static int transcode_audio(InputStream *ist, AVPacket *pkt, int *got_output)

{

    AVFrame *decoded_frame;

    AVCodecContext *avctx = ist->st->codec;

    int bps = av_get_bytes_per_sample(ist->st->codec->sample_fmt);

    int i, ret;



    if (!ist->decoded_frame && !(ist->decoded_frame = avcodec_alloc_frame()))

        return AVERROR(ENOMEM);

    else

        avcodec_get_frame_defaults(ist->decoded_frame);

    decoded_frame = ist->decoded_frame;



    ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt);

    if (ret < 0) {

        return ret;

    }



    if (!*got_output) {

        /* no audio frame */

        return ret;

    }



    /* if the decoder provides a pts, use it instead of the last packet pts.

       the decoder could be delaying output by a packet or more. */

    if (decoded_frame->pts != AV_NOPTS_VALUE)

        ist->next_dts = decoded_frame->pts;



    /* increment next_dts to use for the case where the input stream does not

       have timestamps or there are multiple frames in the packet */

    ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) /

                     avctx->sample_rate;



    // preprocess audio (volume)

    if (audio_volume != 256) {

        int decoded_data_size = decoded_frame->nb_samples * avctx->channels * bps;

        void *samples = decoded_frame->data[0];

        switch (avctx->sample_fmt) {

        case AV_SAMPLE_FMT_U8:

        {

            uint8_t *volp = samples;

            for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

                int v = (((*volp - 128) * audio_volume + 128) >> 8) + 128;

                *volp++ = av_clip_uint8(v);

            }

            break;

        }

        case AV_SAMPLE_FMT_S16:

        {

            int16_t *volp = samples;

            for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

                int v = ((*volp) * audio_volume + 128) >> 8;

                *volp++ = av_clip_int16(v);

            }

            break;

        }

        case AV_SAMPLE_FMT_S32:

        {

            int32_t *volp = samples;

            for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

                int64_t v = (((int64_t)*volp * audio_volume + 128) >> 8);

                *volp++ = av_clipl_int32(v);

            }

            break;

        }

        case AV_SAMPLE_FMT_FLT:

        {

            float *volp = samples;

            float scale = audio_volume / 256.f;

            for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

                *volp++ *= scale;

            }

            break;

        }

        case AV_SAMPLE_FMT_DBL:

        {

            double *volp = samples;

            double scale = audio_volume / 256.;

            for (i = 0; i < (decoded_data_size / sizeof(*volp)); i++) {

                *volp++ *= scale;

            }

            break;

        }

        default:

            av_log(NULL, AV_LOG_FATAL,

                   "Audio volume adjustment on sample format %s is not supported.\n",

                   av_get_sample_fmt_name(ist->st->codec->sample_fmt));

            exit_program(1);

        }

    }



    rate_emu_sleep(ist);



    for (i = 0; i < nb_output_streams; i++) {

        OutputStream *ost = output_streams[i];



        if (!check_output_constraints(ist, ost) || !ost->encoding_needed)

            continue;

        do_audio_out(output_files[ost->file_index]->ctx, ost, ist, decoded_frame);

    }



    return ret;

}
