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e1c1d7d1d7
godot/editor/import/resource_importer_wav.cpp
Rémi Verschelde e1c1d7d1d7 Add a bunch of missing Godot headers in own files
2017-03-05 15:47:28 +01:00

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/*************************************************************************/
/* resource_importer_wav.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 "resource_importer_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()
{
}
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