/* * 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. * @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/frame.h"#include "libavutil/opt.h" #include "libswresample/swresample.h" /** The output bit rate in kbit/s */#define OUTPUT_BIT_RATE 96000/** The number of output channels */#define OUTPUT_CHANNELS 2 /** Open an input file and the required decoder. */static int open_input_file(const char *filename,                           AVFormatContext **input_format_context,                           AVCodecContext **input_codec_context){    AVCodecContext *avctx;    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. */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;    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;    }     av_strlcpy((*output_format_context)->filename, filename,               sizeof((*output_format_context)->filename));     /** 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. */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 */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. */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. */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. */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. */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;        }    }     /**     * Decode the audio frame stored in the temporary packet.     * The input audio stream decoder is used to do this.     * If we are at the end of the file, pass an empty packet to the decoder     * to flush it.     */    if ((error = avcodec_decode_audio4(input_codec_context, frame,                                       data_present, &input_packet)) < 0) {        fprintf(stderr, "Could not decode frame (error '%s')\n",                av_err2str(error));        av_packet_unref(&input_packet);        return error;    }     /**     * If the decoder has not been flushed completely, we are not finished,     * so that this function has to be called again.     */    if (*finished && *data_present)        *finished = 0;    av_packet_unref(&input_packet);    return 0;} /** * 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. */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. */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. */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, decodes, converts and stores * it in the FIFO buffer. */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 && !data_present) {        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.

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