transcode_aac.c

Convert an input audio file to AAC in an MP4 container using FFmpeg.

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 (dusts.nosp@m.igns.nosp@m.@gmai.nosp@m.l.co.nosp@m.m)

/*
 * Copyright (c) 2013-2022 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>
 
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
 
#include "libavcodec/avcodec.h"
 
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
 
#include "libswresample/swresample.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;
    const AVCodec *input_codec;
    const AVStream *stream;
    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;
    }
 
    stream = (*input_format_context)->streams[0];
 
    /* Find a decoder for the audio stream. */
    if (!(input_codec = avcodec_find_decoder(stream->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, stream->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;
    }
 
    /* Set the packet timebase for the decoder. */
    avctx->pkt_timebase = stream->time_base;
 
    /* 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;
    const AVCodec *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. */
    av_channel_layout_default(&avctx->ch_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;
 
    /* 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[out] packet Packet to be initialized
 * @return Error code (0 if successful)
 */
static int init_packet(AVPacket **packet)
{
    if (!(*packet = av_packet_alloc())) {
        fprintf(stderr, "Could not allocate packet\n");
        return AVERROR(ENOMEM);
    }
    return 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.
         */
        error = swr_alloc_set_opts2(resample_context,
                                             &output_codec_context->ch_layout,
                                              output_codec_context->sample_fmt,
                                              output_codec_context->sample_rate,
                                             &input_codec_context->ch_layout,
                                              input_codec_context->sample_fmt,
                                              input_codec_context->sample_rate,
                                              0, NULL);
        if (error < 0) {
            fprintf(stderr, "Could not allocate resample context\n");
            return error;
        }
        /*
        * 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,
                                      output_codec_context->ch_layout.nb_channels, 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;
 
    error = init_packet(&input_packet);
    if (error < 0)
        return error;
 
    *data_present = 0;
    *finished = 0;
    /* 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));
            goto cleanup;
        }
    }
 
    /* 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));
        goto cleanup;
    }
 
    /* 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_free(&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 = calloc(output_codec_context->ch_layout.nb_channels,
                                            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 = av_samples_alloc(*converted_input_samples, NULL,
                                  output_codec_context->ch_layout.nb_channels,
                                  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;
    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;
    av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
    (*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;
 
    error = init_packet(&output_packet);
    if (error < 0)
        return error;
 
    /* Set a timestamp based on the sample rate for the container. */
    if (frame) {
        frame->pts = pts;
        pts += frame->nb_samples;
    }
 
    *data_present = 0;
    /* 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);
    /* Check for errors, but proceed with fetching encoded samples if the
     *  encoder signals that it has nothing more to encode. */
    if (error < 0 && error != AVERROR_EOF) {
      fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
              av_err2str(error));
      goto cleanup;
    }
 
    /* 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_free(&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 main(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);
    }
 
    /* Open the input file for reading. */
    if (open_input_file(argv[1], &input_format_context,
                        &input_codec_context))
        goto cleanup;
    /* Open the output file for writing. */
    if (open_output_file(argv[2], 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 {
                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;
}