mirror of
https://github.com/godotengine/godot
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550 lines
18 KiB
C++
550 lines
18 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) 2014-present Godot Engine contributors (see AUTHORS.md). */
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/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
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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_wav.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 "AudioStreamWAV";
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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 AudioStreamWAV::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),QOA (Quite OK Audio)"), 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_MSG(ERR_FILE_UNRECOGNIZED, vformat("Not a WAV file. File should start with 'RIFF', but found '%s', in file of size %d bytes", riff, file->get_length()));
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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[5];
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wave[4] = 0;
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file->get_buffer((uint8_t *)&wave, 4); //WAVE
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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, vformat("Not a WAV file. Header should contain 'WAVE', but found '%s', in file of size %d bytes", wave, file->get_length()));
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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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AudioStreamWAV::LoopMode loop_mode = AudioStreamWAV::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 or IEEE float 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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if (compression_code == 3 && format_bits % 32) {
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ERR_FAIL_V_MSG(ERR_INVALID_DATA, "Invalid amount of bits in the IEEE float sample (should be 32 or 64).");
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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 (compression_code == 1) {
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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 == 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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} else if (compression_code == 3) {
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if (format_bits == 32) {
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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 == 64) {
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for (int i = 0; i < frames * format_channels; i++) {
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//64 bit IEEE Float
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data.write[i] = file->get_double();
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}
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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 = AudioStreamWAV::LOOP_FORWARD;
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} else if (loop_type == 0x01) {
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loop_mode = AudioStreamWAV::LOOP_PINGPONG;
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} else if (loop_type == 0x02) {
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loop_mode = AudioStreamWAV::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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// Move to the start of the next chunk. Note that RIFF requires a padding byte for odd
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// chunk sizes.
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file->seek(file_pos + chunksize + (chunksize & 1));
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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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// Cubic interpolation should be enough.
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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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new_data.write[i * format_channels + c] = Math::cubic_interpolate(y1, y2, y0, y3, frac);
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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 == AudioStreamWAV::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::db_to_linear(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 = (AudioStreamWAV::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;
|
|
}
|
|
|
|
Vector<uint8_t> pcm_data;
|
|
AudioStreamWAV::Format dst_format;
|
|
|
|
if (compression == 1) {
|
|
dst_format = AudioStreamWAV::FORMAT_IMA_ADPCM;
|
|
if (format_channels == 1) {
|
|
_compress_ima_adpcm(data, pcm_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();
|
|
pcm_data.resize(dl * 2);
|
|
|
|
uint8_t *w = pcm_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;
|
|
bool enforce16 = is16 || compression == 2;
|
|
pcm_data.resize(data.size() * (enforce16 ? 2 : 1));
|
|
{
|
|
uint8_t *w = pcm_data.ptrw();
|
|
|
|
int ds = data.size();
|
|
for (int i = 0; i < ds; i++) {
|
|
if (enforce16) {
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
Vector<uint8_t> dst_data;
|
|
if (compression == 2) {
|
|
dst_format = AudioStreamWAV::FORMAT_QOA;
|
|
qoa_desc desc = { 0, 0, 0, { { { 0 }, { 0 } } } };
|
|
uint32_t qoa_len = 0;
|
|
|
|
desc.samplerate = rate;
|
|
desc.samples = frames;
|
|
desc.channels = format_channels;
|
|
|
|
void *encoded = qoa_encode((short *)pcm_data.ptrw(), &desc, &qoa_len);
|
|
dst_data.resize(qoa_len);
|
|
memcpy(dst_data.ptrw(), encoded, qoa_len);
|
|
} else {
|
|
dst_data = pcm_data;
|
|
}
|
|
|
|
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() {
|
|
}
|