast-huffman size to set initial
+//     parameters for the binary search
+
+// STB_VORBIS_FAST_HUFFMAN_INT
+//     The fast huffman tables are much more efficient if they can be
+//     stored as 16-bit results instead of 32-bit results. This restricts
+//     the codebooks to having only 65535 possible outcomes, though.
+//     (At least, accelerated by the huffman table.)
+#ifndef STB_VORBIS_FAST_HUFFMAN_INT
+#define STB_VORBIS_FAST_HUFFMAN_SHORT
+#endif
+
+// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+//     If the 'fast huffman' search doesn't succeed, then stb_vorbis falls
+//     back on binary searching for the correct one. This requires storing
+//     extra tables with the huffman codes in sorted order. Defining this
+//     symbol trades off space for speed by forcing a linear search in the
+//     non-fast case, except for "sparse" codebooks.
+// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+
+// STB_VORBIS_DIVIDES_IN_RESIDUE
+//     stb_vorbis precomputes the result of the scalar residue decoding
+//     that would otherwise require a divide per chunk. you can trade off
+//     space for time by defining this symbol.
+// #define STB_VORBIS_DIVIDES_IN_RESIDUE
+
+// STB_VORBIS_DIVIDES_IN_CODEBOOK
+//     vorbis VQ codebooks can be encoded two ways: with every case explicitly
+//     stored, or with all elements being chosen from a small range of values,
+//     and all values possible in all elements. By default, stb_vorbis expands
+//     this latter kind out to look like the former kind for ease of decoding,
+//     because otherwise an integer divide-per-vector-element is required to
+//     unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can
+//     trade off storage for speed.
+//#define STB_VORBIS_DIVIDES_IN_CODEBOOK
+
+#ifdef STB_VORBIS_CODEBOOK_SHORTS
+#error "STB_VORBIS_CODEBOOK_SHORTS is no longer supported as it produced incorrect results for some input formats"
+#endif
+
+// STB_VORBIS_DIVIDE_TABLE
+//     this replaces small integer divides in the floor decode loop with
+//     table lookups. made less than 1% difference, so disabled by default.
+
+// STB_VORBIS_NO_INLINE_DECODE
+//     disables the inlining of the scalar codebook fast-huffman decode.
+//     might save a little codespace; useful for debugging
+// #define STB_VORBIS_NO_INLINE_DECODE
+
+// STB_VORBIS_NO_DEFER_FLOOR
+//     Normally we only decode the floor without synthesizing the actual
+//     full curve. We can instead synthesize the curve immediately. This
+//     requires more memory and is very likely slower, so I don't think
+//     you'd ever want to do it except for debugging.
+// #define STB_VORBIS_NO_DEFER_FLOOR
+
+
+
+
+//////////////////////////////////////////////////////////////////////////////
+
+#ifdef STB_VORBIS_NO_PULLDATA_API
+   #define STB_VORBIS_NO_INTEGER_CONVERSION
+   #define STB_VORBIS_NO_STDIO
+#endif
+
+#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO)
+   #define STB_VORBIS_NO_STDIO 1
+#endif
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+
+   // only need endianness for fast-float-to-int, which we don't
+   // use for pushdata
+
+   #ifndef STB_VORBIS_BIG_ENDIAN
+     #define STB_VORBIS_ENDIAN  0
+   #else
+     #define STB_VORBIS_ENDIAN  1
+   #endif
+
+#endif
+#endif
+
+
+#ifndef STB_VORBIS_NO_STDIO
+#include <stdio.h>
+#endif
+
+#ifndef STB_VORBIS_NO_CRT
+   #include <stdlib.h>
+   #include <string.h>
+   #include <assert.h>
+   #include <math.h>
+
+   // find definition of alloca if it's not in stdlib.h:
+   #ifdef _MSC_VER
+      #include <malloc.h>
+   #endif
+   #if defined(__linux__) || defined(__linux) || defined(__EMSCRIPTEN__)
+      #include <alloca.h>
+   #endif
+#else // STB_VORBIS_NO_CRT
+   #define NULL 0
+   #define malloc(s)   0
+   #define free(s)     ((void) 0)
+   #define realloc(s)  0
+#endif // STB_VORBIS_NO_CRT
+
+#include <limits.h>
+
+#ifdef __MINGW32__
+   // eff you mingw:
+   //     "fixed":
+   //         http://sourceforge.net/p/mingw-w64/mailman/message/32882927/
+   //     "no that broke the build, reverted, who cares about C":
+   //         http://sourceforge.net/p/mingw-w64/mailman/message/32890381/
+   #ifdef __forceinline
+   #undef __forceinline
+   #endif
+   #define __forceinline
+#elif !defined(_MSC_VER)
+   #if __GNUC__
+      #define __forceinline inline
+   #else
+      #define __forceinline
+   #endif
+#endif
+
+#if STB_VORBIS_MAX_CHANNELS > 256
+#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range"
+#endif
+
+#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24
+#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range"
+#endif
+
+
+#if 0
+#include <crtdbg.h>
+#define CHECK(f)   _CrtIsValidHeapPointer(f->channel_buffers[1])
+#else
+#define CHECK(f)   ((void) 0)
+#endif
+
+#define MAX_BLOCKSIZE_LOG  13   // from specification
+#define MAX_BLOCKSIZE      (1 << MAX_BLOCKSIZE_LOG)
+
+
+typedef unsigned char  uint8;
+typedef   signed char   int8;
+typedef unsigned short uint16;
+typedef   signed short  int16;
+typedef unsigned int   uint32;
+typedef   signed int    int32;
+
+#ifndef TRUE
+#define TRUE 1
+#define FALSE 0
+#endif
+
+typedef float codetype;
+
+// @NOTE
+//
+// Some arrays below are tagged "//varies", which means it's actually
+// a variable-sized piece of data, but rather than malloc I assume it's
+// small enough it's better to just allocate it all together with the
+// main thing
+//
+// Most of the variables are specified with the smallest size I could pack
+// them into. It might give better performance to make them all full-sized
+// integers. It should be safe to freely rearrange the structures or change
+// the sizes larger--nothing relies on silently truncating etc., nor the
+// order of variables.
+
+#define FAST_HUFFMAN_TABLE_SIZE   (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH)
+#define FAST_HUFFMAN_TABLE_MASK   (FAST_HUFFMAN_TABLE_SIZE - 1)
+
+typedef struct
+{
+   int dimensions, entries;
+   uint8 *codeword_lengths;
+   float  minimum_value;
+   float  delta_value;
+   uint8  value_bits;
+   uint8  lookup_type;
+   uint8  sequence_p;
+   uint8  sparse;
+   uint32 lookup_values;
+   codetype *multiplicands;
+   uint32 *codewords;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+    int16  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #else
+    int32  fast_huffman[FAST_HUFFMAN_TABLE_SIZE];
+   #endif
+   uint32 *sorted_codewords;
+   int    *sorted_values;
+   int     sorted_entries;
+} Codebook;
+
+typedef struct
+{
+   uint8 order;
+   uint16 rate;
+   uint16 bark_map_size;
+   uint8 amplitude_bits;
+   uint8 amplitude_offset;
+   uint8 number_of_books;
+   uint8 book_list[16]; // varies
+} Floor0;
+
+typedef struct
+{
+   uint8 partitions;
+   uint8 partition_class_list[32]; // varies
+   uint8 class_dimensions[16]; // varies
+   uint8 class_subclasses[16]; // varies
+   uint8 class_masterbooks[16]; // varies
+   int16 subclass_books[16][8]; // varies
+   uint16 Xlist[31*8+2]; // varies
+   uint8 sorted_order[31*8+2];
+   uint8 neighbors[31*8+2][2];
+   uint8 floor1_multiplier;
+   uint8 rangebits;
+   int values;
+} Floor1;
+
+typedef union
+{
+   Floor0 floor0;
+   Floor1 floor1;
+} Floor;
+
+typedef struct
+{
+   uint32 begin, end;
+   uint32 part_size;
+   uint8 classifications;
+   uint8 classbook;
+   uint8 **classdata;
+   int16 (*residue_books)[8];
+} Residue;
+
+typedef struct
+{
+   uint8 magnitude;
+   uint8 angle;
+   uint8 mux;
+} MappingChannel;
+
+typedef struct
+{
+   uint16 coupling_steps;
+   MappingChannel *chan;
+   uint8  submaps;
+   uint8  submap_floor[15]; // varies
+   uint8  submap_residue[15]; // varies
+} Mapping;
+
+typedef struct
+{
+   uint8 blockflag;
+   uint8 mapping;
+   uint16 windowtype;
+   uint16 transformtype;
+} Mode;
+
+typedef struct
+{
+   uint32  goal_crc;    // expected crc if match
+   int     bytes_left;  // bytes left in packet
+   uint32  crc_so_far;  // running crc
+   int     bytes_done;  // bytes processed in _current_ chunk
+   uint32  sample_loc;  // granule pos encoded in page
+} CRCscan;
+
+typedef struct
+{
+   uint32 page_start, page_end;
+   uint32 last_decoded_sample;
+} ProbedPage;
+
+struct stb_vorbis
+{
+  // user-accessible info
+   unsigned int sample_rate;
+   int channels;
+
+   unsigned int setup_memory_required;
+   unsigned int temp_memory_required;
+   unsigned int setup_temp_memory_required;
+
+  // input config
+#ifndef STB_VORBIS_NO_STDIO
+   FILE *f;
+   uint32 f_start;
+   int close_on_free;
+#endif
+
+   uint8 *stream;
+   uint8 *stream_start;
+   uint8 *stream_end;
+
+   uint32 stream_len;
+
+   uint8  push_mode;
+
+   uint32 first_audio_page_offset;
+
+   ProbedPage p_first, p_last;
+
+  // memory management
+   stb_vorbis_alloc alloc;
+   int setup_offset;
+   int temp_offset;
+
+  // run-time results
+   int eof;
+   enum STBVorbisError error;
+
+  // user-useful data
+
+  // header info
+   int blocksize[2];
+   int blocksize_0, blocksize_1;
+   int codebook_count;
+   Codebook *codebooks;
+   int floor_count;
+   uint16 floor_types[64]; // varies
+   Floor *floor_config;
+   int residue_count;
+   uint16 residue_types[64]; // varies
+   Residue *residue_config;
+   int mapping_count;
+   Mapping *mapping;
+   int mode_count;
+   Mode mode_config[64];  // varies
+
+   uint32 total_samples;
+
+  // decode buffer
+   float *channel_buffers[STB_VORBIS_MAX_CHANNELS];
+   float *outputs        [STB_VORBIS_MAX_CHANNELS];
+
+   float *previous_window[STB_VORBIS_MAX_CHANNELS];
+   int previous_length;
+
+   #ifndef STB_VORBIS_NO_DEFER_FLOOR
+   int16 *finalY[STB_VORBIS_MAX_CHANNELS];
+   #else
+   float *floor_buffers[STB_VORBIS_MAX_CHANNELS];
+   #endif
+
+   uint32 current_loc; // sample location of next frame to decode
+   int    current_loc_valid;
+
+  // per-blocksize precomputed data
+   
+   // twiddle factors
+   float *A[2],*B[2],*C[2];
+   float *window[2];
+   uint16 *bit_reverse[2];
+
+  // current page/packet/segment streaming info
+   uint32 serial; // stream serial number for verification
+   int last_page;
+   int segment_count;
+   uint8 segments[255];
+   uint8 page_flag;
+   uint8 bytes_in_seg;
+   uint8 first_decode;
+   int next_seg;
+   int last_seg;  // flag that we're on the last segment
+   int last_seg_which; // what was the segment number of the last seg?
+   uint32 acc;
+   int valid_bits;
+   int packet_bytes;
+   int end_seg_with_known_loc;
+   uint32 known_loc_for_packet;
+   int discard_samples_deferred;
+   uint32 samples_output;
+
+  // push mode scanning
+   int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+   CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT];
+#endif
+
+  // sample-access
+   int channel_buffer_start;
+   int channel_buffer_end;
+};
+
+#if defined(STB_VORBIS_NO_PUSHDATA_API)
+   #define IS_PUSH_MODE(f)   FALSE
+#elif defined(STB_VORBIS_NO_PULLDATA_API)
+   #define IS_PUSH_MODE(f)   TRUE
+#else
+   #define IS_PUSH_MODE(f)   ((f)->push_mode)
+#endif
+
+typedef struct stb_vorbis vorb;
+
+static int error(vorb *f, enum STBVorbisError e)
+{
+   f->error = e;
+   if (!f->eof && e != VORBIS_need_more_data) {
+      f->error=e; // breakpoint for debugging
+   }
+   return 0;
+}
+
+
+// these functions are used for allocating temporary memory
+// while decoding. if you can afford the stack space, use
+// alloca(); otherwise, provide a temp buffer and it will
+// allocate out of those.
+
+#define array_size_required(count,size)  (count*(sizeof(void *)+(size)))
+
+#define temp_alloc(f,size)              (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size))
+#ifdef dealloca
+#define temp_free(f,p)                  (f->alloc.alloc_buffer ? 0 : dealloca(size))
+#else
+#define temp_free(f,p)                  0
+#endif
+#define temp_alloc_save(f)              ((f)->temp_offset)
+#define temp_alloc_restore(f,p)         ((f)->temp_offset = (p))
+
+#define temp_block_array(f,count,size)  make_block_array(temp_alloc(f,array_size_required(count,size)), count, size)
+
+// given a sufficiently large block of memory, make an array of pointers to subblocks of it
+static void *make_block_array(void *mem, int count, int size)
+{
+   int i;
+   void ** p = (void **) mem;
+   char *q = (char *) (p + count);
+   for (i=0; i < count; ++i) {
+      p[i] = q;
+      q += size;
+   }
+   return p;
+}
+
+static void *setup_malloc(vorb *f, int sz)
+{
+   sz = (sz+3) & ~3;
+   f->setup_memory_required += sz;
+   if (f->alloc.alloc_buffer) {
+      void *p = (char *) f->alloc.alloc_buffer + f->setup_offset;
+      if (f->setup_offset + sz > f->temp_offset) return NULL;
+      f->setup_offset += sz;
+      return p;
+   }
+   return sz ? malloc(sz) : NULL;
+}
+
+static void setup_free(vorb *f, void *p)
+{
+   if (f->alloc.alloc_buffer) return; // do nothing; setup mem is a stack
+   free(p);
+}
+
+static void *setup_temp_malloc(vorb *f, int sz)
+{
+   sz = (sz+3) & ~3;
+   if (f->alloc.alloc_buffer) {
+      if (f->temp_offset - sz < f->setup_offset) return NULL;
+      f->temp_offset -= sz;
+      return (char *) f->alloc.alloc_buffer + f->temp_offset;
+   }
+   return malloc(sz);
+}
+
+static void setup_temp_free(vorb *f, void *p, int sz)
+{
+   if (f->alloc.alloc_buffer) {
+      f->temp_offset += (sz+3)&~3;
+      return;
+   }
+   free(p);
+}
+
+#define CRC32_POLY    0x04c11db7   // from spec
+
+static uint32 crc_table[256];
+static void crc32_init(void)
+{
+   int i,j;
+   uint32 s;
+   for(i=0; i < 256; i++) {
+      for (s=(uint32) i << 24, j=0; j < 8; ++j)
+         s = (s << 1) ^ (s >= (1U<<31) ? CRC32_POLY : 0);
+      crc_table[i] = s;
+   }
+}
+
+static __forceinline uint32 crc32_update(uint32 crc, uint8 byte)
+{
+   return (crc << 8) ^ crc_table[byte ^ (crc >> 24)];
+}
+
+
+// used in setup, and for huffman that doesn't go fast path
+static unsigned int bit_reverse(unsigned int n)
+{
+  n = ((n & 0xAAAAAAAA) >>  1) | ((n & 0x55555555) << 1);
+  n = ((n & 0xCCCCCCCC) >>  2) | ((n & 0x33333333) << 2);
+  n = ((n & 0xF0F0F0F0) >>  4) | ((n & 0x0F0F0F0F) << 4);
+  n = ((n & 0xFF00FF00) >>  8) | ((n & 0x00FF00FF) << 8);
+  return (n >> 16) | (n << 16);
+}
+
+static float square(float x)
+{
+   return x*x;
+}
+
+// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3
+// as required by the specification. fast(?) implementation from stb.h
+// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup
+static int ilog(int32 n)
+{
+   static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 };
+
+   // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29)
+   if (n < (1 << 14))
+        if (n < (1 <<  4))        return     0 + log2_4[n      ];
+        else if (n < (1 <<  9))      return  5 + log2_4[n >>  5];
+             else                     return 10 + log2_4[n >> 10];
+   else if (n < (1 << 24))
+             if (n < (1 << 19))      return 15 + log2_4[n >> 15];
+             else                     return 20 + log2_4[n >> 20];
+        else if (n < (1 << 29))      return 25 + log2_4[n >> 25];
+             else if (n < (1 << 31)) return 30 + log2_4[n >> 30];
+                  else                return 0; // signed n returns 0
+}
+
+#ifndef M_PI
+  #define M_PI  3.14159265358979323846264f  // from CRC
+#endif
+
+// code length assigned to a value with no huffman encoding
+#define NO_CODE   255
+
+/////////////////////// LEAF SETUP FUNCTIONS //////////////////////////
+//
+// these functions are only called at setup, and only a few times
+// per file
+
+static float float32_unpack(uint32 x)
+{
+   // from the specification
+   uint32 mantissa = x & 0x1fffff;
+   uint32 sign = x & 0x80000000;
+   uint32 exp = (x & 0x7fe00000) >> 21;
+   double res = sign ? -(double)mantissa : (double)mantissa;
+   return (float) ldexp((float)res, exp-788);
+}
+
+
+// zlib & jpeg huffman tables assume that the output symbols
+// can either be arbitrarily arranged, or have monotonically
+// increasing frequencies--they rely on the lengths being sorted;
+// this makes for a very simple generation algorithm.
+// vorbis allows a huffman table with non-sorted lengths. This
+// requires a more sophisticated construction, since symbols in
+// order do not map to huffman codes "in order".
+static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values)
+{
+   if (!c->sparse) {
+      c->codewords      [symbol] = huff_code;
+   } else {
+      c->codewords       [count] = huff_code;
+      c->codeword_lengths[count] = len;
+      values             [count] = symbol;
+   }
+}
+
+static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values)
+{
+   int i,k,m=0;
+   uint32 available[32];
+
+   memset(available, 0, sizeof(available));
+   // find the first entry
+   for (k=0; k < n; ++k) if (len[k] < NO_CODE) break;
+   if (k == n) { assert(c->sorted_entries == 0); return TRUE; }
+   // add to the list
+   add_entry(c, 0, k, m++, len[k], values);
+   // add all available leaves
+   for (i=1; i <= len[k]; ++i)
+      available[i] = 1U << (32-i);
+   // note that the above code treats the first case specially,
+   // but it's really the same as the following code, so they
+   // could probably be combined (except the initial code is 0,
+   // and I use 0 in available[] to mean 'empty')
+   for (i=k+1; i < n; ++i) {
+      uint32 res;
+      int z = len[i], y;
+      if (z == NO_CODE) continue;
+      // find lowest available leaf (should always be earliest,
+      // which is what the specification calls for)
+      // note that this property, and the fact we can never have
+      // more than one free leaf at a given level, isn't totally
+      // trivial to prove, but it seems true and the assert never
+      // fires, so!
+      while (z > 0 && !available[z]) --z;
+      if (z == 0) { return FALSE; }
+      res = available[z];
+      assert(z >= 0 && z < 32);
+      available[z] = 0;
+      add_entry(c, bit_reverse(res), i, m++, len[i], values);
+      // propogate availability up the tree
+      if (z != len[i]) {
+         assert(len[i] >= 0 && len[i] < 32);
+         for (y=len[i]; y > z; --y) {
+            assert(available[y] == 0);
+            available[y] = res + (1 << (32-y));
+         }
+      }
+   }
+   return TRUE;
+}
+
+// accelerated huffman table allows fast O(1) match of all symbols
+// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH
+static void compute_accelerated_huffman(Codebook *c)
+{
+   int i, len;
+   for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i)
+      c->fast_huffman[i] = -1;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT
+   if (len > 32767) len = 32767; // largest possible value we can encode!
+   #endif
+   for (i=0; i < len; ++i) {
+      if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) {
+         uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i];
+         // set table entries for all bit combinations in the higher bits
+         while (z < FAST_HUFFMAN_TABLE_SIZE) {
+             c->fast_huffman[z] = i;
+             z += 1 << c->codeword_lengths[i];
+         }
+      }
+   }
+}
+
+#ifdef _MSC_VER
+#define STBV_CDECL __cdecl
+#else
+#define STBV_CDECL
+#endif
+
+static int STBV_CDECL uint32_compare(const void *p, const void *q)
+{
+   uint32 x = * (uint32 *) p;
+   uint32 y = * (uint32 *) q;
+   return x < y ? -1 : x > y;
+}
+
+static int include_in_sort(Codebook *c, uint8 len)
+{
+   if (c->sparse) { assert(len != NO_CODE); return TRUE; }
+   if (len == NO_CODE) return FALSE;
+   if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE;
+   return FALSE;
+}
+
+// if the fast table above doesn't work, we want to binary
+// search them... need to reverse the bits
+static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values)
+{
+   int i, len;
+   // build a list of all the entries
+   // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN.
+   // this is kind of a frivolous optimization--I don't see any performance improvement,
+   // but it's like 4 extra lines of code, so.
+   if (!c->sparse) {
+      int k = 0;
+      for (i=0; i < c->entries; ++i)
+         if (include_in_sort(c, lengths[i])) 
+            c->sorted_codewords[k++] = bit_reverse(c->codewords[i]);
+      assert(k == c->sorted_entries);
+   } else {
+      for (i=0; i < c->sorted_entries; ++i)
+         c->sorted_codewords[i] = bit_reverse(c->codewords[i]);
+   }
+
+   qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare);
+   c->sorted_codewords[c->sorted_entries] = 0xffffffff;
+
+   len = c->sparse ? c->sorted_entries : c->entries;
+   // now we need to indicate how they correspond; we could either
+   //   #1: sort a different data structure that says who they correspond to
+   //   #2: for each sorted entry, search the original list to find who corresponds
+   //   #3: for each original entry, find the sorted entry
+   // #1 requires extra storage, #2 is slow, #3 can use binary search!
+   for (i=0; i < len; ++i) {
+      int huff_len = c->sparse ? lengths[values[i]] : lengths[i];
+      if (include_in_sort(c,huff_len)) {
+         uint32 code = bit_reverse(c->codewords[i]);
+         int x=0, n=c->sorted_entries;
+         while (n > 1) {
+            // invariant: sc[x] <= code < sc[x+n]
+            int m = x + (n >> 1);
+            if (c->sorted_codewords[m] <= code) {
+               x = m;
+               n -= (n>>1);
+            } else {
+               n >>= 1;
+            }
+         }
+         assert(c->sorted_codewords[x] == code);
+         if (c->sparse) {
+            c->sorted_values[x] = values[i];
+            c->codeword_lengths[x] = huff_len;
+         } else {
+            c->sorted_values[x] = i;
+         }
+      }
+   }
+}
+
+// only run while parsing the header (3 times)
+static int vorbis_validate(uint8 *data)
+{
+   static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' };
+   return memcmp(data, vorbis, 6) == 0;
+}
+
+// called from setup only, once per code book
+// (formula implied by specification)
+static int lookup1_values(int entries, int dim)
+{
+   int r = (int) floor(exp((float) log((float) entries) / dim));
+   if ((int) floor(pow((float) r+1, dim)) <= entries)   // (int) cast for MinGW warning;
+      ++r;                                              // floor() to avoid _ftol() when non-CRT
+   assert(pow((float) r+1, dim) > entries);
+   assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above
+   return r;
+}
+
+// called twice per file
+static void compute_twiddle_factors(int n, float *A, float *B, float *C)
+{
+   int n4 = n >> 2, n8 = n >> 3;
+   int k,k2;
+
+   for (k=k2=0; k < n4; ++k,k2+=2) {
+      A[k2  ] = (float)  cos(4*k*M_PI/n);
+      A[k2+1] = (float) -sin(4*k*M_PI/n);
+      B[k2  ] = (float)  cos((k2+1)*M_PI/n/2) * 0.5f;
+      B[k2+1] = (float)  sin((k2+1)*M_PI/n/2) * 0.5f;
+   }
+   for (k=k2=0; k < n8; ++k,k2+=2) {
+      C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+      C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+   }
+}
+
+static void compute_window(int n, float *window)
+{
+   int n2 = n >> 1, i;
+   for (i=0; i < n2; ++i)
+      window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI)));
+}
+
+static void compute_bitreverse(int n, uint16 *rev)
+{
+   int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+   int i, n8 = n >> 3;
+   for (i=0; i < n8; ++i)
+      rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2;
+}
+
+static int init_blocksize(vorb *f, int b, int n)
+{
+   int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3;
+   f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4);
+   if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem);
+   compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]);
+   f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2);
+   if (!f->window[b]) return error(f, VORBIS_outofmem);
+   compute_window(n, f->window[b]);
+   f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8);
+   if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem);
+   compute_bitreverse(n, f->bit_reverse[b]);
+   return TRUE;
+}
+
+static void neighbors(uint16 *x, int n, int *plow, int *phigh)
+{
+   int low = -1;
+   int high = 65536;
+   int i;
+   for (i=0; i < n; ++i) {
+      if (x[i] > low  && x[i] < x[n]) { *plow  = i; low = x[i]; }
+      if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; }
+   }
+}
+
+// this has been repurposed so y is now the original index instead of y
+typedef struct
+{
+   uint16 x,y;
+} Point;
+
+static int STBV_CDECL point_compare(const void *p, const void *q)
+{
+   Point *a = (Point *) p;
+   Point *b = (Point *) q;
+   return a->x < b->x ? -1 : a->x > b->x;
+}
+
+//
+/////////////////////// END LEAF SETUP FUNCTIONS //////////////////////////
+
+
+#if defined(STB_VORBIS_NO_STDIO)
+   #define USE_MEMORY(z)    TRUE
+#else
+   #define USE_MEMORY(z)    ((z)->stream)
+#endif
+
+static uint8 get8(vorb *z)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; }
+      return *z->stream++;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+   int c = fgetc(z->f);
+   if (c == EOF) { z->eof = TRUE; return 0; }
+   return c;
+   }
+   #endif
+}
+
+static uint32 get32(vorb *f)
+{
+   uint32 x;
+   x = get8(f);
+   x += get8(f) << 8;
+   x += get8(f) << 16;
+   x += (uint32) get8(f) << 24;
+   return x;
+}
+
+static int getn(vorb *z, uint8 *data, int n)
+{
+   if (USE_MEMORY(z)) {
+      if (z->stream+n > z->stream_end) { z->eof = 1; return 0; }
+      memcpy(data, z->stream, n);
+      z->stream += n;
+      return 1;
+   }
+
+   #ifndef STB_VORBIS_NO_STDIO   
+   if (fread(data, n, 1, z->f) == 1)
+      return 1;
+   else {
+      z->eof = 1;
+      return 0;
+   }
+   #endif
+}
+
+static void skip(vorb *z, int n)
+{
+   if (USE_MEMORY(z)) {
+      z->stream += n;
+      if (z->stream >= z->stream_end) z->eof = 1;
+      return;
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   {
+      long x = ftell(z->f);
+      fseek(z->f, x+n, SEEK_SET);
+   }
+   #endif
+}
+
+static int set_file_offset(stb_vorbis *f, unsigned int loc)
+{
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (f->push_mode) return 0;
+   #endif
+   f->eof = 0;
+   if (USE_MEMORY(f)) {
+      if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) {
+         f->stream = f->stream_end;
+         f->eof = 1;
+         return 0;
+      } else {
+         f->stream = f->stream_start + loc;
+         return 1;
+      }
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   if (loc + f->f_start < loc || loc >= 0x80000000) {
+      loc = 0x7fffffff;
+      f->eof = 1;
+   } else {
+      loc += f->f_start;
+   }
+   if (!fseek(f->f, loc, SEEK_SET))
+      return 1;
+   f->eof = 1;
+   fseek(f->f, f->f_start, SEEK_END);
+   return 0;
+   #endif
+}
+
+
+static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 };
+
+static int capture_pattern(vorb *f)
+{
+   if (0x4f != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x67 != get8(f)) return FALSE;
+   if (0x53 != get8(f)) return FALSE;
+   return TRUE;
+}
+
+#define PAGEFLAG_continued_packet   1
+#define PAGEFLAG_first_page         2
+#define PAGEFLAG_last_page          4
+
+static int start_page_no_capturepattern(vorb *f)
+{
+   uint32 loc0,loc1,n;
+   // stream structure version
+   if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version);
+   // header flag
+   f->page_flag = get8(f);
+   // absolute granule position
+   loc0 = get32(f); 
+   loc1 = get32(f);
+   // @TODO: validate loc0,loc1 as valid positions?
+   // stream serial number -- vorbis doesn't interleave, so discard
+   get32(f);
+   //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number);
+   // page sequence number
+   n = get32(f);
+   f->last_page = n;
+   // CRC32
+   get32(f);
+   // page_segments
+   f->segment_count = get8(f);
+   if (!getn(f, f->segments, f->segment_count))
+      return error(f, VORBIS_unexpected_eof);
+   // assume we _don't_ know any the sample position of any segments
+   f->end_seg_with_known_loc = -2;
+   if (loc0 != ~0U || loc1 != ~0U) {
+      int i;
+      // determine which packet is the last one that will complete
+      for (i=f->segment_count-1; i >= 0; --i)
+         if (f->segments[i] < 255)
+            break;
+      // 'i' is now the index of the _last_ segment of a packet that ends
+      if (i >= 0) {
+         f->end_seg_with_known_loc = i;
+         f->known_loc_for_packet   = loc0;
+      }
+   }
+   if (f->first_decode) {
+      int i,len;
+      ProbedPage p;
+      len = 0;
+      for (i=0; i < f->segment_count; ++i)
+         len += f->segments[i];
+      len += 27 + f->segment_count;
+      p.page_start = f->first_audio_page_offset;
+      p.page_end = p.page_start + len;
+      p.last_decoded_sample = loc0;
+      f->p_first = p;
+   }
+   f->next_seg = 0;
+   return TRUE;
+}
+
+static int start_page(vorb *f)
+{
+   if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern);
+   return start_page_no_capturepattern(f);
+}
+
+static int start_packet(vorb *f)
+{
+   while (f->next_seg == -1) {
+      if (!start_page(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet)
+         return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   f->last_seg = FALSE;
+   f->valid_bits = 0;
+   f->packet_bytes = 0;
+   f->bytes_in_seg = 0;
+   // f->next_seg is now valid
+   return TRUE;
+}
+
+static int maybe_start_packet(vorb *f)
+{
+   if (f->next_seg == -1) {
+      int x = get8(f);
+      if (f->eof) return FALSE; // EOF at page boundary is not an error!
+      if (0x4f != x      ) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern);
+      if (!start_page_no_capturepattern(f)) return FALSE;
+      if (f->page_flag & PAGEFLAG_continued_packet) {
+         // set up enough state that we can read this packet if we want,
+         // e.g. during recovery
+         f->last_seg = FALSE;
+         f->bytes_in_seg = 0;
+         return error(f, VORBIS_continued_packet_flag_invalid);
+      }
+   }
+   return start_packet(f);
+}
+
+static int next_segment(vorb *f)
+{
+   int len;
+   if (f->last_seg) return 0;
+   if (f->next_seg == -1) {
+      f->last_seg_which = f->segment_count-1; // in case start_page fails
+      if (!start_page(f)) { f->last_seg = 1; return 0; }
+      if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid);
+   }
+   len = f->segments[f->next_seg++];
+   if (len < 255) {
+      f->last_seg = TRUE;
+      f->last_seg_which = f->next_seg-1;
+   }
+   if (f->next_seg >= f->segment_count)
+      f->next_seg = -1;
+   assert(f->bytes_in_seg == 0);
+   f->bytes_in_seg = len;
+   return len;
+}
+
+#define EOP    (-1)
+#define INVALID_BITS  (-1)
+
+static int get8_packet_raw(vorb *f)
+{
+   if (!f->bytes_in_seg) {  // CLANG!
+      if (f->last_seg) return EOP;
+      else if (!next_segment(f)) return EOP;
+   }
+   assert(f->bytes_in_seg > 0);
+   --f->bytes_in_seg;
+   ++f->packet_bytes;
+   return get8(f);
+}
+
+static int get8_packet(vorb *f)
+{
+   int x = get8_packet_raw(f);
+   f->valid_bits = 0;
+   return x;
+}
+
+static void flush_packet(vorb *f)
+{
+   while (get8_packet_raw(f) != EOP);
+}
+
+// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important
+// as the huffman decoder?
+static uint32 get_bits(vorb *f, int n)
+{
+   uint32 z;
+
+   if (f->valid_bits < 0) return 0;
+   if (f->valid_bits < n) {
+      if (n > 24) {
+         // the accumulator technique below would not work correctly in this case
+         z = get_bits(f, 24);
+         z += get_bits(f, n-24) << 24;
+         return z;
+      }
+      if (f->valid_bits == 0) f->acc = 0;
+      while (f->valid_bits < n) {
+         int z = get8_packet_raw(f);
+         if (z == EOP) {
+            f->valid_bits = INVALID_BITS;
+            return 0;
+         }
+         f->acc += z << f->valid_bits;
+         f->valid_bits += 8;
+      }
+   }
+   if (f->valid_bits < 0) return 0;
+   z = f->acc & ((1 << n)-1);
+   f->acc >>= n;
+   f->valid_bits -= n;
+   return z;
+}
+
+// @OPTIMIZE: primary accumulator for huffman
+// expand the buffer to as many bits as possible without reading off end of packet
+// it might be nice to allow f->valid_bits and f->acc to be stored in registers,
+// e.g. cache them locally and decode locally
+static __forceinline void prep_huffman(vorb *f)
+{
+   if (f->valid_bits <= 24) {
+      if (f->valid_bits == 0) f->acc = 0;
+      do {
+         int z;
+         if (f->last_seg && !f->bytes_in_seg) return;
+         z = get8_packet_raw(f);
+         if (z == EOP) return;
+         f->acc += (unsigned) z << f->valid_bits;
+         f->valid_bits += 8;
+      } while (f->valid_bits <= 24);
+   }
+}
+
+enum
+{
+   VORBIS_packet_id = 1,
+   VORBIS_packet_comment = 3,
+   VORBIS_packet_setup = 5
+};
+
+static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
+{
+   int i;
+   prep_huffman(f);
+
+   if (c->codewords == NULL && c->sorted_codewords == NULL)
+      return -1;
+
+   // cases to use binary search: sorted_codewords && !c->codewords
+   //                             sorted_codewords && c->entries > 8
+   if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) {
+      // binary search
+      uint32 code = bit_reverse(f->acc);
+      int x=0, n=c->sorted_entries, len;
+
+      while (n > 1) {
+         // invariant: sc[x] <= code < sc[x+n]
+         int m = x + (n >> 1);
+         if (c->sorted_codewords[m] <= code) {
+            x = m;
+            n -= (n>>1);
+         } else {
+            n >>= 1;
+         }
+      }
+      // x is now the sorted index
+      if (!c->sparse) x = c->sorted_values[x];
+      // x is now sorted index if sparse, or symbol otherwise
+      len = c->codeword_lengths[x];
+      if (f->valid_bits >= len) {
+         f->acc >>= len;
+         f->valid_bits -= len;
+         return x;
+      }
+
+      f->valid_bits = 0;
+      return -1;
+   }
+
+   // if small, linear search
+   assert(!c->sparse);
+   for (i=0; i < c->entries; ++i) {
+      if (c->codeword_lengths[i] == NO_CODE) continue;
+      if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
+         if (f->valid_bits >= c->codeword_lengths[i]) {
+            f->acc >>= c->codeword_lengths[i];
+            f->valid_bits -= c->codeword_lengths[i];
+            return i;
+         }
+         f->valid_bits = 0;
+         return -1;
+      }
+   }
+
+   error(f, VORBIS_invalid_stream);
+   f->valid_bits = 0;
+   return -1;
+}
+
+#ifndef STB_VORBIS_NO_INLINE_DECODE
+
+#define DECODE_RAW(var, f,c)                                  \
+   if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)        \
+      prep_huffman(f);                                        \
+   var = f->acc & FAST_HUFFMAN_TABLE_MASK;                    \
+   var = c->fast_huffman[var];                                \
+   if (var >= 0) {                                            \
+      int n = c->codeword_lengths[var];                       \
+      f->acc >>= n;                                           \
+      f->valid_bits -= n;                                     \
+      if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \
+   } else {                                                   \
+      var = codebook_decode_scalar_raw(f,c);                  \
+   }
+
+#else
+
+static int codebook_decode_scalar(vorb *f, Codebook *c)
+{
+   int i;
+   if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH)
+      prep_huffman(f);
+   // fast huffman table lookup
+   i = f->acc & FAST_HUFFMAN_TABLE_MASK;
+   i = c->fast_huffman[i];
+   if (i >= 0) {
+      f->acc >>= c->codeword_lengths[i];
+      f->valid_bits -= c->codeword_lengths[i];
+      if (f->valid_bits < 0) { f->valid_bits = 0; return -1; }
+      return i;
+   }
+   return codebook_decode_scalar_raw(f,c);
+}
+
+#define DECODE_RAW(var,f,c)    var = codebook_decode_scalar(f,c);
+
+#endif
+
+#define DECODE(var,f,c)                                       \
+   DECODE_RAW(var,f,c)                                        \
+   if (c->sparse) var = c->sorted_values[var];
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+  #define DECODE_VQ(var,f,c)   DECODE_RAW(var,f,c)
+#else
+  #define DECODE_VQ(var,f,c)   DECODE(var,f,c)
+#endif
+
+
+
+
+
+
+// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case
+// where we avoid one addition
+#define CODEBOOK_ELEMENT(c,off)          (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_FAST(c,off)     (c->multiplicands[off])
+#define CODEBOOK_ELEMENT_BASE(c)         (0)
+
+static int codebook_decode_start(vorb *f, Codebook *c)
+{
+   int z = -1;
+
+   // type 0 is only legal in a scalar context
+   if (c->lookup_type == 0)
+      error(f, VORBIS_invalid_stream);
+   else {
+      DECODE_VQ(z,f,c);
+      if (c->sparse) assert(z < c->sorted_entries);
+      if (z < 0) {  // check for EOP
+         if (!f->bytes_in_seg)
+            if (f->last_seg)
+               return z;
+         error(f, VORBIS_invalid_stream);
+      }
+   }
+   return z;
+}
+
+static int codebook_decode(vorb *f, Codebook *c, float *output, int len)
+{
+   int i,z = codebook_decode_start(f,c);
+   if (z < 0) return FALSE;
+   if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+   if (c->lookup_type == 1) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      int div = 1;
+      for (i=0; i < len; ++i) {
+         int off = (z / div) % c->lookup_values;
+         float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+         output[i] += val;
+         if (c->sequence_p) last = val + c->minimum_value;
+         div *= c->lookup_values;
+      }
+      return TRUE;
+   }
+#endif
+
+   z *= c->dimensions;
+   if (c->sequence_p) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      for (i=0; i < len; ++i) {
+         float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+         output[i] += val;
+         last = val + c->minimum_value;
+      }
+   } else {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      for (i=0; i < len; ++i) {
+         output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+      }
+   }
+
+   return TRUE;
+}
+
+static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step)
+{
+   int i,z = codebook_decode_start(f,c);
+   float last = CODEBOOK_ELEMENT_BASE(c);
+   if (z < 0) return FALSE;
+   if (len > c->dimensions) len = c->dimensions;
+
+#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+   if (c->lookup_type == 1) {
+      int div = 1;
+      for (i=0; i < len; ++i) {
+         int off = (z / div) % c->lookup_values;
+         float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+         output[i*step] += val;
+         if (c->sequence_p) last = val;
+         div *= c->lookup_values;
+      }
+      return TRUE;
+   }
+#endif
+
+   z *= c->dimensions;
+   for (i=0; i < len; ++i) {
+      float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+      output[i*step] += val;
+      if (c->sequence_p) last = val;
+   }
+
+   return TRUE;
+}
+
+static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode)
+{
+   int c_inter = *c_inter_p;
+   int p_inter = *p_inter_p;
+   int i,z, effective = c->dimensions;
+
+   // type 0 is only legal in a scalar context
+   if (c->lookup_type == 0)   return error(f, VORBIS_invalid_stream);
+
+   while (total_decode > 0) {
+      float last = CODEBOOK_ELEMENT_BASE(c);
+      DECODE_VQ(z,f,c);
+      #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+      assert(!c->sparse || z < c->sorted_entries);
+      #endif
+      if (z < 0) {
+         if (!f->bytes_in_seg)
+            if (f->last_seg) return FALSE;
+         return error(f, VORBIS_invalid_stream);
+      }
+
+      // if this will take us off the end of the buffers, stop short!
+      // we check by computing the length of the virtual interleaved
+      // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter),
+      // and the length we'll be using (effective)
+      if (c_inter + p_inter*ch + effective > len * ch) {
+         effective = len*ch - (p_inter*ch - c_inter);
+      }
+
+   #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+      if (c->lookup_type == 1) {
+         int div = 1;
+         for (i=0; i < effective; ++i) {
+            int off = (z / div) % c->lookup_values;
+            float val = CODEBOOK_ELEMENT_FAST(c,off) + last;
+            if (outputs[c_inter])
+               outputs[c_inter][p_inter] += val;
+            if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+            if (c->sequence_p) last = val;
+            div *= c->lookup_values;
+         }
+      } else
+   #endif
+      {
+         z *= c->dimensions;
+         if (c->sequence_p) {
+            for (i=0; i < effective; ++i) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               if (outputs[c_inter])
+                  outputs[c_inter][p_inter] += val;
+               if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+               last = val;
+            }
+         } else {
+            for (i=0; i < effective; ++i) {
+               float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last;
+               if (outputs[c_inter])
+                  outputs[c_inter][p_inter] += val;
+               if (++c_inter == ch) { c_inter = 0; ++p_inter; }
+            }
+         }
+      }
+
+      total_decode -= effective;
+   }
+   *c_inter_p = c_inter;
+   *p_inter_p = p_inter;
+   return TRUE;
+}
+
+static int predict_point(int x, int x0, int x1, int y0, int y1)
+{
+   int dy = y1 - y0;
+   int adx = x1 - x0;
+   // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86?
+   int err = abs(dy) * (x - x0);
+   int off = err / adx;
+   return dy < 0 ? y0 - off : y0 + off;
+}
+
+// the following table is block-copied from the specification
+static float inverse_db_table[256] =
+{
+  1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, 
+  1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, 
+  1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, 
+  2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, 
+  2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, 
+  3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, 
+  4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, 
+  6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, 
+  7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, 
+  1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, 
+  1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, 
+  1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, 
+  2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, 
+  2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, 
+  3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, 
+  4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, 
+  5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, 
+  7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, 
+  9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, 
+  1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, 
+  1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, 
+  2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, 
+  2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, 
+  3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, 
+  4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, 
+  5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, 
+  7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, 
+  9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, 
+  0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, 
+  0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, 
+  0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, 
+  0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, 
+  0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, 
+  0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, 
+  0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, 
+  0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, 
+  0.00092223983f, 0.00098217216f, 0.0010459992f,  0.0011139742f, 
+  0.0011863665f,  0.0012634633f,  0.0013455702f,  0.0014330129f, 
+  0.0015261382f,  0.0016253153f,  0.0017309374f,  0.0018434235f, 
+  0.0019632195f,  0.0020908006f,  0.0022266726f,  0.0023713743f, 
+  0.0025254795f,  0.0026895994f,  0.0028643847f,  0.0030505286f, 
+  0.0032487691f,  0.0034598925f,  0.0036847358f,  0.0039241906f, 
+  0.0041792066f,  0.0044507950f,  0.0047400328f,  0.0050480668f, 
+  0.0053761186f,  0.0057254891f,  0.0060975636f,  0.0064938176f, 
+  0.0069158225f,  0.0073652516f,  0.0078438871f,  0.0083536271f, 
+  0.0088964928f,  0.009474637f,   0.010090352f,   0.010746080f, 
+  0.011444421f,   0.012188144f,   0.012980198f,   0.013823725f, 
+  0.014722068f,   0.015678791f,   0.016697687f,   0.017782797f, 
+  0.018938423f,   0.020169149f,   0.021479854f,   0.022875735f, 
+  0.024362330f,   0.025945531f,   0.027631618f,   0.029427276f, 
+  0.031339626f,   0.033376252f,   0.035545228f,   0.037855157f, 
+  0.040315199f,   0.042935108f,   0.045725273f,   0.048696758f, 
+  0.051861348f,   0.055231591f,   0.058820850f,   0.062643361f, 
+  0.066714279f,   0.071049749f,   0.075666962f,   0.080584227f, 
+  0.085821044f,   0.091398179f,   0.097337747f,   0.10366330f, 
+  0.11039993f,    0.11757434f,    0.12521498f,    0.13335215f, 
+  0.14201813f,    0.15124727f,    0.16107617f,    0.17154380f, 
+  0.18269168f,    0.19456402f,    0.20720788f,    0.22067342f, 
+  0.23501402f,    0.25028656f,    0.26655159f,    0.28387361f, 
+  0.30232132f,    0.32196786f,    0.34289114f,    0.36517414f, 
+  0.38890521f,    0.41417847f,    0.44109412f,    0.46975890f, 
+  0.50028648f,    0.53279791f,    0.56742212f,    0.60429640f, 
+  0.64356699f,    0.68538959f,    0.72993007f,    0.77736504f, 
+  0.82788260f,    0.88168307f,    0.9389798f,     1.0f
+};
+
+
+// @OPTIMIZE: if you want to replace this bresenham line-drawing routine,
+// note that you must produce bit-identical output to decode correctly;
+// this specific sequence of operations is specified in the spec (it's
+// drawing integer-quantized frequency-space lines that the encoder
+// expects to be exactly the same)
+//     ... also, isn't the whole point of Bresenham's algorithm to NOT
+// have to divide in the setup? sigh.
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+#define LINE_OP(a,b)   a *= b
+#else
+#define LINE_OP(a,b)   a = b
+#endif
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+#define DIVTAB_NUMER   32
+#define DIVTAB_DENOM   64
+int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB
+#endif
+
+static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n)
+{
+   int dy = y1 - y0;
+   int adx = x1 - x0;
+   int ady = abs(dy);
+   int base;
+   int x=x0,y=y0;
+   int err = 0;
+   int sy;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+   if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) {
+      if (dy < 0) {
+         base = -integer_divide_table[ady][adx];
+         sy = base-1;
+      } else {
+         base =  integer_divide_table[ady][adx];
+         sy = base+1;
+      }
+   } else {
+      base = dy / adx;
+      if (dy < 0)
+         sy = base - 1;
+      else
+         sy = base+1;
+   }
+#else
+   base = dy / adx;
+   if (dy < 0)
+      sy = base - 1;
+   else
+      sy = base+1;
+#endif
+   ady -= abs(base) * adx;
+   if (x1 > n) x1 = n;
+   if (x < x1) {
+      LINE_OP(output[x], inverse_db_table[y]);
+      for (++x; x < x1; ++x) {
+         err += ady;
+         if (err >= adx) {
+            err -= adx;
+            y += sy;
+         } else
+            y += base;
+         LINE_OP(output[x], inverse_db_table[y]);
+      }
+   }
+}
+
+static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype)
+{
+   int k;
+   if (rtype == 0) {
+      int step = n / book->dimensions;
+      for (k=0; k < step; ++k)
+         if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step))
+            return FALSE;
+   } else {
+      for (k=0; k < n; ) {
+         if (!codebook_decode(f, book, target+offset, n-k))
+            return FALSE;
+         k += book->dimensions;
+         offset += book->dimensions;
+      }
+   }
+   return TRUE;
+}
+
+static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode)
+{
+   int i,j,pass;
+   Residue *r = f->residue_config + rn;
+   int rtype = f->residue_types[rn];
+   int c = r->classbook;
+   int classwords = f->codebooks[c].dimensions;
+   int n_read = r->end - r->begin;
+   int part_read = n_read / r->part_size;
+   int temp_alloc_point = temp_alloc_save(f);
+   #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+   uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata));
+   #else
+   int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications));
+   #endif
+
+   CHECK(f);
+
+   for (i=0; i < ch; ++i)
+      if (!do_not_decode[i])
+         memset(residue_buffers[i], 0, sizeof(float) * n);
+
+   if (rtype == 2 && ch != 1) {
+      for (j=0; j < ch; ++j)
+         if (!do_not_decode[j])
+            break;
+      if (j == ch)
+         goto done;
+
+      for (pass=0; pass < 8; ++pass) {
+         int pcount = 0, class_set = 0;
+         if (ch == 2) {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = (z & 1), p_inter = z>>1;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                     #else
+                     // saves 1%
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                     #endif
+                  } else {
+                     z += r->part_size;
+                     c_inter = z & 1;
+                     p_inter = z >> 1;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         } else if (ch == 1) {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = 0, p_inter = z;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                  } else {
+                     z += r->part_size;
+                     c_inter = 0;
+                     p_inter = z;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         } else {
+            while (pcount < part_read) {
+               int z = r->begin + pcount*r->part_size;
+               int c_inter = z % ch, p_inter = z/ch;
+               if (pass == 0) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int q;
+                  DECODE(q,f,c);
+                  if (q == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[0][class_set] = r->classdata[q];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[0][i+pcount] = q % r->classifications;
+                     q /= r->classifications;
+                  }
+                  #endif
+               }
+               for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+                  int z = r->begin + pcount*r->part_size;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[0][class_set][i];
+                  #else
+                  int c = classifications[0][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     Codebook *book = f->codebooks + b;
+                     if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size))
+                        goto done;
+                  } else {
+                     z += r->part_size;
+                     c_inter = z % ch;
+                     p_inter = z / ch;
+                  }
+               }
+               #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+               ++class_set;
+               #endif
+            }
+         }
+      }
+      goto done;
+   }
+   CHECK(f);
+
+   for (pass=0; pass < 8; ++pass) {
+      int pcount = 0, class_set=0;
+      while (pcount < part_read) {
+         if (pass == 0) {
+            for (j=0; j < ch; ++j) {
+               if (!do_not_decode[j]) {
+                  Codebook *c = f->codebooks+r->classbook;
+                  int temp;
+                  DECODE(temp,f,c);
+                  if (temp == EOP) goto done;
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  part_classdata[j][class_set] = r->classdata[temp];
+                  #else
+                  for (i=classwords-1; i >= 0; --i) {
+                     classifications[j][i+pcount] = temp % r->classifications;
+                     temp /= r->classifications;
+                  }
+                  #endif
+               }
+            }
+         }
+         for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) {
+            for (j=0; j < ch; ++j) {
+               if (!do_not_decode[j]) {
+                  #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+                  int c = part_classdata[j][class_set][i];
+                  #else
+                  int c = classifications[j][pcount];
+                  #endif
+                  int b = r->residue_books[c][pass];
+                  if (b >= 0) {
+                     float *target = residue_buffers[j];
+                     int offset = r->begin + pcount * r->part_size;
+                     int n = r->part_size;
+                     Codebook *book = f->codebooks + b;
+                     if (!residue_decode(f, book, target, offset, n, rtype))
+                        goto done;
+                  }
+               }
+            }
+         }
+         #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+         ++class_set;
+         #endif
+      }
+   }
+  done:
+   CHECK(f);
+   #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+   temp_free(f,part_classdata);
+   #else
+   temp_free(f,classifications);
+   #endif
+   temp_alloc_restore(f,temp_alloc_point);
+}
+
+
+#if 0
+// slow way for debugging
+void inverse_mdct_slow(float *buffer, int n)
+{
+   int i,j;
+   int n2 = n >> 1;
+   float *x = (float *) malloc(sizeof(*x) * n2);
+   memcpy(x, buffer, sizeof(*x) * n2);
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n2; ++j)
+         // formula from paper:
+         //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+         // formula from wikipedia
+         //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+         // these are equivalent, except the formula from the paper inverts the multiplier!
+         // however, what actually works is NO MULTIPLIER!?!
+         //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5));
+         acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1));
+      buffer[i] = acc;
+   }
+   free(x);
+}
+#elif 0
+// same as above, but just barely able to run in real time on modern machines
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+   float mcos[16384];
+   int i,j;
+   int n2 = n >> 1, nmask = (n << 2) -1;
+   float *x = (float *) malloc(sizeof(*x) * n2);
+   memcpy(x, buffer, sizeof(*x) * n2);
+   for (i=0; i < 4*n; ++i)
+      mcos[i] = (float) cos(M_PI / 2 * i / n);
+
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n2; ++j)
+         acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask];
+      buffer[i] = acc;
+   }
+   free(x);
+}
+#elif 0
+// transform to use a slow dct-iv; this is STILL basically trivial,
+// but only requires half as many ops
+void dct_iv_slow(float *buffer, int n)
+{
+   float mcos[16384];
+   float x[2048];
+   int i,j;
+   int n2 = n >> 1, nmask = (n << 3) - 1;
+   memcpy(x, buffer, sizeof(*x) * n);
+   for (i=0; i < 8*n; ++i)
+      mcos[i] = (float) cos(M_PI / 4 * i / n);
+   for (i=0; i < n; ++i) {
+      float acc = 0;
+      for (j=0; j < n; ++j)
+         acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask];
+      buffer[i] = acc;
+   }
+}
+
+void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype)
+{
+   int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4;
+   float temp[4096];
+
+   memcpy(temp, buffer, n2 * sizeof(float));
+   dct_iv_slow(temp, n2);  // returns -c'-d, a-b'
+
+   for (i=0; i < n4  ; ++i) buffer[i] = temp[i+n4];            // a-b'
+   for (   ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1];   // b-a', c+d'
+   for (   ; i < n   ; ++i) buffer[i] = -temp[i - n3_4];       // c'+d
+}
+#endif
+
+#ifndef LIBVORBIS_MDCT
+#define LIBVORBIS_MDCT 0
+#endif
+
+#if LIBVORBIS_MDCT
+// directly call the vorbis MDCT using an interface documented
+// by Jeff Roberts... useful for performance comparison
+typedef struct 
+{
+  int n;
+  int log2n;
+  
+  float *trig;
+  int   *bitrev;
+
+  float scale;
+} mdct_lookup;
+
+extern void mdct_init(mdct_lookup *lookup, int n);
+extern void mdct_clear(mdct_lookup *l);
+extern void mdct_backward(mdct_lookup *init, float *in, float *out);
+
+mdct_lookup M1,M2;
+
+void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+   mdct_lookup *M;
+   if (M1.n == n) M = &M1;
+   else if (M2.n == n) M = &M2;
+   else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; }
+   else { 
+      if (M2.n) __asm int 3;
+      mdct_init(&M2, n);
+      M = &M2;
+   }
+
+   mdct_backward(M, buffer, buffer);
+}
+#endif
+
+
+// the following were split out into separate functions while optimizing;
+// they could be pushed back up but eh. __forceinline showed no change;
+// they're probably already being inlined.
+static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A)
+{
+   float *ee0 = e + i_off;
+   float *ee2 = ee0 + k_off;
+   int i;
+
+   assert((n & 3) == 0);
+   for (i=(n>>2); i > 0; --i) {
+      float k00_20, k01_21;
+      k00_20  = ee0[ 0] - ee2[ 0];
+      k01_21  = ee0[-1] - ee2[-1];
+      ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0];
+      ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1];
+      ee2[ 0] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-1] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-2] - ee2[-2];
+      k01_21  = ee0[-3] - ee2[-3];
+      ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2];
+      ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3];
+      ee2[-2] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-3] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-4] - ee2[-4];
+      k01_21  = ee0[-5] - ee2[-5];
+      ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4];
+      ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5];
+      ee2[-4] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-5] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+
+      k00_20  = ee0[-6] - ee2[-6];
+      k01_21  = ee0[-7] - ee2[-7];
+      ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6];
+      ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7];
+      ee2[-6] = k00_20 * A[0] - k01_21 * A[1];
+      ee2[-7] = k01_21 * A[0] + k00_20 * A[1];
+      A += 8;
+      ee0 -= 8;
+      ee2 -= 8;
+   }
+}
+
+static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1)
+{
+   int i;
+   float k00_20, k01_21;
+
+   float *e0 = e + d0;
+   float *e2 = e0 + k_off;
+
+   for (i=lim >> 2; i > 0; --i) {
+      k00_20 = e0[-0] - e2[-0];
+      k01_21 = e0[-1] - e2[-1];
+      e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0];
+      e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1];
+      e2[-0] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-1] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-2] - e2[-2];
+      k01_21 = e0[-3] - e2[-3];
+      e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2];
+      e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3];
+      e2[-2] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-3] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-4] - e2[-4];
+      k01_21 = e0[-5] - e2[-5];
+      e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4];
+      e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5];
+      e2[-4] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-5] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      A += k1;
+
+      k00_20 = e0[-6] - e2[-6];
+      k01_21 = e0[-7] - e2[-7];
+      e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6];
+      e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7];
+      e2[-6] = (k00_20)*A[0] - (k01_21) * A[1];
+      e2[-7] = (k01_21)*A[0] + (k00_20) * A[1];
+
+      e0 -= 8;
+      e2 -= 8;
+
+      A += k1;
+   }
+}
+
+static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0)
+{
+   int i;
+   float A0 = A[0];
+   float A1 = A[0+1];
+   float A2 = A[0+a_off];
+   float A3 = A[0+a_off+1];
+   float A4 = A[0+a_off*2+0];
+   float A5 = A[0+a_off*2+1];
+   float A6 = A[0+a_off*3+0];
+   float A7 = A[0+a_off*3+1];
+
+   float k00,k11;
+
+   float *ee0 = e  +i_off;
+   float *ee2 = ee0+k_off;
+
+   for (i=n; i > 0; --i) {
+      k00     = ee0[ 0] - ee2[ 0];
+      k11     = ee0[-1] - ee2[-1];
+      ee0[ 0] =  ee0[ 0] + ee2[ 0];
+      ee0[-1] =  ee0[-1] + ee2[-1];
+      ee2[ 0] = (k00) * A0 - (k11) * A1;
+      ee2[-1] = (k11) * A0 + (k00) * A1;
+
+      k00     = ee0[-2] - ee2[-2];
+      k11     = ee0[-3] - ee2[-3];
+      ee0[-2] =  ee0[-2] + ee2[-2];
+      ee0[-3] =  ee0[-3] + ee2[-3];
+      ee2[-2] = (k00) * A2 - (k11) * A3;
+      ee2[-3] = (k11) * A2 + (k00) * A3;
+
+      k00     = ee0[-4] - ee2[-4];
+      k11     = ee0[-5] - ee2[-5];
+      ee0[-4] =  ee0[-4] + ee2[-4];
+      ee0[-5] =  ee0[-5] + ee2[-5];
+      ee2[-4] = (k00) * A4 - (k11) * A5;
+      ee2[-5] = (k11) * A4 + (k00) * A5;
+
+      k00     = ee0[-6] - ee2[-6];
+      k11     = ee0[-7] - ee2[-7];
+      ee0[-6] =  ee0[-6] + ee2[-6];
+      ee0[-7] =  ee0[-7] + ee2[-7];
+      ee2[-6] = (k00) * A6 - (k11) * A7;
+      ee2[-7] = (k11) * A6 + (k00) * A7;
+
+      ee0 -= k0;
+      ee2 -= k0;
+   }
+}
+
+static __forceinline void iter_54(float *z)
+{
+   float k00,k11,k22,k33;
+   float y0,y1,y2,y3;
+
+   k00  = z[ 0] - z[-4];
+   y0   = z[ 0] + z[-4];
+   y2   = z[-2] + z[-6];
+   k22  = z[-2] - z[-6];
+
+   z[-0] = y0 + y2;      // z0 + z4 + z2 + z6
+   z[-2] = y0 - y2;      // z0 + z4 - z2 - z6
+
+   // done with y0,y2
+
+   k33  = z[-3] - z[-7];
+
+   z[-4] = k00 + k33;    // z0 - z4 + z3 - z7
+   z[-6] = k00 - k33;    // z0 - z4 - z3 + z7
+
+   // done with k33
+
+   k11  = z[-1] - z[-5];
+   y1   = z[-1] + z[-5];
+   y3   = z[-3] + z[-7];
+
+   z[-1] = y1 + y3;      // z1 + z5 + z3 + z7
+   z[-3] = y1 - y3;      // z1 + z5 - z3 - z7
+   z[-5] = k11 - k22;    // z1 - z5 + z2 - z6
+   z[-7] = k11 + k22;    // z1 - z5 - z2 + z6
+}
+
+static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n)
+{
+   int a_off = base_n >> 3;
+   float A2 = A[0+a_off];
+   float *z = e + i_off;
+   float *base = z - 16 * n;
+
+   while (z > base) {
+      float k00,k11;
+
+      k00   = z[-0] - z[-8];
+      k11   = z[-1] - z[-9];
+      z[-0] = z[-0] + z[-8];
+      z[-1] = z[-1] + z[-9];
+      z[-8] =  k00;
+      z[-9] =  k11 ;
+
+      k00    = z[ -2] - z[-10];
+      k11    = z[ -3] - z[-11];
+      z[ -2] = z[ -2] + z[-10];
+      z[ -3] = z[ -3] + z[-11];
+      z[-10] = (k00+k11) * A2;
+      z[-11] = (k11-k00) * A2;
+
+      k00    = z[-12] - z[ -4];  // reverse to avoid a unary negation
+      k11    = z[ -5] - z[-13];
+      z[ -4] = z[ -4] + z[-12];
+      z[ -5] = z[ -5] + z[-13];
+      z[-12] = k11;
+      z[-13] = k00;
+
+      k00    = z[-14] - z[ -6];  // reverse to avoid a unary negation
+      k11    = z[ -7] - z[-15];
+      z[ -6] = z[ -6] + z[-14];
+      z[ -7] = z[ -7] + z[-15];
+      z[-14] = (k00+k11) * A2;
+      z[-15] = (k00-k11) * A2;
+
+      iter_54(z);
+      iter_54(z-8);
+      z -= 16;
+   }
+}
+
+static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype)
+{
+   int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+   int ld;
+   // @OPTIMIZE: reduce register pressure by using fewer variables?
+   int save_point = temp_alloc_save(f);
+   float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2));
+   float *u=NULL,*v=NULL;
+   // twiddle factors
+   float *A = f->A[blocktype];
+
+   // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+   // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function.
+
+   // kernel from paper
+
+
+   // merged:
+   //   copy and reflect spectral data
+   //   step 0
+
+   // note that it turns out that the items added together during
+   // this step are, in fact, being added to themselves (as reflected
+   // by step 0). inexplicable inefficiency! this became obvious
+   // once I combined the passes.
+
+   // so there's a missing 'times 2' here (for adding X to itself).
+   // this propogates through linearly to the end, where the numbers
+   // are 1/2 too small, and need to be compensated for.
+
+   {
+      float *d,*e, *AA, *e_stop;
+      d = &buf2[n2-2];
+      AA = A;
+      e = &buffer[0];
+      e_stop = &buffer[n2];
+      while (e != e_stop) {
+         d[1] = (e[0] * AA[0] - e[2]*AA[1]);
+         d[0] = (e[0] * AA[1] + e[2]*AA[0]);
+         d -= 2;
+         AA += 2;
+         e += 4;
+      }
+
+      e = &buffer[n2-3];
+      while (d >= buf2) {
+         d[1] = (-e[2] * AA[0] - -e[0]*AA[1]);
+         d[0] = (-e[2] * AA[1] + -e[0]*AA[0]);
+         d -= 2;
+         AA += 2;
+         e -= 4;
+      }
+   }
+
+   // now we use symbolic names for these, so that we can
+   // possibly swap their meaning as we change which operations
+   // are in place
+
+   u = buffer;
+   v = buf2;
+
+   // step 2    (paper output is w, now u)
+   // this could be in place, but the data ends up in the wrong
+   // place... _somebody_'s got to swap it, so this is nominated
+   {
+      float *AA = &A[n2-8];
+      float *d0,*d1, *e0, *e1;
+
+      e0 = &v[n4];
+      e1 = &v[0];
+
+      d0 = &u[n4];
+      d1 = &u[0];
+
+      while (AA >= A) {
+         float v40_20, v41_21;
+
+         v41_21 = e0[1] - e1[1];
+         v40_20 = e0[0] - e1[0];
+         d0[1]  = e0[1] + e1[1];
+         d0[0]  = e0[0] + e1[0];
+         d1[1]  = v41_21*AA[4] - v40_20*AA[5];
+         d1[0]  = v40_20*AA[4] + v41_21*AA[5];
+
+         v41_21 = e0[3] - e1[3];
+         v40_20 = e0[2] - e1[2];
+         d0[3]  = e0[3] + e1[3];
+         d0[2]  = e0[2] + e1[2];
+         d1[3]  = v41_21*AA[0] - v40_20*AA[1];
+         d1[2]  = v40_20*AA[0] + v41_21*AA[1];
+
+         AA -= 8;
+
+         d0 += 4;
+         d1 += 4;
+         e0 += 4;
+         e1 += 4;
+      }
+   }
+
+   // step 3
+   ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+
+   // optimized step 3:
+
+   // the original step3 loop can be nested r inside s or s inside r;
+   // it's written originally as s inside r, but this is dumb when r
+   // iterates many times, and s few. So I have two copies of it and
+   // switch between them halfway.
+
+   // this is iteration 0 of step 3
+   imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A);
+   imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A);
+
+   // this is iteration 1 of step 3
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16);
+   imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16);
+
+   l=2;
+   for (; l < (ld-3)>>1; ++l) {
+      int k0 = n >> (l+2), k0_2 = k0>>1;
+      int lim = 1 << (l+1);
+      int i;
+      for (i=0; i < lim; ++i)
+         imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3));
+   }
+
+   for (; l < ld-6; ++l) {
+      int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1;
+      int rlim = n >> (l+6), r;
+      int lim = 1 << (l+1);
+      int i_off;
+      float *A0 = A;
+      i_off = n2-1;
+      for (r=rlim; r > 0; --r) {
+         imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0);
+         A0 += k1*4;
+         i_off -= 8;
+      }
+   }
+
+   // iterations with count:
+   //   ld-6,-5,-4 all interleaved together
+   //       the big win comes from getting rid of needless flops
+   //         due to the constants on pass 5 & 4 being all 1 and 0;
+   //       combining them to be simultaneous to improve cache made little difference
+   imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n);
+
+   // output is u
+
+   // step 4, 5, and 6
+   // cannot be in-place because of step 5
+   {
+      uint16 *bitrev = f->bit_reverse[blocktype];
+      // weirdly, I'd have thought reading sequentially and writing
+      // erratically would have been better than vice-versa, but in
+      // fact that's not what my testing showed. (That is, with
+      // j = bitreverse(i), do you read i and write j, or read j and write i.)
+
+      float *d0 = &v[n4-4];
+      float *d1 = &v[n2-4];
+      while (d0 >= v) {
+         int k4;
+
+         k4 = bitrev[0];
+         d1[3] = u[k4+0];
+         d1[2] = u[k4+1];
+         d0[3] = u[k4+2];
+         d0[2] = u[k4+3];
+
+         k4 = bitrev[1];
+         d1[1] = u[k4+0];
+         d1[0] = u[k4+1];
+         d0[1] = u[k4+2];
+         d0[0] = u[k4+3];
+         
+         d0 -= 4;
+         d1 -= 4;
+         bitrev += 2;
+      }
+   }
+   // (paper output is u, now v)
+
+
+   // data must be in buf2
+   assert(v == buf2);
+
+   // step 7   (paper output is v, now v)
+   // this is now in place
+   {
+      float *C = f->C[blocktype];
+      float *d, *e;
+
+      d = v;
+      e = v + n2 - 4;
+
+      while (d < e) {
+         float a02,a11,b0,b1,b2,b3;
+
+         a02 = d[0] - e[2];
+         a11 = d[1] + e[3];
+
+         b0 = C[1]*a02 + C[0]*a11;
+         b1 = C[1]*a11 - C[0]*a02;
+
+         b2 = d[0] + e[ 2];
+         b3 = d[1] - e[ 3];
+
+         d[0] = b2 + b0;
+         d[1] = b3 + b1;
+         e[2] = b2 - b0;
+         e[3] = b1 - b3;
+
+         a02 = d[2] - e[0];
+         a11 = d[3] + e[1];
+
+         b0 = C[3]*a02 + C[2]*a11;
+         b1 = C[3]*a11 - C[2]*a02;
+
+         b2 = d[2] + e[ 0];
+         b3 = d[3] - e[ 1];
+
+         d[2] = b2 + b0;
+         d[3] = b3 + b1;
+         e[0] = b2 - b0;
+         e[1] = b1 - b3;
+
+         C += 4;
+         d += 4;
+         e -= 4;
+      }
+   }
+
+   // data must be in buf2
+
+
+   // step 8+decode   (paper output is X, now buffer)
+   // this generates pairs of data a la 8 and pushes them directly through
+   // the decode kernel (pushing rather than pulling) to avoid having
+   // to make another pass later
+
+   // this cannot POSSIBLY be in place, so we refer to the buffers directly
+
+   {
+      float *d0,*d1,*d2,*d3;
+
+      float *B = f->B[blocktype] + n2 - 8;
+      float *e = buf2 + n2 - 8;
+      d0 = &buffer[0];
+      d1 = &buffer[n2-4];
+      d2 = &buffer[n2];
+      d3 = &buffer[n-4];
+      while (e >= v) {
+         float p0,p1,p2,p3;
+
+         p3 =  e[6]*B[7] - e[7]*B[6];
+         p2 = -e[6]*B[6] - e[7]*B[7]; 
+
+         d0[0] =   p3;
+         d1[3] = - p3;
+         d2[0] =   p2;
+         d3[3] =   p2;
+
+         p1 =  e[4]*B[5] - e[5]*B[4];
+         p0 = -e[4]*B[4] - e[5]*B[5]; 
+
+         d0[1] =   p1;
+         d1[2] = - p1;
+         d2[1] =   p0;
+         d3[2] =   p0;
+
+         p3 =  e[2]*B[3] - e[3]*B[2];
+         p2 = -e[2]*B[2] - e[3]*B[3]; 
+
+         d0[2] =   p3;
+         d1[1] = - p3;
+         d2[2] =   p2;
+         d3[1] =   p2;
+
+         p1 =  e[0]*B[1] - e[1]*B[0];
+         p0 = -e[0]*B[0] - e[1]*B[1]; 
+
+         d0[3] =   p1;
+         d1[0] = - p1;
+         d2[3] =   p0;
+         d3[0] =   p0;
+
+         B -= 8;
+         e -= 8;
+         d0 += 4;
+         d2 += 4;
+         d1 -= 4;
+         d3 -= 4;
+      }
+   }
+
+   temp_free(f,buf2);
+   temp_alloc_restore(f,save_point);
+}
+
+#if 0
+// this is the original version of the above code, if you want to optimize it from scratch
+void inverse_mdct_naive(float *buffer, int n)
+{
+   float s;
+   float A[1 << 12], B[1 << 12], C[1 << 11];
+   int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l;
+   int n3_4 = n - n4, ld;
+   // how can they claim this only uses N words?!
+   // oh, because they're only used sparsely, whoops
+   float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13];
+   // set up twiddle factors
+
+   for (k=k2=0; k < n4; ++k,k2+=2) {
+      A[k2  ] = (float)  cos(4*k*M_PI/n);
+      A[k2+1] = (float) -sin(4*k*M_PI/n);
+      B[k2  ] = (float)  cos((k2+1)*M_PI/n/2);
+      B[k2+1] = (float)  sin((k2+1)*M_PI/n/2);
+   }
+   for (k=k2=0; k < n8; ++k,k2+=2) {
+      C[k2  ] = (float)  cos(2*(k2+1)*M_PI/n);
+      C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n);
+   }
+
+   // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio"
+   // Note there are bugs in that pseudocode, presumably due to them attempting
+   // to rename the arrays nicely rather than representing the way their actual
+   // implementation bounces buffers back and forth. As a result, even in the
+   // "some formulars corrected" version, a direct implementation fails. These
+   // are noted below as "paper bug".
+
+   // copy and reflect spectral data
+   for (k=0; k < n2; ++k) u[k] = buffer[k];
+   for (   ; k < n ; ++k) u[k] = -buffer[n - k - 1];
+   // kernel from paper
+   // step 1
+   for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) {
+      v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2]   - (u[k4+2] - u[n-k4-3])*A[k2+1];
+      v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2];
+   }
+   // step 2
+   for (k=k4=0; k < n8; k+=1, k4+=4) {
+      w[n2+3+k4] = v[n2+3+k4] + v[k4+3];
+      w[n2+1+k4] = v[n2+1+k4] + v[k4+1];
+      w[k4+3]    = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4];
+      w[k4+1]    = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4];
+   }
+   // step 3
+   ld = ilog(n) - 1; // ilog is off-by-one from normal definitions
+   for (l=0; l < ld-3; ++l) {
+      int k0 = n >> (l+2), k1 = 1 << (l+3);
+      int rlim = n >> (l+4), r4, r;
+      int s2lim = 1 << (l+2), s2;
+      for (r=r4=0; r < rlim; r4+=4,++r) {
+         for (s2=0; s2 < s2lim; s2+=2) {
+            u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4];
+            u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4];
+            u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1]
+                                - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1];
+            u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1]
+                                + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1];
+         }
+      }
+      if (l+1 < ld-3) {
+         // paper bug: ping-ponging of u&w here is omitted
+         memcpy(w, u, sizeof(u));
+      }
+   }
+
+   // step 4
+   for (i=0; i < n8; ++i) {
+      int j = bit_reverse(i) >> (32-ld+3);
+      assert(j < n8);
+      if (i == j) {
+         // paper bug: original code probably swapped in place; if copying,
+         //            need to directly copy in this case
+         int i8 = i << 3;
+         v[i8+1] = u[i8+1];
+         v[i8+3] = u[i8+3];
+         v[i8+5] = u[i8+5];
+         v[i8+7] = u[i8+7];
+      } else if (i < j) {
+         int i8 = i << 3, j8 = j << 3;
+         v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1];
+         v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3];
+         v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5];
+         v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7];
+      }
+   }
+   // step 5
+   for (k=0; k < n2; ++k) {
+      w[k] = v[k*2+1];
+   }
+   // step 6
+   for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) {
+      u[n-1-k2] = w[k4];
+      u[n-2-k2] = w[k4+1];
+      u[n3_4 - 1 - k2] = w[k4+2];
+      u[n3_4 - 2 - k2] = w[k4+3];
+   }
+   // step 7
+   for (k=k2=0; k < n8; ++k, k2 += 2) {
+      v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+      v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2;
+      v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+      v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2;
+   }
+   // step 8
+   for (k=k2=0; k < n4; ++k,k2 += 2) {
+      X[k]      = v[k2+n2]*B[k2  ] + v[k2+1+n2]*B[k2+1];
+      X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2  ];
+   }
+
+   // decode kernel to output
+   // determined the following value experimentally
+   // (by first figuring out what made inverse_mdct_slow work); then matching that here
+   // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?)
+   s = 0.5; // theoretically would be n4
+
+   // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code,
+   //     so it needs to use the "old" B values to behave correctly, or else
+   //     set s to 1.0 ]]]
+   for (i=0; i < n4  ; ++i) buffer[i] = s * X[i+n4];
+   for (   ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1];
+   for (   ; i < n   ; ++i) buffer[i] = -s * X[i - n3_4];
+}
+#endif
+
+static float *get_window(vorb *f, int len)
+{
+   len <<= 1;
+   if (len == f->blocksize_0) return f->window[0];
+   if (len == f->blocksize_1) return f->window[1];
+   assert(0);
+   return NULL;
+}
+
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+typedef int16 YTYPE;
+#else
+typedef int YTYPE;
+#endif
+static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag)
+{
+   int n2 = n >> 1;
+   int s = map->chan[i].mux, floor;
+   floor = map->submap_floor[s];
+   if (f->floor_types[floor] == 0) {
+      return error(f, VORBIS_invalid_stream);
+   } else {
+      Floor1 *g = &f->floor_config[floor].floor1;
+      int j,q;
+      int lx = 0, ly = finalY[0] * g->floor1_multiplier;
+      for (q=1; q < g->values; ++q) {
+         j = g->sorted_order[q];
+         #ifndef STB_VORBIS_NO_DEFER_FLOOR
+         if (finalY[j] >= 0)
+         #else
+         if (step2_flag[j])
+         #endif
+         {
+            int hy = finalY[j] * g->floor1_multiplier;
+            int hx = g->Xlist[j];
+            if (lx != hx)
+               draw_line(target, lx,ly, hx,hy, n2);
+            CHECK(f);
+            lx = hx, ly = hy;
+         }
+      }
+      if (lx < n2) {
+         // optimization of: draw_line(target, lx,ly, n,ly, n2);
+         for (j=lx; j < n2; ++j)
+            LINE_OP(target[j], inverse_db_table[ly]);
+         CHECK(f);
+      }
+   }
+   return TRUE;
+}
+
+// The meaning of "left" and "right"
+//
+// For a given frame:
+//     we compute samples from 0..n
+//     window_center is n/2
+//     we'll window and mix the samples from left_start to left_end with data from the previous frame
+//     all of the samples from left_end to right_start can be output without mixing; however,
+//        this interval is 0-length except when transitioning between short and long frames
+//     all of the samples from right_start to right_end need to be mixed with the next frame,
+//        which we don't have, so those get saved in a buffer
+//     frame N's right_end-right_start, the number of samples to mix with the next frame,
+//        has to be the same as frame N+1's left_end-left_start (which they are by
+//        construction)
+
+static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+   Mode *m;
+   int i, n, prev, next, window_center;
+   f->channel_buffer_start = f->channel_buffer_end = 0;
+
+  retry:
+   if (f->eof) return FALSE;
+   if (!maybe_start_packet(f))
+      return FALSE;
+   // check packet type
+   if (get_bits(f,1) != 0) {
+      if (IS_PUSH_MODE(f))
+         return error(f,VORBIS_bad_packet_type);
+      while (EOP != get8_packet(f));
+      goto retry;
+   }
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+   i = get_bits(f, ilog(f->mode_count-1));
+   if (i == EOP) return FALSE;
+   if (i >= f->mode_count) return FALSE;
+   *mode = i;
+   m = f->mode_config + i;
+   if (m->blockflag) {
+      n = f->blocksize_1;
+      prev = get_bits(f,1);
+      next = get_bits(f,1);
+   } else {
+      prev = next = 0;
+      n = f->blocksize_0;
+   }
+
+// WINDOWING
+
+   window_center = n >> 1;
+   if (m->blockflag && !prev) {
+      *p_left_start = (n - f->blocksize_0) >> 2;
+      *p_left_end   = (n + f->blocksize_0) >> 2;
+   } else {
+      *p_left_start = 0;
+      *p_left_end   = window_center;
+   }
+   if (m->blockflag && !next) {
+      *p_right_start = (n*3 - f->blocksize_0) >> 2;
+      *p_right_end   = (n*3 + f->blocksize_0) >> 2;
+   } else {
+      *p_right_start = window_center;
+      *p_right_end   = n;
+   }
+
+   return TRUE;
+}
+
+static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left)
+{
+   Mapping *map;
+   int i,j,k,n,n2;
+   int zero_channel[256];
+   int really_zero_channel[256];
+
+// WINDOWING
+
+   n = f->blocksize[m->blockflag];
+   map = &f->mapping[m->mapping];
+
+// FLOORS
+   n2 = n >> 1;
+
+   CHECK(f);
+
+   for (i=0; i < f->channels; ++i) {
+      int s = map->chan[i].mux, floor;
+      zero_channel[i] = FALSE;
+      floor = map->submap_floor[s];
+      if (f->floor_types[floor] == 0) {
+         return error(f, VORBIS_invalid_stream);
+      } else {
+         Floor1 *g = &f->floor_config[floor].floor1;
+         if (get_bits(f, 1)) {
+            short *finalY;
+            uint8 step2_flag[256];
+            static int range_list[4] = { 256, 128, 86, 64 };
+            int range = range_list[g->floor1_multiplier-1];
+            int offset = 2;
+            finalY = f->finalY[i];
+            finalY[0] = get_bits(f, ilog(range)-1);
+            finalY[1] = get_bits(f, ilog(range)-1);
+            for (j=0; j < g->partitions; ++j) {
+               int pclass = g->partition_class_list[j];
+               int cdim = g->class_dimensions[pclass];
+               int cbits = g->class_subclasses[pclass];
+               int csub = (1 << cbits)-1;
+               int cval = 0;
+               if (cbits) {
+                  Codebook *c = f->codebooks + g->class_masterbooks[pclass];
+                  DECODE(cval,f,c);
+               }
+               for (k=0; k < cdim; ++k) {
+                  int book = g->subclass_books[pclass][cval & csub];
+                  cval = cval >> cbits;
+                  if (book >= 0) {
+                     int temp;
+                     Codebook *c = f->codebooks + book;
+                     DECODE(temp,f,c);
+                     finalY[offset++] = temp;
+                  } else
+                     finalY[offset++] = 0;
+               }
+            }
+            if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec
+            step2_flag[0] = step2_flag[1] = 1;
+            for (j=2; j < g->values; ++j) {
+               int low, high, pred, highroom, lowroom, room, val;
+               low = g->neighbors[j][0];
+               high = g->neighbors[j][1];
+               //neighbors(g->Xlist, j, &low, &high);
+               pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]);
+               val = finalY[j];
+               highroom = range - pred;
+               lowroom = pred;
+               if (highroom < lowroom)
+                  room = highroom * 2;
+               else
+                  room = lowroom * 2;
+               if (val) {
+                  step2_flag[low] = step2_flag[high] = 1;
+                  step2_flag[j] = 1;
+                  if (val >= room)
+                     if (highroom > lowroom)
+                        finalY[j] = val - lowroom + pred;
+                     else
+                        finalY[j] = pred - val + highroom - 1;
+                  else
+                     if (val & 1)
+                        finalY[j] = pred - ((val+1)>>1);
+                     else
+                        finalY[j] = pred + (val>>1);
+               } else {
+                  step2_flag[j] = 0;
+                  finalY[j] = pred;
+               }
+            }
+
+#ifdef STB_VORBIS_NO_DEFER_FLOOR
+            do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag);
+#else
+            // defer final floor computation until _after_ residue
+            for (j=0; j < g->values; ++j) {
+               if (!step2_flag[j])
+                  finalY[j] = -1;
+            }
+#endif
+         } else {
+           error:
+            zero_channel[i] = TRUE;
+         }
+         // So we just defer everything else to later
+
+         // at this point we've decoded the floor into buffer
+      }
+   }
+   CHECK(f);
+   // at this point we've decoded all floors
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+
+   // re-enable coupled channels if necessary
+   memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels);
+   for (i=0; i < map->coupling_steps; ++i)
+      if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) {
+         zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE;
+      }
+
+   CHECK(f);
+// RESIDUE DECODE
+   for (i=0; i < map->submaps; ++i) {
+      float *residue_buffers[STB_VORBIS_MAX_CHANNELS];
+      int r;
+      uint8 do_not_decode[256];
+      int ch = 0;
+      for (j=0; j < f->channels; ++j) {
+         if (map->chan[j].mux == i) {
+            if (zero_channel[j]) {
+               do_not_decode[ch] = TRUE;
+               residue_buffers[ch] = NULL;
+            } else {
+               do_not_decode[ch] = FALSE;
+               residue_buffers[ch] = f->channel_buffers[j];
+            }
+            ++ch;
+         }
+      }
+      r = map->submap_residue[i];
+      decode_residue(f, residue_buffers, ch, n2, r, do_not_decode);
+   }
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+   CHECK(f);
+
+// INVERSE COUPLING
+   for (i = map->coupling_steps-1; i >= 0; --i) {
+      int n2 = n >> 1;
+      float *m = f->channel_buffers[map->chan[i].magnitude];
+      float *a = f->channel_buffers[map->chan[i].angle    ];
+      for (j=0; j < n2; ++j) {
+         float a2,m2;
+         if (m[j] > 0)
+            if (a[j] > 0)
+               m2 = m[j], a2 = m[j] - a[j];
+            else
+               a2 = m[j], m2 = m[j] + a[j];
+         else
+            if (a[j] > 0)
+               m2 = m[j], a2 = m[j] + a[j];
+            else
+               a2 = m[j], m2 = m[j] - a[j];
+         m[j] = m2;
+         a[j] = a2;
+      }
+   }
+   CHECK(f);
+
+   // finish decoding the floors
+#ifndef STB_VORBIS_NO_DEFER_FLOOR
+   for (i=0; i < f->channels; ++i) {
+      if (really_zero_channel[i]) {
+         memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+      } else {
+         do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL);
+      }
+   }
+#else
+   for (i=0; i < f->channels; ++i) {
+      if (really_zero_channel[i]) {
+         memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2);
+      } else {
+         for (j=0; j < n2; ++j)
+            f->channel_buffers[i][j] *= f->floor_buffers[i][j];
+      }
+   }
+#endif
+
+// INVERSE MDCT
+   CHECK(f);
+   for (i=0; i < f->channels; ++i)
+      inverse_mdct(f->channel_buffers[i], n, f, m->blockflag);
+   CHECK(f);
+
+   // this shouldn't be necessary, unless we exited on an error
+   // and want to flush to get to the next packet
+   flush_packet(f);
+
+   if (f->first_decode) {
+      // assume we start so first non-discarded sample is sample 0
+      // this isn't to spec, but spec would require us to read ahead
+      // and decode the size of all current frames--could be done,
+      // but presumably it's not a commonly used feature
+      f->current_loc = -n2; // start of first frame is positioned for discard
+      // we might have to discard samples "from" the next frame too,
+      // if we're lapping a large block then a small at the start?
+      f->discard_samples_deferred = n - right_end;
+      f->current_loc_valid = TRUE;
+      f->first_decode = FALSE;
+   } else if (f->discard_samples_deferred) {
+      if (f->discard_samples_deferred >= right_start - left_start) {
+         f->discard_samples_deferred -= (right_start - left_start);
+         left_start = right_start;
+         *p_left = left_start;
+      } else {
+         left_start += f->discard_samples_deferred;
+         *p_left = left_start;
+         f->discard_samples_deferred = 0;
+      }
+   } else if (f->previous_length == 0 && f->current_loc_valid) {
+      // we're recovering from a seek... that means we're going to discard
+      // the samples from this packet even though we know our position from
+      // the last page header, so we need to update the position based on
+      // the discarded samples here
+      // but wait, the code below is going to add this in itself even
+      // on a discard, so we don't need to do it here...
+   }
+
+   // check if we have ogg information about the sample # for this packet
+   if (f->last_seg_which == f->end_seg_with_known_loc) {
+      // if we have a valid current loc, and this is final:
+      if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) {
+         uint32 current_end = f->known_loc_for_packet - (n-right_end);
+         // then let's infer the size of the (probably) short final frame
+         if (current_end < f->current_loc + (right_end-left_start)) {
+            if (current_end < f->current_loc) {
+               // negative truncation, that's impossible!
+               *len = 0;
+            } else {
+               *len = current_end - f->current_loc;
+            }
+            *len += left_start;
+            if (*len > right_end) *len = right_end; // this should never happen
+            f->current_loc += *len;
+            return TRUE;
+         }
+      }
+      // otherwise, just set our sample loc
+      // guess that the ogg granule pos refers to the _middle_ of the
+      // last frame?
+      // set f->current_loc to the position of left_start
+      f->current_loc = f->known_loc_for_packet - (n2-left_start);
+      f->current_loc_valid = TRUE;
+   }
+   if (f->current_loc_valid)
+      f->current_loc += (right_start - left_start);
+
+   if (f->alloc.alloc_buffer)
+      assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset);
+   *len = right_end;  // ignore samples after the window goes to 0
+   CHECK(f);
+
+   return TRUE;
+}
+
+static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right)
+{
+   int mode, left_end, right_end;
+   if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0;
+   return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left);
+}
+
+static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right)
+{
+   int prev,i,j;
+   // we use right&left (the start of the right- and left-window sin()-regions)
+   // to determine how much to return, rather than inferring from the rules
+   // (same result, clearer code); 'left' indicates where our sin() window
+   // starts, therefore where the previous window's right edge starts, and
+   // therefore where to start mixing from the previous buffer. 'right'
+   // indicates where our sin() ending-window starts, therefore that's where
+   // we start saving, and where our returned-data ends.
+
+   // mixin from previous window
+   if (f->previous_length) {
+      int i,j, n = f->previous_length;
+      float *w = get_window(f, n);
+      for (i=0; i < f->channels; ++i) {
+         for (j=0; j < n; ++j)
+            f->channel_buffers[i][left+j] =
+               f->channel_buffers[i][left+j]*w[    j] +
+               f->previous_window[i][     j]*w[n-1-j];
+      }
+   }
+
+   prev = f->previous_length;
+
+   // last half of this data becomes previous window
+   f->previous_length = len - right;
+
+   // @OPTIMIZE: could avoid this copy by double-buffering the
+   // output (flipping previous_window with channel_buffers), but
+   // then previous_window would have to be 2x as large, and
+   // channel_buffers couldn't be temp mem (although they're NOT
+   // currently temp mem, they could be (unless we want to level
+   // performance by spreading out the computation))
+   for (i=0; i < f->channels; ++i)
+      for (j=0; right+j < len; ++j)
+         f->previous_window[i][j] = f->channel_buffers[i][right+j];
+
+   if (!prev)
+      // there was no previous packet, so this data isn't valid...
+      // this isn't entirely true, only the would-have-overlapped data
+      // isn't valid, but this seems to be what the spec requires
+      return 0;
+
+   // truncate a short frame
+   if (len < right) right = len;
+
+   f->samples_output += right-left;
+
+   return right - left;
+}
+
+static void vorbis_pump_first_frame(stb_vorbis *f)
+{
+   int len, right, left;
+   if (vorbis_decode_packet(f, &len, &left, &right))
+      vorbis_finish_frame(f, len, left, right);
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+static int is_whole_packet_present(stb_vorbis *f, int end_page)
+{
+   // make sure that we have the packet available before continuing...
+   // this requires a full ogg parse, but we know we can fetch from f->stream
+
+   // instead of coding this out explicitly, we could save the current read state,
+   // read the next packet with get8() until end-of-packet, check f->eof, then
+   // reset the state? but that would be slower, esp. since we'd have over 256 bytes
+   // of state to restore (primarily the page segment table)
+
+   int s = f->next_seg, first = TRUE;
+   uint8 *p = f->stream;
+
+   if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag
+      for (; s < f->segment_count; ++s) {
+         p += f->segments[s];
+         if (f->segments[s] < 255)               // stop at first short segment
+            break;
+      }
+      // either this continues, or it ends it...
+      if (end_page)
+         if (s < f->segment_count-1)             return error(f, VORBIS_invalid_stream);
+      if (s == f->segment_count)
+         s = -1; // set 'crosses page' flag
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      first = FALSE;
+   }
+   for (; s == -1;) {
+      uint8 *q; 
+      int n;
+
+      // check that we have the page header ready
+      if (p + 26 >= f->stream_end)               return error(f, VORBIS_need_more_data);
+      // validate the page
+      if (memcmp(p, ogg_page_header, 4))         return error(f, VORBIS_invalid_stream);
+      if (p[4] != 0)                             return error(f, VORBIS_invalid_stream);
+      if (first) { // the first segment must NOT have 'continued_packet', later ones MUST
+         if (f->previous_length)
+            if ((p[5] & PAGEFLAG_continued_packet))  return error(f, VORBIS_invalid_stream);
+         // if no previous length, we're resynching, so we can come in on a continued-packet,
+         // which we'll just drop
+      } else {
+         if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream);
+      }
+      n = p[26]; // segment counts
+      q = p+27;  // q points to segment table
+      p = q + n; // advance past header
+      // make sure we've read the segment table
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      for (s=0; s < n; ++s) {
+         p += q[s];
+         if (q[s] < 255)
+            break;
+      }
+      if (end_page)
+         if (s < n-1)                            return error(f, VORBIS_invalid_stream);
+      if (s == n)
+         s = -1; // set 'crosses page' flag
+      if (p > f->stream_end)                     return error(f, VORBIS_need_more_data);
+      first = FALSE;
+   }
+   return TRUE;
+}
+#endif // !STB_VORBIS_NO_PUSHDATA_API
+
+static int start_decoder(vorb *f)
+{
+   uint8 header[6], x,y;
+   int len,i,j,k, max_submaps = 0;
+   int longest_floorlist=0;
+
+   // first page, first packet
+
+   if (!start_page(f))                              return FALSE;
+   // validate page flag
+   if (!(f->page_flag & PAGEFLAG_first_page))       return error(f, VORBIS_invalid_first_page);
+   if (f->page_flag & PAGEFLAG_last_page)           return error(f, VORBIS_invalid_first_page);
+   if (f->page_flag & PAGEFLAG_continued_packet)    return error(f, VORBIS_invalid_first_page);
+   // check for expected packet length
+   if (f->segment_count != 1)                       return error(f, VORBIS_invalid_first_page);
+   if (f->segments[0] != 30)                        return error(f, VORBIS_invalid_first_page);
+   // read packet
+   // check packet header
+   if (get8(f) != VORBIS_packet_id)                 return error(f, VORBIS_invalid_first_page);
+   if (!getn(f, header, 6))                         return error(f, VORBIS_unexpected_eof);
+   if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_first_page);
+   // vorbis_version
+   if (get32(f) != 0)                               return error(f, VORBIS_invalid_first_page);
+   f->channels = get8(f); if (!f->channels)         return error(f, VORBIS_invalid_first_page);
+   if (f->channels > STB_VORBIS_MAX_CHANNELS)       return error(f, VORBIS_too_many_channels);
+   f->sample_rate = get32(f); if (!f->sample_rate)  return error(f, VORBIS_invalid_first_page);
+   get32(f); // bitrate_maximum
+   get32(f); // bitrate_nominal
+   get32(f); // bitrate_minimum
+   x = get8(f);
+   {
+      int log0,log1;
+      log0 = x & 15;
+      log1 = x >> 4;
+      f->blocksize_0 = 1 << log0;
+      f->blocksize_1 = 1 << log1;
+      if (log0 < 6 || log0 > 13)                       return error(f, VORBIS_invalid_setup);
+      if (log1 < 6 || log1 > 13)                       return error(f, VORBIS_invalid_setup);
+      if (log0 > log1)                                 return error(f, VORBIS_invalid_setup);
+   }
+
+   // framing_flag
+   x = get8(f);
+   if (!(x & 1))                                    return error(f, VORBIS_invalid_first_page);
+
+   // second packet!
+   if (!start_page(f))                              return FALSE;
+
+   if (!start_packet(f))                            return FALSE;
+   do {
+      len = next_segment(f);
+      skip(f, len);
+      f->bytes_in_seg = 0;
+   } while (len);
+
+   // third packet!
+   if (!start_packet(f))                            return FALSE;
+
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (IS_PUSH_MODE(f)) {
+      if (!is_whole_packet_present(f, TRUE)) {
+         // convert error in ogg header to write type
+         if (f->error == VORBIS_invalid_stream)
+            f->error = VORBIS_invalid_setup;
+         return FALSE;
+      }
+   }
+   #endif
+
+   crc32_init(); // always init it, to avoid multithread race conditions
+
+   if (get8_packet(f) != VORBIS_packet_setup)       return error(f, VORBIS_invalid_setup);
+   for (i=0; i < 6; ++i) header[i] = get8_packet(f);
+   if (!vorbis_validate(header))                    return error(f, VORBIS_invalid_setup);
+
+   // codebooks
+
+   f->codebook_count = get_bits(f,8) + 1;
+   f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count);
+   if (f->codebooks == NULL)                        return error(f, VORBIS_outofmem);
+   memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count);
+   for (i=0; i < f->codebook_count; ++i) {
+      uint32 *values;
+      int ordered, sorted_count;
+      int total=0;
+      uint8 *lengths;
+      Codebook *c = f->codebooks+i;
+      CHECK(f);
+      x = get_bits(f, 8); if (x != 0x42)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8); if (x != 0x43)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8); if (x != 0x56)            return error(f, VORBIS_invalid_setup);
+      x = get_bits(f, 8);
+      c->dimensions = (get_bits(f, 8)<<8) + x;
+      x = get_bits(f, 8);
+      y = get_bits(f, 8);
+      c->entries = (get_bits(f, 8)<<16) + (y<<8) + x;
+      ordered = get_bits(f,1);
+      c->sparse = ordered ? 0 : get_bits(f,1);
+
+      if (c->dimensions == 0 && c->entries != 0)    return error(f, VORBIS_invalid_setup);
+
+      if (c->sparse)
+         lengths = (uint8 *) setup_temp_malloc(f, c->entries);
+      else
+         lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+
+      if (!lengths) return error(f, VORBIS_outofmem);
+
+      if (ordered) {
+         int current_entry = 0;
+         int current_length = get_bits(f,5) + 1;
+         while (current_entry < c->entries) {
+            int limit = c->entries - current_entry;
+            int n = get_bits(f, ilog(limit));
+            if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); }
+            memset(lengths + current_entry, current_length, n);
+            current_entry += n;
+            ++current_length;
+         }
+      } else {
+         for (j=0; j < c->entries; ++j) {
+            int present = c->sparse ? get_bits(f,1) : 1;
+            if (present) {
+               lengths[j] = get_bits(f, 5) + 1;
+               ++total;
+               if (lengths[j] == 32)
+                  return error(f, VORBIS_invalid_setup);
+            } else {
+               lengths[j] = NO_CODE;
+            }
+         }
+      }
+
+      if (c->sparse && total >= c->entries >> 2) {
+         // convert sparse items to non-sparse!
+         if (c->entries > (int) f->setup_temp_memory_required)
+            f->setup_temp_memory_required = c->entries;
+
+         c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries);
+         if (c->codeword_lengths == NULL) return error(f, VORBIS_outofmem);
+         memcpy(c->codeword_lengths, lengths, c->entries);
+         setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs!
+         lengths = c->codeword_lengths;
+         c->sparse = 0;
+      }
+
+      // compute the size of the sorted tables
+      if (c->sparse) {
+         sorted_count = total;
+      } else {
+         sorted_count = 0;
+         #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH
+         for (j=0; j < c->entries; ++j)
+            if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE)
+               ++sorted_count;
+         #endif
+      }
+
+      c->sorted_entries = sorted_count;
+      values = NULL;
+
+      CHECK(f);
+      if (!c->sparse) {
+         c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries);
+         if (!c->codewords)                  return error(f, VORBIS_outofmem);
+      } else {
+         unsigned int size;
+         if (c->sorted_entries) {
+            c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries);
+            if (!c->codeword_lengths)           return error(f, VORBIS_outofmem);
+            c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries);
+            if (!c->codewords)                  return error(f, VORBIS_outofmem);
+            values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries);
+            if (!values)                        return error(f, VORBIS_outofmem);
+         }
+         size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries;
+         if (size > f->setup_temp_memory_required)
+            f->setup_temp_memory_required = size;
+      }
+
+      if (!compute_codewords(c, lengths, c->entries, values)) {
+         if (c->sparse) setup_temp_free(f, values, 0);
+         return error(f, VORBIS_invalid_setup);
+      }
+
+      if (c->sorted_entries) {
+         // allocate an extra slot for sentinels
+         c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1));
+         if (c->sorted_codewords == NULL) return error(f, VORBIS_outofmem);
+         // allocate an extra slot at the front so that c->sorted_values[-1] is defined
+         // so that we can catch that case without an extra if
+         c->sorted_values    = ( int   *) setup_malloc(f, sizeof(*c->sorted_values   ) * (c->sorted_entries+1));
+         if (c->sorted_values == NULL) return error(f, VORBIS_outofmem);
+         ++c->sorted_values;
+         c->sorted_values[-1] = -1;
+         compute_sorted_huffman(c, lengths, values);
+      }
+
+      if (c->sparse) {
+         setup_temp_free(f, values, sizeof(*values)*c->sorted_entries);
+         setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries);
+         setup_temp_free(f, lengths, c->entries);
+         c->codewords = NULL;
+      }
+
+      compute_accelerated_huffman(c);
+
+      CHECK(f);
+      c->lookup_type = get_bits(f, 4);
+      if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup);
+      if (c->lookup_type > 0) {
+         uint16 *mults;
+         c->minimum_value = float32_unpack(get_bits(f, 32));
+         c->delta_value = float32_unpack(get_bits(f, 32));
+         c->value_bits = get_bits(f, 4)+1;
+         c->sequence_p = get_bits(f,1);
+         if (c->lookup_type == 1) {
+            c->lookup_values = lookup1_values(c->entries, c->dimensions);
+         } else {
+            c->lookup_values = c->entries * c->dimensions;
+         }
+         if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
+         mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
+         if (mults == NULL) return error(f, VORBIS_outofmem);
+         for (j=0; j < (int) c->lookup_values; ++j) {
+            int q = get_bits(f, c->value_bits);
+            if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); }
+            mults[j] = q;
+         }
+
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+         if (c->lookup_type == 1) {
+            int len, sparse = c->sparse;
+            float last=0;
+            // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop
+            if (sparse) {
+               if (c->sorted_entries == 0) goto skip;
+               c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions);
+            } else
+               c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries        * c->dimensions);
+            if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+            len = sparse ? c->sorted_entries : c->entries;
+            for (j=0; j < len; ++j) {
+               unsigned int z = sparse ? c->sorted_values[j] : j;
+               unsigned int div=1;
+               for (k=0; k < c->dimensions; ++k) {
+                  int off = (z / div) % c->lookup_values;
+                  float val = mults[off];
+                  val = mults[off]*c->delta_value + c->minimum_value + last;
+                  c->multiplicands[j*c->dimensions + k] = val;
+                  if (c->sequence_p)
+                     last = val;
+                  if (k+1 < c->dimensions) {
+                     if (div > UINT_MAX / (unsigned int) c->lookup_values) {
+                        setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values);
+                        return error(f, VORBIS_invalid_setup);
+                     }
+                     div *= c->lookup_values;
+                  }
+               }
+            }
+            c->lookup_type = 2;
+         }
+         else
+#endif
+         {
+            float last=0;
+            CHECK(f);
+            c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values);
+            if (c->multiplicands == NULL) { setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); }
+            for (j=0; j < (int) c->lookup_values; ++j) {
+               float val = mults[j] * c->delta_value + c->minimum_value + last;
+               c->multiplicands[j] = val;
+               if (c->sequence_p)
+                  last = val;
+            }
+         }
+#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK
+        skip:;
+#endif
+         setup_temp_free(f, mults, sizeof(mults[0])*c->lookup_values);
+
+         CHECK(f);
+      }
+      CHECK(f);
+   }
+
+   // time domain transfers (notused)
+
+   x = get_bits(f, 6) + 1;
+   for (i=0; i < x; ++i) {
+      uint32 z = get_bits(f, 16);
+      if (z != 0) return error(f, VORBIS_invalid_setup);
+   }
+
+   // Floors
+   f->floor_count = get_bits(f, 6)+1;
+   f->floor_config = (Floor *)  setup_malloc(f, f->floor_count * sizeof(*f->floor_config));
+   if (f->floor_config == NULL) return error(f, VORBIS_outofmem);
+   for (i=0; i < f->floor_count; ++i) {
+      f->floor_types[i] = get_bits(f, 16);
+      if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup);
+      if (f->floor_types[i] == 0) {
+         Floor0 *g = &f->floor_config[i].floor0;
+         g->order = get_bits(f,8);
+         g->rate = get_bits(f,16);
+         g->bark_map_size = get_bits(f,16);
+         g->amplitude_bits = get_bits(f,6);
+         g->amplitude_offset = get_bits(f,8);
+         g->number_of_books = get_bits(f,4) + 1;
+         for (j=0; j < g->number_of_books; ++j)
+            g->book_list[j] = get_bits(f,8);
+         return error(f, VORBIS_feature_not_supported);
+      } else {
+         Point p[31*8+2];
+         Floor1 *g = &f->floor_config[i].floor1;
+         int max_class = -1; 
+         g->partitions = get_bits(f, 5);
+         for (j=0; j < g->partitions; ++j) {
+            g->partition_class_list[j] = get_bits(f, 4);
+            if (g->partition_class_list[j] > max_class)
+               max_class = g->partition_class_list[j];
+         }
+         for (j=0; j <= max_class; ++j) {
+            g->class_dimensions[j] = get_bits(f, 3)+1;
+            g->class_subclasses[j] = get_bits(f, 2);
+            if (g->class_subclasses[j]) {
+               g->class_masterbooks[j] = get_bits(f, 8);
+               if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            }
+            for (k=0; k < 1 << g->class_subclasses[j]; ++k) {
+               g->subclass_books[j][k] = get_bits(f,8)-1;
+               if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            }
+         }
+         g->floor1_multiplier = get_bits(f,2)+1;
+         g->rangebits = get_bits(f,4);
+         g->Xlist[0] = 0;
+         g->Xlist[1] = 1 << g->rangebits;
+         g->values = 2;
+         for (j=0; j < g->partitions; ++j) {
+            int c = g->partition_class_list[j];
+            for (k=0; k < g->class_dimensions[c]; ++k) {
+               g->Xlist[g->values] = get_bits(f, g->rangebits);
+               ++g->values;
+            }
+         }
+         // precompute the sorting
+         for (j=0; j < g->values; ++j) {
+            p[j].x = g->Xlist[j];
+            p[j].y = j;
+         }
+         qsort(p, g->values, sizeof(p[0]), point_compare);
+         for (j=0; j < g->values; ++j)
+            g->sorted_order[j] = (uint8) p[j].y;
+         // precompute the neighbors
+         for (j=2; j < g->values; ++j) {
+            int low,hi;
+            neighbors(g->Xlist, j, &low,&hi);
+            g->neighbors[j][0] = low;
+            g->neighbors[j][1] = hi;
+         }
+
+         if (g->values > longest_floorlist)
+            longest_floorlist = g->values;
+      }
+   }
+
+   // Residue
+   f->residue_count = get_bits(f, 6)+1;
+   f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(f->residue_config[0]));
+   if (f->residue_config == NULL) return error(f, VORBIS_outofmem);
+   memset(f->residue_config, 0, f->residue_count * sizeof(f->residue_config[0]));
+   for (i=0; i < f->residue_count; ++i) {
+      uint8 residue_cascade[64];
+      Residue *r = f->residue_config+i;
+      f->residue_types[i] = get_bits(f, 16);
+      if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup);
+      r->begin = get_bits(f, 24);
+      r->end = get_bits(f, 24);
+      if (r->end < r->begin) return error(f, VORBIS_invalid_setup);
+      r->part_size = get_bits(f,24)+1;
+      r->classifications = get_bits(f,6)+1;
+      r->classbook = get_bits(f,8);
+      if (r->classbook >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+      for (j=0; j < r->classifications; ++j) {
+         uint8 high_bits=0;
+         uint8 low_bits=get_bits(f,3);
+         if (get_bits(f,1))
+            high_bits = get_bits(f,5);
+         residue_cascade[j] = high_bits*8 + low_bits;
+      }
+      r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications);
+      if (r->residue_books == NULL) return error(f, VORBIS_outofmem);
+      for (j=0; j < r->classifications; ++j) {
+         for (k=0; k < 8; ++k) {
+            if (residue_cascade[j] & (1 << k)) {
+               r->residue_books[j][k] = get_bits(f, 8);
+               if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup);
+            } else {
+               r->residue_books[j][k] = -1;
+            }
+         }
+      }
+      // precompute the classifications[] array to avoid inner-loop mod/divide
+      // call it 'classdata' since we already have r->classifications
+      r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+      if (!r->classdata) return error(f, VORBIS_outofmem);
+      memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries);
+      for (j=0; j < f->codebooks[r->classbook].entries; ++j) {
+         int classwords = f->codebooks[r->classbook].dimensions;
+         int temp = j;
+         r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords);
+         if (r->classdata[j] == NULL) return error(f, VORBIS_outofmem);
+         for (k=classwords-1; k >= 0; --k) {
+            r->classdata[j][k] = temp % r->classifications;
+            temp /= r->classifications;
+         }
+      }
+   }
+
+   f->mapping_count = get_bits(f,6)+1;
+   f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping));
+   if (f->mapping == NULL) return error(f, VORBIS_outofmem);
+   memset(f->mapping, 0, f->mapping_count * sizeof(*f->mapping));
+   for (i=0; i < f->mapping_count; ++i) {
+      Mapping *m = f->mapping + i;      
+      int mapping_type = get_bits(f,16);
+      if (mapping_type != 0) return error(f, VORBIS_invalid_setup);
+      m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan));
+      if (m->chan == NULL) return error(f, VORBIS_outofmem);
+      if (get_bits(f,1))
+         m->submaps = get_bits(f,4)+1;
+      else
+         m->submaps = 1;
+      if (m->submaps > max_submaps)
+         max_submaps = m->submaps;
+      if (get_bits(f,1)) {
+         m->coupling_steps = get_bits(f,8)+1;
+         for (k=0; k < m->coupling_steps; ++k) {
+            m->chan[k].magnitude = get_bits(f, ilog(f->channels-1));
+            m->chan[k].angle = get_bits(f, ilog(f->channels-1));
+            if (m->chan[k].magnitude >= f->channels)        return error(f, VORBIS_invalid_setup);
+            if (m->chan[k].angle     >= f->channels)        return error(f, VORBIS_invalid_setup);
+            if (m->chan[k].magnitude == m->chan[k].angle)   return error(f, VORBIS_invalid_setup);
+         }
+      } else
+         m->coupling_steps = 0;
+
+      // reserved field
+      if (get_bits(f,2)) return error(f, VORBIS_invalid_setup);
+      if (m->submaps > 1) {
+         for (j=0; j < f->channels; ++j) {
+            m->chan[j].mux = get_bits(f, 4);
+            if (m->chan[j].mux >= m->submaps)                return error(f, VORBIS_invalid_setup);
+         }
+      } else
+         // @SPECIFICATION: this case is missing from the spec
+         for (j=0; j < f->channels; ++j)
+            m->chan[j].mux = 0;
+
+      for (j=0; j < m->submaps; ++j) {
+         get_bits(f,8); // discard
+         m->submap_floor[j] = get_bits(f,8);
+         m->submap_residue[j] = get_bits(f,8);
+         if (m->submap_floor[j] >= f->floor_count)      return error(f, VORBIS_invalid_setup);
+         if (m->submap_residue[j] >= f->residue_count)  return error(f, VORBIS_invalid_setup);
+      }
+   }
+
+   // Modes
+   f->mode_count = get_bits(f, 6)+1;
+   for (i=0; i < f->mode_count; ++i) {
+      Mode *m = f->mode_config+i;
+      m->blockflag = get_bits(f,1);
+      m->windowtype = get_bits(f,16);
+      m->transformtype = get_bits(f,16);
+      m->mapping = get_bits(f,8);
+      if (m->windowtype != 0)                 return error(f, VORBIS_invalid_setup);
+      if (m->transformtype != 0)              return error(f, VORBIS_invalid_setup);
+      if (m->mapping >= f->mapping_count)     return error(f, VORBIS_invalid_setup);
+   }
+
+   flush_packet(f);
+
+   f->previous_length = 0;
+
+   for (i=0; i < f->channels; ++i) {
+      f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1);
+      f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+      f->finalY[i]          = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist);
+      if (f->channel_buffers[i] == NULL || f->previous_window[i] == NULL || f->finalY[i] == NULL) return error(f, VORBIS_outofmem);
+      #ifdef STB_VORBIS_NO_DEFER_FLOOR
+      f->floor_buffers[i]   = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2);
+      if (f->floor_buffers[i] == NULL) return error(f, VORBIS_outofmem);
+      #endif
+   }
+
+   if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE;
+   if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE;
+   f->blocksize[0] = f->blocksize_0;
+   f->blocksize[1] = f->blocksize_1;
+
+#ifdef STB_VORBIS_DIVIDE_TABLE
+   if (integer_divide_table[1][1]==0)
+      for (i=0; i < DIVTAB_NUMER; ++i)
+         for (j=1; j < DIVTAB_DENOM; ++j)
+            integer_divide_table[i][j] = i / j;
+#endif
+
+   // compute how much temporary memory is needed
+
+   // 1.
+   {
+      uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1);
+      uint32 classify_mem;
+      int i,max_part_read=0;
+      for (i=0; i < f->residue_count; ++i) {
+         Residue *r = f->residue_config + i;
+         int n_read = r->end - r->begin;
+         int part_read = n_read / r->part_size;
+         if (part_read > max_part_read)
+            max_part_read = part_read;
+      }
+      #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE
+      classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *));
+      #else
+      classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *));
+      #endif
+
+      f->temp_memory_required = classify_mem;
+      if (imdct_mem > f->temp_memory_required)
+         f->temp_memory_required = imdct_mem;
+   }
+
+   f->first_decode = TRUE;
+
+   if (f->alloc.alloc_buffer) {
+      assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes);
+      // check if there's enough temp memory so we don't error later
+      if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset)
+         return error(f, VORBIS_outofmem);
+   }
+
+   f->first_audio_page_offset = stb_vorbis_get_file_offset(f);
+
+   return TRUE;
+}
+
+static void vorbis_deinit(stb_vorbis *p)
+{
+   int i,j;
+   if (p->residue_config) {
+      for (i=0; i < p->residue_count; ++i) {
+         Residue *r = p->residue_config+i;
+         if (r->classdata) {
+            for (j=0; j < p->codebooks[r->classbook].entries; ++j)
+               setup_free(p, r->classdata[j]);
+            setup_free(p, r->classdata);
+         }
+         setup_free(p, r->residue_books);
+      }
+   }
+
+   if (p->codebooks) {
+      CHECK(p);
+      for (i=0; i < p->codebook_count; ++i) {
+         Codebook *c = p->codebooks + i;
+         setup_free(p, c->codeword_lengths);
+         setup_free(p, c->multiplicands);
+         setup_free(p, c->codewords);
+         setup_free(p, c->sorted_codewords);
+         // c->sorted_values[-1] is the first entry in the array
+         setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL);
+      }
+      setup_free(p, p->codebooks);
+   }
+   setup_free(p, p->floor_config);
+   setup_free(p, p->residue_config);
+   if (p->mapping) {
+      for (i=0; i < p->mapping_count; ++i)
+         setup_free(p, p->mapping[i].chan);
+      setup_free(p, p->mapping);
+   }
+   CHECK(p);
+   for (i=0; i < p->channels && i < STB_VORBIS_MAX_CHANNELS; ++i) {
+      setup_free(p, p->channel_buffers[i]);
+      setup_free(p, p->previous_window[i]);
+      #ifdef STB_VORBIS_NO_DEFER_FLOOR
+      setup_free(p, p->floor_buffers[i]);
+      #endif
+      setup_free(p, p->finalY[i]);
+   }
+   for (i=0; i < 2; ++i) {
+      setup_free(p, p->A[i]);
+      setup_free(p, p->B[i]);
+      setup_free(p, p->C[i]);
+      setup_free(p, p->window[i]);
+      setup_free(p, p->bit_reverse[i]);
+   }
+   #ifndef STB_VORBIS_NO_STDIO
+   if (p->close_on_free) fclose(p->f);
+   #endif
+}
+
+void stb_vorbis_close(stb_vorbis *p)
+{
+   if (p == NULL) return;
+   vorbis_deinit(p);
+   setup_free(p,p);
+}
+
+static void vorbis_init(stb_vorbis *p, const stb_vorbis_alloc *z)
+{
+   memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start
+   if (z) {
+      p->alloc = *z;
+      p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3;
+      p->temp_offset = p->alloc.alloc_buffer_length_in_bytes;
+   }
+   p->eof = 0;
+   p->error = VORBIS__no_error;
+   p->stream = NULL;
+   p->codebooks = NULL;
+   p->page_crc_tests = -1;
+   #ifndef STB_VORBIS_NO_STDIO
+   p->close_on_free = FALSE;
+   p->f = NULL;
+   #endif
+}
+
+int stb_vorbis_get_sample_offset(stb_vorbis *f)
+{
+   if (f->current_loc_valid)
+      return f->current_loc;
+   else
+      return -1;
+}
+
+stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f)
+{
+   stb_vorbis_info d;
+   d.channels = f->channels;
+   d.sample_rate = f->sample_rate;
+   d.setup_memory_required = f->setup_memory_required;
+   d.setup_temp_memory_required = f->setup_temp_memory_required;
+   d.temp_memory_required = f->temp_memory_required;
+   d.max_frame_size = f->blocksize_1 >> 1;
+   return d;
+}
+
+int stb_vorbis_get_error(stb_vorbis *f)
+{
+   int e = f->error;
+   f->error = VORBIS__no_error;
+   return e;
+}
+
+static stb_vorbis * vorbis_alloc(stb_vorbis *f)
+{
+   stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p));
+   return p;
+}
+
+#ifndef STB_VORBIS_NO_PUSHDATA_API
+
+void stb_vorbis_flush_pushdata(stb_vorbis *f)
+{
+   f->previous_length = 0;
+   f->page_crc_tests  = 0;
+   f->discard_samples_deferred = 0;
+   f->current_loc_valid = FALSE;
+   f->first_decode = FALSE;
+   f->samples_output = 0;
+   f->channel_buffer_start = 0;
+   f->channel_buffer_end = 0;
+}
+
+static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len)
+{
+   int i,n;
+   for (i=0; i < f->page_crc_tests; ++i)
+      f->scan[i].bytes_done = 0;
+
+   // if we have room for more scans, search for them first, because
+   // they may cause us to stop early if their header is incomplete
+   if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) {
+      if (data_len < 4) return 0;
+      data_len -= 3; // need to look for 4-byte sequence, so don't miss
+                     // one that straddles a boundary
+      for (i=0; i < data_len; ++i) {
+         if (data[i] == 0x4f) {
+            if (0==memcmp(data+i, ogg_page_header, 4)) {
+               int j,len;
+               uint32 crc;
+               // make sure we have the whole page header
+               if (i+26 >= data_len || i+27+data[i+26] >= data_len) {
+                  // only read up to this page start, so hopefully we'll
+                  // have the whole page header start next time
+                  data_len = i;
+                  break;
+               }
+               // ok, we have it all; compute the length of the page
+               len = 27 + data[i+26];
+               for (j=0; j < data[i+26]; ++j)
+                  len += data[i+27+j];
+               // scan everything up to the embedded crc (which we must 0)
+               crc = 0;
+               for (j=0; j < 22; ++j)
+                  crc = crc32_update(crc, data[i+j]);
+               // now process 4 0-bytes
+               for (   ; j < 26; ++j)
+                  crc = crc32_update(crc, 0);
+               // len is the total number of bytes we need to scan
+               n = f->page_crc_tests++;
+               f->scan[n].bytes_left = len-j;
+               f->scan[n].crc_so_far = crc;
+               f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24);
+               // if the last frame on a page is continued to the next, then
+               // we can't recover the sample_loc immediately
+               if (data[i+27+data[i+26]-1] == 255)
+                  f->scan[n].sample_loc = ~0;
+               else
+                  f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24);
+               f->scan[n].bytes_done = i+j;
+               if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT)
+                  break;
+               // keep going if we still have room for more
+            }
+         }
+      }
+   }
+
+   for (i=0; i < f->page_crc_tests;) {
+      uint32 crc;
+      int j;
+      int n = f->scan[i].bytes_done;
+      int m = f->scan[i].bytes_left;
+      if (m > data_len - n) m = data_len - n;
+      // m is the bytes to scan in the current chunk
+      crc = f->scan[i].crc_so_far;
+      for (j=0; j < m; ++j)
+         crc = crc32_update(crc, data[n+j]);
+      f->scan[i].bytes_left -= m;
+      f->scan[i].crc_so_far = crc;
+      if (f->scan[i].bytes_left == 0) {
+         // does it match?
+         if (f->scan[i].crc_so_far == f->scan[i].goal_crc) {
+            // Houston, we have page
+            data_len = n+m; // consumption amount is wherever that scan ended
+            f->page_crc_tests = -1; // drop out of page scan mode
+            f->previous_length = 0; // decode-but-don't-output one frame
+            f->next_seg = -1;       // start a new page
+            f->current_loc = f->scan[i].sample_loc; // set the current sample location
+                                    // to the amount we'd have decoded had we decoded this page
+            f->current_loc_valid = f->current_loc != ~0U;
+            return data_len;
+         }
+         // delete entry
+         f->scan[i] = f->scan[--f->page_crc_tests];
+      } else {
+         ++i;
+      }
+   }
+
+   return data_len;
+}
+
+// return value: number of bytes we used
+int stb_vorbis_decode_frame_pushdata(
+         stb_vorbis *f,                   // the file we're decoding
+         const uint8 *data, int data_len, // the memory available for decoding
+         int *channels,                   // place to write number of float * buffers
+         float ***output,                 // place to write float ** array of float * buffers
+         int *samples                     // place to write number of output samples
+     )
+{
+   int i;
+   int len,right,left;
+
+   if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   if (f->page_crc_tests >= 0) {
+      *samples = 0;
+      return vorbis_search_for_page_pushdata(f, (uint8 *) data, data_len);
+   }
+
+   f->stream     = (uint8 *) data;
+   f->stream_end = (uint8 *) data + data_len;
+   f->error      = VORBIS__no_error;
+
+   // check that we have the entire packet in memory
+   if (!is_whole_packet_present(f, FALSE)) {
+      *samples = 0;
+      return 0;
+   }
+
+   if (!vorbis_decode_packet(f, &len, &left, &right)) {
+      // save the actual error we encountered
+      enum STBVorbisError error = f->error;
+      if (error == VORBIS_bad_packet_type) {
+         // flush and resynch
+         f->error = VORBIS__no_error;
+         while (get8_packet(f) != EOP)
+            if (f->eof) break;
+         *samples = 0;
+         return (int) (f->stream - data);
+      }
+      if (error == VORBIS_continued_packet_flag_invalid) {
+         if (f->previous_length == 0) {
+            // we may be resynching, in which case it's ok to hit one
+            // of these; just discard the packet
+            f->error = VORBIS__no_error;
+            while (get8_packet(f) != EOP)
+               if (f->eof) break;
+            *samples = 0;
+            return (int) (f->stream - data);
+         }
+      }
+      // if we get an error while parsing, what to do?
+      // well, it DEFINITELY won't work to continue from where we are!
+      stb_vorbis_flush_pushdata(f);
+      // restore the error that actually made us bail
+      f->error = error;
+      *samples = 0;
+      return 1;
+   }
+
+   // success!
+   len = vorbis_finish_frame(f, len, left, right);
+   for (i=0; i < f->channels; ++i)
+      f->outputs[i] = f->channel_buffers[i] + left;
+
+   if (channels) *channels = f->channels;
+   *samples = len;
+   *output = f->outputs;
+   return (int) (f->stream - data);
+}
+
+stb_vorbis *stb_vorbis_open_pushdata(
+         const unsigned char *data, int data_len, // the memory available for decoding
+         int *data_used,              // only defined if result is not NULL
+         int *error, const stb_vorbis_alloc *alloc)
+{
+   stb_vorbis *f, p;
+   vorbis_init(&p, alloc);
+   p.stream     = (uint8 *) data;
+   p.stream_end = (uint8 *) data + data_len;
+   p.push_mode  = TRUE;
+   if (!start_decoder(&p)) {
+      if (p.eof)
+         *error = VORBIS_need_more_data;
+      else
+         *error = p.error;
+      return NULL;
+   }
+   f = vorbis_alloc(&p);
+   if (f) {
+      *f = p;
+      *data_used = (int) (f->stream - data);
+      *error = 0;
+      return f;
+   } else {
+      vorbis_deinit(&p);
+      return NULL;
+   }
+}
+#endif // STB_VORBIS_NO_PUSHDATA_API
+
+unsigned int stb_vorbis_get_file_offset(stb_vorbis *f)
+{
+   #ifndef STB_VORBIS_NO_PUSHDATA_API
+   if (f->push_mode) return 0;
+   #endif
+   if (USE_MEMORY(f)) return (unsigned int) (f->stream - f->stream_start);
+   #ifndef STB_VORBIS_NO_STDIO
+   return (unsigned int) (ftell(f->f) - f->f_start);
+   #endif
+}
+
+#ifndef STB_VORBIS_NO_PULLDATA_API
+//
+// DATA-PULLING API
+//
+
+static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
+{
+   for(;;) {
+      int n;
+      if (f->eof) return 0;
+      n = get8(f);
+      if (n == 0x4f) { // page header candidate
+         unsigned int retry_loc = stb_vorbis_get_file_offset(f);
+         int i;
+         // check if we're off the end of a file_section stream
+         if (retry_loc - 25 > f->stream_len)
+            return 0;
+         // check the rest of the header
+         for (i=1; i < 4; ++i)
+            if (get8(f) != ogg_page_header[i])
+               break;
+         if (f->eof) return 0;
+         if (i == 4) {
+            uint8 header[27];
+            uint32 i, crc, goal, len;
+            for (i=0; i < 4; ++i)
+               header[i] = ogg_page_header[i];
+            for (; i < 27; ++i)
+               header[i] = get8(f);
+            if (f->eof) return 0;
+            if (header[4] != 0) goto invalid;
+            goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24);
+            for (i=22; i < 26; ++i)
+               header[i] = 0;
+            crc = 0;
+            for (i=0; i < 27; ++i)
+               crc = crc32_update(crc, header[i]);
+            len = 0;
+            for (i=0; i < header[26]; ++i) {
+               int s = get8(f);
+               crc = crc32_update(crc, s);
+               len += s;
+            }
+            if (len && f->eof) return 0;
+            for (i=0; i < len; ++i)
+               crc = crc32_update(crc, get8(f));
+            // finished parsing probable page
+            if (crc == goal) {
+               // we could now check that it's either got the last
+               // page flag set, OR it's followed by the capture
+               // pattern, but I guess TECHNICALLY you could have
+               // a file with garbage between each ogg page and recover
+               // from it automatically? So even though that paranoia
+               // might decrease the chance of an invalid decode by
+               // another 2^32, not worth it since it would hose those
+               // invalid-but-useful files?
+               if (end)
+                  *end = stb_vorbis_get_file_offset(f);
+               if (last) {
+                  if (header[5] & 0x04)
+                     *last = 1;
+                  else
+                     *last = 0;
+               }
+               set_file_offset(f, retry_loc-1);
+               return 1;
+            }
+         }
+        invalid:
+         // not a valid page, so rewind and look for next one
+         set_file_offset(f, retry_loc);
+      }
+   }
+}
+
+
+#define SAMPLE_unknown  0xffffffff
+
+// seeking is implemented with a binary search, which narrows down the range to
+// 64K, before using a linear search (because finding the synchronization
+// pattern can be expensive, and the chance we'd find the end page again is
+// relatively high for small ranges)
+//
+// two initial interpolation-style probes are used at the start of the search
+// to try to bound either side of the binary search sensibly, while still
+// working in O(log n) time if they fail.
+
+static int get_seek_page_info(stb_vorbis *f, ProbedPage *z)
+{
+   uint8 header[27], lacing[255];
+   int i,len;
+
+   // record where the page starts
+   z->page_start = stb_vorbis_get_file_offset(f);
+
+   // parse the header
+   getn(f, header, 27);
+   if (header[0] != 'O' || header[1] != 'g' || header[2] != 'g' || header[3] != 'S')
+      return 0;
+   getn(f, lacing, header[26]);
+
+   // determine the length of the payload
+   len = 0;
+   for (i=0; i < header[26]; ++i)
+      len += lacing[i];
+
+   // this implies where the page ends
+   z->page_end = z->page_start + 27 + header[26] + len;
+
+   // read the last-decoded sample out of the data
+   z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 24);
+
+   // restore file state to where we were
+   set_file_offset(f, z->page_start);
+   return 1;
+}
+
+// rarely used function to seek back to the preceeding page while finding the
+// start of a packet
+static int go_to_page_before(stb_vorbis *f, unsigned int limit_offset)
+{
+   unsigned int previous_safe, end;
+
+   // now we want to seek back 64K from the limit
+   if (limit_offset >= 65536 && limit_offset-65536 >= f->first_audio_page_offset)
+      previous_safe = limit_offset - 65536;
+   else
+      previous_safe = f->first_audio_page_offset;
+
+   set_file_offset(f, previous_safe);
+
+   while (vorbis_find_page(f, &end, NULL)) {
+      if (end >= limit_offset && stb_vorbis_get_file_offset(f) < limit_offset)
+         return 1;
+      set_file_offset(f, end);
+   }
+
+   return 0;
+}
+
+// implements the search logic for finding a page and starting decoding. if
+// the function succeeds, current_loc_valid will be true and current_loc will
+// be less than or equal to the provided sample number (the closer the
+// better).
+static int seek_to_sample_coarse(stb_vorbis *f, uint32 sample_number)
+{
+   ProbedPage left, right, mid;
+   int i, start_seg_with_known_loc, end_pos, page_start;
+   uint32 delta, stream_length, padding;
+   double offset, bytes_per_sample;
+   int probe = 0;
+
+   // find the last page and validate the target sample
+   stream_length = stb_vorbis_stream_length_in_samples(f);
+   if (stream_length == 0)            return error(f, VORBIS_seek_without_length);
+   if (sample_number > stream_length) return error(f, VORBIS_seek_invalid);
+
+   // this is the maximum difference between the window-center (which is the
+   // actual granule position value), and the right-start (which the spec
+   // indicates should be the granule position (give or take one)).
+   padding = ((f->blocksize_1 - f->blocksize_0) >> 2);
+   if (sample_number < padding)
+      sample_number = 0;
+   else
+      sample_number -= padding;
+
+   left = f->p_first;
+   while (left.last_decoded_sample == ~0U) {
+      // (untested) the first page does not have a 'last_decoded_sample'
+      set_file_offset(f, left.page_end);
+      if (!get_seek_page_info(f, &left)) goto error;
+   }
+
+   right = f->p_last;
+   assert(right.last_decoded_sample != ~0U);
+
+   // starting from the start is handled differently
+   if (sample_number <= left.last_decoded_sample) {
+      stb_vorbis_seek_start(f);
+      return 1;
+   }
+
+   while (left.page_end != right.page_start) {
+      assert(left.page_end < right.page_start);
+      // search range in bytes
+      delta = right.page_start - left.page_end;
+      if (delta <= 65536) {
+         // there's only 64K left to search - handle it linearly
+         set_file_offset(f, left.page_end);
+      } else {
+         if (probe < 2) {
+            if (probe == 0) {
+               // first probe (interpolate)
+               double data_bytes = right.page_end - left.page_start;
+               bytes_per_sample = data_bytes / right.last_decoded_sample;
+               offset = left.page_start + bytes_per_sample * (sample_number - left.last_decoded_sample);
+            } else {
+               // second probe (try to bound the other side)
+               double error = ((double) sample_number - mid.last_decoded_sample) * bytes_per_sample;
+               if (error >= 0 && error <  8000) error =  8000;
+               if (error <  0 && error > -8000) error = -8000;
+               offset += error * 2;
+            }
+
+            // ensure the offset is valid
+            if (offset < left.page_end)
+               offset = left.page_end;
+            if (offset > right.page_start - 65536)
+               offset = right.page_start - 65536;
+
+            set_file_offset(f, (unsigned int) offset);
+         } else {
+            // binary search for large ranges (offset by 32K to ensure
+            // we don't hit the right page)
+            set_file_offset(f, left.page_end + (delta / 2) - 32768);
+         }
+
+         if (!vorbis_find_page(f, NULL, NULL)) goto error;
+      }
+
+      for (;;) {
+         if (!get_seek_page_info(f, &mid)) goto error;
+         if (mid.last_decoded_sample != ~0U) break;
+         // (untested) no frames end on this page
+         set_file_offset(f, mid.page_end);
+         assert(mid.page_start < right.page_start);
+      }
+
+      // if we've just found the last page again then we're in a tricky file,
+      // and we're close enough.
+      if (mid.page_start == right.page_start)
+         break;
+
+      if (sample_number < mid.last_decoded_sample)
+         right = mid;
+      else
+         left = mid;
+
+      ++probe;
+   }
+
+   // seek back to start of the last packet
+   page_start = left.page_start;
+   set_file_offset(f, page_start);
+   if (!start_page(f)) return error(f, VORBIS_seek_failed);
+   end_pos = f->end_seg_with_known_loc;
+   assert(end_pos >= 0);
+
+   for (;;) {
+      for (i = end_pos; i > 0; --i)
+         if (f->segments[i-1] != 255)
+            break;
+
+      start_seg_with_known_loc = i;
+
+      if (start_seg_with_known_loc > 0 || !(f->page_flag & PAGEFLAG_continued_packet))
+         break;
+
+      // (untested) the final packet begins on an earlier page
+      if (!go_to_page_before(f, page_start))
+         goto error;
+
+      page_start = stb_vorbis_get_file_offset(f);
+      if (!start_page(f)) goto error;
+      end_pos = f->segment_count - 1;
+   }
+
+   // prepare to start decoding
+   f->current_loc_valid = FALSE;
+   f->last_seg = FALSE;
+   f->valid_bits = 0;
+   f->packet_bytes = 0;
+   f->bytes_in_seg = 0;
+   f->previous_length = 0;
+   f->next_seg = start_seg_with_known_loc;
+
+   for (i = 0; i < start_seg_with_known_loc; i++)
+      skip(f, f->segments[i]);
+
+   // start decoding (optimizable - this frame is generally discarded)
+   vorbis_pump_first_frame(f);
+   return 1;
+
+error:
+   // try to restore the file to a valid state
+   stb_vorbis_seek_start(f);
+   return error(f, VORBIS_seek_failed);
+}
+
+// the same as vorbis_decode_initial, but without advancing
+static int peek_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode)
+{
+   int bits_read, bytes_read;
+
+   if (!vorbis_decode_initial(f, p_left_start, p_left_end, p_right_start, p_right_end, mode))
+      return 0;
+
+   // either 1 or 2 bytes were read, figure out which so we can rewind
+   bits_read = 1 + ilog(f->mode_count-1);
+   if (f->mode_config[*mode].blockflag)
+      bits_read += 2;
+   bytes_read = (bits_read + 7) / 8;
+
+   f->bytes_in_seg += bytes_read;
+   f->packet_bytes -= bytes_read;
+   skip(f, -bytes_read);
+   if (f->next_seg == -1)
+      f->next_seg = f->segment_count - 1;
+   else
+      f->next_seg--;
+   f->valid_bits = 0;
+
+   return 1;
+}
+
+int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number)
+{
+   uint32 max_frame_samples;
+
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   // fast page-level search
+   if (!seek_to_sample_coarse(f, sample_number))
+      return 0;
+
+   assert(f->current_loc_valid);
+   assert(f->current_loc <= sample_number);
+
+   // linear search for the relevant packet
+   max_frame_samples = (f->blocksize_1*3 - f->blocksize_0) >> 2;
+   while (f->current_loc < sample_number) {
+      int left_start, left_end, right_start, right_end, mode, frame_samples;
+      if (!peek_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode))
+         return error(f, VORBIS_seek_failed);
+      // calculate the number of samples returned by the next frame
+      frame_samples = right_start - left_start;
+      if (f->current_loc + frame_samples > sample_number) {
+         return 1; // the next frame will contain the sample
+      } else if (f->current_loc + frame_samples + max_frame_samples > sample_number) {
+         // there's a chance the frame after this could contain the sample
+         vorbis_pump_first_frame(f);
+      } else {
+         // this frame is too early to be relevant
+         f->current_loc += frame_samples;
+         f->previous_length = 0;
+         maybe_start_packet(f);
+         flush_packet(f);
+      }
+   }
+   // the next frame will start with the sample
+   assert(f->current_loc == sample_number);
+   return 1;
+}
+
+int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number)
+{
+   if (!stb_vorbis_seek_frame(f, sample_number))
+      return 0;
+
+   if (sample_number != f->current_loc) {
+      int n;
+      uint32 frame_start = f->current_loc;
+      stb_vorbis_get_frame_float(f, &n, NULL);
+      assert(sample_number > frame_start);
+      assert(f->channel_buffer_start + (int) (sample_number-frame_start) <= f->channel_buffer_end);
+      f->channel_buffer_start += (sample_number - frame_start);
+   }
+
+   return 1;
+}
+
+void stb_vorbis_seek_start(stb_vorbis *f)
+{
+   if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; }
+   set_file_offset(f, f->first_audio_page_offset);
+   f->previous_length = 0;
+   f->first_decode = TRUE;
+   f->next_seg = -1;
+   vorbis_pump_first_frame(f);
+}
+
+unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f)
+{
+   unsigned int restore_offset, previous_safe;
+   unsigned int end, last_page_loc;
+
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+   if (!f->total_samples) {
+      unsigned int last;
+      uint32 lo,hi;
+      char header[6];
+
+      // first, store the current decode position so we can restore it
+      restore_offset = stb_vorbis_get_file_offset(f);
+
+      // now we want to seek back 64K from the end (the last page must
+      // be at most a little less than 64K, but let's allow a little slop)
+      if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset)
+         previous_safe = f->stream_len - 65536;
+      else
+         previous_safe = f->first_audio_page_offset;
+
+      set_file_offset(f, previous_safe);
+      // previous_safe is now our candidate 'earliest known place that seeking
+      // to will lead to the final page'
+
+      if (!vorbis_find_page(f, &end, &last)) {
+         // if we can't find a page, we're hosed!
+         f->error = VORBIS_cant_find_last_page;
+         f->total_samples = 0xffffffff;
+         goto done;
+      }
+
+      // check if there are more pages
+      last_page_loc = stb_vorbis_get_file_offset(f);
+
+      // stop when the last_page flag is set, not when we reach eof;
+      // this allows us to stop short of a 'file_section' end without
+      // explicitly checking the length of the section
+      while (!last) {
+         set_file_offset(f, end);
+         if (!vorbis_find_page(f, &end, &last)) {
+            // the last page we found didn't have the 'last page' flag
+            // set. whoops!
+            break;
+         }
+         previous_safe = last_page_loc+1;
+         last_page_loc = stb_vorbis_get_file_offset(f);
+      }
+
+      set_file_offset(f, last_page_loc);
+
+      // parse the header
+      getn(f, (unsigned char *)header, 6);
+      // extract the absolute granule position
+      lo = get32(f);
+      hi = get32(f);
+      if (lo == 0xffffffff && hi == 0xffffffff) {
+         f->error = VORBIS_cant_find_last_page;
+         f->total_samples = SAMPLE_unknown;
+         goto done;
+      }
+      if (hi)
+         lo = 0xfffffffe; // saturate
+      f->total_samples = lo;
+
+      f->p_last.page_start = last_page_loc;
+      f->p_last.page_end   = end;
+      f->p_last.last_decoded_sample = lo;
+
+     done:
+      set_file_offset(f, restore_offset);
+   }
+   return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples;
+}
+
+float stb_vorbis_stream_length_in_seconds(stb_vorbis *f)
+{
+   return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate;
+}
+
+
+
+int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output)
+{
+   int len, right,left,i;
+   if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing);
+
+   if (!vorbis_decode_packet(f, &len, &left, &right)) {
+      f->channel_buffer_start = f->channel_buffer_end = 0;
+      return 0;
+   }
+
+   len = vorbis_finish_frame(f, len, left, right);
+   for (i=0; i < f->channels; ++i)
+      f->outputs[i] = f->channel_buffers[i] + left;
+
+   f->channel_buffer_start = left;
+   f->channel_buffer_end   = left+len;
+
+   if (channels) *channels = f->channels;
+   if (output)   *output = f->outputs;
+   return len;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc, unsigned int length)
+{
+   stb_vorbis *f, p;
+   vorbis_init(&p, alloc);
+   p.f = file;
+   p.f_start = (uint32) ftell(file);
+   p.stream_len   = length;
+   p.close_on_free = close_on_free;
+   if (start_decoder(&p)) {
+      f = vorbis_alloc(&p);
+      if (f) {
+         *f = p;
+         vorbis_pump_first_frame(f);
+         return f;
+      }
+   }
+   if (error) *error = p.error;
+   vorbis_deinit(&p);
+   return NULL;
+}
+
+stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, const stb_vorbis_alloc *alloc)
+{
+   unsigned int len, start;
+   start = (unsigned int) ftell(file);
+   fseek(file, 0, SEEK_END);
+   len = (unsigned int) (ftell(file) - start);
+   fseek(file, start, SEEK_SET);
+   return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len);
+}
+
+stb_vorbis * stb_vorbis_open_filename(const char *filename, int *error, const stb_vorbis_alloc *alloc)
+{
+   FILE *f = fopen(filename, "rb");
+   if (f) 
+      return stb_vorbis_open_file(f, TRUE, error, alloc);
+   if (error) *error = VORBIS_file_open_failure;
+   return NULL;
+}
+#endif // STB_VORBIS_NO_STDIO
+
+stb_vorbis * stb_vorbis_open_memory(const unsigned char *data, int len, int *error, const stb_vorbis_alloc *alloc)
+{
+   stb_vorbis *f, p;
+   if (data == NULL) return NULL;
+   vorbis_init(&p, alloc);
+   p.stream = (uint8 *) data;
+   p.stream_end = (uint8 *) data + len;
+   p.stream_start = (uint8 *) p.stream;
+   p.stream_len = len;
+   p.push_mode = FALSE;
+   if (start_decoder(&p)) {
+      f = vorbis_alloc(&p);
+      if (f) {
+         *f = p;
+         vorbis_pump_first_frame(f);
+         return f;
+      }
+   }
+   if (error) *error = p.error;
+   vorbis_deinit(&p);
+   return NULL;
+}
+
+#ifndef STB_VORBIS_NO_INTEGER_CONVERSION
+#define PLAYBACK_MONO     1
+#define PLAYBACK_LEFT     2
+#define PLAYBACK_RIGHT    4
+
+#define L  (PLAYBACK_LEFT  | PLAYBACK_MONO)
+#define C  (PLAYBACK_LEFT  | PLAYBACK_RIGHT | PLAYBACK_MONO)
+#define R  (PLAYBACK_RIGHT | PLAYBACK_MONO)
+
+static int8 channel_position[7][6] =
+{
+   { 0 },
+   { C },
+   { L, R },
+   { L, C, R },
+   { L, R, L, R },
+   { L, C, R, L, R },
+   { L, C, R, L, R, C },
+};
+
+
+#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
+   typedef union {
+      float f;
+      int i;
+   } float_conv;
+   typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];
+   #define FASTDEF(x) float_conv x
+   // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round
+   #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT))
+   #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22))
+   #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) (temp.f = (x) + MAGIC(s), temp.i - ADDEND(s))
+   #define check_endianness()  
+#else
+   #define FAST_SCALED_FLOAT_TO_INT(temp,x,s) ((int) ((x) * (1 << (s))))
+   #define check_endianness()
+   #define FASTDEF(x)
+#endif
+
+static void copy_samples(short *dest, float *src, int len)
+{
+   int i;
+   check_endianness();
+   for (i=0; i < len; ++i) {
+      FASTDEF(temp);
+      int v = FAST_SCALED_FLOAT_TO_INT(temp, src[i],15);
+      if ((unsigned int) (v + 32768) > 65535)
+         v = v < 0 ? -32768 : 32767;
+      dest[i] = v;
+   }
+}
+
+static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len)
+{
+   #define BUFFER_SIZE  32
+   float buffer[BUFFER_SIZE];
+   int i,j,o,n = BUFFER_SIZE;
+   check_endianness();
+   for (o = 0; o < len; o += BUFFER_SIZE) {
+      memset(buffer, 0, sizeof(buffer));
+      if (o + n > len) n = len - o;
+      for (j=0; j < num_c; ++j) {
+         if (channel_position[num_c][j] & mask) {
+            for (i=0; i < n; ++i)
+               buffer[i] += data[j][d_offset+o+i];
+         }
+      }
+      for (i=0; i < n; ++i) {
+         FASTDEF(temp);
+         int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+         if ((unsigned int) (v + 32768) > 65535)
+            v = v < 0 ? -32768 : 32767;
+         output[o+i] = v;
+      }
+   }
+}
+
+static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len)
+{
+   #define BUFFER_SIZE  32
+   float buffer[BUFFER_SIZE];
+   int i,j,o,n = BUFFER_SIZE >> 1;
+   // o is the offset in the source data
+   check_endianness();
+   for (o = 0; o < len; o += BUFFER_SIZE >> 1) {
+      // o2 is the offset in the output data
+      int o2 = o << 1;
+      memset(buffer, 0, sizeof(buffer));
+      if (o + n > len) n = len - o;
+      for (j=0; j < num_c; ++j) {
+         int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT);
+         if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+0] += data[j][d_offset+o+i];
+               buffer[i*2+1] += data[j][d_offset+o+i];
+            }
+         } else if (m == PLAYBACK_LEFT) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+0] += data[j][d_offset+o+i];
+            }
+         } else if (m == PLAYBACK_RIGHT) {
+            for (i=0; i < n; ++i) {
+               buffer[i*2+1] += data[j][d_offset+o+i];
+            }
+         }
+      }
+      for (i=0; i < (n<<1); ++i) {
+         FASTDEF(temp);
+         int v = FAST_SCALED_FLOAT_TO_INT(temp,buffer[i],15);
+         if ((unsigned int) (v + 32768) > 65535)
+            v = v < 0 ? -32768 : 32767;
+         output[o2+i] = v;
+      }
+   }
+}
+
+static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples)
+{
+   int i;
+   if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+      static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} };
+      for (i=0; i < buf_c; ++i)
+         compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples);
+   } else {
+      int limit = buf_c < data_c ? buf_c : data_c;
+      for (i=0; i < limit; ++i)
+         copy_samples(buffer[i]+b_offset, data[i]+d_offset, samples);
+      for (   ; i < buf_c; ++i)
+         memset(buffer[i]+b_offset, 0, sizeof(short) * samples);
+   }
+}
+
+int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
+{
+   float **output;
+   int len = stb_vorbis_get_frame_float(f, NULL, &output);
+   if (len > num_samples) len = num_samples;
+   if (len)
+      convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);
+   return len;
+}
+
+static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len)
+{
+   int i;
+   check_endianness();
+   if (buf_c != data_c && buf_c <= 2 && data_c <= 6) {
+      assert(buf_c == 2);
+      for (i=0; i < buf_c; ++i)
+         compute_stereo_samples(buffer, data_c, data, d_offset, len);
+   } else {
+      int limit = buf_c < data_c ? buf_c : data_c;
+      int j;
+      for (j=0; j < len; ++j) {
+         for (i=0; i < limit; ++i) {
+            FASTDEF(temp);
+            float f = data[i][d_offset+j];
+            int v = FAST_SCALED_FLOAT_TO_INT(temp, f,15);//data[i][d_offset+j],15);
+            if ((unsigned int) (v + 32768) > 65535)
+               v = v < 0 ? -32768 : 32767;
+            *buffer++ = v;
+         }
+         for (   ; i < buf_c; ++i)
+            *buffer++ = 0;
+      }
+   }
+}
+
+int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts)
+{
+   float **output;
+   int len;
+   if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts);
+   len = stb_vorbis_get_frame_float(f, NULL, &output);
+   if (len) {
+      if (len*num_c > num_shorts) len = num_shorts / num_c;
+      convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len);
+   }
+   return len;
+}
+
+int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts)
+{
+   float **outputs;
+   int len = num_shorts / channels;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      if (k)
+         convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+      buffer += k*channels;
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+
+int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len)
+{
+   float **outputs;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      if (k)
+         convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k);
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len) break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break;
+   }
+   return n;
+}
+
+#ifndef STB_VORBIS_NO_STDIO
+int stb_vorbis_decode_filename(const char *filename, int *channels, int *sample_rate, short **output)
+{
+   int data_len, offset, total, limit, error;
+   short *data;
+   stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL);
+   if (v == NULL) return -1;
+   limit = v->channels * 4096;
+   *channels = v->channels;
+   if (sample_rate)
+      *sample_rate = v->sample_rate;
+   offset = data_len = 0;
+   total = limit;
+   data = (short *) malloc(total * sizeof(*data));
+   if (data == NULL) {
+      stb_vorbis_close(v);
+      return -2;
+   }
+   for (;;) {
+      int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+      if (n == 0) break;
+      data_len += n;
+      offset += n * v->channels;
+      if (offset + limit > total) {
+         short *data2;
+         total *= 2;
+         data2 = (short *) realloc(data, total * sizeof(*data));
+         if (data2 == NULL) {
+            free(data);
+            stb_vorbis_close(v);
+            return -2;
+         }
+         data = data2;
+      }
+   }
+   *output = data;
+   stb_vorbis_close(v);
+   return data_len;
+}
+#endif // NO_STDIO
+
+int stb_vorbis_decode_memory(const uint8 *mem, int len, int *channels, int *sample_rate, short **output)
+{
+   int data_len, offset, total, limit, error;
+   short *data;
+   stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL);
+   if (v == NULL) return -1;
+   limit = v->channels * 4096;
+   *channels = v->channels;
+   if (sample_rate)
+      *sample_rate = v->sample_rate;
+   offset = data_len = 0;
+   total = limit;
+   data = (short *) malloc(total * sizeof(*data));
+   if (data == NULL) {
+      stb_vorbis_close(v);
+      return -2;
+   }
+   for (;;) {
+      int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset);
+      if (n == 0) break;
+      data_len += n;
+      offset += n * v->channels;
+      if (offset + limit > total) {
+         short *data2;
+         total *= 2;
+         data2 = (short *) realloc(data, total * sizeof(*data));
+         if (data2 == NULL) {
+            free(data);
+            stb_vorbis_close(v);
+            return -2;
+         }
+         data = data2;
+      }
+   }
+   *output = data;
+   stb_vorbis_close(v);
+   return data_len;
+}
+#endif // STB_VORBIS_NO_INTEGER_CONVERSION
+
+int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats)
+{
+   float **outputs;
+   int len = num_floats / channels;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < len) {
+      int i,j;
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= len) k = len - n;
+      for (j=0; j < k; ++j) {
+         for (i=0; i < z; ++i)
+            *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j];
+         for (   ; i < channels; ++i)
+            *buffer++ = 0;
+      }
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == len)
+         break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+         break;
+   }
+   return n;
+}
+
+int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples)
+{
+   float **outputs;
+   int n=0;
+   int z = f->channels;
+   if (z > channels) z = channels;
+   while (n < num_samples) {
+      int i;
+      int k = f->channel_buffer_end - f->channel_buffer_start;
+      if (n+k >= num_samples) k = num_samples - n;
+      if (k) {
+         for (i=0; i < z; ++i)
+            memcpy(buffer[i]+n, f->channel_buffers[i]+f->channel_buffer_start, sizeof(float)*k);
+         for (   ; i < channels; ++i)
+            memset(buffer[i]+n, 0, sizeof(float) * k);
+      }
+      n += k;
+      f->channel_buffer_start += k;
+      if (n == num_samples)
+         break;
+      if (!stb_vorbis_get_frame_float(f, NULL, &outputs))
+         break;
+   }
+   return n;
+}
+#endif // STB_VORBIS_NO_PULLDATA_API
+
+/* Version history
+    1.09    - 2016/04/04 - back out 'avoid discarding last frame' fix from previous version
+    1.08    - 2016/04/02 - fixed multiple warnings; fix setup memory leaks;
+                           avoid discarding last frame of audio data
+    1.07    - 2015/01/16 - fixed some warnings, fix mingw, const-correct API
+                           some more crash fixes when out of memory or with corrupt files 
+    1.06    - 2015/08/31 - full, correct support for seeking API (Dougall Johnson)
+                           some crash fixes when out of memory or with corrupt files
+    1.05    - 2015/04/19 - don't define __forceinline if it's redundant
+    1.04    - 2014/08/27 - fix missing const-correct case in API
+    1.03    - 2014/08/07 - Warning fixes
+    1.02    - 2014/07/09 - Declare qsort compare function _cdecl on windows
+    1.01    - 2014/06/18 - fix stb_vorbis_get_samples_float
+    1.0     - 2014/05/26 - fix memory leaks; fix warnings; fix bugs in multichannel
+                           (API change) report sample rate for decode-full-file funcs
+    0.99996 - bracket #include <malloc.h> for macintosh compilation by Laurent Gomila
+    0.99995 - use union instead of pointer-cast for fast-float-to-int to avoid alias-optimization problem
+    0.99994 - change fast-float-to-int to work in single-precision FPU mode, remove endian-dependence
+    0.99993 - remove assert that fired on legal files with empty tables
+    0.99992 - rewind-to-start
+    0.99991 - bugfix to stb_vorbis_get_samples_short by Bernhard Wodo
+    0.9999 - (should have been 0.99990) fix no-CRT support, compiling as C++
+    0.9998 - add a full-decode function with a memory source
+    0.9997 - fix a bug in the read-from-FILE case in 0.9996 addition
+    0.9996 - query length of vorbis stream in samples/seconds
+    0.9995 - bugfix to another optimization that only happened in certain files
+    0.9994 - bugfix to one of the optimizations that caused significant (but inaudible?) errors
+    0.9993 - performance improvements; runs in 99% to 104% of time of reference implementation
+    0.9992 - performance improvement of IMDCT; now performs close to reference implementation
+    0.9991 - performance improvement of IMDCT
+    0.999 - (should have been 0.9990) performance improvement of IMDCT
+    0.998 - no-CRT support from Casey Muratori
+    0.997 - bugfixes for bugs found by Terje Mathisen
+    0.996 - bugfix: fast-huffman decode initialized incorrectly for sparse codebooks; fixing gives 10% speedup - found by Terje Mathisen
+    0.995 - bugfix: fix to 'effective' overrun detection - found by Terje Mathisen
+    0.994 - bugfix: garbage decode on final VQ symbol of a non-multiple - found by Terje Mathisen
+    0.993 - bugfix: pushdata API required 1 extra byte for empty page (failed to consume final page if empty) - found by Terje Mathisen
+    0.992 - fixes for MinGW warning
+    0.991 - turn fast-float-conversion on by default
+    0.990 - fix push-mode seek recovery if you seek into the headers
+    0.98b - fix to bad release of 0.98
+    0.98 - fix push-mode seek recovery; robustify float-to-int and support non-fast mode
+    0.97 - builds under c++ (typecasting, don't use 'class' keyword)
+    0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code
+    0.95 - clamping code for 16-bit functions
+    0.94 - not publically released
+    0.93 - fixed all-zero-floor case (was decoding garbage)
+    0.92 - fixed a memory leak
+    0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION
+    0.90 - first public release
+*/
+
+#endif // STB_VORBIS_HEADER_ONLY
diff --git a/tools/editor/editor_audio_buses.cpp b/tools/editor/editor_audio_buses.cpp
new file mode 100644
index 0000000000..d570abcc82
--- /dev/null
+++ b/tools/editor/editor_audio_buses.cpp
@@ -0,0 +1,619 @@
+#include "editor_audio_buses.h"
+#include "editor_node.h"
+#include "servers/audio_server.h"
+
+
+void EditorAudioBus::_notification(int p_what) {
+
+	if (p_what==NOTIFICATION_READY) {
+
+		vu_l->set_under_texture(get_icon("BusVuEmpty","EditorIcons"));
+		vu_l->set_progress_texture(get_icon("BusVuFull","EditorIcons"));
+		vu_r->set_under_texture(get_icon("BusVuEmpty","EditorIcons"));
+		vu_r->set_progress_texture(get_icon("BusVuFull","EditorIcons"));
+		scale->set_texture( get_icon("BusVuDb","EditorIcons"));
+
+		disabled_vu = get_icon("BusVuFrozen","EditorIcons");
+
+		prev_active=true;
+		update_bus();
+		set_process(true);
+	}
+
+	if (p_what==NOTIFICATION_DRAW) {
+
+		if (has_focus()) {
+			draw_style_box(get_stylebox("focus","Button"),Rect2(Vector2(),get_size()));
+		}
+	}
+
+	if (p_what==NOTIFICATION_PROCESS) {
+
+		float real_peak[2]={-100,-100};
+		bool activity_found=false;
+
+		int cc;
+		switch(AudioServer::get_singleton()->get_speaker_mode()) {
+			case AudioServer::SPEAKER_MODE_STEREO: cc = 1; break;
+			case AudioServer::SPEAKER_SURROUND_51: cc = 4; break;
+			case AudioServer::SPEAKER_SURROUND_71: cc = 5; break;
+		}
+
+		for(int i=0;i<cc;i++) {
+			if (AudioServer::get_singleton()->is_bus_channel_active(get_index(),i)) {
+				activity_found=true;
+				real_peak[0]=MAX(real_peak[0],AudioServer::get_singleton()->get_bus_peak_volume_left_db(get_index(),i));
+				real_peak[1]=MAX(real_peak[1],AudioServer::get_singleton()->get_bus_peak_volume_right_db(get_index(),i));
+			}
+		}
+
+
+		if (real_peak[0]>peak_l) {
+			peak_l = real_peak[0];
+		} else {
+			peak_l-=get_process_delta_time()*60.0;
+		}
+
+		if (real_peak[1]>peak_r) {
+			peak_r = real_peak[1];
+		} else {
+			peak_r-=get_process_delta_time()*60.0;
+
+		}
+
+		vu_l->set_value(peak_l);
+		vu_r->set_value(peak_r);
+
+		if (activity_found!=prev_active) {
+			if (activity_found) {
+				vu_l->set_over_texture(Ref<Texture>());
+				vu_r->set_over_texture(Ref<Texture>());
+			} else {
+				vu_l->set_over_texture(disabled_vu);
+				vu_r->set_over_texture(disabled_vu);
+
+			}
+
+			prev_active=activity_found;
+		}
+
+	}
+
+	if (p_what==NOTIFICATION_VISIBILITY_CHANGED) {
+
+		peak_l=-100;
+		peak_r=-100;
+		prev_active=true;
+
+		set_process(is_visible_in_tree());
+	}
+
+}
+
+void EditorAudioBus::update_send() {
+
+	send->clear();
+	if (get_index()==0) {
+		send->set_disabled(true);
+		send->set_text("Speakers");
+	} else {
+		send->set_disabled(false);
+		StringName current_send = AudioServer::get_singleton()->get_bus_send(get_index());
+		int current_send_index=0; //by default to master
+
+		for(int i=0;i<get_index();i++) {
+			StringName send_name = AudioServer::get_singleton()->get_bus_name(i);
+			send->add_item(send_name);
+			if (send_name==current_send) {
+				current_send_index=i;
+			}
+		}
+
+		send->select(current_send_index);
+	}
+}
+
+void EditorAudioBus::update_bus() {
+
+	if (updating_bus)
+		return;
+
+	updating_bus=true;
+
+	int index = get_index();
+
+	slider->set_value(AudioServer::get_singleton()->get_bus_volume_db(index));
+	track_name->set_text(AudioServer::get_singleton()->get_bus_name(index));
+	if (get_index()==0)
+		track_name->set_editable(false);
+
+	solo->set_pressed( AudioServer::get_singleton()->is_bus_solo(index));
+	mute->set_pressed( AudioServer::get_singleton()->is_bus_mute(index));
+	bypass->set_pressed( AudioServer::get_singleton()->is_bus_bypassing_effects(index));
+	// effects..
+	effects->clear();
+
+	TreeItem *root = effects->create_item();
+	for(int i=0;i<AudioServer::get_singleton()->get_bus_effect_count(index);i++) {
+
+		Ref<AudioEffect> afx = AudioServer::get_singleton()->get_bus_effect(index,i);
+
+		TreeItem *fx = effects->create_item(root);
+		fx->set_cell_mode(0,TreeItem::CELL_MODE_CHECK);
+		fx->set_editable(0,true);
+		fx->set_checked(0,AudioServer::get_singleton()->is_bus_effect_enabled(index,i));
+		fx->set_text(0,afx->get_name());
+		fx->set_metadata(0,i);
+
+	}
+
+	TreeItem *add = effects->create_item(root);
+	add->set_cell_mode(0,TreeItem::CELL_MODE_CUSTOM);
+	add->set_editable(0,true);
+	add->set_selectable(0,false);
+	add->set_text(0,"Add Effect");
+
+	update_send();
+
+	updating_bus=false;
+
+}
+
+
+void EditorAudioBus::_name_changed(const String& p_new_name) {
+
+	if (p_new_name==AudioServer::get_singleton()->get_bus_name(get_index()))
+		return;
+
+	String attempt=p_new_name;
+	int attempts=1;
+
+	while(true) {
+
+		bool name_free=true;
+		for(int i=0;i<AudioServer::get_singleton()->get_bus_count();i++) {
+
+			if (AudioServer::get_singleton()->get_bus_name(i)==attempt) {
+				name_free=false;
+				break;
+			}
+		}
+
+		if (name_free) {
+			break;
+		}
+
+		attempts++;
+		attempt=p_new_name+" "+itos(attempts);
+	}
+	updating_bus=true;
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+
+	StringName current = AudioServer::get_singleton()->get_bus_name(get_index());
+	ur->create_action("Rename Audio Bus");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_name",get_index(),attempt);
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_name",get_index(),current);
+
+	for(int i=0;i<AudioServer::get_singleton()->get_bus_count();i++) {
+		if (AudioServer::get_singleton()->get_bus_send(i)==current) {
+			ur->add_do_method(AudioServer::get_singleton(),"set_bus_send",i,attempt);
+			ur->add_undo_method(AudioServer::get_singleton(),"set_bus_send",i,current);
+		}
+	}
+
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+
+
+	ur->add_do_method(buses,"_update_sends");
+	ur->add_undo_method(buses,"_update_sends");
+	ur->commit_action();
+
+	updating_bus=false;
+
+}
+
+void EditorAudioBus::_volume_db_changed(float p_db){
+
+	if (updating_bus)
+		return;
+
+	updating_bus=true;
+
+	print_line("new volume: "+rtos(p_db));
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Change Audio Bus Volume",UndoRedo::MERGE_ENDS);
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_volume_db",get_index(),p_db);
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_volume_db",get_index(),AudioServer::get_singleton()->get_bus_volume_db(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+
+	updating_bus=false;
+
+}
+void EditorAudioBus::_solo_toggled(){
+
+	updating_bus=true;
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Toggle Audio Bus Solo");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_solo",get_index(),solo->is_pressed());
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_solo",get_index(),AudioServer::get_singleton()->is_bus_solo(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+
+	updating_bus=false;
+
+}
+void EditorAudioBus::_mute_toggled(){
+
+	updating_bus=true;
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Toggle Audio Bus Mute");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_mute",get_index(),mute->is_pressed());
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_mute",get_index(),AudioServer::get_singleton()->is_bus_mute(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+
+	updating_bus=false;
+
+}
+void EditorAudioBus::_bypass_toggled(){
+
+	updating_bus=true;
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Toggle Audio Bus Bypass Effects");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_bypass_effects",get_index(),bypass->is_pressed());
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_bypass_effects",get_index(),AudioServer::get_singleton()->is_bus_bypassing_effects(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+
+	updating_bus=false;
+
+
+}
+
+void EditorAudioBus::_send_selected(int p_which) {
+
+	updating_bus=true;
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Select Audio Bus Send");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_send",get_index(),send->get_item_text(p_which));
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_send",get_index(),AudioServer::get_singleton()->get_bus_send(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+
+	updating_bus=false;
+}
+
+void EditorAudioBus::_effect_selected() {
+
+	TreeItem *effect = effects->get_selected();
+	if (!effect)
+		return;
+	updating_bus=true;
+
+	if (effect->get_metadata(0)!=Variant()) {
+
+		int index = effect->get_metadata(0);
+		Ref<AudioEffect> effect = AudioServer::get_singleton()->get_bus_effect(get_index(),index);
+		if (effect.is_valid()) {
+			EditorNode::get_singleton()->push_item(effect.ptr());
+		}
+	}
+
+	updating_bus=false;
+
+}
+
+void EditorAudioBus::_effect_edited() {
+
+	if (updating_bus)
+		return;
+
+	TreeItem *effect = effects->get_edited();
+	if (!effect)
+		return;
+
+	if (effect->get_metadata(0)==Variant()) {
+		Rect2 area = effects->get_item_rect(effect);
+
+		effect_options->set_pos(effects->get_global_pos()+area.pos+Vector2(0,area.size.y));
+		effect_options->popup();
+		//add effect
+	} else  {
+		int index = effect->get_metadata(0);
+		updating_bus=true;
+
+		UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+		ur->create_action("Select Audio Bus Send");
+		ur->add_do_method(AudioServer::get_singleton(),"set_bus_effect_enabled",get_index(),index,effect->is_checked(0));
+		ur->add_undo_method(AudioServer::get_singleton(),"set_bus_effect_enabled",get_index(),index,AudioServer::get_singleton()->is_bus_effect_enabled(get_index(),index));
+		ur->add_do_method(buses,"_update_bus",get_index());
+		ur->add_undo_method(buses,"_update_bus",get_index());
+		ur->commit_action();
+
+		updating_bus=false;
+
+	}
+
+}
+
+void EditorAudioBus::_effect_add(int p_which) {
+
+	if (updating_bus)
+		return;
+
+	StringName name = effect_options->get_item_metadata(p_which);
+
+	Object *fx = ClassDB::instance(name);
+	ERR_FAIL_COND(!fx);
+	AudioEffect *afx = fx->cast_to<AudioEffect>();
+	ERR_FAIL_COND(!afx);
+	Ref<AudioEffect> afxr = Ref<AudioEffect>(afx);
+
+	afxr->set_name(effect_options->get_item_text(p_which));
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+	ur->create_action("Add Audio Bus Effect");
+	ur->add_do_method(AudioServer::get_singleton(),"add_bus_effect",get_index(),afxr,-1);
+	ur->add_undo_method(AudioServer::get_singleton(),"remove_bus_effect",get_index(),AudioServer::get_singleton()->get_bus_effect_count(get_index()));
+	ur->add_do_method(buses,"_update_bus",get_index());
+	ur->add_undo_method(buses,"_update_bus",get_index());
+	ur->commit_action();
+}
+
+void EditorAudioBus::_bind_methods() {
+
+	ClassDB::bind_method("update_bus",&EditorAudioBus::update_bus);
+	ClassDB::bind_method("update_send",&EditorAudioBus::update_send);
+	ClassDB::bind_method("_name_changed",&EditorAudioBus::_name_changed);
+	ClassDB::bind_method("_volume_db_changed",&EditorAudioBus::_volume_db_changed);
+	ClassDB::bind_method("_solo_toggled",&EditorAudioBus::_solo_toggled);
+	ClassDB::bind_method("_mute_toggled",&EditorAudioBus::_mute_toggled);
+	ClassDB::bind_method("_bypass_toggled",&EditorAudioBus::_bypass_toggled);
+	ClassDB::bind_method("_name_focus_exit",&EditorAudioBus::_name_focus_exit);
+	ClassDB::bind_method("_send_selected",&EditorAudioBus::_send_selected);
+	ClassDB::bind_method("_effect_edited",&EditorAudioBus::_effect_edited);
+	ClassDB::bind_method("_effect_selected",&EditorAudioBus::_effect_selected);
+	ClassDB::bind_method("_effect_add",&EditorAudioBus::_effect_add);
+}
+
+EditorAudioBus::EditorAudioBus(EditorAudioBuses *p_buses) {
+
+	buses=p_buses;
+	updating_bus=false;
+
+	VBoxContainer *vb = memnew( VBoxContainer );
+	add_child(vb);
+
+	set_v_size_flags(SIZE_EXPAND_FILL);
+
+	track_name = memnew( LineEdit );
+	vb->add_child(track_name);
+	track_name->connect("text_entered",this,"_name_changed");
+	track_name->connect("focus_exited",this,"_name_focus_exit");
+
+	HBoxContainer *hbc = memnew( HBoxContainer);
+	vb->add_child(hbc);
+	hbc->add_spacer();
+	solo = memnew( ToolButton );
+	solo->set_text("S");
+	solo->set_toggle_mode(true);
+	solo->set_modulate(Color(0.8,1.2,0.8));
+	solo->set_focus_mode(FOCUS_NONE);
+	solo->connect("pressed",this,"_solo_toggled");
+	hbc->add_child(solo);
+	mute = memnew( ToolButton );
+	mute->set_text("M");
+	mute->set_toggle_mode(true);
+	mute->set_modulate(Color(1.2,0.8,0.8));
+	mute->set_focus_mode(FOCUS_NONE);
+	mute->connect("pressed",this,"_mute_toggled");
+	hbc->add_child(mute);
+	bypass = memnew( ToolButton );
+	bypass->set_text("B");
+	bypass->set_toggle_mode(true);
+	bypass->set_modulate(Color(1.1,1.1,0.8));
+	bypass->set_focus_mode(FOCUS_NONE);
+	bypass->connect("pressed",this,"_bypass_toggled");
+	hbc->add_child(bypass);
+	hbc->add_spacer();
+
+	HBoxContainer *hb = memnew( HBoxContainer );
+	vb->add_child(hb);
+	slider = memnew( VSlider );
+	slider->set_min(-80);
+	slider->set_max(24);
+	slider->set_step(0.1);
+
+	slider->connect("value_changed",this,"_volume_db_changed");
+	hb->add_child(slider);
+	vu_l = memnew( TextureProgress );
+	vu_l->set_fill_mode(TextureProgress::FILL_BOTTOM_TO_TOP);
+	hb->add_child(vu_l);
+	vu_l->set_min(-80);
+	vu_l->set_max(24);
+	vu_l->set_step(0.1);
+
+	vu_r = memnew( TextureProgress );
+	vu_r->set_fill_mode(TextureProgress::FILL_BOTTOM_TO_TOP);
+	hb->add_child(vu_r);
+	vu_r->set_min(-80);
+	vu_r->set_max(24);
+	vu_r->set_step(0.1);
+
+	scale = memnew( TextureRect );
+	hb->add_child(scale);
+
+	add_child(hb);
+
+	effects = memnew( Tree );
+	effects->set_hide_root(true);
+	effects->set_custom_minimum_size(Size2(0,90)*EDSCALE);
+	effects->set_hide_folding(true);
+	vb->add_child(effects);
+	effects->connect("item_edited",this,"_effect_edited");
+	effects->connect("cell_selected",this,"_effect_selected");
+	effects->set_edit_checkbox_cell_only_when_checkbox_is_pressed(true);
+
+
+	send = memnew( OptionButton );
+	send->set_clip_text(true);
+	send->connect("item_selected",this,"_send_selected");
+	vb->add_child(send);
+
+	set_focus_mode(FOCUS_CLICK);
+
+	effect_options = memnew( PopupMenu );
+	effect_options->connect("index_pressed",this,"_effect_add");
+	add_child(effect_options);
+	List<StringName> effects;
+	ClassDB::get_inheriters_from_class("AudioEffect",&effects);
+	effects.sort_custom<StringName::AlphCompare>();
+	for (List<StringName>::Element *E=effects.front();E;E=E->next()) {
+		if (!ClassDB::can_instance(E->get()))
+			continue;
+
+		Ref<Texture> icon;
+		if (has_icon(E->get(),"EditorIcons")) {
+			icon = get_icon(E->get(),"EditorIcons");
+		}
+		String name = E->get().operator String().replace("AudioEffect","");
+		effect_options->add_item(name);
+		effect_options->set_item_metadata(effect_options->get_item_count()-1,E->get());
+		effect_options->set_item_icon(effect_options->get_item_count()-1,icon);
+	}
+
+
+}
+
+
+void EditorAudioBuses::_update_buses() {
+
+	while(bus_hb->get_child_count()>0) {
+		memdelete(bus_hb->get_child(0));
+	}
+
+	for(int i=0;i<AudioServer::get_singleton()->get_bus_count();i++) {
+
+		EditorAudioBus *audio_bus = memnew( EditorAudioBus(this) );
+		if (i==0) {
+			audio_bus->set_self_modulate(Color(1,0.9,0.9));
+		}
+		bus_hb->add_child(audio_bus);
+
+	}
+}
+
+void EditorAudioBuses::register_editor() {
+
+	EditorAudioBuses * audio_buses = memnew( EditorAudioBuses );
+	EditorNode::get_singleton()->add_bottom_panel_item("Audio",audio_buses);
+}
+
+void EditorAudioBuses::_notification(int p_what) {
+
+	if (p_what==NOTIFICATION_READY) {
+		_update_buses();
+	}
+}
+
+
+void EditorAudioBuses::_add_bus() {
+
+	UndoRedo *ur = EditorNode::get_singleton()->get_undo_redo();
+
+	//need to simulate new name, so we can undi :(
+	ur->create_action("Add Audio Bus");
+	ur->add_do_method(AudioServer::get_singleton(),"set_bus_count",AudioServer::get_singleton()->get_bus_count()+1);
+	ur->add_undo_method(AudioServer::get_singleton(),"set_bus_count",AudioServer::get_singleton()->get_bus_count());
+	ur->add_do_method(this,"_update_buses");
+	ur->add_undo_method(this,"_update_buses");
+	ur->commit_action();
+
+}
+
+void EditorAudioBuses::_update_bus(int p_index) {
+
+	if (p_index>=bus_hb->get_child_count())
+		return;
+
+	bus_hb->get_child(p_index)->call("update_bus");
+}
+
+void EditorAudioBuses::_update_sends() {
+
+	for(int i=0;i<bus_hb->get_child_count();i++) {
+		bus_hb->get_child(i)->call("update_send");
+	}
+}
+
+void EditorAudioBuses::_bind_methods() {
+
+	ClassDB::bind_method("_add_bus",&EditorAudioBuses::_add_bus);
+	ClassDB::bind_method("_update_buses",&EditorAudioBuses::_update_buses);
+	ClassDB::bind_method("_update_bus",&EditorAudioBuses::_update_bus);
+	ClassDB::bind_method("_update_sends",&EditorAudioBuses::_update_sends);
+}
+
+EditorAudioBuses::EditorAudioBuses()
+{
+
+	top_hb = memnew( HBoxContainer );
+	add_child(top_hb);
+
+	add = memnew( Button );
+	top_hb->add_child(add);;
+	add->set_text(TTR("Add"));
+
+	add->connect("pressed",this,"_add_bus");
+
+	Ref<ButtonGroup> bg;
+	bg.instance();
+
+	buses = memnew( ToolButton );
+	top_hb->add_child(buses);
+	buses->set_text(TTR("Buses"));
+	buses->set_button_group(bg);
+	buses->set_toggle_mode(true);
+	buses->set_pressed(true);
+
+	groups = memnew( ToolButton );
+	top_hb->add_child(groups);
+	groups->set_text(TTR("Groups"));
+	groups->set_button_group(bg);
+	groups->set_toggle_mode(true);
+
+	bus_scroll = memnew( ScrollContainer );
+	bus_scroll->set_v_size_flags(SIZE_EXPAND_FILL);
+	bus_scroll->set_enable_h_scroll(true);
+	bus_scroll->set_enable_v_scroll(false);
+	add_child(bus_scroll);
+	bus_hb = memnew( HBoxContainer );
+	bus_scroll->add_child(bus_hb);
+
+	group_scroll = memnew( ScrollContainer );
+	group_scroll->set_v_size_flags(SIZE_EXPAND_FILL);
+	group_scroll->set_enable_h_scroll(true);
+	group_scroll->set_enable_v_scroll(false);
+	add_child(group_scroll);
+	group_hb = memnew( HBoxContainer );
+	group_scroll->add_child(group_hb);
+
+	group_scroll->hide();
+
+
+	set_v_size_flags(SIZE_EXPAND_FILL);
+
+
+}
diff --git a/tools/editor/editor_audio_buses.h b/tools/editor/editor_audio_buses.h
new file mode 100644
index 0000000000..8787101393
--- /dev/null
+++ b/tools/editor/editor_audio_buses.h
@@ -0,0 +1,106 @@
+#ifndef EDITORAUDIOBUSES_H
+#define EDITORAUDIOBUSES_H
+
+
+#include "scene/gui/box_container.h"
+#include "scene/gui/button.h"
+#include "scene/gui/tool_button.h"
+#include "scene/gui/scroll_container.h"
+#include "scene/gui/panel_container.h"
+#include "scene/gui/slider.h"
+#include "scene/gui/texture_progress.h"
+#include "scene/gui/texture_rect.h"
+#include "scene/gui/line_edit.h"
+#include "scene/gui/tree.h"
+#include "scene/gui/option_button.h"
+
+class EditorAudioBuses;
+
+class EditorAudioBus : public PanelContainer {
+
+	GDCLASS( EditorAudioBus, PanelContainer )
+
+	bool prev_active;
+	float peak_l;
+	float peak_r;
+
+	Ref<Texture> disabled_vu;
+	LineEdit *track_name;
+	VSlider *slider;
+	TextureProgress *vu_l;
+	TextureProgress *vu_r;
+	TextureRect *scale;
+	OptionButton *send;
+
+	PopupMenu *effect_options;
+
+	Button *solo;
+	Button *mute;
+	Button *bypass;
+
+	Tree *effects;
+
+	bool updating_bus;
+
+	void _name_changed(const String& p_new_name);
+	void _name_focus_exit() { _name_changed(track_name->get_text()); }
+	void _volume_db_changed(float p_db);
+	void _solo_toggled();
+	void _mute_toggled();
+	void _bypass_toggled();
+	void _send_selected(int p_which);
+	void _effect_edited();
+	void _effect_add(int p_which);
+	void _effect_selected();
+
+friend class EditorAudioBuses;
+
+	EditorAudioBuses *buses;
+
+protected:
+
+	static void _bind_methods();
+	void _notification(int p_what);
+public:
+
+	void update_bus();
+	void update_send();
+
+	EditorAudioBus(EditorAudioBuses *p_buses=NULL);
+};
+
+
+class EditorAudioBuses : public VBoxContainer  {
+
+	GDCLASS(EditorAudioBuses,VBoxContainer)
+
+	HBoxContainer *top_hb;
+
+	Button *add;
+	ToolButton *buses;
+	ToolButton *groups;
+	ScrollContainer *bus_scroll;
+	HBoxContainer *bus_hb;
+	ScrollContainer *group_scroll;
+	HBoxContainer *group_hb;
+
+	void _add_bus();
+	void _update_buses();
+	void _update_bus(int p_index);
+	void _update_sends();
+
+
+protected:
+
+	static void _bind_methods();
+	void _notification(int p_what);
+public:
+
+
+
+	static void register_editor();
+
+	EditorAudioBuses();
+};
+
+#endif // EDITORAUDIOBUSES_H
diff --git a/tools/editor/editor_node.cpp b/tools/editor/editor_node.cpp
index 505e898336..9adb82a3b4 100644
--- a/tools/editor/editor_node.cpp
+++ b/tools/editor/editor_node.cpp
@@ -115,6 +115,7 @@
 
 #include "plugins/editor_preview_plugins.h"
 #include "editor_initialize_ssl.h"
+#include "editor_audio_buses.h"
 #include "script_editor_debugger.h"
 
 EditorNode *EditorNode::singleton=NULL;
@@ -1937,7 +1938,7 @@ void EditorNode::_run(bool p_current,const String& p_custom) {
 
 	List<String> breakpoints;
 	editor_data.get_editor_breakpoints(&breakpoints);
-    
+
 	args = GlobalConfig::get_singleton()->get("editor/main_run_args");
 
 	Error error = editor_run.run(run_filename,args,breakpoints,current_filename);
@@ -2802,10 +2803,10 @@ void EditorNode::_menu_option_confirm(int p_option,bool p_confirmed) {
 			update_menu->get_popup()->set_item_checked(1,true);
 			OS::get_singleton()->set_low_processor_usage_mode(true);
 		} break;
-        case SETTINGS_UPDATE_SPINNER_HIDE: {
+	case SETTINGS_UPDATE_SPINNER_HIDE: {
 			update_menu->set_icon(gui_base->get_icon("Collapse","EditorIcons"));
-            update_menu->get_popup()->toggle_item_checked(3);
-        } break;
+	    update_menu->get_popup()->toggle_item_checked(3);
+	} break;
 		case SETTINGS_PREFERENCES: {
 
 			settings_config_dialog->popup_edit_settings();
@@ -2930,16 +2931,7 @@ void EditorNode::_menu_option_confirm(int p_option,bool p_confirmed) {
 
 		default: {
 
-			if (p_option>=TOOL_MENU_BASE) {
-				int idx = p_option - TOOL_MENU_BASE;
-
-				if (tool_menu_items[idx].submenu != "")
-					break;
-
-				Object *handler = ObjectDB::get_instance(tool_menu_items[idx].handler);
-				ERR_FAIL_COND(!handler);
-				handler->call(tool_menu_items[idx].callback, tool_menu_items[idx].ud);
-			} else if (p_option>=OBJECT_METHOD_BASE) {
+			if (p_option>=OBJECT_METHOD_BASE) {
 
 				ERR_FAIL_COND(!current);
 
@@ -5274,100 +5266,6 @@ void EditorNode::add_plugin_init_callback(EditorPluginInitializeCallback p_callb
 
 EditorPluginInitializeCallback EditorNode::plugin_init_callbacks[EditorNode::MAX_INIT_CALLBACKS];
 
-void EditorNode::_tool_menu_insert_item(const ToolMenuItem& p_item) {
-
-	int idx = tool_menu_items.size();
-
-	String cat;
-	if (p_item.name.find("/") >= 0) {
-		cat = p_item.name.get_slice("/", 0);
-	} else {
-		idx = 0;
-		cat = "";
-	}
-
-	for (int i = tool_menu_items.size() - 1; i >= 0; i--) {
-		String name = tool_menu_items[i].name;
-
-		if (name.begins_with(cat) && (cat != "" || name.find("/") < 0)) {
-			idx = i + 1;
-			break;
-		}
-	}
-
-	tool_menu_items.insert(idx, p_item);
-}
-
-void EditorNode::_rebuild_tool_menu() const {
-
-	if (_initializing_tool_menu)
-		return;
-
-	PopupMenu *menu = tool_menu->get_popup();
-	menu->clear();
-
-	for (int i = 0; i < tool_menu_items.size(); i++) {
-		menu->add_item(tool_menu_items[i].name.get_slice("/", 1), TOOL_MENU_BASE + i);
-
-		if (tool_menu_items[i].submenu != "")
-			menu->set_item_submenu(i, tool_menu_items[i].submenu);
-	}
-}
-
-void EditorNode::add_tool_menu_item(const String& p_name, Object *p_handler, const String& p_callback, const Variant& p_ud) {
-
-	ERR_FAIL_COND(!p_handler);
-
-	ToolMenuItem tmi;
-	tmi.name = p_name;
-	tmi.submenu = "";
-	tmi.ud = p_ud;
-	tmi.handler = p_handler->get_instance_ID();
-	tmi.callback = p_callback;
-	_tool_menu_insert_item(tmi);
-
-	_rebuild_tool_menu();
-}
-
-void EditorNode::add_tool_submenu_item(const String& p_name, PopupMenu *p_submenu) {
-
-	ERR_FAIL_COND(!p_submenu);
-	ERR_FAIL_COND(p_submenu->get_parent() != NULL);
-
-	ToolMenuItem tmi;
-	tmi.name = p_name;
-	tmi.submenu = p_submenu->get_name();
-	tmi.ud = Variant();
-	tmi.handler = -1;
-	tmi.callback = "";
-	_tool_menu_insert_item(tmi);
-
-	tool_menu->get_popup()->add_child(p_submenu);
-
-	_rebuild_tool_menu();
-}
-
-void EditorNode::remove_tool_menu_item(const String& p_name) {
-
-	for (int i = 0; i < tool_menu_items.size(); i++) {
-		if (tool_menu_items[i].name == p_name) {
-			String submenu = tool_menu_items[i].submenu;
-
-			if (submenu != "") {
-				Node *n = tool_menu->get_popup()->get_node(submenu);
-
-				if (n) {
-					tool_menu->get_popup()->remove_child(n);
-					memdelete(n);
-				}
-			}
-
-			tool_menu_items.remove(i);
-		}
-	}
-
-	_rebuild_tool_menu();
-}
 
 int EditorNode::build_callback_count=0;
 
@@ -5511,8 +5409,6 @@ EditorNode::EditorNode() {
 	docks_visible = true;
 
 
-	_initializing_tool_menu = true;
-
 	FileAccess::set_backup_save(true);
 
 	PathRemap::get_singleton()->clear_remaps(); //editor uses no remaps
@@ -5972,9 +5868,10 @@ EditorNode::EditorNode() {
 
 	//tool_menu->set_icon(gui_base->get_icon("Save","EditorIcons"));
 	left_menu_hb->add_child( tool_menu );
-	tool_menu->get_popup()->connect("id_pressed", this, "_menu_option");
 
-	add_tool_menu_item(TTR("Orphan Resource Explorer"), this, "_menu_option", TOOLS_ORPHAN_RESOURCES);
+	p=tool_menu->get_popup();
+	p->connect("id_pressed",this,"_menu_option");
+	p->add_item(TTR("Orphan Resource Explorer"),TOOLS_ORPHAN_RESOURCES);
 
 	export_button = memnew( ToolButton );
 	export_button->set_tooltip(TTR("Export the project to many platforms."));
@@ -6658,6 +6555,9 @@ EditorNode::EditorNode() {
 	add_editor_plugin( memnew( SpatialEditorPlugin(this) ) );
 	add_editor_plugin( memnew( ScriptEditorPlugin(this) ) );
 
+
+	EditorAudioBuses::register_editor();
+
 	ScriptTextEditor::register_editor(); //register one for text scripts
 
 	if (StreamPeerSSL::is_available()) {
@@ -6855,8 +6755,7 @@ EditorNode::EditorNode() {
 		_initializing_addons=false;
 	}
 
-	_initializing_tool_menu = false;
-	_rebuild_tool_menu();
+
 
 	_load_docks();
 
diff --git a/tools/editor/editor_plugin.cpp b/tools/editor/editor_plugin.cpp
index 69be7f8a4d..2f59b0bd07 100644
--- a/tools/editor/editor_plugin.cpp
+++ b/tools/editor/editor_plugin.cpp
@@ -136,7 +136,7 @@ void EditorPlugin::add_control_to_container(CustomControlContainer p_location,Co
 
 void EditorPlugin::add_tool_menu_item(const String& p_name, Object *p_handler, const String& p_callback, const Variant& p_ud) {
 
-	EditorNode::get_singleton()->add_tool_menu_item(p_name, p_handler, p_callback, p_ud);
+	//EditorNode::get_singleton()->add_tool_menu_item(p_name, p_handler, p_callback, p_ud);
 }
 
 void EditorPlugin::add_tool_submenu_item(const String& p_name, Object *p_submenu) {
@@ -144,12 +144,12 @@ void EditorPlugin::add_tool_submenu_item(const String& p_name, Object *p_submenu
 	ERR_FAIL_NULL(p_submenu);
 	PopupMenu *submenu = p_submenu->cast_to<PopupMenu>();
 	ERR_FAIL_NULL(submenu);
-	EditorNode::get_singleton()->add_tool_submenu_item(p_name, submenu);
+	//EditorNode::get_singleton()->add_tool_submenu_item(p_name, submenu);
 }
 
 void EditorPlugin::remove_tool_menu_item(const String& p_name) {
 
-	EditorNode::get_singleton()->remove_tool_menu_item(p_name);
+	//EditorNode::get_singleton()->remove_tool_menu_item(p_name);
 }
 
 Ref<SpatialEditorGizmo> EditorPlugin::create_spatial_gizmo(Spatial* p_spatial) {
@@ -371,9 +371,9 @@ void EditorPlugin::_bind_methods() {
 	ClassDB::bind_method(_MD("add_control_to_dock","slot","control:Control"),&EditorPlugin::add_control_to_dock);
 	ClassDB::bind_method(_MD("remove_control_from_docks","control:Control"),&EditorPlugin::remove_control_from_docks);
 	ClassDB::bind_method(_MD("remove_control_from_bottom_panel","control:Control"),&EditorPlugin::remove_control_from_bottom_panel);
-	ClassDB::bind_method(_MD("add_tool_menu_item", "name", "handler", "callback", "ud"),&EditorPlugin::add_tool_menu_item,DEFVAL(Variant()));
+	//ClassDB::bind_method(_MD("add_tool_menu_item", "name", "handler", "callback", "ud"),&EditorPlugin::add_tool_menu_item,DEFVAL(Variant()));
 	ClassDB::bind_method(_MD("add_tool_submenu_item", "name", "submenu:PopupMenu"),&EditorPlugin::add_tool_submenu_item);
-	ClassDB::bind_method(_MD("remove_tool_menu_item", "name"),&EditorPlugin::remove_tool_menu_item);
+	//ClassDB::bind_method(_MD("remove_tool_menu_item", "name"),&EditorPlugin::remove_tool_menu_item);
 	ClassDB::bind_method(_MD("add_custom_type","type","base","script:Script","icon:Texture"),&EditorPlugin::add_custom_type);
 	ClassDB::bind_method(_MD("remove_custom_type","type"),&EditorPlugin::remove_custom_type);
 	ClassDB::bind_method(_MD("get_editor_viewport:Control"), &EditorPlugin::get_editor_viewport);
diff --git a/tools/editor/icons/icon_audio_effect_amplify.png b/tools/editor/icons/icon_audio_effect_amplify.png
new file mode 100644
index 0000000000..9af3227d40
--- /dev/null
+++ b/tools/editor/icons/icon_audio_effect_amplify.png
diff --git a/tools/editor/icons/icon_bus_vu_db.png b/tools/editor/icons/icon_bus_vu_db.png
new file mode 100644
index 0000000000..52507cae52
--- /dev/null
+++ b/tools/editor/icons/icon_bus_vu_db.png
diff --git a/tools/editor/icons/icon_bus_vu_empty.png b/tools/editor/icons/icon_bus_vu_empty.png
new file mode 100644
index 0000000000..6fc3143a55
--- /dev/null
+++ b/tools/editor/icons/icon_bus_vu_empty.png
diff --git a/tools/editor/icons/icon_bus_vu_frozen.png b/tools/editor/icons/icon_bus_vu_frozen.png
new file mode 100644
index 0000000000..cf128afa91
--- /dev/null
+++ b/tools/editor/icons/icon_bus_vu_frozen.png
diff --git a/tools/editor/icons/icon_bus_vu_full.png b/tools/editor/icons/icon_bus_vu_full.png
new file mode 100644
index 0000000000..9e3d7a93e3
--- /dev/null
+++ b/tools/editor/icons/icon_bus_vu_full.png
diff --git a/tools/editor/icons/icon_vu_db.png b/tools/editor/icons/icon_vu_db.png
new file mode 100644
index 0000000000..405a929e2a
--- /dev/null
+++ b/tools/editor/icons/icon_vu_db.png