886 lines
34 KiB
C
886 lines
34 KiB
C
/*
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* Copyright (c) 2013-2018 Andreas Unterweger
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Simple audio converter
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*
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* @example transcode_aac.c
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* Convert an input audio file to AAC in an MP4 container using FFmpeg.
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* Formats other than MP4 are supported based on the output file extension.
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* @author Andreas Unterweger (dustsigns@gmail.com)
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*/
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#include <stdio.h>
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#include "libavformat/avformat.h"
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#include "libavformat/avio.h"
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#include "libavcodec/avcodec.h"
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#include "libavutil/audio_fifo.h"
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#include "libavutil/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/frame.h"
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#include "libavutil/opt.h"
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#include "libswresample/swresample.h"
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/* The output bit rate in bit/s */
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#define OUTPUT_BIT_RATE 96000
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/* The number of output channels */
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#define OUTPUT_CHANNELS 2
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/**
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* Open an input file and the required decoder.
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* @param filename File to be opened
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* @param[out] input_format_context Format context of opened file
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* @param[out] input_codec_context Codec context of opened file
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* @return Error code (0 if successful)
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*/
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static int open_input_file(const char *filename,
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AVFormatContext **input_format_context,
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AVCodecContext **input_codec_context)
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{
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AVCodecContext *avctx;
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AVCodec *input_codec;
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int error;
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/* Open the input file to read from it. */
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if ((error = avformat_open_input(input_format_context, filename, NULL,
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NULL)) < 0) {
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fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
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filename, av_err2str(error));
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*input_format_context = NULL;
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return error;
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}
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/* Get information on the input file (number of streams etc.). */
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if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
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fprintf(stderr, "Could not open find stream info (error '%s')\n",
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av_err2str(error));
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avformat_close_input(input_format_context);
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return error;
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}
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/* Make sure that there is only one stream in the input file. */
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if ((*input_format_context)->nb_streams != 1) {
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fprintf(stderr, "Expected one audio input stream, but found %d\n",
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(*input_format_context)->nb_streams);
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avformat_close_input(input_format_context);
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return AVERROR_EXIT;
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}
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/* Find a decoder for the audio stream. */
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if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
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fprintf(stderr, "Could not find input codec\n");
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avformat_close_input(input_format_context);
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return AVERROR_EXIT;
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}
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/* Allocate a new decoding context. */
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avctx = avcodec_alloc_context3(input_codec);
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if (!avctx) {
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fprintf(stderr, "Could not allocate a decoding context\n");
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avformat_close_input(input_format_context);
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return AVERROR(ENOMEM);
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}
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/* Initialize the stream parameters with demuxer information. */
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error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
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if (error < 0) {
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avformat_close_input(input_format_context);
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avcodec_free_context(&avctx);
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return error;
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}
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/* Open the decoder for the audio stream to use it later. */
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if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
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fprintf(stderr, "Could not open input codec (error '%s')\n",
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av_err2str(error));
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avcodec_free_context(&avctx);
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avformat_close_input(input_format_context);
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return error;
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}
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/* Save the decoder context for easier access later. */
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*input_codec_context = avctx;
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return 0;
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}
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/**
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* Open an output file and the required encoder.
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* Also set some basic encoder parameters.
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* Some of these parameters are based on the input file's parameters.
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* @param filename File to be opened
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* @param input_codec_context Codec context of input file
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* @param[out] output_format_context Format context of output file
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* @param[out] output_codec_context Codec context of output file
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* @return Error code (0 if successful)
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*/
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static int open_output_file(const char *filename,
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AVCodecContext *input_codec_context,
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AVFormatContext **output_format_context,
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AVCodecContext **output_codec_context)
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{
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AVCodecContext *avctx = NULL;
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AVIOContext *output_io_context = NULL;
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AVStream *stream = NULL;
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AVCodec *output_codec = NULL;
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int error;
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/* Open the output file to write to it. */
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if ((error = avio_open(&output_io_context, filename,
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AVIO_FLAG_WRITE)) < 0) {
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fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
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filename, av_err2str(error));
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return error;
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}
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/* Create a new format context for the output container format. */
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if (!(*output_format_context = avformat_alloc_context())) {
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fprintf(stderr, "Could not allocate output format context\n");
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return AVERROR(ENOMEM);
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}
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/* Associate the output file (pointer) with the container format context. */
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(*output_format_context)->pb = output_io_context;
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/* Guess the desired container format based on the file extension. */
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if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
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NULL))) {
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fprintf(stderr, "Could not find output file format\n");
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goto cleanup;
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}
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if (!((*output_format_context)->url = av_strdup(filename))) {
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fprintf(stderr, "Could not allocate url.\n");
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error = AVERROR(ENOMEM);
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goto cleanup;
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}
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/* Find the encoder to be used by its name. */
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if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
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fprintf(stderr, "Could not find an AAC encoder.\n");
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goto cleanup;
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}
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/* Create a new audio stream in the output file container. */
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if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
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fprintf(stderr, "Could not create new stream\n");
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error = AVERROR(ENOMEM);
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goto cleanup;
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}
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avctx = avcodec_alloc_context3(output_codec);
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if (!avctx) {
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fprintf(stderr, "Could not allocate an encoding context\n");
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error = AVERROR(ENOMEM);
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goto cleanup;
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}
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/* Set the basic encoder parameters.
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* The input file's sample rate is used to avoid a sample rate conversion. */
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avctx->channels = OUTPUT_CHANNELS;
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avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
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avctx->sample_rate = input_codec_context->sample_rate;
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avctx->sample_fmt = output_codec->sample_fmts[0];
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avctx->bit_rate = OUTPUT_BIT_RATE;
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/* Allow the use of the experimental AAC encoder. */
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avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
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/* Set the sample rate for the container. */
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stream->time_base.den = input_codec_context->sample_rate;
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stream->time_base.num = 1;
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/* Some container formats (like MP4) require global headers to be present.
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* Mark the encoder so that it behaves accordingly. */
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if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
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avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
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/* Open the encoder for the audio stream to use it later. */
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if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
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fprintf(stderr, "Could not open output codec (error '%s')\n",
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av_err2str(error));
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goto cleanup;
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}
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error = avcodec_parameters_from_context(stream->codecpar, avctx);
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if (error < 0) {
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fprintf(stderr, "Could not initialize stream parameters\n");
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goto cleanup;
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}
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/* Save the encoder context for easier access later. */
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*output_codec_context = avctx;
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return 0;
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cleanup:
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avcodec_free_context(&avctx);
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avio_closep(&(*output_format_context)->pb);
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avformat_free_context(*output_format_context);
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*output_format_context = NULL;
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return error < 0 ? error : AVERROR_EXIT;
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}
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/**
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* Initialize one data packet for reading or writing.
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* @param packet Packet to be initialized
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*/
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static void init_packet(AVPacket *packet)
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{
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av_init_packet(packet);
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/* Set the packet data and size so that it is recognized as being empty. */
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packet->data = NULL;
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packet->size = 0;
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}
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/**
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* Initialize one audio frame for reading from the input file.
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* @param[out] frame Frame to be initialized
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* @return Error code (0 if successful)
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*/
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static int init_input_frame(AVFrame **frame)
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{
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if (!(*frame = av_frame_alloc())) {
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fprintf(stderr, "Could not allocate input frame\n");
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return AVERROR(ENOMEM);
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}
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return 0;
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}
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/**
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* Initialize the audio resampler based on the input and output codec settings.
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* If the input and output sample formats differ, a conversion is required
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* libswresample takes care of this, but requires initialization.
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* @param input_codec_context Codec context of the input file
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* @param output_codec_context Codec context of the output file
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* @param[out] resample_context Resample context for the required conversion
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* @return Error code (0 if successful)
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*/
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static int init_resampler(AVCodecContext *input_codec_context,
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AVCodecContext *output_codec_context,
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SwrContext **resample_context)
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{
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int error;
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/*
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* Create a resampler context for the conversion.
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* Set the conversion parameters.
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* Default channel layouts based on the number of channels
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* are assumed for simplicity (they are sometimes not detected
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* properly by the demuxer and/or decoder).
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*/
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*resample_context = swr_alloc_set_opts(NULL,
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av_get_default_channel_layout(output_codec_context->channels),
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output_codec_context->sample_fmt,
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output_codec_context->sample_rate,
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av_get_default_channel_layout(input_codec_context->channels),
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input_codec_context->sample_fmt,
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input_codec_context->sample_rate,
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0, NULL);
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if (!*resample_context) {
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fprintf(stderr, "Could not allocate resample context\n");
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return AVERROR(ENOMEM);
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}
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/*
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* Perform a sanity check so that the number of converted samples is
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* not greater than the number of samples to be converted.
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* If the sample rates differ, this case has to be handled differently
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*/
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av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
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/* Open the resampler with the specified parameters. */
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if ((error = swr_init(*resample_context)) < 0) {
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fprintf(stderr, "Could not open resample context\n");
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swr_free(resample_context);
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return error;
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}
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return 0;
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}
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/**
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* Initialize a FIFO buffer for the audio samples to be encoded.
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* @param[out] fifo Sample buffer
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* @param output_codec_context Codec context of the output file
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* @return Error code (0 if successful)
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*/
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static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
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{
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/* Create the FIFO buffer based on the specified output sample format. */
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if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
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output_codec_context->channels, 1))) {
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fprintf(stderr, "Could not allocate FIFO\n");
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return AVERROR(ENOMEM);
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}
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return 0;
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}
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/**
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* Write the header of the output file container.
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* @param output_format_context Format context of the output file
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* @return Error code (0 if successful)
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*/
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static int write_output_file_header(AVFormatContext *output_format_context)
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{
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int error;
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if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
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fprintf(stderr, "Could not write output file header (error '%s')\n",
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av_err2str(error));
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return error;
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}
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return 0;
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}
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/**
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* Decode one audio frame from the input file.
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* @param frame Audio frame to be decoded
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* @param input_format_context Format context of the input file
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* @param input_codec_context Codec context of the input file
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* @param[out] data_present Indicates whether data has been decoded
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* @param[out] finished Indicates whether the end of file has
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* been reached and all data has been
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* decoded. If this flag is false, there
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* is more data to be decoded, i.e., this
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* function has to be called again.
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* @return Error code (0 if successful)
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*/
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static int decode_audio_frame(AVFrame *frame,
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AVFormatContext *input_format_context,
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AVCodecContext *input_codec_context,
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int *data_present, int *finished)
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{
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/* Packet used for temporary storage. */
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AVPacket input_packet;
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int error;
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init_packet(&input_packet);
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/* Read one audio frame from the input file into a temporary packet. */
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if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
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/* If we are at the end of the file, flush the decoder below. */
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if (error == AVERROR_EOF)
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*finished = 1;
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else {
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fprintf(stderr, "Could not read frame (error '%s')\n",
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av_err2str(error));
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return error;
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}
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}
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/* Send the audio frame stored in the temporary packet to the decoder.
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* The input audio stream decoder is used to do this. */
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if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
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fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
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av_err2str(error));
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return error;
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}
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/* Receive one frame from the decoder. */
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error = avcodec_receive_frame(input_codec_context, frame);
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/* If the decoder asks for more data to be able to decode a frame,
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* return indicating that no data is present. */
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if (error == AVERROR(EAGAIN)) {
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error = 0;
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goto cleanup;
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/* If the end of the input file is reached, stop decoding. */
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} else if (error == AVERROR_EOF) {
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*finished = 1;
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error = 0;
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goto cleanup;
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} else if (error < 0) {
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fprintf(stderr, "Could not decode frame (error '%s')\n",
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av_err2str(error));
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goto cleanup;
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/* Default case: Return decoded data. */
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} else {
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*data_present = 1;
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goto cleanup;
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}
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cleanup:
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av_packet_unref(&input_packet);
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return error;
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}
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/**
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* Initialize a temporary storage for the specified number of audio samples.
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* The conversion requires temporary storage due to the different format.
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* The number of audio samples to be allocated is specified in frame_size.
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* @param[out] converted_input_samples Array of converted samples. The
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* dimensions are reference, channel
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* (for multi-channel audio), sample.
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* @param output_codec_context Codec context of the output file
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* @param frame_size Number of samples to be converted in
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* each round
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* @return Error code (0 if successful)
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*/
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static int init_converted_samples(uint8_t ***converted_input_samples,
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AVCodecContext *output_codec_context,
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int frame_size)
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{
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int error;
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/* Allocate as many pointers as there are audio channels.
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* Each pointer will later point to the audio samples of the corresponding
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* channels (although it may be NULL for interleaved formats).
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*/
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if (!(*converted_input_samples = calloc(output_codec_context->channels,
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sizeof(**converted_input_samples)))) {
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fprintf(stderr, "Could not allocate converted input sample pointers\n");
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return AVERROR(ENOMEM);
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}
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/* Allocate memory for the samples of all channels in one consecutive
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* block for convenience. */
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if ((error = av_samples_alloc(*converted_input_samples, NULL,
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output_codec_context->channels,
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frame_size,
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output_codec_context->sample_fmt, 0)) < 0) {
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fprintf(stderr,
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"Could not allocate converted input samples (error '%s')\n",
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av_err2str(error));
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av_freep(&(*converted_input_samples)[0]);
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free(*converted_input_samples);
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return error;
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}
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return 0;
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}
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/**
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* Convert the input audio samples into the output sample format.
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* The conversion happens on a per-frame basis, the size of which is
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* specified by frame_size.
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* @param input_data Samples to be decoded. The dimensions are
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* channel (for multi-channel audio), sample.
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* @param[out] converted_data Converted samples. The dimensions are channel
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* (for multi-channel audio), sample.
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* @param frame_size Number of samples to be converted
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* @param resample_context Resample context for the conversion
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* @return Error code (0 if successful)
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*/
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static int convert_samples(const uint8_t **input_data,
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uint8_t **converted_data, const int frame_size,
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SwrContext *resample_context)
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{
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int error;
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/* Convert the samples using the resampler. */
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if ((error = swr_convert(resample_context,
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converted_data, frame_size,
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input_data , frame_size)) < 0) {
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fprintf(stderr, "Could not convert input samples (error '%s')\n",
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av_err2str(error));
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return error;
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}
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return 0;
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}
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/**
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* 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;
|
|
(*frame)->channel_layout = output_codec_context->channel_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;
|
|
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 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 {
|
|
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;
|
|
}
|