Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

1 files changed, 237 insertions, 0 deletions

diff --git a/fdk-aac/aac-enc.c b/fdk-aac/aac-enc.c
new file mode 100644
index 0000000..c90ff12
--- /dev/null
+++ b/fdk-aac/aac-enc.c
@@ -0,0 +1,237 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *	  http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
+ * express or implied.
+ * See the License for the specific language governing permissions
+ * and limitations under the License.
+ * -------------------------------------------------------------------
+ */
+
+#include <stdio.h>
+#include <stdint.h>
+
+#if defined(_MSC_VER)
+#include <getopt.h>
+#else
+#include <unistd.h>
+#endif
+
+#include <stdlib.h>
+#include "libAACenc/include/aacenc_lib.h"
+#include "wavreader.h"
+
+void usage(const char* name) {
+	fprintf(stderr, "%s [-r bitrate] [-t aot] [-a afterburner] [-s sbr] [-v vbr] in.wav out.aac\n", name);
+	fprintf(stderr, "Supported AOTs:\n");
+	fprintf(stderr, "\t2\tAAC-LC\n");
+	fprintf(stderr, "\t5\tHE-AAC\n");
+	fprintf(stderr, "\t29\tHE-AAC v2\n");
+	fprintf(stderr, "\t23\tAAC-LD\n");
+	fprintf(stderr, "\t39\tAAC-ELD\n");
+}
+
+int main(int argc, char *argv[]) {
+	int bitrate = 64000;
+	int ch;
+	const char *infile, *outfile;
+	FILE *out;
+	void *wav;
+	int format, sample_rate, channels, bits_per_sample;
+	int input_size;
+	uint8_t* input_buf;
+	int16_t* convert_buf;
+	int aot = 2;
+	int afterburner = 1;
+	int eld_sbr = 0;
+	int vbr = 0;
+	HANDLE_AACENCODER handle;
+	CHANNEL_MODE mode;
+	AACENC_InfoStruct info = { 0 };
+	while ((ch = getopt(argc, argv, "r:t:a:s:v:")) != -1) {
+		switch (ch) {
+		case 'r':
+			bitrate = atoi(optarg);
+			break;
+		case 't':
+			aot = atoi(optarg);
+			break;
+		case 'a':
+			afterburner = atoi(optarg);
+			break;
+		case 's':
+			eld_sbr = atoi(optarg);
+			break;
+		case 'v':
+			vbr = atoi(optarg);
+			break;
+		case '?':
+		default:
+			usage(argv[0]);
+			return 1;
+		}
+	}
+	if (argc - optind < 2) {
+		usage(argv[0]);
+		return 1;
+	}
+	infile = argv[optind];
+	outfile = argv[optind + 1];
+
+	wav = wav_read_open(infile);
+	if (!wav) {
+		fprintf(stderr, "Unable to open wav file %s\n", infile);
+		return 1;
+	}
+	if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) {
+		fprintf(stderr, "Bad wav file %s\n", infile);
+		return 1;
+	}
+	if (format != 1) {
+		fprintf(stderr, "Unsupported WAV format %d\n", format);
+		return 1;
+	}
+	if (bits_per_sample != 16) {
+		fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+		return 1;
+	}
+	switch (channels) {
+	case 1: mode = MODE_1;       break;
+	case 2: mode = MODE_2;       break;
+	case 3: mode = MODE_1_2;     break;
+	case 4: mode = MODE_1_2_1;   break;
+	case 5: mode = MODE_1_2_2;   break;
+	case 6: mode = MODE_1_2_2_1; break;
+	default:
+		fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+		return 1;
+	}
+	if (aacEncOpen(&handle, 0, channels) != AACENC_OK) {
+		fprintf(stderr, "Unable to open encoder\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the AOT\n");
+		return 1;
+	}
+	if (aot == 39 && eld_sbr) {
+		if (aacEncoder_SetParam(handle, AACENC_SBR_MODE, 1) != AACENC_OK) {
+			fprintf(stderr, "Unable to set SBR mode for ELD\n");
+			return 1;
+		}
+	}
+	if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the AOT\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the channel mode\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the wav channel order\n");
+		return 1;
+	}
+	if (vbr) {
+		if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, vbr) != AACENC_OK) {
+			fprintf(stderr, "Unable to set the VBR bitrate mode\n");
+			return 1;
+		}
+	} else {
+		if (aacEncoder_SetParam(handle, AACENC_BITRATE, bitrate) != AACENC_OK) {
+			fprintf(stderr, "Unable to set the bitrate\n");
+			return 1;
+		}
+	}
+	if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_MP4_ADTS) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the ADTS transmux\n");
+		return 1;
+	}
+	if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+		fprintf(stderr, "Unable to set the afterburner mode\n");
+		return 1;
+	}
+	if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+		fprintf(stderr, "Unable to initialize the encoder\n");
+		return 1;
+	}
+	if (aacEncInfo(handle, &info) != AACENC_OK) {
+		fprintf(stderr, "Unable to get the encoder info\n");
+		return 1;
+	}
+
+	out = fopen(outfile, "wb");
+	if (!out) {
+		perror(outfile);
+		return 1;
+	}
+
+	input_size = channels*2*info.frameLength;
+	input_buf = (uint8_t*) malloc(input_size);
+	convert_buf = (int16_t*) malloc(input_size);
+
+	while (1) {
+		AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+		AACENC_InArgs in_args = { 0 };
+		AACENC_OutArgs out_args = { 0 };
+		int in_identifier = IN_AUDIO_DATA;
+		int in_size, in_elem_size;
+		int out_identifier = OUT_BITSTREAM_DATA;
+		int out_size, out_elem_size;
+		int read, i;
+		void *in_ptr, *out_ptr;
+		uint8_t outbuf[20480];
+		AACENC_ERROR err;
+
+		read = wav_read_data(wav, input_buf, input_size);
+		for (i = 0; i < read/2; i++) {
+			const uint8_t* in = &input_buf[2*i];
+			convert_buf[i] = in[0] | (in[1] << 8);
+		}
+		in_ptr = convert_buf;
+		in_size = read;
+		in_elem_size = 2;
+
+		in_args.numInSamples = read <= 0 ? -1 : read/2;
+		in_buf.numBufs = 1;
+		in_buf.bufs = &in_ptr;
+		in_buf.bufferIdentifiers = &in_identifier;
+		in_buf.bufSizes = &in_size;
+		in_buf.bufElSizes = &in_elem_size;
+
+		out_ptr = outbuf;
+		out_size = sizeof(outbuf);
+		out_elem_size = 1;
+		out_buf.numBufs = 1;
+		out_buf.bufs = &out_ptr;
+		out_buf.bufferIdentifiers = &out_identifier;
+		out_buf.bufSizes = &out_size;
+		out_buf.bufElSizes = &out_elem_size;
+
+		if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
+			if (err == AACENC_ENCODE_EOF)
+				break;
+			fprintf(stderr, "Encoding failed\n");
+			return 1;
+		}
+		if (out_args.numOutBytes == 0)
+			continue;
+		fwrite(outbuf, 1, out_args.numOutBytes, out);
+	}
+	free(input_buf);
+	free(convert_buf);
+	fclose(out);
+	wav_read_close(wav);
+	aacEncClose(&handle);
+
+	return 0;
+}
+