FFMPEG- Duration of audio file is inaccurate

Question

I have video file (mp4). I want to detach audio stream (AAC format) from that file and save in PC. With below code, Generated aac file canplay now on KM player, but can not play on VLC player. Information of duration displays on player is wrong. Please help me with this problem.

 err = avformat_open_input(input_format_context, filename, NULL, NULL);
if (err < 0) {
    return err;
}

/* If not enough info to get the stream parameters, we decode the
   first frames to get it. (used in mpeg case for example) */
ret = avformat_find_stream_info(*input_format_context, 0);
if (ret < 0) {
    av_log(NULL, AV_LOG_FATAL, "%s: could not find codec parameters\n", filename);
    return ret;
}

/* dump the file content */
av_dump_format(*input_format_context, 0, filename, 0);

for (size_t i = 0; i < (*input_format_context)->nb_streams; i++) {
    AVStream *st = (*input_format_context)->streams[i];
    if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
        *input_codec_context = st->codec;
        *input_audio_stream = st;

        FILE *file = NULL;
        file = fopen("C:\\Users\\MyPC\\Downloads\\Test.aac", "wb");
        AVPacket reading_packet;
        av_init_packet(&reading_packet);
        while (av_read_frame(*input_format_context, &reading_packet) == 0) {
            if (reading_packet.stream_index == (int) i) {

                uint8_t adts_header[7];
                unsigned int obj_type = 0;
                unsigned int num_data_block = (reading_packet.size)/1024;
                int rate_idx = st->codec->sample_rate, channels = st->codec->channels;

                 uint16_t frame_length;

                // include the header length also
                 frame_length = reading_packet.size + 7;

                /* We want the same metadata */
                /* Generate ADTS header */
                if(adts_header == NULL) return -1;
                /* Sync point over a full byte */
                adts_header[0] = 0xFF;
                /* Sync point continued over first 4 bits + static 4 bits
                * (ID, layer, protection)*/
                adts_header[1] = 0xF1;
                /* Object type over first 2 bits */
                adts_header[2] = obj_type << 6;
                /* rate index over next 4 bits */
                adts_header[2] |= (rate_idx << 2);
                /* channels over last 2 bits */
                adts_header[2] |= (channels & 0x4) >> 2;
                /* channels continued over next 2 bits + 4 bits at zero */
                adts_header[3] = (channels & 0x3) << 6;
                /* frame size over last 2 bits */
                adts_header[3] |= (frame_length & 0x1800) >> 11;
                /* frame size continued over full byte */
                adts_header[4] = (frame_length & 0x1FF8) >> 3;
                /* frame size continued first 3 bits */
                adts_header[5] = (frame_length & 0x7) << 5;
                /* buffer fullness (0x7FF for VBR) over 5 last bits*/
                adts_header[5] |= 0x1F;
                /* buffer fullness (0x7FF for VBR) continued over 6 first bits + 2 zeros
                * number of raw data blocks */
                adts_header[6] = 0xFA;
                adts_header[6] |= num_data_block & 0x03; // Set raw Data blocks.

                fwrite(adts_header, 1, 7, file);
                fwrite(reading_packet.data, 1, reading_packet.size, file);
            }
            av_free_packet(&reading_packet);   
        }
        fclose(file);

        return 0;
    }
}

Answer 1

Object type and sample rate index must be set to the real, correct values. Both values can be parsed out of the audio specific config in the extradata field in the codec context. All the information you need is here: http://wiki.multimedia.cx/index.php?title=MPEG-4_Audio