Add runtime file loading to AudioStreamWAV

scene/resources/audio_stream_wav.cpp

@@ -30,9 +30,12 @@
 
 #include "audio_stream_wav.h"
 
-#include "core/io/file_access.h"
+#include "core/io/file_access_memory.h"
 #include "core/io/marshalls.h"
 
+const float TRIM_DB_LIMIT = -50;
+const int TRIM_FADE_OUT_FRAMES = 500;
+
 void AudioStreamPlaybackWAV::start(double p_from_pos) {
 	if (base->format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
 		//no seeking in IMA_ADPCM
@@ -721,6 +724,9 @@ Ref<AudioSample> AudioStreamWAV::generate_sample() const {
 }
 
 void AudioStreamWAV::_bind_methods() {
+	ClassDB::bind_static_method("AudioStreamWAV", D_METHOD("load_from_file", "path", "options"), &AudioStreamWAV::load_from_file, DEFVAL(Dictionary()));
+	ClassDB::bind_static_method("AudioStreamWAV", D_METHOD("load_from_buffer", "buffer", "options"), &AudioStreamWAV::load_from_buffer, DEFVAL(Dictionary()));
+
 	ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamWAV::set_data);
 	ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamWAV::get_data);
 
@@ -763,6 +769,477 @@ void AudioStreamWAV::_bind_methods() {
 	BIND_ENUM_CONSTANT(LOOP_BACKWARD);
 }
 
+Ref<AudioStreamWAV> AudioStreamWAV::load_from_buffer(const Vector<uint8_t> &p_file_data, const Dictionary &p_options) {
+	// /* STEP 1, READ WAVE FILE */
+
+	Ref<FileAccessMemory> file;
+	file.instantiate();
+	Error err = file->open_custom(p_file_data.ptr(), p_file_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 */
+
+	// 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));
+	}
+
+	/* 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;
+
+	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;
+
+	Vector<float> data;
+
+	while (!file->eof_reached()) {
+		/* chunk */
+		char chunk_id[4];
+		file->get_buffer((uint8_t *)&chunk_id, 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 (chunk_id[0] == 'f' && chunk_id[1] == 'm' && chunk_id[2] == 't' && chunk_id[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) {
+				ERR_FAIL_V_MSG(Ref<AudioStreamWAV>(), "Format not supported for WAVE file (not PCM). Save WAVE files as uncompressed PCM or IEEE float instead.");
+			}
+
+			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 (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).");
+			}
+
+			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).");
+			}
+
+			/* Don't need anything else, continue */
+			format_found = true;
+		}
+
+		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;
+
+			if (!format_found) {
+				ERR_PRINT("'data' chunk before 'format' chunk found.");
+				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));
+	}
+
+	// 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 = 0.0;
+			int ipos = 0;
+
+			for (int i = 0; i < new_data_frames; i++) {
+				// Cubic interpolation should be enough.
+
+				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];
+
+				new_data.write[i * format_channels + c] = Math::cubic_interpolate(y1, y2, y0, y3, frac);
+
+				// 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_mode) {
+			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 = 0.0;
+		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_mode == AudioStreamWAV::LOOP_DISABLED) && format_channels > 0) {
+		int first = 0;
+		int last = (frames / format_channels) - 1;
+		bool found = false;
+		float limit = Math::db_to_linear(TRIM_DB_LIMIT);
+
+		for (int i = 0; i < data.size() / format_channels; i++) {
+			float amp_channel_sum = 0.0;
+			for (int j = 0; j < format_channels; j++) {
+				amp_channel_sum += Math::abs(data[(i * format_channels) + j]);
+			}
+
+			float amp = Math::abs(amp_channel_sum / (float)format_channels);
+
+			if (!found && amp > limit) {
+				first = i;
+				found = true;
+			}
+
+			if (found && amp > limit) {
+				last = i;
+			}
+		}
+
+		if (first < last) {
+			Vector<float> new_data;
+			new_data.resize((last - first) * format_channels);
+			for (int i = first; i < last; i++) {
+				float fade_out_mult = 1.0;
+
+				if (last - i < TRIM_FADE_OUT_FRAMES) {
+					fade_out_mult = ((float)(last - i - 1) / (float)TRIM_FADE_OUT_FRAMES);
+				}
+
+				for (int j = 0; j < format_channels; j++) {
+					new_data.write[((i - first) * format_channels) + j] = data[(i * format_channels) + j] * fade_out_mult;
+				}
+			}
+
+			data = new_data;
+			frames = data.size() / format_channels;
+		}
+	}
+
+	if (import_loop_mode >= 2) {
+		loop_mode = (AudioStreamWAV::LoopMode)(import_loop_mode - 1);
+		loop_begin = p_options["edit/loop_begin"];
+		loop_end = p_options["edit/loop_end"];
+		// Wrap around to max frames, so `-1` can be used to select the end, etc.
+		if (loop_begin < 0) {
+			loop_begin = CLAMP(loop_begin + frames, 0, frames - 1);
+		}
+		if (loop_end < 0) {
+			loop_end = CLAMP(loop_end + frames, 0, frames - 1);
+		}
+	}
+
+	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;
+	}
+
+	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 = {};
+		uint32_t qoa_len = 0;
+
+		desc.samplerate = rate;
+		desc.samples = frames;
+		desc.channels = format_channels;
+
+		void *encoded = qoa_encode((short *)pcm_data.ptr(), &desc, &qoa_len);
+		if (encoded) {
+			dst_data.resize(qoa_len);
+			memcpy(dst_data.ptrw(), encoded, qoa_len);
+			QOA_FREE(encoded);
+		}
+	} 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);
+	return sample;
+}
+
+Ref<AudioStreamWAV> AudioStreamWAV::load_from_file(const String &p_path, const Dictionary &p_options) {
+	Vector<uint8_t> file_data = FileAccess::get_file_as_bytes(p_path);
+	ERR_FAIL_COND_V_MSG(file_data.is_empty(), Ref<AudioStreamWAV>(), vformat("Cannot open file '%s'.", p_path));
+	return load_from_buffer(file_data, p_options);
+}
+
 AudioStreamWAV::AudioStreamWAV() {}
 
 AudioStreamWAV::~AudioStreamWAV() {}