wav file importing!

scene/register_scene_types.cpp

@@ -85,6 +85,7 @@
 #include "scene/gui/graph_node.h"
 #include "scene/gui/graph_edit.h"
 #include "scene/gui/tool_button.h"
+#include "scene/resources/audio_stream_sample.h"
 #include "scene/resources/video_stream.h"
 #include "scene/2d/particles_2d.h"
 #include "scene/2d/path_2d.h"
@@ -596,6 +597,7 @@ void register_scene_types() {
 
 	ClassDB::register_class<AudioPlayer>();
 	ClassDB::register_virtual_class<VideoStream>();
+	ClassDB::register_class<AudioStreamSample>();
 
 	OS::get_singleton()->yield(); //may take time to init
 

scene/resources/audio_stream_sample.cpp

@@ -0,0 +1,557 @@
+#include "audio_stream_sample.h"
+
+void AudioStreamPlaybackSample::start(float p_from_pos) {
+
+	for(int i=0;i<2;i++) {
+		ima_adpcm[i].step_index=0;
+		ima_adpcm[i].predictor=0;
+		ima_adpcm[i].loop_step_index=0;
+		ima_adpcm[i].loop_predictor=0;
+		ima_adpcm[i].last_nibble=-1;
+		ima_adpcm[i].loop_pos=0x7FFFFFFF;
+		ima_adpcm[i].window_ofs=0;
+		ima_adpcm[i].ptr=(const uint8_t*)base->data;
+		ima_adpcm[i].ptr+=AudioStreamSample::DATA_PAD;
+	}
+
+	seek_pos(p_from_pos);
+	sign=1;
+	active=true;
+}
+
+void AudioStreamPlaybackSample::stop() {
+
+	active=false;
+}
+
+bool AudioStreamPlaybackSample::is_playing() const {
+
+	return active;
+}
+
+int AudioStreamPlaybackSample::get_loop_count() const {
+
+	return 0;
+}
+
+float AudioStreamPlaybackSample::get_pos() const {
+
+	return float(offset>>MIX_FRAC_BITS)/base->mix_rate;
+}
+void AudioStreamPlaybackSample::seek_pos(float p_time) {
+
+	if (base->format==AudioStreamSample::FORMAT_IMA_ADPCM)
+		return; //no seeking in ima-adpcm
+
+	float max=get_length();
+	if (p_time<0) {
+		p_time=0;
+	} else if (p_time>=max) {
+		p_time=max-0.001;
+	}
+
+	offset = uint64_t(p_time * base->mix_rate)<<MIX_FRAC_BITS;
+}
+
+
+template<class Depth,bool is_stereo,bool is_ima_adpcm>
+void AudioStreamPlaybackSample::do_resample(const Depth* p_src, AudioFrame *p_dst,int64_t &offset,int32_t &increment,uint32_t amount,IMA_ADPCM_State *ima_adpcm) {
+
+	// this function will be compiled branchless by any decent compiler
+
+	int32_t final,final_r,next,next_r;
+	while (amount--) {
+
+		int64_t pos=offset >> MIX_FRAC_BITS;
+		if (is_stereo && !is_ima_adpcm)
+			pos<<=1;
+
+		if (is_ima_adpcm) {
+
+			int64_t sample_pos = pos + ima_adpcm[0].window_ofs;
+
+			while(sample_pos>ima_adpcm[0].last_nibble) {
+
+
+				static const int16_t _ima_adpcm_step_table[89] = {
+					7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+					19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+					50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+					130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+					337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+					876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+					2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+					5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+					15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+				};
+
+				static const int8_t _ima_adpcm_index_table[16] = {
+					-1, -1, -1, -1, 2, 4, 6, 8,
+					-1, -1, -1, -1, 2, 4, 6, 8
+				};
+
+				for(int i=0;i<(is_stereo?2:1);i++) {
+
+
+					int16_t nibble,diff,step;
+
+					ima_adpcm[i].last_nibble++;
+					const uint8_t *src_ptr=ima_adpcm[i].ptr;
+
+
+					uint8_t nbb = src_ptr[ (ima_adpcm[i].last_nibble>>1) *  (is_stereo?2:1) + i ];
+					nibble = (ima_adpcm[i].last_nibble&1)?(nbb>>4):(nbb&0xF);
+					step=_ima_adpcm_step_table[ima_adpcm[i].step_index];
+
+
+					ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
+					if (ima_adpcm[i].step_index<0)
+						ima_adpcm[i].step_index=0;
+					if (ima_adpcm[i].step_index>88)
+						ima_adpcm[i].step_index=88;
+
+					diff = step >> 3 ;
+					if (nibble & 1)
+						diff += step >> 2 ;
+					if (nibble & 2)
+						diff += step >> 1 ;
+					if (nibble & 4)
+						diff += step ;
+					if (nibble & 8)
+						diff = -diff ;
+
+					ima_adpcm[i].predictor+=diff;
+					if (ima_adpcm[i].predictor<-0x8000)
+						ima_adpcm[i].predictor=-0x8000;
+					else if (ima_adpcm[i].predictor>0x7FFF)
+						ima_adpcm[i].predictor=0x7FFF;
+
+
+					/* store loop if there */
+					if (ima_adpcm[i].last_nibble==ima_adpcm[i].loop_pos) {
+
+						ima_adpcm[i].loop_step_index = ima_adpcm[i].step_index;
+						ima_adpcm[i].loop_predictor = ima_adpcm[i].predictor;
+					}
+
+					//printf("%i - %i - pred %i\n",int(ima_adpcm[i].last_nibble),int(nibble),int(ima_adpcm[i].predictor));
+
+				}
+
+			}
+
+			final=ima_adpcm[0].predictor;
+			if (is_stereo) {
+				final_r=ima_adpcm[1].predictor;
+			}
+
+		} else {
+			final=p_src[pos];
+			if (is_stereo)
+				final_r=p_src[pos+1];
+
+			if (sizeof(Depth)==1) { /* conditions will not exist anymore when compiled! */
+				final<<=8;
+				if (is_stereo)
+					final_r<<=8;
+			}
+
+			if (is_stereo) {
+
+				next=p_src[pos+2];
+				next_r=p_src[pos+3];
+			} else {
+				next=p_src[pos+1];
+			}
+
+			if (sizeof(Depth)==1) {
+				next<<=8;
+				if (is_stereo)
+					next_r<<=8;
+			}
+
+			int32_t frac=int64_t(offset&MIX_FRAC_MASK);
+
+			final=final+((next-final)*frac >> MIX_FRAC_BITS);
+			if (is_stereo)
+				final_r=final_r+((next_r-final_r)*frac >> MIX_FRAC_BITS);
+
+		}
+
+
+		if (!is_stereo) {
+			final_r=final; //copy to right channel if stereo
+		}
+
+		p_dst->l=final/32767.0;
+		p_dst->r=final_r/32767.0;
+		p_dst++;
+
+		offset+=increment;
+	}
+}
+
+void AudioStreamPlaybackSample::mix(AudioFrame* p_buffer,float p_rate_scale,int p_frames) {
+
+	if (!base->data || !active) {
+		for(int i=0;i<p_frames;i++) {
+			p_buffer[i]=AudioFrame(0,0);
+		}
+		return;
+	}
+
+	int len = base->data_bytes;
+	switch(base->format) {
+		case AudioStreamSample::FORMAT_8_BITS: len/=1; break;
+		case AudioStreamSample::FORMAT_16_BITS: len/=2; break;
+		case AudioStreamSample::FORMAT_IMA_ADPCM: len*=2; break;
+	}
+
+	if (base->stereo) {
+		len/=2;
+	}
+
+	/* some 64-bit fixed point precaches */
+
+	int64_t loop_begin_fp=((int64_t)len<< MIX_FRAC_BITS);
+	int64_t loop_end_fp=((int64_t)base->loop_end << MIX_FRAC_BITS);
+	int64_t length_fp=((int64_t)len << MIX_FRAC_BITS);
+	int64_t begin_limit=(base->loop_mode!=AudioStreamSample::LOOP_DISABLED)?loop_begin_fp:0;
+	int64_t end_limit=(base->loop_mode!=AudioStreamSample::LOOP_DISABLED)?loop_end_fp:length_fp;
+	bool is_stereo=base->stereo;
+
+	int32_t todo=p_frames;
+
+	float base_rate = AudioServer::get_singleton()->get_mix_rate();
+	float srate = base->mix_rate;
+	srate*=p_rate_scale;
+	float fincrement = srate / base_rate;
+	int32_t increment = int32_t(fincrement * MIX_FRAC_LEN);
+	increment*=sign;
+
+
+	//looping
+
+	AudioStreamSample::LoopMode loop_format=base->loop_mode;
+	AudioStreamSample::Format format = base->format;
+
+
+	/* audio data */
+
+	uint8_t *dataptr=(uint8_t*)base->data;
+	const void *data=dataptr+AudioStreamSample::DATA_PAD;
+	AudioFrame *dst_buff=p_buffer;
+
+
+	if (format==AudioStreamSample::FORMAT_IMA_ADPCM) {
+
+		if (loop_format!=AudioStreamSample::LOOP_DISABLED) {
+			ima_adpcm[0].loop_pos=loop_begin_fp>>MIX_FRAC_BITS;
+			ima_adpcm[1].loop_pos=loop_begin_fp>>MIX_FRAC_BITS;
+			loop_format=AudioStreamSample::LOOP_FORWARD;
+		}
+	}
+
+	while (todo>0) {
+
+		int64_t limit=0;
+		int32_t target=0,aux=0;
+
+		/** LOOP CHECKING **/
+
+		if ( increment < 0 ) {
+			/* going backwards */
+
+			if (  loop_format!=AudioStreamSample::LOOP_DISABLED && offset < loop_begin_fp ) {
+				/* loopstart reached */
+				if ( loop_format==AudioStreamSample::LOOP_PING_PONG ) {
+					/* bounce ping pong */
+					offset= loop_begin_fp + ( loop_begin_fp-offset );
+					increment=-increment;
+					sign*=-1;
+				} else {
+					/* go to loop-end */
+					offset=loop_end_fp-(loop_begin_fp-offset);
+				}
+			} else {
+				/* check for sample not reaching begining */
+				if(offset < 0) {
+
+					active=false;
+					break;
+				}
+			}
+		} else {
+			/* going forward */
+			if(  loop_format!=AudioStreamSample::LOOP_DISABLED && offset >= loop_end_fp ) {
+				/* loopend reached */
+
+				if ( loop_format==AudioStreamSample::LOOP_PING_PONG ) {
+					/* bounce ping pong */
+					offset=loop_end_fp-(offset-loop_end_fp);
+					increment=-increment;
+					sign*=-1;
+				} else {
+					/* go to loop-begin */
+
+					if (format==AudioStreamSample::FORMAT_IMA_ADPCM) {
+						for(int i=0;i<2;i++) {
+							ima_adpcm[i].step_index=ima_adpcm[i].loop_step_index;
+							ima_adpcm[i].predictor=ima_adpcm[i].loop_predictor;
+							ima_adpcm[i].last_nibble=loop_begin_fp>>MIX_FRAC_BITS;
+						}
+						offset=loop_begin_fp;
+					} else {
+						offset=loop_begin_fp+(offset-loop_end_fp);
+					}
+
+				}
+			} else {
+				/* no loop, check for end of sample */
+				if(offset >= length_fp) {
+
+					active=false;
+					break;
+				}
+			}
+		}
+
+		/** MIXCOUNT COMPUTING **/
+
+		/* next possible limit (looppoints or sample begin/end */
+		limit=(increment < 0) ?begin_limit:end_limit;
+
+		/* compute what is shorter, the todo or the limit? */
+		aux=(limit-offset)/increment+1;
+		target=(aux<todo)?aux:todo; /* mix target is the shorter buffer */
+
+		/* check just in case */
+		if ( target<=0 ) {
+			active=false;
+			break;
+		}
+
+		todo-=target;
+
+		switch(base->format) {
+			case AudioStreamSample::FORMAT_8_BITS: {
+
+				if (is_stereo)
+					do_resample<int8_t,true,false>((int8_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+				else
+					do_resample<int8_t,false,false>((int8_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+			} break;
+			case AudioStreamSample::FORMAT_16_BITS: {
+				if (is_stereo)
+					do_resample<int16_t,true,false>((int16_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+				else
+					do_resample<int16_t,false,false>((int16_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+
+			} break;
+			case AudioStreamSample::FORMAT_IMA_ADPCM: {
+				if (is_stereo)
+					do_resample<int8_t,true,true>((int8_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+				else
+					do_resample<int8_t,false,true>((int8_t*)data,dst_buff,offset,increment,target,ima_adpcm);
+
+			} break;
+		}
+
+		dst_buff+=target;
+
+	}
+
+
+}
+
+float AudioStreamPlaybackSample::get_length() const {
+
+	int len = base->data_bytes;
+	switch(base->format) {
+		case AudioStreamSample::FORMAT_8_BITS: len/=1; break;
+		case AudioStreamSample::FORMAT_16_BITS: len/=2; break;
+		case AudioStreamSample::FORMAT_IMA_ADPCM: len*=2; break;
+	}
+
+	if (base->stereo) {
+		len/=2;
+	}
+
+
+	return float(len)/base->mix_rate;
+}
+
+
+AudioStreamPlaybackSample::AudioStreamPlaybackSample() {
+
+	active=false;
+	offset=0;
+	sign=1;
+}
+
+
+/////////////////////
+
+
+void AudioStreamSample::set_format(Format p_format) {
+
+	format=p_format;
+}
+
+AudioStreamSample::Format AudioStreamSample::get_format() const{
+
+	return format;
+}
+
+void AudioStreamSample::set_loop_mode(LoopMode p_loop_mode){
+
+	loop_mode=p_loop_mode;
+}
+AudioStreamSample::LoopMode AudioStreamSample::get_loop_mode() const{
+
+	return loop_mode;
+}
+
+void AudioStreamSample::set_loop_begin(int p_frame){
+
+	loop_begin=p_frame;
+}
+int AudioStreamSample::get_loop_begin() const{
+
+	return loop_begin;
+}
+
+void AudioStreamSample::set_loop_end(int p_frame){
+
+	loop_end=p_frame;
+}
+int AudioStreamSample::get_loop_end() const{
+
+	return loop_end;
+}
+
+
+void AudioStreamSample::set_mix_rate(int p_hz){
+
+	mix_rate=p_hz;
+}
+int AudioStreamSample::get_mix_rate() const{
+
+	return mix_rate;
+}
+void AudioStreamSample::set_stereo(bool p_enable){
+
+	stereo=p_enable;
+}
+bool AudioStreamSample::is_stereo() const{
+
+	return stereo;
+}
+
+void AudioStreamSample::set_data(const PoolVector<uint8_t>& p_data) {
+
+	AudioServer::get_singleton()->lock();
+	if (data) {
+		AudioServer::get_singleton()->audio_data_free(data);
+		data=NULL;
+		data_bytes=0;
+	}
+
+	int datalen = p_data.size();
+	if (datalen) {
+
+		PoolVector<uint8_t>::Read r = p_data.read();
+		int alloc_len = datalen+DATA_PAD*2;
+		data = AudioServer::get_singleton()->audio_data_alloc(alloc_len); //alloc with some padding for interpolation
+		zeromem(data,alloc_len);
+		uint8_t *dataptr=(uint8_t*)data;
+		copymem(dataptr+DATA_PAD,r.ptr(),datalen);
+		data_bytes=datalen;
+	}
+
+	AudioServer::get_singleton()->unlock();
+
+}
+PoolVector<uint8_t> AudioStreamSample::get_data() const{
+
+	PoolVector<uint8_t> pv;
+
+	if (data) {
+		pv.resize(data_bytes);
+		{
+
+			PoolVector<uint8_t>::Write w =pv.write();
+			copymem(w.ptr(),data,data_bytes);
+		}
+	}
+
+	return pv;
+}
+
+
+Ref<AudioStreamPlayback> AudioStreamSample::instance_playback() {
+
+	Ref<AudioStreamPlaybackSample> sample;
+	sample.instance();
+	sample->base=Ref<AudioStreamSample>(this);
+	return sample;
+}
+
+String AudioStreamSample::get_stream_name() const {
+
+	return "";
+}
+
+void AudioStreamSample::_bind_methods() {
+
+	ClassDB::bind_method(_MD("set_format","format"),&AudioStreamSample::set_format);
+	ClassDB::bind_method(_MD("get_format"),&AudioStreamSample::get_format);
+
+	ClassDB::bind_method(_MD("set_loop_mode","loop_mode"),&AudioStreamSample::set_loop_mode);
+	ClassDB::bind_method(_MD("get_loop_mode"),&AudioStreamSample::get_loop_mode);
+
+	ClassDB::bind_method(_MD("set_loop_begin","loop_begin"),&AudioStreamSample::set_loop_begin);
+	ClassDB::bind_method(_MD("get_loop_begin"),&AudioStreamSample::get_loop_begin);
+
+	ClassDB::bind_method(_MD("set_loop_end","loop_end"),&AudioStreamSample::set_loop_end);
+	ClassDB::bind_method(_MD("get_loop_end"),&AudioStreamSample::get_loop_end);
+
+	ClassDB::bind_method(_MD("set_mix_rate","mix_rate"),&AudioStreamSample::set_mix_rate);
+	ClassDB::bind_method(_MD("get_mix_rate"),&AudioStreamSample::get_mix_rate);
+
+	ClassDB::bind_method(_MD("set_stereo","stereo"),&AudioStreamSample::set_stereo);
+	ClassDB::bind_method(_MD("is_stereo"),&AudioStreamSample::is_stereo);
+
+	ClassDB::bind_method(_MD("set_data","data"),&AudioStreamSample::set_data);
+	ClassDB::bind_method(_MD("get_data"),&AudioStreamSample::get_data);
+
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"format",PROPERTY_HINT_ENUM,"8-Bit,16-Bit,IMA-ADPCM"),_SCS("set_format"),_SCS("get_format"));
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"loop_mode",PROPERTY_HINT_ENUM,"Disabled,Forward,Ping-Pong"),_SCS("set_loop_mode"),_SCS("get_loop_mode"));
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"loop_begin"),_SCS("set_loop_begin"),_SCS("get_loop_begin"));
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"loop_end"),_SCS("set_loop_end"),_SCS("get_loop_end"));
+	ADD_PROPERTY(PropertyInfo(Variant::INT,"mix_rate"),_SCS("set_mix_rate"),_SCS("get_mix_rate"));
+	ADD_PROPERTY(PropertyInfo(Variant::BOOL,"stereo"),_SCS("set_stereo"),_SCS("is_stereo"));
+	ADD_PROPERTY(PropertyInfo(Variant::POOL_BYTE_ARRAY,"data",PROPERTY_HINT_NONE,"",PROPERTY_USAGE_NOEDITOR),_SCS("set_data"),_SCS("get_data"));
+
+}
+
+AudioStreamSample::AudioStreamSample()
+{
+	format=FORMAT_8_BITS;
+	loop_mode=LOOP_DISABLED;
+	stereo=false;
+	loop_begin=0;
+	loop_end=0;
+	mix_rate=44100;
+	data=NULL;
+	data_bytes=0;
+}
+AudioStreamSample::~AudioStreamSample() {
+
+
+	if (data) {
+		AudioServer::get_singleton()->audio_data_free(data);
+		data=NULL;
+		data_bytes=0;
+	}
+}

scene/resources/audio_stream_sample.h

@@ -0,0 +1,128 @@
+#ifndef AUDIOSTREAMSAMPLE_H
+#define AUDIOSTREAMSAMPLE_H
+
+#include "servers/audio/audio_stream.h"
+
+
+class AudioStreamSample;
+
+class AudioStreamPlaybackSample : public AudioStreamPlayback {
+
+	GDCLASS( AudioStreamPlaybackSample, AudioStreamPlayback )
+	enum {
+		MIX_FRAC_BITS=13,
+		MIX_FRAC_LEN=(1<<MIX_FRAC_BITS),
+		MIX_FRAC_MASK=MIX_FRAC_LEN-1,
+	};
+
+	struct IMA_ADPCM_State {
+
+		int16_t step_index;
+		int32_t predictor;
+		/* values at loop point */
+		int16_t loop_step_index;
+		int32_t loop_predictor;
+		int32_t last_nibble;
+		int32_t loop_pos;
+		int32_t window_ofs;
+		const uint8_t *ptr;
+	} ima_adpcm[2];
+
+	int64_t offset;
+	int sign;
+	bool active;
+friend class AudioStreamSample;
+	Ref<AudioStreamSample> base;
+
+	template<class Depth,bool is_stereo,bool is_ima_adpcm>
+	void do_resample(const Depth* p_src, AudioFrame *p_dst,int64_t &offset,int32_t &increment,uint32_t amount,IMA_ADPCM_State *ima_adpcm);
+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 void mix(AudioFrame* p_buffer,float p_rate_scale,int p_frames);
+
+	virtual float get_length() const; //if supported, otherwise return 0
+
+
+	AudioStreamPlaybackSample();
+};
+
+class AudioStreamSample : public AudioStream {
+	GDCLASS(AudioStreamSample,AudioStream)
+	RES_BASE_EXTENSION("smp")
+
+public:
+
+	enum Format {
+		FORMAT_8_BITS,
+		FORMAT_16_BITS,
+		FORMAT_IMA_ADPCM
+	};
+
+	enum LoopMode {
+		LOOP_DISABLED,
+		LOOP_FORWARD,
+		LOOP_PING_PONG
+	};
+
+
+private:
+friend class AudioStreamPlaybackSample;
+
+	enum {
+		DATA_PAD=16 //padding for interpolation
+	};
+
+	Format format;
+	LoopMode loop_mode;
+	bool stereo;
+	int loop_begin;
+	int loop_end;
+	int mix_rate;
+	void *data;
+	uint32_t data_bytes;
+protected:
+
+	static void _bind_methods();
+public:
+	void set_format(Format p_format);
+	Format get_format() const;
+
+	void set_loop_mode(LoopMode p_loop_mode);
+	LoopMode get_loop_mode() const;
+
+	void set_loop_begin(int p_frame);
+	int get_loop_begin() const;
+
+	void set_loop_end(int p_frame);
+	int get_loop_end() const;
+
+	void set_mix_rate(int p_hz);
+	int get_mix_rate() const;
+
+	void set_stereo(bool p_enable);
+	bool is_stereo() const;
+
+	void set_data(const PoolVector<uint8_t>& p_data);
+	PoolVector<uint8_t> get_data() const;
+
+
+	virtual Ref<AudioStreamPlayback> instance_playback();
+	virtual String get_stream_name() const;
+
+	AudioStreamSample();
+	~AudioStreamSample();
+};
+
+VARIANT_ENUM_CAST(AudioStreamSample::Format)
+VARIANT_ENUM_CAST(AudioStreamSample::LoopMode)
+
+#endif // AUDIOSTREAMSample_H

tools/editor/editor_node.cpp

@@ -101,6 +101,7 @@
 #include "plugins/gi_probe_editor_plugin.h"
 #include "import/resource_import_texture.h"
 #include "import/resource_importer_csv_translation.h"
+#include "import/resource_import_wav.h"
 // end
 #include "editor_settings.h"
 #include "io_plugins/editor_texture_import_plugin.h"
@@ -5126,6 +5127,10 @@ EditorNode::EditorNode() {
 		import_csv_translation.instance();
 		ResourceFormatImporter::get_singleton()->add_importer(import_csv_translation);
 
+		Ref<ResourceImporterWAV> import_wav;
+		import_wav.instance();
+		ResourceFormatImporter::get_singleton()->add_importer(import_wav);
+
 	}
 
 	_pvrtc_register_compressors();

tools/editor/import/resource_import_wav.cpp

@@ -0,0 +1,619 @@
+#include "resource_import_wav.h"
+
+#include "scene/resources/audio_stream_sample.h"
+#include "os/file_access.h"
+#include "io/marshalls.h"
+#include "io/resource_saver.h"
+
+String ResourceImporterWAV::get_importer_name() const {
+
+	return "wav";
+}
+
+String ResourceImporterWAV::get_visible_name() const{
+
+	return "Microsoft WAV";
+}
+void ResourceImporterWAV::get_recognized_extensions(List<String> *p_extensions) const{
+
+	p_extensions->push_back("wav");
+}
+String ResourceImporterWAV::get_save_extension() const {
+	return "smp";
+}
+
+String ResourceImporterWAV::get_resource_type() const{
+
+	return "AudioStreamSample";
+}
+
+bool ResourceImporterWAV::get_option_visibility(const String& p_option,const Map<StringName,Variant>& p_options) const {
+
+	return true;
+}
+
+int ResourceImporterWAV::get_preset_count() const {
+	return 0;
+}
+String ResourceImporterWAV::get_preset_name(int p_idx) const {
+
+	return String();
+}
+
+
+void ResourceImporterWAV::get_import_options(List<ImportOption> *r_options,int p_preset) const {
+
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"force/8_bit"),false));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"force/mono"),false));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"force/max_rate"),false));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::REAL,"force/max_rate_hz",PROPERTY_HINT_EXP_RANGE,"11025,192000,1"),44100));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"edit/trim"),true));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"edit/normalize"),true));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL,"edit/loop"),false));
+	r_options->push_back(ImportOption(PropertyInfo(Variant::INT,"compress/mode",PROPERTY_HINT_ENUM,"Disabled,RAM (Ima-ADPCM)"),0));
+
+}
+
+
+Error ResourceImporterWAV::import(const String& p_source_file, const String& p_save_path, const Map<StringName,Variant>& p_options, List<String>* r_platform_variants, List<String> *r_gen_files) {
+
+	/* STEP 1, READ WAVE FILE */
+
+	Error err;
+	FileAccess *file=FileAccess::open(p_source_file, FileAccess::READ,&err);
+
+	ERR_FAIL_COND_V( err!=OK, ERR_CANT_OPEN );
+
+	/* CHECK RIFF */
+	char riff[5];
+	riff[4]=0;
+	file->get_buffer((uint8_t*)&riff,4); //RIFF
+
+	if (riff[0]!='R' || riff[1]!='I' || riff[2]!='F' || riff[3]!='F') {
+
+		file->close();
+		memdelete(file);
+		ERR_FAIL_V( ERR_FILE_UNRECOGNIZED );
+	}
+
+
+	/* GET FILESIZE */
+	uint32_t filesize=file->get_32();
+
+	/* CHECK WAVE */
+
+	char wave[4];
+
+	file->get_buffer((uint8_t*)&wave,4); //RIFF
+
+	if (wave[0]!='W' || wave[1]!='A' || wave[2]!='V' || wave[3]!='E') {
+
+
+		file->close();
+		memdelete(file);
+		ERR_EXPLAIN("Not a WAV file (no WAVE RIFF Header)")
+		ERR_FAIL_V( ERR_FILE_UNRECOGNIZED );
+	}
+
+	int format_bits=0;
+	int format_channels=0;
+
+	AudioStreamSample::LoopMode loop=AudioStreamSample::LOOP_DISABLED;
+	bool format_found=false;
+	bool data_found=false;
+	int format_freq=0;
+	int loop_begin=0;
+	int loop_end=0;
+	int frames;
+
+	Vector<float> data;
+
+	while (!file->eof_reached()) {
+
+
+		/* chunk */
+		char chunkID[4];
+		file->get_buffer((uint8_t*)&chunkID,4); //RIFF
+
+		/* chunk size */
+		uint32_t chunksize=file->get_32();
+		uint32_t file_pos=file->get_pos(); //save file pos, so we can skip to next chunk safely
+
+		if (file->eof_reached()) {
+
+			//ERR_PRINT("EOF REACH");
+			break;
+		}
+
+		if (chunkID[0]=='f' && chunkID[1]=='m' && chunkID[2]=='t' && chunkID[3]==' ' && !format_found) {
+			/* IS FORMAT CHUNK */
+
+			uint16_t compression_code=file->get_16();
+
+
+			if (compression_code!=1) {
+				ERR_PRINT("Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM instead.");
+				break;
+			}
+
+			format_channels=file->get_16();
+			if (format_channels!=1 && format_channels !=2) {
+
+				ERR_PRINT("Format not supported for WAVE file (not stereo or mono)");
+				break;
+
+			}
+
+			format_freq=file->get_32(); //sampling rate
+
+			file->get_32(); // average bits/second (unused)
+			file->get_16(); // block align (unused)
+			format_bits=file->get_16(); // bits per sample
+
+			if (format_bits%8) {
+
+				ERR_PRINT("Strange number of bits in sample (not 8,16,24,32)");
+				break;
+			}
+
+			/* Dont need anything else, continue */
+			format_found=true;
+		}
+
+
+		if (chunkID[0]=='d' && chunkID[1]=='a' && chunkID[2]=='t' && chunkID[3]=='a' && !data_found) {
+			/* IS FORMAT CHUNK */
+			data_found=true;
+
+			if (!format_found) {
+				ERR_PRINT("'data' chunk before 'format' chunk found.");
+				break;
+
+			}
+
+			frames=chunksize;
+
+			frames/=format_channels;
+			frames/=(format_bits>>3);
+
+			/*print_line("chunksize: "+itos(chunksize));
+			print_line("channels: "+itos(format_channels));
+			print_line("bits: "+itos(format_bits));
+*/
+
+			int len=frames;
+			if (format_channels==2)
+				len*=2;
+			if (format_bits>8)
+				len*=2;
+
+
+			data.resize(frames*format_channels);
+
+			for (int i=0;i<frames;i++) {
+
+
+				for (int c=0;c<format_channels;c++) {
+
+
+					if (format_bits==8) {
+						// 8 bit samples are UNSIGNED
+
+						uint8_t s = file->get_8();
+						s-=128;
+						int8_t *sp=(int8_t*)&s;
+
+						data[i*format_channels+c]=float(*sp)/128.0;
+
+					} else {
+						//16+ bits samples are SIGNED
+						// if sample is > 16 bits, just read extra bytes
+
+						uint32_t s=0;
+						for (int b=0;b<(format_bits>>3);b++) {
+
+							s|=((uint32_t)file->get_8())<<(b*8);
+						}
+						s<<=(32-format_bits);
+						int32_t ss=s;
+
+
+						data[i*format_channels+c]=(ss>>16)/32768.0;
+					}
+				}
+
+			}
+
+
+
+			if (file->eof_reached()) {
+				file->close();
+				memdelete(file);
+				ERR_EXPLAIN("Premature end of file.");
+				ERR_FAIL_V(ERR_FILE_CORRUPT);
+			}
+		}
+
+		if (chunkID[0]=='s' && chunkID[1]=='m' && chunkID[2]=='p' && chunkID[3]=='l') {
+			//loop point info!
+
+			for(int i=0;i<10;i++)
+				file->get_32(); // i wish to know why should i do this... no doc!
+
+			loop=file->get_32()?AudioStreamSample::LOOP_PING_PONG:AudioStreamSample::LOOP_FORWARD;
+			loop_begin=file->get_32();
+			loop_end=file->get_32();
+
+		}
+		file->seek( file_pos+chunksize );
+	}
+
+	file->close();
+	memdelete(file);
+
+	// STEP 2, APPLY CONVERSIONS
+
+
+	bool is16=format_bits!=8;
+	int rate=format_freq;
+
+	print_line("Input Sample: ");
+	print_line("\tframes: "+itos(frames));
+	print_line("\tformat_channels: "+itos(format_channels));
+	print_line("\t16bits: "+itos(is16));
+	print_line("\trate: "+itos(rate));
+	print_line("\tloop: "+itos(loop));
+	print_line("\tloop begin: "+itos(loop_begin));
+	print_line("\tloop end: "+itos(loop_end));
+
+
+	//apply frequency limit
+
+	bool limit_rate = p_options["force/max_rate"];
+	int limit_rate_hz = p_options["force/max_rate_hz"];
+	if (limit_rate && rate > limit_rate_hz) {
+		//resampleeee!!!
+		int new_data_frames = frames * limit_rate_hz / rate;
+		Vector<float> new_data;
+		new_data.resize( new_data_frames * format_channels );
+		for(int c=0;c<format_channels;c++) {
+
+			for(int i=0;i<new_data_frames;i++) {
+
+				//simple cubic interpolation should be enough.
+				float pos = float(i) * frames / new_data_frames;
+				float mu = pos-Math::floor(pos);
+				int ipos = int(Math::floor(pos));
+
+				float y0=data[MAX(0,ipos-1)*format_channels+c];
+				float y1=data[ipos*format_channels+c];
+				float y2=data[MIN(frames-1,ipos+1)*format_channels+c];
+				float y3=data[MIN(frames-1,ipos+2)*format_channels+c];
+
+				float mu2 = mu*mu;
+				float a0 = y3 - y2 - y0 + y1;
+				float a1 = y0 - y1 - a0;
+				float a2 = y2 - y0;
+				float a3 = y1;
+
+				float res=(a0*mu*mu2+a1*mu2+a2*mu+a3);
+
+				new_data[i*format_channels+c]=res;
+			}
+		}
+
+		if (loop) {
+
+			loop_begin=loop_begin*new_data_frames/frames;
+			loop_end=loop_end*new_data_frames/frames;
+		}
+		data=new_data;
+		rate=limit_rate_hz;
+		frames=new_data_frames;
+	}
+
+
+	bool normalize = p_options["edit/normalize"];
+
+	if (normalize) {
+
+		float max=0;
+		for(int i=0;i<data.size();i++) {
+
+			float amp = Math::abs(data[i]);
+			if (amp>max)
+				max=amp;
+		}
+
+		if (max>0) {
+
+			float mult=1.0/max;
+			for(int i=0;i<data.size();i++) {
+
+				data[i]*=mult;
+			}
+
+		}
+	}
+
+	bool trim = p_options["edit/trim"];
+
+	if (trim && !loop) {
+
+		int first=0;
+		int last=(frames*format_channels)-1;
+		bool found=false;
+		float limit = Math::db2linear((float)-30);
+		for(int i=0;i<data.size();i++) {
+			float amp = Math::abs(data[i]);
+
+			if (!found && amp > limit) {
+				first=i;
+				found=true;
+			}
+
+			if (found && amp > limit) {
+				last=i;
+			}
+		}
+
+		first/=format_channels;
+		last/=format_channels;
+
+		if (first<last) {
+
+			Vector<float> new_data;
+			new_data.resize((last-first+1)*format_channels);
+			for(int i=first*format_channels;i<=last*format_channels;i++) {
+				new_data[i-first*format_channels]=data[i];
+			}
+
+			data=new_data;
+			frames=data.size()/format_channels;
+		}
+
+	}
+
+	bool make_loop = p_options["edit/loop"];
+
+	if (make_loop && !loop) {
+
+		loop=AudioStreamSample::LOOP_FORWARD;
+		loop_begin=0;
+		loop_end=frames;
+	}
+
+	int compression = p_options["compress/mode"];
+	bool force_mono = p_options["force/mono"];
+
+
+	if (force_mono && format_channels==2) {
+
+		Vector<float> new_data;
+		new_data.resize(data.size()/2);
+		for(int i=0;i<frames;i++) {
+			new_data[i]=(data[i*2+0]+data[i*2+1])/2.0;
+		}
+
+		data=new_data;
+		format_channels=1;
+	}
+
+	bool force_8_bit = p_options["force/8_bit"];
+	if (force_8_bit) {
+
+		is16=false;
+	}
+
+
+	PoolVector<uint8_t> dst_data;
+	AudioStreamSample::Format dst_format;
+
+	if ( compression == 1) {
+
+		dst_format=AudioStreamSample::FORMAT_IMA_ADPCM;
+		if (format_channels==1) {
+			_compress_ima_adpcm(data,dst_data);
+		} else {
+
+			//byte interleave
+			Vector<float> left;
+			Vector<float> right;
+
+			int tframes = data.size()/2;
+			left.resize(tframes);
+			right.resize(tframes);
+
+			for(int i=0;i<tframes;i++) {
+				left[i]=data[i*2+0];
+				right[i]=data[i*2+1];
+			}
+
+			PoolVector<uint8_t> bleft;
+			PoolVector<uint8_t> bright;
+
+			_compress_ima_adpcm(left,bleft);
+			_compress_ima_adpcm(right,bright);
+
+			int dl = bleft.size();
+			dst_data.resize( dl *2 );
+
+			PoolVector<uint8_t>::Write w=dst_data.write();
+			PoolVector<uint8_t>::Read rl=bleft.read();
+			PoolVector<uint8_t>::Read rr=bright.read();
+
+			for(int i=0;i<dl;i++) {
+				w[i*2+0]=rl[i];
+				w[i*2+1]=rr[i];
+			}
+		}
+
+		//print_line("compressing ima-adpcm, resulting buffersize is "+itos(dst_data.size())+" from "+itos(data.size()));
+
+	} else {
+
+		dst_format=is16?AudioStreamSample::FORMAT_16_BITS:AudioStreamSample::FORMAT_8_BITS;
+		dst_data.resize( data.size() * (is16?2:1));
+		{
+			PoolVector<uint8_t>::Write w = dst_data.write();
+
+			int ds=data.size();
+			for(int i=0;i<ds;i++) {
+
+				if (is16) {
+					int16_t v = CLAMP(data[i]*32768,-32768,32767);
+					encode_uint16(v,&w[i*2]);
+				} else {
+					int8_t v = CLAMP(data[i]*128,-128,127);
+					w[i]=v;
+				}
+			}
+		}
+	}
+
+
+	Ref<AudioStreamSample> sample;
+	sample.instance();
+	sample->set_data(dst_data);
+	sample->set_format(dst_format);
+	sample->set_mix_rate(rate);
+	sample->set_loop_mode(loop);
+	sample->set_loop_begin(loop_begin);
+	sample->set_loop_end(loop_end);
+	sample->set_stereo(format_channels==2);
+
+	ResourceSaver::save(p_save_path+".smp",sample);
+
+
+	return OK;
+
+}
+
+void ResourceImporterWAV::_compress_ima_adpcm(const Vector<float>& p_data,PoolVector<uint8_t>& dst_data) {
+
+
+	/*p_sample_data->data = (void*)malloc(len);
+	xm_s8 *dataptr=(xm_s8*)p_sample_data->data;*/
+
+	static const int16_t _ima_adpcm_step_table[89] = {
+		7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+		19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+		50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+		130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+		337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+		876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+		2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+		5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+		15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+	};
+
+	static const int8_t _ima_adpcm_index_table[16] = {
+		-1, -1, -1, -1, 2, 4, 6, 8,
+		-1, -1, -1, -1, 2, 4, 6, 8
+	};
+
+
+	int datalen = p_data.size();
+	int datamax=datalen;
+	if (datalen&1)
+		datalen++;
+
+	dst_data.resize(datalen/2+4);
+	PoolVector<uint8_t>::Write w = dst_data.write();
+
+
+	int i,step_idx=0,prev=0;
+	uint8_t *out = w.ptr();
+	//int16_t xm_prev=0;
+	const float *in=p_data.ptr();
+
+
+	/* initial value is zero */
+	*(out++) =0;
+	*(out++) =0;
+	/* Table index initial value */
+	*(out++) =0;
+	/* unused */
+	*(out++) =0;
+
+	for (i=0;i<datalen;i++) {
+		int step,diff,vpdiff,mask;
+		uint8_t nibble;
+		int16_t xm_sample;
+
+		if (i>=datamax)
+			xm_sample=0;
+		else {
+
+
+			xm_sample=CLAMP(in[i]*32767.0,-32768,32767);
+			/*
+			if (xm_sample==32767 || xm_sample==-32768)
+				printf("clippy!\n",xm_sample);
+			*/
+		}
+
+		//xm_sample=xm_sample+xm_prev;
+		//xm_prev=xm_sample;
+
+		diff = (int)xm_sample - prev ;
+
+		nibble=0 ;
+		step =  _ima_adpcm_step_table[ step_idx ];
+		vpdiff = step >> 3 ;
+		if (diff < 0) {
+			nibble=8;
+			diff=-diff ;
+		}
+		mask = 4 ;
+		while (mask) {
+
+			if (diff >= step) {
+
+				nibble |= mask;
+				diff -= step;
+				vpdiff += step;
+			}
+
+			step >>= 1 ;
+			mask >>= 1 ;
+		};
+
+		if (nibble&8)
+			prev-=vpdiff ;
+		else
+			prev+=vpdiff ;
+
+		if (prev > 32767) {
+			//printf("%i,xms %i, prev %i,diff %i, vpdiff %i, clip up %i\n",i,xm_sample,prev,diff,vpdiff,prev);
+			prev=32767;
+		} else if (prev < -32768) {
+			//printf("%i,xms %i, prev %i,diff %i, vpdiff %i, clip down %i\n",i,xm_sample,prev,diff,vpdiff,prev);
+			prev = -32768 ;
+		}
+
+		step_idx += _ima_adpcm_index_table[nibble];
+		if (step_idx< 0)
+			step_idx= 0 ;
+		else if (step_idx> 88)
+			step_idx= 88 ;
+
+
+		if (i&1) {
+			*out|=nibble<<4;
+			out++;
+		} else {
+			*out=nibble;
+		}
+		/*dataptr[i]=prev>>8;*/
+	}
+
+
+
+
+}
+
+ResourceImporterWAV::ResourceImporterWAV()
+{
+
+}

tools/editor/import/resource_import_wav.h

@@ -0,0 +1,30 @@
+#ifndef RESOURCEIMPORTWAV_H
+#define RESOURCEIMPORTWAV_H
+
+
+#include "io/resource_import.h"
+
+class ResourceImporterWAV : public ResourceImporter {
+	GDCLASS(ResourceImporterWAV,ResourceImporter)
+public:
+	virtual String get_importer_name() const;
+	virtual String get_visible_name() const;
+	virtual void get_recognized_extensions(List<String> *p_extensions) const;
+	virtual String get_save_extension() const;
+	virtual String get_resource_type() const;
+
+
+	virtual int get_preset_count() const;
+	virtual String get_preset_name(int p_idx) const;
+
+	virtual void get_import_options(List<ImportOption> *r_options,int p_preset=0) const;
+	virtual bool get_option_visibility(const String& p_option,const Map<StringName,Variant>& p_options) const;
+
+	void _compress_ima_adpcm(const Vector<float>& p_data,PoolVector<uint8_t>& dst_data);
+
+	virtual Error import(const String& p_source_file,const String& p_save_path,const Map<StringName,Variant>& p_options,List<String>* r_platform_variants,List<String>* r_gen_files=NULL);
+
+	ResourceImporterWAV();
+};
+
+#endif // RESOURCEIMPORTWAV_H