541 lines
17 KiB
C++
541 lines
17 KiB
C++
/*************************************************************************/
|
|
/* resource_importer_wav.cpp */
|
|
/*************************************************************************/
|
|
/* This file is part of: */
|
|
/* GODOT ENGINE */
|
|
/* https://godotengine.org */
|
|
/*************************************************************************/
|
|
/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
|
|
/* Copyright (c) 2014-2022 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 "core/io/file_access.h"
|
|
#include "core/io/marshalls.h"
|
|
#include "core/io/resource_saver.h"
|
|
#include "scene/resources/audio_stream_wav.h"
|
|
|
|
const float TRIM_DB_LIMIT = -50;
|
|
const int TRIM_FADE_OUT_FRAMES = 500;
|
|
|
|
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 "AudioStreamWAV";
|
|
}
|
|
|
|
bool ResourceImporterWAV::get_option_visibility(const String &p_path, const String &p_option, const HashMap<StringName, Variant> &p_options) const {
|
|
if (p_option == "force/max_rate_hz" && !bool(p_options["force/max_rate"])) {
|
|
return false;
|
|
}
|
|
|
|
// Don't show begin/end loop points if loop mode is auto-detected or disabled.
|
|
if ((int)p_options["edit/loop_mode"] < 2 && (p_option == "edit/loop_begin" || p_option == "edit/loop_end")) {
|
|
return false;
|
|
}
|
|
|
|
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(const String &p_path, 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", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_DEFAULT | PROPERTY_USAGE_UPDATE_ALL_IF_MODIFIED), false));
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::FLOAT, "force/max_rate_hz", PROPERTY_HINT_RANGE, "11025,192000,1,exp"), 44100));
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/trim"), false));
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/normalize"), false));
|
|
// Keep the `edit/loop_mode` enum in sync with AudioStreamWAV::LoopMode (note: +1 offset due to "Detect From WAV").
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_mode", PROPERTY_HINT_ENUM, "Detect From WAV,Disabled,Forward,Ping-Pong,Backward", PROPERTY_USAGE_DEFAULT | PROPERTY_USAGE_UPDATE_ALL_IF_MODIFIED), 0));
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_begin"), 0));
|
|
r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_end"), -1));
|
|
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 HashMap<StringName, Variant> &p_options, List<String> *r_platform_variants, List<String> *r_gen_files, Variant *r_metadata) {
|
|
/* STEP 1, READ WAVE FILE */
|
|
|
|
Error err;
|
|
Ref<FileAccess> file = FileAccess::open(p_source_file, FileAccess::READ, &err);
|
|
|
|
ERR_FAIL_COND_V_MSG(err != OK, ERR_CANT_OPEN, "Cannot open file '" + p_source_file + "'.");
|
|
|
|
/* 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') {
|
|
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') {
|
|
ERR_FAIL_V_MSG(ERR_FILE_UNRECOGNIZED, "Not a WAV file (no WAVE RIFF header).");
|
|
}
|
|
|
|
// Let users override potential loop points from the WAV.
|
|
// We parse the WAV loop points only with "Detect From WAV" (0).
|
|
int import_loop_mode = p_options["edit/loop_mode"];
|
|
|
|
int format_bits = 0;
|
|
int format_channels = 0;
|
|
|
|
AudioStreamWAV::LoopMode loop_mode = AudioStreamWAV::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_FAIL_V_MSG(ERR_INVALID_DATA, "Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM or IEEE float instead.");
|
|
}
|
|
|
|
format_channels = file->get_16();
|
|
if (format_channels != 1 && format_channels != 2) {
|
|
ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Format not supported for WAVE file (not stereo or mono).");
|
|
}
|
|
|
|
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 || format_bits == 0) {
|
|
ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Invalid amount of bits in the sample (should be one of 8, 16, 24 or 32).");
|
|
}
|
|
|
|
if (compression_code == 3 && format_bits % 32) {
|
|
ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Invalid amount of bits in the IEEE float sample (should be 32 or 64).");
|
|
}
|
|
|
|
/* 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 DATA CHUNK */
|
|
data_found = true;
|
|
|
|
if (!format_found) {
|
|
ERR_PRINT("'data' chunk before 'format' chunk found.");
|
|
break;
|
|
}
|
|
|
|
frames = chunksize;
|
|
|
|
if (format_channels == 0) {
|
|
ERR_FAIL_COND_V(format_channels == 0, ERR_INVALID_DATA);
|
|
}
|
|
frames /= format_channels;
|
|
frames /= (format_bits >> 3);
|
|
|
|
/*print_line("chunksize: "+itos(chunksize));
|
|
print_line("channels: "+itos(format_channels));
|
|
print_line("bits: "+itos(format_bits));
|
|
*/
|
|
|
|
data.resize(frames * format_channels);
|
|
|
|
if (compression_code == 1) {
|
|
if (format_bits == 8) {
|
|
for (int i = 0; i < frames * format_channels; i++) {
|
|
// 8 bit samples are UNSIGNED
|
|
|
|
data.write[i] = int8_t(file->get_8() - 128) / 128.f;
|
|
}
|
|
} else if (format_bits == 16) {
|
|
for (int i = 0; i < frames * format_channels; i++) {
|
|
//16 bit SIGNED
|
|
|
|
data.write[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.write[i] = (int32_t(s) >> 16) / 32768.f;
|
|
}
|
|
}
|
|
} else if (compression_code == 3) {
|
|
if (format_bits == 32) {
|
|
for (int i = 0; i < frames * format_channels; i++) {
|
|
//32 bit IEEE Float
|
|
|
|
data.write[i] = file->get_float();
|
|
}
|
|
} else if (format_bits == 64) {
|
|
for (int i = 0; i < frames * format_channels; i++) {
|
|
//64 bit IEEE Float
|
|
|
|
data.write[i] = file->get_double();
|
|
}
|
|
}
|
|
}
|
|
|
|
if (file->eof_reached()) {
|
|
ERR_FAIL_V_MSG(ERR_FILE_CORRUPT, "Premature end of file.");
|
|
}
|
|
}
|
|
|
|
if (import_loop_mode == 0 && 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!
|
|
}
|
|
|
|
// only read 0x00 (loop forward), 0x01 (loop ping-pong) and 0x02 (loop backward)
|
|
// Skip anything else because it's not supported, reserved for future uses or sampler specific
|
|
// from https://sites.google.com/site/musicgapi/technical-documents/wav-file-format#smpl (loop type values table)
|
|
int loop_type = file->get_32();
|
|
if (loop_type == 0x00 || loop_type == 0x01 || loop_type == 0x02) {
|
|
if (loop_type == 0x00) {
|
|
loop_mode = AudioStreamWAV::LOOP_FORWARD;
|
|
} else if (loop_type == 0x01) {
|
|
loop_mode = AudioStreamWAV::LOOP_PINGPONG;
|
|
} else if (loop_type == 0x02) {
|
|
loop_mode = AudioStreamWAV::LOOP_BACKWARD;
|
|
}
|
|
loop_begin = file->get_32();
|
|
loop_end = file->get_32();
|
|
}
|
|
}
|
|
file->seek(file_pos + chunksize);
|
|
}
|
|
|
|
// 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) {
|
|
// resample!
|
|
int new_data_frames = (int)(frames * (float)limit_rate_hz / (float)rate);
|
|
|
|
Vector<float> new_data;
|
|
new_data.resize(new_data_frames * format_channels);
|
|
for (int c = 0; c < format_channels; c++) {
|
|
float frac = .0f;
|
|
int ipos = 0;
|
|
|
|
for (int i = 0; i < new_data_frames; i++) {
|
|
//simple cubic interpolation should be enough.
|
|
|
|
float mu = frac;
|
|
|
|
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.write[i * format_channels + c] = res;
|
|
|
|
// update position and always keep fractional part within ]0...1]
|
|
// in order to avoid 32bit floating point precision errors
|
|
|
|
frac += (float)rate / (float)limit_rate_hz;
|
|
int tpos = (int)Math::floor(frac);
|
|
ipos += tpos;
|
|
frac -= tpos;
|
|
}
|
|
}
|
|
|
|
if (loop_mode) {
|
|
loop_begin = (int)(loop_begin * (float)new_data_frames / (float)frames);
|
|
loop_end = (int)(loop_end * (float)new_data_frames / (float)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.write[i] *= mult;
|
|
}
|
|
}
|
|
}
|
|
|
|
bool trim = p_options["edit/trim"];
|
|
|
|
if (trim && (loop_mode != AudioStreamWAV::LOOP_DISABLED) && format_channels > 0) {
|
|
int first = 0;
|
|
int last = (frames / format_channels) - 1;
|
|
bool found = false;
|
|
float limit = Math::db2linear(TRIM_DB_LIMIT);
|
|
|
|
for (int i = 0; i < data.size() / format_channels; i++) {
|
|
float ampChannelSum = 0;
|
|
for (int j = 0; j < format_channels; j++) {
|
|
ampChannelSum += Math::abs(data[(i * format_channels) + j]);
|
|
}
|
|
|
|
float amp = Math::abs(ampChannelSum / (float)format_channels);
|
|
|
|
if (!found && amp > limit) {
|
|
first = i;
|
|
found = true;
|
|
}
|
|
|
|
if (found && amp > limit) {
|
|
last = i;
|
|
}
|
|
}
|
|
|
|
if (first < last) {
|
|
Vector<float> new_data;
|
|
new_data.resize((last - first) * format_channels);
|
|
for (int i = first; i < last; i++) {
|
|
float fadeOutMult = 1;
|
|
|
|
if (last - i < TRIM_FADE_OUT_FRAMES) {
|
|
fadeOutMult = ((float)(last - i - 1) / (float)TRIM_FADE_OUT_FRAMES);
|
|
}
|
|
|
|
for (int j = 0; j < format_channels; j++) {
|
|
new_data.write[((i - first) * format_channels) + j] = data[(i * format_channels) + j] * fadeOutMult;
|
|
}
|
|
}
|
|
|
|
data = new_data;
|
|
frames = data.size() / format_channels;
|
|
}
|
|
}
|
|
|
|
if (import_loop_mode >= 2) {
|
|
loop_mode = (AudioStreamWAV::LoopMode)(import_loop_mode - 1);
|
|
loop_begin = p_options["edit/loop_begin"];
|
|
loop_end = p_options["edit/loop_end"];
|
|
// Wrap around to max frames, so `-1` can be used to select the end, etc.
|
|
if (loop_begin < 0) {
|
|
loop_begin = CLAMP(loop_begin + frames + 1, 0, frames);
|
|
}
|
|
if (loop_end < 0) {
|
|
loop_end = CLAMP(loop_end + frames + 1, 0, 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.write[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;
|
|
}
|
|
|
|
Vector<uint8_t> dst_data;
|
|
AudioStreamWAV::Format dst_format;
|
|
|
|
if (compression == 1) {
|
|
dst_format = AudioStreamWAV::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.write[i] = data[i * 2 + 0];
|
|
right.write[i] = data[i * 2 + 1];
|
|
}
|
|
|
|
Vector<uint8_t> bleft;
|
|
Vector<uint8_t> bright;
|
|
|
|
_compress_ima_adpcm(left, bleft);
|
|
_compress_ima_adpcm(right, bright);
|
|
|
|
int dl = bleft.size();
|
|
dst_data.resize(dl * 2);
|
|
|
|
uint8_t *w = dst_data.ptrw();
|
|
const uint8_t *rl = bleft.ptr();
|
|
const uint8_t *rr = bright.ptr();
|
|
|
|
for (int i = 0; i < dl; i++) {
|
|
w[i * 2 + 0] = rl[i];
|
|
w[i * 2 + 1] = rr[i];
|
|
}
|
|
}
|
|
|
|
} else {
|
|
dst_format = is16 ? AudioStreamWAV::FORMAT_16_BITS : AudioStreamWAV::FORMAT_8_BITS;
|
|
dst_data.resize(data.size() * (is16 ? 2 : 1));
|
|
{
|
|
uint8_t *w = dst_data.ptrw();
|
|
|
|
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<AudioStreamWAV> sample;
|
|
sample.instantiate();
|
|
sample->set_data(dst_data);
|
|
sample->set_format(dst_format);
|
|
sample->set_mix_rate(rate);
|
|
sample->set_loop_mode(loop_mode);
|
|
sample->set_loop_begin(loop_begin);
|
|
sample->set_loop_end(loop_end);
|
|
sample->set_stereo(format_channels == 2);
|
|
|
|
ResourceSaver::save(sample, p_save_path + ".sample");
|
|
|
|
return OK;
|
|
}
|
|
|
|
ResourceImporterWAV::ResourceImporterWAV() {
|
|
}
|