/* * 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> #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;    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;    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. */    avctx->channels       = OUTPUT_CHANNELS;    avctx->channel_layout = av_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[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.         * 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 = swr_alloc_set_opts(NULL,                                              av_get_default_channel_layout(output_codec_context->channels),                                              output_codec_context->sample_fmt,                                              output_codec_context->sample_rate,                                              av_get_default_channel_layout(input_codec_context->channels),                                              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,                                      output_codec_context->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;     /* 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->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->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;    }

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