video-pc/libs/Recorder/transcode_aac.cpp

/*
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/

/**
* @file
* Simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/

#include <stdio.h>

extern "C" {
#include <libavformat/avformat.h>
#include <libavformat/avio.h>

#include <libavutil/audio_fifo.h>
#include <libswresample/swresample.h>
}
#include "headers_ffmpeg.h"

/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
/* The number of output channels */
#define OUTPUT_CHANNELS 2

/**
* Open an input file and the required decoder.
* @param      filename             File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context  Codec context of opened file
* @return Error code (0 if successful)
*/
static int open_input_file(const char *filename,
                           AVFormatContext **input_format_context,
                           AVCodecContext **input_codec_context)
{
    AVCodecContext *avctx;
    FFmpegCodec *input_codec;
    int error;

    /* Open the input file to read from it. */
    if ((error = avformat_open_input(input_format_context, filename, NULL, NULL)) < 0) {
        fprintf(stderr,
                "Could not open input file '%s' (error '%s')\n",
                filename,
                av_err2str(error));
        *input_format_context = NULL;
        return error;
    }

    /* Get information on the input file (number of streams etc.). */
    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
        fprintf(stderr, "Could not open find stream info (error '%s')\n", av_err2str(error));
        avformat_close_input(input_format_context);
        return error;
    }

    /* Make sure that there is only one stream in the input file. */
    if ((*input_format_context)->nb_streams != 1) {
        fprintf(stderr,
                "Expected one audio input stream, but found %d\n",
                (*input_format_context)->nb_streams);
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
    }

    /* Find a decoder for the audio stream. */
    if (!(input_codec = avcodec_find_decoder(
              (*input_format_context)->streams[0]->codecpar->codec_id))) {
        fprintf(stderr, "Could not find input codec\n");
        avformat_close_input(input_format_context);
        return AVERROR_EXIT;
    }

    /* Allocate a new decoding context. */
    avctx = avcodec_alloc_context3(input_codec);
    if (!avctx) {
        fprintf(stderr, "Could not allocate a decoding context\n");
        avformat_close_input(input_format_context);
        return AVERROR(ENOMEM);
    }

    /* Initialize the stream parameters with demuxer information. */
    error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
    if (error < 0) {
        avformat_close_input(input_format_context);
        avcodec_free_context(&avctx);
        return error;
    }

    /* Open the decoder for the audio stream to use it later. */
    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
        fprintf(stderr, "Could not open input codec (error '%s')\n", av_err2str(error));
        avcodec_free_context(&avctx);
        avformat_close_input(input_format_context);
        return error;
    }

    /* Save the decoder context for easier access later. */
    *input_codec_context = avctx;

    return 0;
}

/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param      filename              File to be opened
* @param      input_codec_context   Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context  Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
                            AVCodecContext *input_codec_context,
                            AVFormatContext **output_format_context,
                            AVCodecContext **output_codec_context)
{
    AVCodecContext *avctx = NULL;
    AVIOContext *output_io_context = NULL;
    AVStream *stream = NULL;
    FFmpegCodec *output_codec = NULL;
    int error;

    /* Open the output file to write to it. */
    if ((error = avio_open(&output_io_context, filename, AVIO_FLAG_WRITE)) < 0) {
        fprintf(stderr,
                "Could not open output file '%s' (error '%s')\n",
                filename,
                av_err2str(error));
        return error;
    }

    /* Create a new format context for the output container format. */
    if (!(*output_format_context = avformat_alloc_context())) {
        fprintf(stderr, "Could not allocate output format context\n");
        return AVERROR(ENOMEM);
    }

    /* Associate the output file (pointer) with the container format context. */
    (*output_format_context)->pb = output_io_context;

    /* Guess the desired container format based on the file extension. */
    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename, NULL))) {
        fprintf(stderr, "Could not find output file format\n");
        goto cleanup;
    }

    if (!((*output_format_context)->url = av_strdup(filename))) {
        fprintf(stderr, "Could not allocate url.\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }

    /* Find the encoder to be used by its name. */
    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
        fprintf(stderr, "Could not find an AAC encoder.\n");
        goto cleanup;
    }

    /* Create a new audio stream in the output file container. */
    if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
        fprintf(stderr, "Could not create new stream\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }

    avctx = avcodec_alloc_context3(output_codec);
    if (!avctx) {
        fprintf(stderr, "Could not allocate an encoding context\n");
        error = AVERROR(ENOMEM);
        goto cleanup;
    }

    /* Set the basic encoder parameters.
	* The input file's sample rate is used to avoid a sample rate conversion. */
    ffmpeg_set_codec_channels(avctx, OUTPUT_CHANNELS);
    ffmpeg_set_codec_channel_layout(avctx, ffmpeg_get_default_channel_layout(OUTPUT_CHANNELS));
    avctx->sample_rate = input_codec_context->sample_rate;
    avctx->sample_fmt = output_codec->sample_fmts[0];
    avctx->bit_rate = OUTPUT_BIT_RATE;

    /* Allow the use of the experimental AAC encoder. */
    avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

    /* Set the sample rate for the container. */
    stream->time_base.den = input_codec_context->sample_rate;
    stream->time_base.num = 1;

    /* Some container formats (like MP4) require global headers to be present.
	* Mark the encoder so that it behaves accordingly. */
    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
        avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;

    /* Open the encoder for the audio stream to use it later. */
    if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
        fprintf(stderr, "Could not open output codec (error '%s')\n", av_err2str(error));
        goto cleanup;
    }

    error = avcodec_parameters_from_context(stream->codecpar, avctx);
    if (error < 0) {
        fprintf(stderr, "Could not initialize stream parameters\n");
        goto cleanup;
    }

    /* Save the encoder context for easier access later. */
    *output_codec_context = avctx;

    return 0;

cleanup:
    avcodec_free_context(&avctx);
    avio_closep(&(*output_format_context)->pb);
    avformat_free_context(*output_format_context);
    *output_format_context = NULL;
    return error < 0 ? error : AVERROR_EXIT;
}

/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet)
{
    av_init_packet(packet);
    /* Set the packet data and size so that it is recognized as being empty. */
    packet->data = NULL;
    packet->size = 0;
}

/**
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
static int init_input_frame(AVFrame **frame)
{
    if (!(*frame = av_frame_alloc())) {
        fprintf(stderr, "Could not allocate input frame\n");
        return AVERROR(ENOMEM);
    }
    return 0;
}

/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
* @param      input_codec_context  Codec context of the input file
* @param      output_codec_context Codec context of the output file
* @param[out] resample_context     Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
                          AVCodecContext *output_codec_context,
                          SwrContext **resample_context)
{
    int error;

    /*
	* Create a resampler context for the conversion.
	* Set the conversion parameters.
	* Default channel layouts based on the number of channels
	* are assumed for simplicity (they are sometimes not detected
	* properly by the demuxer and/or decoder).
	*/
    *resample_context
        = ffmpeg_swr_alloc_set_opts(NULL,
                             ffmpeg_get_default_channel_layout(ffmpeg_get_codec_context_channels(output_codec_context)),
                             output_codec_context->sample_fmt,
                             output_codec_context->sample_rate,
                             ffmpeg_get_default_channel_layout(ffmpeg_get_codec_context_channels(input_codec_context)),
                             input_codec_context->sample_fmt,
                             input_codec_context->sample_rate,
                             0,
                             NULL);
    if (!*resample_context) {
        fprintf(stderr, "Could not allocate resample context\n");
        return AVERROR(ENOMEM);
    }
    /*
	* Perform a sanity check so that the number of converted samples is
	* not greater than the number of samples to be converted.
	* If the sample rates differ, this case has to be handled differently
	*/
    av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);

    /* Open the resampler with the specified parameters. */
    if ((error = swr_init(*resample_context)) < 0) {
        fprintf(stderr, "Could not open resample context\n");
        swr_free(resample_context);
        return error;
    }
    return 0;
}

/**
* Initialize a FIFO buffer for the audio samples to be encoded.
* @param[out] fifo                 Sample buffer
* @param      output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
    /* Create the FIFO buffer based on the specified output sample format. */
    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
                                      ffmpeg_get_codec_context_channels(output_codec_context),
                                      1))) {
        fprintf(stderr, "Could not allocate FIFO\n");
        return AVERROR(ENOMEM);
    }
    return 0;
}

/**
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_header(AVFormatContext *output_format_context)
{
    int error;
    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
        fprintf(stderr, "Could not write output file header (error '%s')\n", av_err2str(error));
        return error;
    }
    return 0;
}

/**
* Decode one audio frame from the input file.
* @param      frame                Audio frame to be decoded
* @param      input_format_context Format context of the input file
* @param      input_codec_context  Codec context of the input file
* @param[out] data_present         Indicates whether data has been decoded
* @param[out] finished             Indicates whether the end of file has
*                                  been reached and all data has been
*                                  decoded. If this flag is false, there
*                                  is more data to be decoded, i.e., this
*                                  function has to be called again.
* @return Error code (0 if successful)
*/
static int decode_audio_frame(AVFrame *frame,
                              AVFormatContext *input_format_context,
                              AVCodecContext *input_codec_context,
                              int *data_present,
                              int *finished)
{
    /* Packet used for temporary storage. */
    AVPacket input_packet;
    int error;
    init_packet(&input_packet);

    /* Read one audio frame from the input file into a temporary packet. */
    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
        /* If we are at the end of the file, flush the decoder below. */
        if (error == AVERROR_EOF)
            *finished = 1;
        else {
            fprintf(stderr, "Could not read frame (error '%s')\n", av_err2str(error));
            return error;
        }
    }

    /* Send the audio frame stored in the temporary packet to the decoder.
	* The input audio stream decoder is used to do this. */
    if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
        fprintf(stderr, "Could not send packet for decoding (error '%s')\n", av_err2str(error));
        return error;
    }

    /* Receive one frame from the decoder. */
    error = avcodec_receive_frame(input_codec_context, frame);
    /* If the decoder asks for more data to be able to decode a frame,
	* return indicating that no data is present. */
    if (error == AVERROR(EAGAIN)) {
        error = 0;
        goto cleanup;
        /* If the end of the input file is reached, stop decoding. */
    } else if (error == AVERROR_EOF) {
        *finished = 1;
        error = 0;
        goto cleanup;
    } else if (error < 0) {
        fprintf(stderr, "Could not decode frame (error '%s')\n", av_err2str(error));
        goto cleanup;
        /* Default case: Return decoded data. */
    } else {
        *data_present = 1;
        goto cleanup;
    }

cleanup:
    av_packet_unref(&input_packet);
    return error;
}

/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
*                                     dimensions are reference, channel
*                                     (for multi-channel audio), sample.
* @param      output_codec_context    Codec context of the output file
* @param      frame_size              Number of samples to be converted in
*                                     each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
                                  AVCodecContext *output_codec_context,
                                  int frame_size)
{
    int error;

    /* Allocate as many pointers as there are audio channels.
	* Each pointer will later point to the audio samples of the corresponding
	* channels (although it may be NULL for interleaved formats).
	*/
    if (!(*converted_input_samples = (uint8_t **) calloc(ffmpeg_get_codec_context_channels(output_codec_context),
                                                         sizeof(**converted_input_samples)))) {
        fprintf(stderr, "Could not allocate converted input sample pointers\n");
        return AVERROR(ENOMEM);
    }

    /* Allocate memory for the samples of all channels in one consecutive
	* block for convenience. */
    if ((error = ffmpeg_av_samples_alloc(*converted_input_samples,
                                         NULL,
                                         ffmpeg_get_codec_context_channels(output_codec_context),
                                         frame_size,
                                         output_codec_context->sample_fmt,
                                         0))
        < 0) {
        fprintf(stderr,
                "Could not allocate converted input samples (error '%s')\n",
                av_err2str(error));
        av_freep(&(*converted_input_samples)[0]);
        free(*converted_input_samples);
        return error;
    }
    return 0;
}

/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param      input_data       Samples to be decoded. The dimensions are
*                              channel (for multi-channel audio), sample.
* @param[out] converted_data   Converted samples. The dimensions are channel
*                              (for multi-channel audio), sample.
* @param      frame_size       Number of samples to be converted
* @param      resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
                           uint8_t **converted_data,
                           const int frame_size,
                           SwrContext *resample_context)
{
    int error;

    /* Convert the samples using the resampler. */
    if ((error = swr_convert(resample_context, converted_data, frame_size, input_data, frame_size))
        < 0) {
        fprintf(stderr, "Could not convert input samples (error '%s')\n", av_err2str(error));
        return error;
    }

    return 0;
}

/**
* Add converted input audio samples to the FIFO buffer for later processing.
* @param fifo                    Buffer to add the samples to
* @param converted_input_samples Samples to be added. The dimensions are channel
*                                (for multi-channel audio), sample.
* @param frame_size              Number of samples to be converted
* @return Error code (0 if successful)
*/
static int add_samples_to_fifo(AVAudioFifo *fifo,
                               uint8_t **converted_input_samples,
                               const int frame_size)
{
    int error;

    /* Make the FIFO as large as it needs to be to hold both,
	* the old and the new samples. */
    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
        fprintf(stderr, "Could not reallocate FIFO\n");
        return error;
    }

    /* Store the new samples in the FIFO buffer. */
    if (av_audio_fifo_write(fifo, (void **) converted_input_samples, frame_size) < frame_size) {
        fprintf(stderr, "Could not write data to FIFO\n");
        return AVERROR_EXIT;
    }
    return 0;
}

/**
* Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
* @param      fifo                 Buffer used for temporary storage
* @param      input_format_context Format context of the input file
* @param      input_codec_context  Codec context of the input file
* @param      output_codec_context Codec context of the output file
* @param      resampler_context    Resample context for the conversion
* @param[out] finished             Indicates whether the end of file has
*                                  been reached and all data has been
*                                  decoded. If this flag is false,
*                                  there is more data to be decoded,
*                                  i.e., this function has to be called
*                                  again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
                                         AVFormatContext *input_format_context,
                                         AVCodecContext *input_codec_context,
                                         AVCodecContext *output_codec_context,
                                         SwrContext *resampler_context,
                                         int *finished)
{
    /* Temporary storage of the input samples of the frame read from the file. */
    AVFrame *input_frame = NULL;
    /* Temporary storage for the converted input samples. */
    uint8_t **converted_input_samples = NULL;
    int data_present = 0;
    int ret = AVERROR_EXIT;

    /* Initialize temporary storage for one input frame. */
    if (init_input_frame(&input_frame))
        goto cleanup;
    /* Decode one frame worth of audio samples. */
    if (decode_audio_frame(input_frame,
                           input_format_context,
                           input_codec_context,
                           &data_present,
                           finished))
        goto cleanup;
    /* If we are at the end of the file and there are no more samples
	* in the decoder which are delayed, we are actually finished.
	* This must not be treated as an error. */
    if (*finished) {
        ret = 0;
        goto cleanup;
    }
    /* If there is decoded data, convert and store it. */
    if (data_present) {
        /* Initialize the temporary storage for the converted input samples. */
        if (init_converted_samples(&converted_input_samples,
                                   output_codec_context,
                                   input_frame->nb_samples))
            goto cleanup;

        /* Convert the input samples to the desired output sample format.
		* This requires a temporary storage provided by converted_input_samples. */
        if (convert_samples((const uint8_t **) input_frame->extended_data,
                            converted_input_samples,
                            input_frame->nb_samples,
                            resampler_context))
            goto cleanup;

        /* Add the converted input samples to the FIFO buffer for later processing. */
        if (add_samples_to_fifo(fifo, converted_input_samples, input_frame->nb_samples))
            goto cleanup;
        ret = 0;
    }
    ret = 0;

cleanup:
    if (converted_input_samples) {
        av_freep(&converted_input_samples[0]);
        free(converted_input_samples);
    }
    av_frame_free(&input_frame);

    return ret;
}

/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame                Frame to be initialized
* @param      output_codec_context Codec context of the output file
* @param      frame_size           Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
{
    int error;

    /* Create a new frame to store the audio samples. */
    if (!(*frame = av_frame_alloc())) {
        fprintf(stderr, "Could not allocate output frame\n");
        return AVERROR_EXIT;
    }

    /* Set the frame's parameters, especially its size and format.
	* av_frame_get_buffer needs this to allocate memory for the
	* audio samples of the frame.
	* Default channel layouts based on the number of channels
	* are assumed for simplicity. */
    (*frame)->nb_samples = frame_size;
    ffmpeg_set_frame_channel_layout(*frame, output_codec_context);
    (*frame)->format = output_codec_context->sample_fmt;
    (*frame)->sample_rate = output_codec_context->sample_rate;

    /* Allocate the samples of the created frame. This call will make
	* sure that the audio frame can hold as many samples as specified. */
    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
        fprintf(stderr, "Could not allocate output frame samples (error '%s')\n", av_err2str(error));
        av_frame_free(frame);
        return error;
    }

    return 0;
}

/* Global timestamp for the audio frames. */
static int64_t pts = 0;

/**
* Encode one frame worth of audio to the output file.
* @param      frame                 Samples to be encoded
* @param      output_format_context Format context of the output file
* @param      output_codec_context  Codec context of the output file
* @param[out] data_present          Indicates whether data has been
*                                   encoded
* @return Error code (0 if successful)
*/
static int encode_audio_frame(AVFrame *frame,
                              AVFormatContext *output_format_context,
                              AVCodecContext *output_codec_context,
                              int *data_present)
{
    /* Packet used for temporary storage. */
    AVPacket output_packet;
    int error;
    init_packet(&output_packet);

    /* Set a timestamp based on the sample rate for the container. */
    if (frame) {
        frame->pts = pts;
        pts += frame->nb_samples;
    }

    /* Send the audio frame stored in the temporary packet to the encoder.
	* The output audio stream encoder is used to do this. */
    error = avcodec_send_frame(output_codec_context, frame);
    /* The encoder signals that it has nothing more to encode. */
    if (error == AVERROR_EOF) {
        error = 0;
        goto cleanup;
    } else if (error < 0) {
        fprintf(stderr, "Could not send packet for encoding (error '%s')\n", av_err2str(error));
        return error;
    }

    /* Receive one encoded frame from the encoder. */
    error = avcodec_receive_packet(output_codec_context, &output_packet);
    /* If the encoder asks for more data to be able to provide an
	* encoded frame, return indicating that no data is present. */
    if (error == AVERROR(EAGAIN)) {
        error = 0;
        goto cleanup;
        /* If the last frame has been encoded, stop encoding. */
    } else if (error == AVERROR_EOF) {
        error = 0;
        goto cleanup;
    } else if (error < 0) {
        fprintf(stderr, "Could not encode frame (error '%s')\n", av_err2str(error));
        goto cleanup;
        /* Default case: Return encoded data. */
    } else {
        *data_present = 1;
    }

    /* Write one audio frame from the temporary packet to the output file. */
    if (*data_present && (error = av_write_frame(output_format_context, &output_packet)) < 0) {
        fprintf(stderr, "Could not write frame (error '%s')\n", av_err2str(error));
        goto cleanup;
    }

cleanup:
    av_packet_unref(&output_packet);
    return error;
}

/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo                  Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context  Codec context of the output file
* @return Error code (0 if successful)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
                                 AVFormatContext *output_format_context,
                                 AVCodecContext *output_codec_context)
{
    /* Temporary storage of the output samples of the frame written to the file. */
    AVFrame *output_frame;
    /* Use the maximum number of possible samples per frame.
	* If there is less than the maximum possible frame size in the FIFO
	* buffer use this number. Otherwise, use the maximum possible frame size. */
    const int frame_size = FFMIN(av_audio_fifo_size(fifo), output_codec_context->frame_size);
    int data_written;

    /* Initialize temporary storage for one output frame. */
    if (init_output_frame(&output_frame, output_codec_context, frame_size))
        return AVERROR_EXIT;

    /* Read as many samples from the FIFO buffer as required to fill the frame.
	* The samples are stored in the frame temporarily. */
    if (av_audio_fifo_read(fifo, (void **) output_frame->data, frame_size) < frame_size) {
        fprintf(stderr, "Could not read data from FIFO\n");
        av_frame_free(&output_frame);
        return AVERROR_EXIT;
    }

    /* Encode one frame worth of audio samples. */
    if (encode_audio_frame(output_frame,
                           output_format_context,
                           output_codec_context,
                           &data_written)) {
        av_frame_free(&output_frame);
        return AVERROR_EXIT;
    }
    av_frame_free(&output_frame);
    return 0;
}

/**
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
    int error;
    if ((error = av_write_trailer(output_format_context)) < 0) {
        fprintf(stderr, "Could not write output file trailer (error '%s')\n", av_err2str(error));
        return error;
    }
    return 0;
}

int test_transcode()
{
    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
    SwrContext *resample_context = NULL;
    AVAudioFifo *fifo = NULL;
    int ret = AVERROR_EXIT;

    //if (argc != 3) {
    //	fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
    //	exit(1);
    //}

    const char *inputfile = "WAS_2019-09-05_14_19_42_109.wav";
    const char *outputfile = "transcode.aac";

    /* Open the input file for reading. */
    if (open_input_file(inputfile, &input_format_context, &input_codec_context))
        goto cleanup;
    /* Open the output file for writing. */
    if (open_output_file(outputfile,
                         input_codec_context,
                         &output_format_context,
                         &output_codec_context))
        goto cleanup;
    /* Initialize the resampler to be able to convert audio sample formats. */
    if (init_resampler(input_codec_context, output_codec_context, &resample_context))
        goto cleanup;
    /* Initialize the FIFO buffer to store audio samples to be encoded. */
    if (init_fifo(&fifo, output_codec_context))
        goto cleanup;
    /* Write the header of the output file container. */
    if (write_output_file_header(output_format_context))
        goto cleanup;

    /* Loop as long as we have input samples to read or output samples
	* to write; abort as soon as we have neither. */
    while (1) {
        /* Use the encoder's desired frame size for processing. */
        const int output_frame_size = output_codec_context->frame_size;
        int finished = 0;

        /* Make sure that there is one frame worth of samples in the FIFO
		* buffer so that the encoder can do its work.
		* Since the decoder's and the encoder's frame size may differ, we
		* need to FIFO buffer to store as many frames worth of input samples
		* that they make up at least one frame worth of output samples. */
        while (av_audio_fifo_size(fifo) < output_frame_size) {
            /* Decode one frame worth of audio samples, convert it to the
			* output sample format and put it into the FIFO buffer. */
            if (read_decode_convert_and_store(fifo,
                                              input_format_context,
                                              input_codec_context,
                                              output_codec_context,
                                              resample_context,
                                              &finished))
                goto cleanup;

            /* If we are at the end of the input file, we continue
			* encoding the remaining audio samples to the output file. */
            if (finished)
                break;
        }

        /* If we have enough samples for the encoder, we encode them.
		* At the end of the file, we pass the remaining samples to
		* the encoder. */
        while (av_audio_fifo_size(fifo) >= output_frame_size
               || (finished && av_audio_fifo_size(fifo) > 0))
            /* Take one frame worth of audio samples from the FIFO buffer,
			* encode it and write it to the output file. */
            if (load_encode_and_write(fifo, output_format_context, output_codec_context))
                goto cleanup;

        /* If we are at the end of the input file and have encoded
		* all remaining samples, we can exit this loop and finish. */
        if (finished) {
            int data_written;
            /* Flush the encoder as it may have delayed frames. */
            do {
                data_written = 0;
                if (encode_audio_frame(NULL,
                                       output_format_context,
                                       output_codec_context,
                                       &data_written))
                    goto cleanup;
            } while (data_written);
            break;
        }
    }

    /* Write the trailer of the output file container. */
    if (write_output_file_trailer(output_format_context))
        goto cleanup;
    ret = 0;

cleanup:
    if (fifo)
        av_audio_fifo_free(fifo);
    swr_free(&resample_context);
    if (output_codec_context)
        avcodec_free_context(&output_codec_context);
    if (output_format_context) {
        avio_closep(&output_format_context->pb);
        avformat_free_context(output_format_context);
    }
    if (input_codec_context)
        avcodec_free_context(&input_codec_context);
    if (input_format_context)
        avformat_close_input(&input_format_context);

    return ret;
}

int main1(int argc, char **argv)
{
    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
    SwrContext *resample_context = NULL;
    AVAudioFifo *fifo = NULL;
    int ret = AVERROR_EXIT;

    //if (argc != 3) {
    //	fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
    //	exit(1);
    //}

    const char *inputfile = "WAS_2019-09-05_14_19_42_109.wav";
    const char *outputfile = "transcode.aac";

    /* Open the input file for reading. */
    if (open_input_file(inputfile, &input_format_context, &input_codec_context))
        goto cleanup;
    /* Open the output file for writing. */
    if (open_output_file(outputfile,
                         input_codec_context,
                         &output_format_context,
                         &output_codec_context))
        goto cleanup;
    /* Initialize the resampler to be able to convert audio sample formats. */
    if (init_resampler(input_codec_context, output_codec_context, &resample_context))
        goto cleanup;
    /* Initialize the FIFO buffer to store audio samples to be encoded. */
    if (init_fifo(&fifo, output_codec_context))
        goto cleanup;
    /* Write the header of the output file container. */
    if (write_output_file_header(output_format_context))
        goto cleanup;

    /* Loop as long as we have input samples to read or output samples
	* to write; abort as soon as we have neither. */
    while (1) {
        /* Use the encoder's desired frame size for processing. */
        const int output_frame_size = output_codec_context->frame_size;
        int finished = 0;

        /* Make sure that there is one frame worth of samples in the FIFO
		* buffer so that the encoder can do its work.
		* Since the decoder's and the encoder's frame size may differ, we
		* need to FIFO buffer to store as many frames worth of input samples
		* that they make up at least one frame worth of output samples. */
        while (av_audio_fifo_size(fifo) < output_frame_size) {
            /* Decode one frame worth of audio samples, convert it to the
			* output sample format and put it into the FIFO buffer. */
            if (read_decode_convert_and_store(fifo,
                                              input_format_context,
                                              input_codec_context,
                                              output_codec_context,
                                              resample_context,
                                              &finished))
                goto cleanup;

            /* If we are at the end of the input file, we continue
			* encoding the remaining audio samples to the output file. */
            if (finished)
                break;
        }

        /* If we have enough samples for the encoder, we encode them.
		* At the end of the file, we pass the remaining samples to
		* the encoder. */
        while (av_audio_fifo_size(fifo) >= output_frame_size
               || (finished && av_audio_fifo_size(fifo) > 0))
            /* Take one frame worth of audio samples from the FIFO buffer,
			* encode it and write it to the output file. */
            if (load_encode_and_write(fifo, output_format_context, output_codec_context))
                goto cleanup;

        /* If we are at the end of the input file and have encoded
		* all remaining samples, we can exit this loop and finish. */
        if (finished) {
            int data_written;
            /* Flush the encoder as it may have delayed frames. */
            do {
                data_written = 0;
                if (encode_audio_frame(NULL,
                                       output_format_context,
                                       output_codec_context,
                                       &data_written))
                    goto cleanup;
            } while (data_written);
            break;
        }
    }

    /* Write the trailer of the output file container. */
    if (write_output_file_trailer(output_format_context))
        goto cleanup;
    ret = 0;

cleanup:
    if (fifo)
        av_audio_fifo_free(fifo);
    swr_free(&resample_context);
    if (output_codec_context)
        avcodec_free_context(&output_codec_context);
    if (output_format_context) {
        avio_closep(&output_format_context->pb);
        avformat_free_context(output_format_context);
    }
    if (input_codec_context)
        avcodec_free_context(&input_codec_context);
    if (input_format_context)
        avformat_close_input(&input_format_context);

    return ret;
}