godot/tools/editor/import/resource_import_wav.cpp

#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()
{

}