godot/editor/import/resource_importer_wav.cpp
Rémi Verschelde 6d16f2f053 Fix error macro calls not ending with semicolon
It's not necessary, but the vast majority of calls of error macros
do have an ending semicolon, so it's best to be consistent.
Most WARN_DEPRECATED calls did *not* have a semicolon, but there's
no reason for them to be treated differently.
2019-06-11 14:49:34 +02:00

552 lines
16 KiB
C++

/*************************************************************************/
/* resource_importer_wav.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2019 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/marshalls.h"
#include "core/io/resource_saver.h"
#include "core/os/file_access.h"
#include "scene/resources/audio_stream_sample.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 "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, Variant *r_metadata) {
/* 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) {
file->close();
memdelete(file);
ERR_EXPLAIN("Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM instead.");
ERR_FAIL_V(ERR_INVALID_DATA);
}
format_channels = file->get_16();
if (format_channels != 1 && format_channels != 2) {
file->close();
memdelete(file);
ERR_EXPLAIN("Format not supported for WAVE file (not stereo or mono).");
ERR_FAIL_V(ERR_INVALID_DATA);
}
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) {
file->close();
memdelete(file);
ERR_EXPLAIN("Invalid amount of bits in the sample (should be one of 8, 16, 24 or 32).");
ERR_FAIL_V(ERR_INVALID_DATA);
}
/* 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;
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.write[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.write[i] = file->get_float();
}
} 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;
}
}
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!
// 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 = AudioStreamSample::LOOP_FORWARD;
} else if (loop_type == 0x01) {
loop = AudioStreamSample::LOOP_PING_PONG;
} else if (loop_type == 0x02) {
loop = AudioStreamSample::LOOP_BACKWARD;
}
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) {
// 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) {
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 && 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;
}
}
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.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;
}
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.write[i] = data[i * 2 + 0];
right.write[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];
}
}
} 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;
}
ResourceImporterWAV::ResourceImporterWAV() {
}