mirror of https://github.com/mstorsjo/fdk-aac.git
Add an example of decoding from an ADTS file
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commit
fc4cd7dd43
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@ -27,3 +27,4 @@ missing
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stamp-h1
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aac-enc
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compile
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aac-dec
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@ -39,12 +39,15 @@ libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \
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-export-symbols $(top_srcdir)/fdk-aac.sym
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if EXAMPLE
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bin_PROGRAMS = aac-enc$(EXEEXT)
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bin_PROGRAMS = aac-enc$(EXEEXT) aac-dec$(EXEEXT)
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aac_enc_LDADD = libfdk-aac.la
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aac_enc_SOURCES = aac-enc.c wavreader.c
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noinst_HEADERS = wavreader.h
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aac_dec_LDADD = libfdk-aac.la
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aac_dec_SOURCES = aac-dec.c wavwriter.c
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noinst_HEADERS = wavreader.h wavwriter.h
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endif
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AACDEC_SRC = \
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@ -0,0 +1,113 @@
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/* ------------------------------------------------------------------
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* Copyright (C) 2013 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 "libAACdec/include/aacdecoder_lib.h"
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#include "wavwriter.h"
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int main(int argc, char *argv[]) {
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const char *infile, *outfile;
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FILE *in;
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void *wav = NULL;
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int output_size;
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uint8_t *output_buf;
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int16_t *decode_buf;
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HANDLE_AACDECODER handle;
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int frame_size = 0;
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if (argc < 3) {
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fprintf(stderr, "%s in.aac out.wav\n", argv[0]);
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return 1;
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}
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infile = argv[1];
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outfile = argv[2];
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handle = aacDecoder_Open(TT_MP4_ADTS, 1);
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in = fopen(infile, "rb");
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if (!in) {
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perror(infile);
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return 1;
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}
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output_size = 8*2*2048;
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output_buf = (uint8_t*) malloc(output_size);
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decode_buf = (int16_t*) malloc(output_size);
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while (1) {
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uint8_t packet[10240], *ptr = packet;
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int n, i;
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UINT valid, packet_size;
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AAC_DECODER_ERROR err;
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n = fread(packet, 1, 7, in);
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if (n != 7)
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break;
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if (packet[0] != 0xff || (packet[1] & 0xf0) != 0xf0) {
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fprintf(stderr, "Not an ADTS packet\n");
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break;
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}
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packet_size = ((packet[3] & 0x03) << 11) | (packet[4] << 3) | (packet[5] >> 5);
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n = fread(packet + 7, 1, packet_size - 7, in);
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if (n != packet_size - 7) {
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fprintf(stderr, "Partial packet\n");
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break;
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}
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valid = packet_size;
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err = aacDecoder_Fill(handle, &ptr, &packet_size, &valid);
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Fill failed: %x\n", err);
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break;
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}
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err = aacDecoder_DecodeFrame(handle, decode_buf, output_size / sizeof(INT_PCM), 0);
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if (err == AAC_DEC_NOT_ENOUGH_BITS)
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continue;
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Decode failed: %x\n", err);
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continue;
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}
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if (!wav) {
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CStreamInfo *info = aacDecoder_GetStreamInfo(handle);
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if (!info || info->sampleRate <= 0) {
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fprintf(stderr, "No stream info\n");
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break;
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}
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frame_size = info->frameSize * info->numChannels;
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// Note, this probably doesn't return channels > 2 in the right order for wav
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wav = wav_write_open(outfile, info->sampleRate, 16, info->numChannels);
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if (!wav) {
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perror(outfile);
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break;
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}
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}
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for (i = 0; i < frame_size; i++) {
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uint8_t* out = &output_buf[2*i];
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out[0] = decode_buf[i] & 0xff;
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out[1] = decode_buf[i] >> 8;
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}
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wav_write_data(wav, output_buf, 2*frame_size);
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}
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free(output_buf);
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free(decode_buf);
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fclose(in);
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if (wav)
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wav_write_close(wav);
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aacDecoder_Close(handle);
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return 0;
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}
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