asgardius/virtualx-engine
1
0
Fork 0
Code Issues Pull requests Projects Releases 1 Packages Wiki Activity Actions 1
virtualx-engine/editor/import/resource_importer_wav.cpp

624 lines
17 KiB
C++
Raw Blame History

/*************************************************************************/
/* resource_importer_wav.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2018 Godot Engine contributors (cf. AUTHORS.md) */
/* */
/* 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 "io/marshalls.h"
#include "io/resource_saver.h"
#include "os/file_access.h"
#include "scene/resources/audio_stream_sample.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 "sample";
}
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 */
file->get_32(); // filesize
/* 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;
uint16_t compression_code = 1;
bool format_found = false;
bool data_found = false;
int format_freq = 0;
int loop_begin = 0;
int loop_end = 0;
int frames = 0;
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_position(); //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 */
//Issue: #7755 : Not a bug - usage of other formats (format codes) are unsupported in current importer version.
//Consider revision for engine version 3.0
compression_code = file->get_16();
if (compression_code != 1 && compression_code != 3) {
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;
}
/* Don't 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);
if (format_bits == 8) {
for (int i = 0; i < frames * format_channels; i++) {
// 8 bit samples are UNSIGNED
data[i] = int8_t(file->get_8() - 128) / 128.f;
}
} else if (format_bits == 32 && compression_code == 3) {
for (int i = 0; i < frames * format_channels; i++) {
//32 bit IEEE Float
data[i] = file->get_float();
}
} else if (format_bits == 16) {
for (int i = 0; i < frames * format_channels; i++) {
//16 bit SIGNED
data[i] = int16_t(file->get_16()) / 32768.f;
}
} else {
for (int i = 0; i < frames * format_channels; i++) {
//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);
data[i] = (int32_t(s) >> 16) / 32768.f;
}
}
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!
/**
* Consider exploring next document:
* http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Docs/RIFFNEW.pdf
* Especially on page:
* 16 - 17
* Timestamp:
* 22:38 06.07.2017 GMT
**/
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 && rate > 0 && frames > 0) {
//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 && format_channels > 0) {
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 + 1) * 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 + ".sample", 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() {
}
Reference in a new issue View git blame Copy permalink