746dddc067
* Map is unnecessary and inefficient in almost every case. * Replaced by the new HashMap. * Renamed Map to RBMap and Set to RBSet for cases that still make sense (order matters) but use is discouraged. There were very few cases where replacing by HashMap was undesired because keeping the key order was intended. I tried to keep those (as RBMap) as much as possible, but might have missed some. Review appreciated!
526 lines
17 KiB
C++
526 lines
17 KiB
C++
/*************************************************************************/
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/* resource_importer_wav.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "resource_importer_wav.h"
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#include "core/io/file_access.h"
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#include "core/io/marshalls.h"
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#include "core/io/resource_saver.h"
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#include "scene/resources/audio_stream_sample.h"
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const float TRIM_DB_LIMIT = -50;
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const int TRIM_FADE_OUT_FRAMES = 500;
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String ResourceImporterWAV::get_importer_name() const {
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return "wav";
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}
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String ResourceImporterWAV::get_visible_name() const {
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return "Microsoft WAV";
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}
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void ResourceImporterWAV::get_recognized_extensions(List<String> *p_extensions) const {
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p_extensions->push_back("wav");
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}
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String ResourceImporterWAV::get_save_extension() const {
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return "sample";
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}
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String ResourceImporterWAV::get_resource_type() const {
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return "AudioStreamSample";
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}
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bool ResourceImporterWAV::get_option_visibility(const String &p_path, const String &p_option, const HashMap<StringName, Variant> &p_options) const {
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if (p_option == "force/max_rate_hz" && !bool(p_options["force/max_rate"])) {
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return false;
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}
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// Don't show begin/end loop points if loop mode is auto-detected or disabled.
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if ((int)p_options["edit/loop_mode"] < 2 && (p_option == "edit/loop_begin" || p_option == "edit/loop_end")) {
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return false;
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}
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return true;
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}
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int ResourceImporterWAV::get_preset_count() const {
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return 0;
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}
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String ResourceImporterWAV::get_preset_name(int p_idx) const {
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return String();
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}
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void ResourceImporterWAV::get_import_options(const String &p_path, List<ImportOption> *r_options, int p_preset) const {
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r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/8_bit"), false));
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r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/mono"), false));
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r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/max_rate", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_DEFAULT | PROPERTY_USAGE_UPDATE_ALL_IF_MODIFIED), false));
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r_options->push_back(ImportOption(PropertyInfo(Variant::FLOAT, "force/max_rate_hz", PROPERTY_HINT_RANGE, "11025,192000,1,exp"), 44100));
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r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/trim"), false));
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r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/normalize"), false));
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// Keep the `edit/loop_mode` enum in sync with AudioStreamSample::LoopMode (note: +1 offset due to "Detect From WAV").
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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));
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r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_begin"), 0));
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r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "edit/loop_end"), -1));
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r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "compress/mode", PROPERTY_HINT_ENUM, "Disabled,RAM (Ima-ADPCM)"), 0));
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}
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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) {
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/* STEP 1, READ WAVE FILE */
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Error err;
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Ref<FileAccess> file = FileAccess::open(p_source_file, FileAccess::READ, &err);
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ERR_FAIL_COND_V_MSG(err != OK, ERR_CANT_OPEN, "Cannot open file '" + p_source_file + "'.");
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/* CHECK RIFF */
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char riff[5];
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riff[4] = 0;
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file->get_buffer((uint8_t *)&riff, 4); //RIFF
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if (riff[0] != 'R' || riff[1] != 'I' || riff[2] != 'F' || riff[3] != 'F') {
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ERR_FAIL_V(ERR_FILE_UNRECOGNIZED);
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}
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/* GET FILESIZE */
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file->get_32(); // filesize
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/* CHECK WAVE */
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char wave[4];
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file->get_buffer((uint8_t *)&wave, 4); //RIFF
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if (wave[0] != 'W' || wave[1] != 'A' || wave[2] != 'V' || wave[3] != 'E') {
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ERR_FAIL_V_MSG(ERR_FILE_UNRECOGNIZED, "Not a WAV file (no WAVE RIFF header).");
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}
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// Let users override potential loop points from the WAV.
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// We parse the WAV loop points only with "Detect From WAV" (0).
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int import_loop_mode = p_options["edit/loop_mode"];
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int format_bits = 0;
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int format_channels = 0;
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AudioStreamSample::LoopMode loop_mode = AudioStreamSample::LOOP_DISABLED;
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uint16_t compression_code = 1;
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bool format_found = false;
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bool data_found = false;
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int format_freq = 0;
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int loop_begin = 0;
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int loop_end = 0;
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int frames = 0;
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Vector<float> data;
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while (!file->eof_reached()) {
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/* chunk */
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char chunkID[4];
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file->get_buffer((uint8_t *)&chunkID, 4); //RIFF
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/* chunk size */
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uint32_t chunksize = file->get_32();
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uint32_t file_pos = file->get_position(); //save file pos, so we can skip to next chunk safely
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if (file->eof_reached()) {
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//ERR_PRINT("EOF REACH");
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break;
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}
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if (chunkID[0] == 'f' && chunkID[1] == 'm' && chunkID[2] == 't' && chunkID[3] == ' ' && !format_found) {
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/* IS FORMAT CHUNK */
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//Issue: #7755 : Not a bug - usage of other formats (format codes) are unsupported in current importer version.
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//Consider revision for engine version 3.0
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compression_code = file->get_16();
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if (compression_code != 1 && compression_code != 3) {
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ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM instead.");
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}
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format_channels = file->get_16();
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if (format_channels != 1 && format_channels != 2) {
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ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Format not supported for WAVE file (not stereo or mono).");
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}
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format_freq = file->get_32(); //sampling rate
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file->get_32(); // average bits/second (unused)
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file->get_16(); // block align (unused)
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format_bits = file->get_16(); // bits per sample
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if (format_bits % 8 || format_bits == 0) {
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ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Invalid amount of bits in the sample (should be one of 8, 16, 24 or 32).");
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}
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/* Don't need anything else, continue */
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format_found = true;
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}
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if (chunkID[0] == 'd' && chunkID[1] == 'a' && chunkID[2] == 't' && chunkID[3] == 'a' && !data_found) {
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/* IS DATA CHUNK */
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data_found = true;
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if (!format_found) {
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ERR_PRINT("'data' chunk before 'format' chunk found.");
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break;
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}
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frames = chunksize;
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if (format_channels == 0) {
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ERR_FAIL_COND_V(format_channels == 0, ERR_INVALID_DATA);
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}
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frames /= format_channels;
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frames /= (format_bits >> 3);
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/*print_line("chunksize: "+itos(chunksize));
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print_line("channels: "+itos(format_channels));
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print_line("bits: "+itos(format_bits));
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*/
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data.resize(frames * format_channels);
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if (format_bits == 8) {
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for (int i = 0; i < frames * format_channels; i++) {
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// 8 bit samples are UNSIGNED
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data.write[i] = int8_t(file->get_8() - 128) / 128.f;
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}
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} else if (format_bits == 32 && compression_code == 3) {
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for (int i = 0; i < frames * format_channels; i++) {
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//32 bit IEEE Float
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data.write[i] = file->get_float();
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}
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} else if (format_bits == 16) {
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for (int i = 0; i < frames * format_channels; i++) {
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//16 bit SIGNED
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data.write[i] = int16_t(file->get_16()) / 32768.f;
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}
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} else {
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for (int i = 0; i < frames * format_channels; i++) {
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//16+ bits samples are SIGNED
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// if sample is > 16 bits, just read extra bytes
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uint32_t s = 0;
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for (int b = 0; b < (format_bits >> 3); b++) {
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s |= ((uint32_t)file->get_8()) << (b * 8);
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}
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s <<= (32 - format_bits);
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data.write[i] = (int32_t(s) >> 16) / 32768.f;
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}
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}
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if (file->eof_reached()) {
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ERR_FAIL_V_MSG(ERR_FILE_CORRUPT, "Premature end of file.");
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}
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}
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if (import_loop_mode == 0 && chunkID[0] == 's' && chunkID[1] == 'm' && chunkID[2] == 'p' && chunkID[3] == 'l') {
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// Loop point info!
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/**
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* Consider exploring next document:
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* http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Docs/RIFFNEW.pdf
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* Especially on page:
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* 16 - 17
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* Timestamp:
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* 22:38 06.07.2017 GMT
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**/
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for (int i = 0; i < 10; i++) {
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file->get_32(); // i wish to know why should i do this... no doc!
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}
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// only read 0x00 (loop forward), 0x01 (loop ping-pong) and 0x02 (loop backward)
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// Skip anything else because it's not supported, reserved for future uses or sampler specific
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// from https://sites.google.com/site/musicgapi/technical-documents/wav-file-format#smpl (loop type values table)
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int loop_type = file->get_32();
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if (loop_type == 0x00 || loop_type == 0x01 || loop_type == 0x02) {
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if (loop_type == 0x00) {
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loop_mode = AudioStreamSample::LOOP_FORWARD;
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} else if (loop_type == 0x01) {
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loop_mode = AudioStreamSample::LOOP_PINGPONG;
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} else if (loop_type == 0x02) {
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loop_mode = AudioStreamSample::LOOP_BACKWARD;
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}
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loop_begin = file->get_32();
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loop_end = file->get_32();
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}
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}
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file->seek(file_pos + chunksize);
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}
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// STEP 2, APPLY CONVERSIONS
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bool is16 = format_bits != 8;
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int rate = format_freq;
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/*
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print_line("Input Sample: ");
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print_line("\tframes: " + itos(frames));
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print_line("\tformat_channels: " + itos(format_channels));
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print_line("\t16bits: " + itos(is16));
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print_line("\trate: " + itos(rate));
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print_line("\tloop: " + itos(loop));
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print_line("\tloop begin: " + itos(loop_begin));
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print_line("\tloop end: " + itos(loop_end));
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*/
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//apply frequency limit
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bool limit_rate = p_options["force/max_rate"];
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int limit_rate_hz = p_options["force/max_rate_hz"];
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if (limit_rate && rate > limit_rate_hz && rate > 0 && frames > 0) {
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// resample!
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int new_data_frames = (int)(frames * (float)limit_rate_hz / (float)rate);
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Vector<float> new_data;
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new_data.resize(new_data_frames * format_channels);
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for (int c = 0; c < format_channels; c++) {
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float frac = .0f;
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int ipos = 0;
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for (int i = 0; i < new_data_frames; i++) {
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//simple cubic interpolation should be enough.
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float mu = frac;
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float y0 = data[MAX(0, ipos - 1) * format_channels + c];
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float y1 = data[ipos * format_channels + c];
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float y2 = data[MIN(frames - 1, ipos + 1) * format_channels + c];
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float y3 = data[MIN(frames - 1, ipos + 2) * format_channels + c];
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float mu2 = mu * mu;
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float a0 = y3 - y2 - y0 + y1;
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float a1 = y0 - y1 - a0;
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float a2 = y2 - y0;
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float a3 = y1;
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float res = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3);
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new_data.write[i * format_channels + c] = res;
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// update position and always keep fractional part within ]0...1]
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// in order to avoid 32bit floating point precision errors
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frac += (float)rate / (float)limit_rate_hz;
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int tpos = (int)Math::floor(frac);
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ipos += tpos;
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frac -= tpos;
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}
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}
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if (loop_mode) {
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loop_begin = (int)(loop_begin * (float)new_data_frames / (float)frames);
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loop_end = (int)(loop_end * (float)new_data_frames / (float)frames);
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}
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data = new_data;
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rate = limit_rate_hz;
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frames = new_data_frames;
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}
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bool normalize = p_options["edit/normalize"];
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if (normalize) {
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float max = 0;
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for (int i = 0; i < data.size(); i++) {
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float amp = Math::abs(data[i]);
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if (amp > max) {
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max = amp;
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}
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}
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if (max > 0) {
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float mult = 1.0 / max;
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for (int i = 0; i < data.size(); i++) {
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data.write[i] *= mult;
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}
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}
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}
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bool trim = p_options["edit/trim"];
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if (trim && (loop_mode != AudioStreamSample::LOOP_DISABLED) && format_channels > 0) {
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int first = 0;
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int last = (frames / format_channels) - 1;
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bool found = false;
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float limit = Math::db2linear(TRIM_DB_LIMIT);
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for (int i = 0; i < data.size() / format_channels; i++) {
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float ampChannelSum = 0;
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for (int j = 0; j < format_channels; j++) {
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ampChannelSum += Math::abs(data[(i * format_channels) + j]);
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}
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float amp = Math::abs(ampChannelSum / (float)format_channels);
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if (!found && amp > limit) {
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first = i;
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found = true;
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}
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if (found && amp > limit) {
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last = i;
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}
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}
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if (first < last) {
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Vector<float> new_data;
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new_data.resize((last - first) * format_channels);
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for (int i = first; i < last; i++) {
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float fadeOutMult = 1;
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if (last - i < TRIM_FADE_OUT_FRAMES) {
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fadeOutMult = ((float)(last - i - 1) / (float)TRIM_FADE_OUT_FRAMES);
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}
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for (int j = 0; j < format_channels; j++) {
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new_data.write[((i - first) * format_channels) + j] = data[(i * format_channels) + j] * fadeOutMult;
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}
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}
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data = new_data;
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frames = data.size() / format_channels;
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}
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}
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if (import_loop_mode >= 2) {
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loop_mode = (AudioStreamSample::LoopMode)(import_loop_mode - 1);
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loop_begin = p_options["edit/loop_begin"];
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loop_end = p_options["edit/loop_end"];
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// Wrap around to max frames, so `-1` can be used to select the end, etc.
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if (loop_begin < 0) {
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loop_begin = CLAMP(loop_begin + frames + 1, 0, frames);
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}
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if (loop_end < 0) {
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loop_end = CLAMP(loop_end + frames + 1, 0, frames);
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}
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}
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int compression = p_options["compress/mode"];
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bool force_mono = p_options["force/mono"];
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if (force_mono && format_channels == 2) {
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Vector<float> new_data;
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new_data.resize(data.size() / 2);
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for (int i = 0; i < frames; i++) {
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new_data.write[i] = (data[i * 2 + 0] + data[i * 2 + 1]) / 2.0;
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}
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data = new_data;
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format_channels = 1;
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}
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bool force_8_bit = p_options["force/8_bit"];
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if (force_8_bit) {
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is16 = false;
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}
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Vector<uint8_t> dst_data;
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AudioStreamSample::Format dst_format;
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if (compression == 1) {
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dst_format = AudioStreamSample::FORMAT_IMA_ADPCM;
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if (format_channels == 1) {
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_compress_ima_adpcm(data, dst_data);
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} else {
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//byte interleave
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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 ? AudioStreamSample::FORMAT_16_BITS : AudioStreamSample::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<AudioStreamSample> 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(p_save_path + ".sample", sample);
|
|
|
|
return OK;
|
|
}
|
|
|
|
ResourceImporterWAV::ResourceImporterWAV() {
|
|
}
|