/*************************************************************************/ /* resource_importer_wav.cpp */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2018 Godot Engine contributors (cf. AUTHORS.md) */ /* */ /* Permission is hereby granted, free of charge, to any person obtaining */ /* a copy of this software and associated documentation files (the */ /* "Software"), to deal in the Software without restriction, including */ /* without limitation the rights to use, copy, modify, merge, publish, */ /* distribute, sublicense, and/or sell copies of the Software, and to */ /* permit persons to whom the Software is furnished to do so, subject to */ /* the following conditions: */ /* */ /* The above copyright notice and this permission notice shall be */ /* included in all copies or substantial portions of the Software. */ /* */ /* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */ /* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */ /* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/ /* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */ /* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */ /* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */ /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ /*************************************************************************/ #include "resource_importer_wav.h" #include "core/io/marshalls.h" #include "core/io/resource_saver.h" #include "core/os/file_access.h" #include "scene/resources/audio_stream_sample.h" String ResourceImporterWAV::get_importer_name() const { return "wav"; } String ResourceImporterWAV::get_visible_name() const { return "Microsoft WAV"; } void ResourceImporterWAV::get_recognized_extensions(List<String> *p_extensions) const { p_extensions->push_back("wav"); } String ResourceImporterWAV::get_save_extension() const { return "sample"; } String ResourceImporterWAV::get_resource_type() const { return "AudioStreamSample"; } bool ResourceImporterWAV::get_option_visibility(const String &p_option, const Map<StringName, Variant> &p_options) const { return true; } int ResourceImporterWAV::get_preset_count() const { return 0; } String ResourceImporterWAV::get_preset_name(int p_idx) const { return String(); } void ResourceImporterWAV::get_import_options(List<ImportOption> *r_options, int p_preset) const { r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/8_bit"), false)); r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/mono"), false)); r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "force/max_rate"), false)); r_options->push_back(ImportOption(PropertyInfo(Variant::REAL, "force/max_rate_hz", PROPERTY_HINT_EXP_RANGE, "11025,192000,1"), 44100)); r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/trim"), true)); r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/normalize"), true)); r_options->push_back(ImportOption(PropertyInfo(Variant::BOOL, "edit/loop"), false)); r_options->push_back(ImportOption(PropertyInfo(Variant::INT, "compress/mode", PROPERTY_HINT_ENUM, "Disabled,RAM (Ima-ADPCM)"), 0)); } Error ResourceImporterWAV::import(const String &p_source_file, const String &p_save_path, const Map<StringName, Variant> &p_options, List<String> *r_platform_variants, List<String> *r_gen_files) { /* STEP 1, READ WAVE FILE */ Error err; FileAccess *file = FileAccess::open(p_source_file, FileAccess::READ, &err); ERR_FAIL_COND_V(err != OK, ERR_CANT_OPEN); /* CHECK RIFF */ char riff[5]; riff[4] = 0; file->get_buffer((uint8_t *)&riff, 4); //RIFF if (riff[0] != 'R' || riff[1] != 'I' || riff[2] != 'F' || riff[3] != 'F') { file->close(); memdelete(file); ERR_FAIL_V(ERR_FILE_UNRECOGNIZED); } /* GET FILESIZE */ file->get_32(); // filesize /* CHECK WAVE */ char wave[4]; file->get_buffer((uint8_t *)&wave, 4); //RIFF if (wave[0] != 'W' || wave[1] != 'A' || wave[2] != 'V' || wave[3] != 'E') { file->close(); memdelete(file); ERR_EXPLAIN("Not a WAV file (no WAVE RIFF Header)") ERR_FAIL_V(ERR_FILE_UNRECOGNIZED); } int format_bits = 0; int format_channels = 0; AudioStreamSample::LoopMode loop = AudioStreamSample::LOOP_DISABLED; uint16_t compression_code = 1; bool format_found = false; bool data_found = false; int format_freq = 0; int loop_begin = 0; int loop_end = 0; int frames = 0; Vector<float> data; while (!file->eof_reached()) { /* chunk */ char chunkID[4]; file->get_buffer((uint8_t *)&chunkID, 4); //RIFF /* chunk size */ uint32_t chunksize = file->get_32(); uint32_t file_pos = file->get_position(); //save file pos, so we can skip to next chunk safely if (file->eof_reached()) { //ERR_PRINT("EOF REACH"); break; } if (chunkID[0] == 'f' && chunkID[1] == 'm' && chunkID[2] == 't' && chunkID[3] == ' ' && !format_found) { /* IS FORMAT CHUNK */ //Issue: #7755 : Not a bug - usage of other formats (format codes) are unsupported in current importer version. //Consider revision for engine version 3.0 compression_code = file->get_16(); if (compression_code != 1 && compression_code != 3) { 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) and 0x01 (loop ping-pong) and skip anything else because // it's not supported (loop backward), 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 = loop_type ? AudioStreamSample::LOOP_PING_PONG : AudioStreamSample::LOOP_FORWARD; loop_begin = file->get_32(); loop_end = file->get_32(); } } file->seek(file_pos + chunksize); } file->close(); memdelete(file); // STEP 2, APPLY CONVERSIONS bool is16 = format_bits != 8; int rate = format_freq; /* print_line("Input Sample: "); print_line("\tframes: " + itos(frames)); print_line("\tformat_channels: " + itos(format_channels)); print_line("\t16bits: " + itos(is16)); print_line("\trate: " + itos(rate)); print_line("\tloop: " + itos(loop)); print_line("\tloop begin: " + itos(loop_begin)); print_line("\tloop end: " + itos(loop_end)); */ //apply frequency limit bool limit_rate = p_options["force/max_rate"]; int limit_rate_hz = p_options["force/max_rate_hz"]; if (limit_rate && rate > limit_rate_hz && rate > 0 && frames > 0) { // 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((float)-30); for (int i = 0; i < data.size(); i++) { float amp = Math::abs(data[i]); if (!found && amp > limit) { first = i; found = true; } if (found && amp > limit) { last = i; } } first /= format_channels; last /= format_channels; if (first < last) { Vector<float> new_data; new_data.resize((last - first + 1) * format_channels); for (int i = first * format_channels; i < (last + 1) * format_channels; i++) { new_data.write[i - first * format_channels] = data[i]; } data = new_data; frames = data.size() / format_channels; } } bool make_loop = p_options["edit/loop"]; if (make_loop && !loop) { loop = AudioStreamSample::LOOP_FORWARD; loop_begin = 0; loop_end = frames; } int compression = p_options["compress/mode"]; bool force_mono = p_options["force/mono"]; if (force_mono && format_channels == 2) { Vector<float> new_data; new_data.resize(data.size() / 2); for (int i = 0; i < frames; i++) { new_data.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() { }