wav file importing!

This commit is contained in:
Juan Linietsky 2017-02-02 22:51:26 -03:00
parent 63673de247
commit a02933bb3c
6 changed files with 1341 additions and 0 deletions

View file

@ -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

View file

@ -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;
}
}

View file

@ -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

View file

@ -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();

View file

@ -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()
{
}

View file

@ -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