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