Add support for AIFF files and other WAV formats

Author: DeeJayLSP

COPYRIGHT.txt

@@ -220,6 +220,11 @@ Comment: doctest
 Copyright: 2016-2023, Viktor Kirilov
 License: Expat
 
+Files: thirdparty/dr_libs/*
+Comment: dr_libs
+Copyright: 2024 David Reid
+License: public-domain or Unlicense or Expat
+
 Files: thirdparty/embree/*
 Comment: Embree
 Copyright: 2009-2021 Intel Corporation

doc/classes/AudioStreamWAV.xml

@@ -1,10 +1,10 @@
 <?xml version="1.0" encoding="UTF-8" ?>
 <class name="AudioStreamWAV" inherits="AudioStream" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="../class.xsd">
 	<brief_description>
-		Stores audio data loaded from WAV files.
+		Stores audio data loaded from WAV and AIFF files.
 	</brief_description>
 	<description>
-		AudioStreamWAV stores sound samples loaded from WAV files. To play the stored sound, use an [AudioStreamPlayer] (for non-positional audio) or [AudioStreamPlayer2D]/[AudioStreamPlayer3D] (for positional audio). The sound can be looped.
+		AudioStreamWAV stores sound samples loaded from WAV and AIFF files. To play the stored sound, use an [AudioStreamPlayer] (for non-positional audio) or [AudioStreamPlayer2D]/[AudioStreamPlayer3D] (for positional audio). The sound can be looped.
 		This class can also be used to store dynamically-generated PCM audio data. See also [AudioStreamGenerator] for procedural audio generation.
 	</description>
 	<tutorials>

doc/classes/ResourceImporterWAV.xml

@@ -1,11 +1,11 @@
 <?xml version="1.0" encoding="UTF-8" ?>
 <class name="ResourceImporterWAV" inherits="ResourceImporter" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="../class.xsd">
 	<brief_description>
-		Imports a WAV audio file for playback.
+		Imports WAV and AIFF audio files for playback.
 	</brief_description>
 	<description>
-		WAV is an uncompressed format, which can provide higher quality compared to Ogg Vorbis and MP3. It also has the lowest CPU cost to decode. This means high numbers of WAV sounds can be played at the same time, even on low-end devices.
-		By default, Godot imports WAV files using the lossy Quite OK Audio compression. You may change this by setting the [member compress/mode] property.
+		WAV and AIFF are uncompressed formats, which can provide higher quality compared to Ogg Vorbis and MP3. It also has the lowest CPU cost to decode. This means high numbers of WAV and AIFF sounds can be played at the same time, even on low-end devices.
+		By default, Godot imports WAV and AIFF files using the lossy Quite OK Audio compression. You may change this by setting the [member compress/mode] property.
 	</description>
 	<tutorials>
 		<link title="Importing audio samples">$DOCS_URL/tutorials/assets_pipeline/importing_audio_samples.html</link>
@@ -25,7 +25,7 @@
 		</member>
 		<member name="edit/loop_mode" type="int" setter="" getter="" default="0">
 			Controls how audio should loop.
-			- [b]Detect From WAV:[/b] Uses loop information from the WAV metadata.
+			- [b]Detect From WAV:[/b] Uses loop information from the WAV metadata (AIFF does not support this).
 			- [b]Disabled:[/b] Don't loop audio, even if the metadata indicates the file playback should loop.
 			- [b]Forward:[/b] Standard audio looping. Plays the audio forward from the beginning to [member edit/loop_end], then returns to [member edit/loop_begin] and repeats.
 			- [b]Ping-Pong:[/b] Plays the audio forward until [member edit/loop_end], then backwards to [member edit/loop_begin], repeating this cycle.

editor/import/resource_importer_wav.cpp

@@ -37,11 +37,15 @@ String ResourceImporterWAV::get_importer_name() const {
 }
 
 String ResourceImporterWAV::get_visible_name() const {
-	return "Microsoft WAV";
+	return "Microsoft WAV/Apple AIFF";
 }
 
 void ResourceImporterWAV::get_recognized_extensions(List<String> *p_extensions) const {
 	p_extensions->push_back("wav");
+	p_extensions->push_back("wave");
+	p_extensions->push_back("aif");
+	p_extensions->push_back("aiff");
+	p_extensions->push_back("aifc");
 }
 
 String ResourceImporterWAV::get_save_extension() const {

scene/resources/audio_stream_wav.cpp

@@ -28,10 +28,10 @@
 /* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                 */
 /**************************************************************************/
 
-#include "audio_stream_wav.h"
-
-#include "core/io/file_access_memory.h"
 #include "core/io/marshalls.h"
+#include "thirdparty/dr_libs/dr_wav_defines.h"
+
+#include "audio_stream_wav.h"
 
 const float TRIM_DB_LIMIT = -50;
 const int TRIM_FADE_OUT_FRAMES = 500;
@@ -724,221 +724,78 @@ Ref<AudioSample> AudioStreamWAV::generate_sample() const {
 }
 
 Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_stream_data, const Dictionary &p_options) {
-	// /* STEP 1, READ WAVE FILE */
-
-	Ref<FileAccessMemory> file;
-	file.instantiate();
-	Error err = file->open_custom(p_stream_data.ptr(), p_stream_data.size());
-	ERR_FAIL_COND_V_MSG(err != OK, Ref<AudioStreamWAV>(), "Cannot create memfile for WAV file buffer.");
-
-	/* CHECK RIFF */
-	char riff[5];
-	riff[4] = 0;
-	file->get_buffer((uint8_t *)&riff, 4); //RIFF
-
-	if (riff[0] != 'R' || riff[1] != 'I' || riff[2] != 'F' || riff[3] != 'F') {
-		ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), vformat("Not a WAV file. File should start with 'RIFF', but found '%s', in file of size %d bytes", riff, file->get_length()));
-	}
-
-	/* GET FILESIZE */
+	drwav wav;
 
-	// The file size in header is 8 bytes less than the actual size.
-	// See https://docs.fileformat.com/audio/wav/
-	const int FILE_SIZE_HEADER_OFFSET = 8;
-	uint32_t file_size_header = file->get_32() + FILE_SIZE_HEADER_OFFSET;
-	uint64_t file_size = file->get_length();
-	if (file_size != file_size_header) {
-		WARN_PRINT(vformat("File size %d is %s than the expected size %d.", file_size, file_size > file_size_header ? "larger" : "smaller", file_size_header));
+	if (!drwav_init_memory_with_metadata(&wav, p_stream_data.ptr(), p_stream_data.size(), DRWAV_WITH_METADATA, nullptr)) {
+		ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Audio data is invalid, corrupted, or an unsupported format.");
 	}
 
-	/* CHECK WAVE */
-
-	char wave[5];
-	wave[4] = 0;
-	file->get_buffer((uint8_t *)&wave, 4); //WAVE
-
-	if (wave[0] != 'W' || wave[1] != 'A' || wave[2] != 'V' || wave[3] != 'E') {
-		ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), vformat("Not a WAV file. Header should contain 'WAVE', but found '%s', in file of size %d bytes", wave, file->get_length()));
-	}
-
-	// Let users override potential loop points from the WAV.
-	// We parse the WAV loop points only with "Detect From WAV" (0).
-	int import_loop_mode = p_options["edit/loop_mode"];
-
-	int format_bits = 0;
-	int format_channels = 0;
+	return _load_from_dr_wav(wav, p_options);
+}
 
-	AudioStreamWAV::LoopMode loop_mode = AudioStreamWAV::LOOP_DISABLED;
-	uint16_t compression_code = 1;
-	bool format_found = false;
-	bool data_found = false;
-	int format_freq = 0;
-	int loop_begin = 0;
-	int loop_end = 0;
-	int frames = 0;
+Ref<AudioStreamWAV> AudioStreamWAV::load_from_file(const String &p_path, const Dictionary &p_options) {
+	Error err;
 
-	Vector<float> data;
+	Ref<FileAccess> file = FileAccess::open(p_path, FileAccess::READ, &err);
+	ERR_FAIL_COND_V_MSG(err != OK, Ref<AudioStreamWAV>(), vformat("Cannot open file '%s'.", p_path));
 
-	while (!file->eof_reached()) {
-		/* chunk */
-		char chunk_id[4];
-		file->get_buffer((uint8_t *)&chunk_id, 4); //RIFF
+	drwav_read_proc read_fa = [](void *p_user_data, void *p_out, size_t to_read) -> size_t {
+		Ref<FileAccess> lfile = *(Ref<FileAccess> *)p_user_data;
+		return lfile->get_buffer((uint8_t *)p_out, to_read);
+	};
 
-		/* 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
+	drwav_seek_proc seek_fa = [](void *p_user_data, int p_offset, drwav_seek_origin origin) -> drwav_bool32 {
+		Ref<FileAccess> lfile = *(Ref<FileAccess> *)p_user_data;
+		uint64_t new_offset = p_offset + (origin == drwav_seek_origin_current ? lfile->get_position() : 0);
 
-		if (file->eof_reached()) {
-			//ERR_PRINT("EOF REACH");
-			break;
+		if (new_offset > lfile->get_length()) {
+			return DRWAV_FALSE;
 		}
 
-		if (chunk_id[0] == 'f' && chunk_id[1] == 'm' && chunk_id[2] == 't' && chunk_id[3] == ' ' && !format_found) {
-			/* IS FORMAT CHUNK */
+		lfile->seek(new_offset);
+		return DRWAV_TRUE;
+	};
 
-			//Issue: #7755 : Not a bug - usage of other formats (format codes) are unsupported in current importer version.
-			//Consider revision for engine version 3.0
-			compression_code = file->get_16();
-			if (compression_code != 1 && compression_code != 3) {
-				ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM or IEEE float instead.");
-			}
+	drwav wav;
 
-			format_channels = file->get_16();
-			if (format_channels != 1 && format_channels != 2) {
-				ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Format not supported for WAVE file (not stereo or mono).");
-			}
-
-			format_freq = file->get_32(); //sampling rate
-
-			file->get_32(); // average bits/second (unused)
-			file->get_16(); // block align (unused)
-			format_bits = file->get_16(); // bits per sample
+	if (!drwav_init_with_metadata(&wav, read_fa, seek_fa, &file, DRWAV_WITH_METADATA, nullptr)) {
+		ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Cannot read data from file '" + p_path + "'. Data is invalid, corrupted, or an unsupported format.");
+	}
 
-			if (format_bits % 8 || format_bits == 0) {
-				ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Invalid amount of bits in the sample (should be one of 8, 16, 24 or 32).");
-			}
+	return _load_from_dr_wav(wav, p_options);
+}
 
-			if (compression_code == 3 && format_bits % 32) {
-				ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Invalid amount of bits in the IEEE float sample (should be 32 or 64).");
-			}
+Ref<AudioStreamWAV> AudioStreamWAV::_load_from_dr_wav(drwav &p_wav, const Dictionary &p_options) {
+	// STEP 1, READ DATA
 
-			/* Don't need anything else, continue */
-			format_found = true;
-		}
+	int format_bits = p_wav.bitsPerSample;
+	int format_channels = p_wav.channels;
+	int format_freq = p_wav.sampleRate;
+	int frames = p_wav.totalPCMFrameCount;
 
-		if (chunk_id[0] == 'd' && chunk_id[1] == 'a' && chunk_id[2] == 't' && chunk_id[3] == 'a' && !data_found) {
-			/* IS DATA CHUNK */
-			data_found = true;
+	int import_loop_mode = p_options["edit/loop_mode"];
 
-			if (!format_found) {
-				ERR_PRINT("'data' chunk before 'format' chunk found.");
+	int loop_begin = 0;
+	int loop_end = 0;
+	LoopMode loop_mode = LOOP_DISABLED;
+	if (import_loop_mode == 0) {
+		for (uint32_t meta = 0; meta < p_wav.metadataCount; ++meta) {
+			drwav_metadata md = p_wav.pMetadata[meta];
+			if (md.type == drwav_metadata_type_smpl && md.data.smpl.sampleLoopCount > 0) {
+				drwav_smpl_loop loop = md.data.smpl.pLoops[0];
+				loop_mode = (LoopMode)(loop.type + 1);
+				loop_begin = loop.firstSampleByteOffset;
+				loop_end = loop.lastSampleByteOffset;
 				break;
 			}
-
-			uint64_t remaining_bytes = file_size - file_pos;
-			frames = chunksize;
-			if (remaining_bytes < chunksize) {
-				WARN_PRINT("Data chunk size is smaller than expected. Proceeding with actual data size.");
-				frames = remaining_bytes;
-			}
-
-			ERR_FAIL_COND_V(format_channels == 0, Ref<AudioStreamWAV>());
-			frames /= format_channels;
-			frames /= (format_bits >> 3);
-
-			/*print_line("chunksize: "+itos(chunksize));
-			print_line("channels: "+itos(format_channels));
-			print_line("bits: "+itos(format_bits));
-			*/
-
-			data.resize(frames * format_channels);
-
-			if (compression_code == 1) {
-				if (format_bits == 8) {
-					for (int i = 0; i < frames * format_channels; i++) {
-						// 8 bit samples are UNSIGNED
-
-						data.write[i] = int8_t(file->get_8() - 128) / 128.f;
-					}
-				} else if (format_bits == 16) {
-					for (int i = 0; i < frames * format_channels; i++) {
-						//16 bit SIGNED
-
-						data.write[i] = int16_t(file->get_16()) / 32768.f;
-					}
-				} else {
-					for (int i = 0; i < frames * format_channels; i++) {
-						//16+ bits samples are SIGNED
-						// if sample is > 16 bits, just read extra bytes
-
-						uint32_t s = 0;
-						for (int b = 0; b < (format_bits >> 3); b++) {
-							s |= ((uint32_t)file->get_8()) << (b * 8);
-						}
-						s <<= (32 - format_bits);
-
-						data.write[i] = (int32_t(s) >> 16) / 32768.f;
-					}
-				}
-			} else if (compression_code == 3) {
-				if (format_bits == 32) {
-					for (int i = 0; i < frames * format_channels; i++) {
-						//32 bit IEEE Float
-
-						data.write[i] = file->get_float();
-					}
-				} else if (format_bits == 64) {
-					for (int i = 0; i < frames * format_channels; i++) {
-						//64 bit IEEE Float
-
-						data.write[i] = file->get_double();
-					}
-				}
-			}
-
-			// This is commented out due to some weird edge case seemingly in FileAccessMemory, doesn't seem to have any side effects though.
-			// if (file->eof_reached()) {
-			// 	ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Premature end of file.");
-			// }
 		}
-
-		if (import_loop_mode == 0 && chunk_id[0] == 's' && chunk_id[1] == 'm' && chunk_id[2] == 'p' && chunk_id[3] == 'l') {
-			// Loop point info!
-
-			/**
-			 *	Consider exploring next document:
-			 *		http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Docs/RIFFNEW.pdf
-			 *	Especially on page:
-			 *		16 - 17
-			 *	Timestamp:
-			 *		22:38 06.07.2017 GMT
-			 **/
-
-			for (int i = 0; i < 10; i++) {
-				file->get_32(); // i wish to know why should i do this... no doc!
-			}
-
-			// only read 0x00 (loop forward), 0x01 (loop ping-pong) and 0x02 (loop backward)
-			// Skip anything else because it's not supported, reserved for future uses or sampler specific
-			// from https://sites.google.com/site/musicgapi/technical-documents/wav-file-format#smpl (loop type values table)
-			int loop_type = file->get_32();
-			if (loop_type == 0x00 || loop_type == 0x01 || loop_type == 0x02) {
-				if (loop_type == 0x00) {
-					loop_mode = AudioStreamWAV::LOOP_FORWARD;
-				} else if (loop_type == 0x01) {
-					loop_mode = AudioStreamWAV::LOOP_PINGPONG;
-				} else if (loop_type == 0x02) {
-					loop_mode = AudioStreamWAV::LOOP_BACKWARD;
-				}
-				loop_begin = file->get_32();
-				loop_end = file->get_32();
-			}
-		}
-		// Move to the start of the next chunk. Note that RIFF requires a padding byte for odd
-		// chunk sizes.
-		file->seek(file_pos + chunksize + (chunksize & 1));
 	}
 
+	Vector<float> data;
+	data.resize(frames * format_channels);
+	drwav_read_pcm_frames_f32(&p_wav, frames, data.ptrw());
+	drwav_uninit(&p_wav);
+
 	// STEP 2, APPLY CONVERSIONS
 
 	bool is16 = format_bits != 8;
@@ -1176,12 +1033,6 @@ Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_st
 	return sample;
 }
 
-Ref<AudioStreamWAV> AudioStreamWAV::load_from_file(const String &p_path, const Dictionary &p_options) {
-	const Vector<uint8_t> stream_data = FileAccess::get_file_as_bytes(p_path);
-	ERR_FAIL_COND_V_MSG(stream_data.is_empty(), Ref<AudioStreamWAV>(), vformat("Cannot open file '%s'.", p_path));
-	return load_from_buffer(stream_data, p_options);
-}
-
 void AudioStreamWAV::_bind_methods() {
 	ClassDB::bind_static_method("AudioStreamWAV", D_METHOD("load_from_buffer", "stream_data", "options"), &AudioStreamWAV::load_from_buffer, DEFVAL(Dictionary()));
 	ClassDB::bind_static_method("AudioStreamWAV", D_METHOD("load_from_file", "path", "options"), &AudioStreamWAV::load_from_file, DEFVAL(Dictionary()));

scene/resources/audio_stream_wav.h

@@ -33,6 +33,7 @@
 
 #include "servers/audio/audio_stream.h"
 
+#include "thirdparty/dr_libs/dr_wav.h"
 #include "thirdparty/misc/qoa.h"
 
 class AudioStreamWAV;
@@ -138,6 +139,8 @@ class AudioStreamWAV : public AudioStream {
 	LocalVector<uint8_t> data;
 	uint32_t data_bytes = 0;
 
+	static Ref<AudioStreamWAV> _load_from_dr_wav(drwav &p_wav, const Dictionary &p_options);
+
 protected:
 	static void _bind_methods();