forked from KolibriOS/kolibrios
771 lines
28 KiB
C
771 lines
28 KiB
C
/*
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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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* @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 kbit/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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* Convert an error code into a text message.
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* @param error Error code to be converted
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* @return Corresponding error text (not thread-safe)
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*/
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static const char *get_error_text(const int error)
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{
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static char error_buffer[255];
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av_strerror(error, error_buffer, sizeof(error_buffer));
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return error_buffer;
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}
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/** Open an input file and the required decoder. */
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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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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, get_error_text(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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get_error_text(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]->codec->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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/** Open the decoder for the audio stream to use it later. */
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if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
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input_codec, NULL)) < 0) {
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fprintf(stderr, "Could not open input codec (error '%s')\n",
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get_error_text(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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/** Save the decoder context for easier access later. */
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*input_codec_context = (*input_format_context)->streams[0]->codec;
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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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*/
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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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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, get_error_text(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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av_strlcpy((*output_format_context)->filename, filename,
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sizeof((*output_format_context)->filename));
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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, output_codec))) {
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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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/** Save the encoder context for easier access later. */
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*output_codec_context = stream->codec;
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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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*/
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(*output_codec_context)->channels = OUTPUT_CHANNELS;
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(*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
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(*output_codec_context)->sample_rate = input_codec_context->sample_rate;
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(*output_codec_context)->sample_fmt = output_codec->sample_fmts[0];
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(*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
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/** Allow the use of the experimental AAC encoder */
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(*output_codec_context)->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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/**
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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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*/
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if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
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(*output_codec_context)->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(*output_codec_context, output_codec, NULL)) < 0) {
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fprintf(stderr, "Could not open output codec (error '%s')\n",
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get_error_text(error));
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goto cleanup;
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}
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return 0;
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cleanup:
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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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/** Initialize one data packet for reading or writing. */
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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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/** Initialize one audio frame for reading from the input file */
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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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*/
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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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/** Initialize a FIFO buffer for the audio samples to be encoded. */
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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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/** Write the header of the output file container. */
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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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get_error_text(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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/** Decode one audio frame from the input file. */
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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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get_error_text(error));
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return error;
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}
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}
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/**
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* Decode the audio frame stored in the temporary packet.
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* The input audio stream decoder is used to do this.
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* If we are at the end of the file, pass an empty packet to the decoder
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* to flush it.
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*/
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if ((error = avcodec_decode_audio4(input_codec_context, frame,
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data_present, &input_packet)) < 0) {
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fprintf(stderr, "Could not decode frame (error '%s')\n",
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get_error_text(error));
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av_free_packet(&input_packet);
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return error;
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}
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/**
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* If the decoder has not been flushed completely, we are not finished,
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* so that this function has to be called again.
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*/
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if (*finished && *data_present)
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*finished = 0;
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av_free_packet(&input_packet);
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return 0;
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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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*/
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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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/**
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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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/**
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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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*/
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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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get_error_text(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 specified
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* by frame_size.
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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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get_error_text(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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/** Add converted input audio samples to the FIFO buffer for later processing. */
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static int add_samples_to_fifo(AVAudioFifo *fifo,
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uint8_t **converted_input_samples,
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const int frame_size)
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{
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int error;
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/**
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* Make the FIFO as large as it needs to be to hold both,
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* the old and the new samples.
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*/
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if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
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fprintf(stderr, "Could not reallocate FIFO\n");
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return error;
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}
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/** Store the new samples in the FIFO buffer. */
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if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
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frame_size) < frame_size) {
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fprintf(stderr, "Could not write data to FIFO\n");
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return AVERROR_EXIT;
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}
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return 0;
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}
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/**
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* Read one audio frame from the input file, decodes, converts and stores
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* it in the FIFO buffer.
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*/
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static int read_decode_convert_and_store(AVAudioFifo *fifo,
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AVFormatContext *input_format_context,
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AVCodecContext *input_codec_context,
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AVCodecContext *output_codec_context,
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SwrContext *resampler_context,
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int *finished)
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{
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/** Temporary storage of the input samples of the frame read from the file. */
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AVFrame *input_frame = NULL;
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/** Temporary storage for the converted input samples. */
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uint8_t **converted_input_samples = NULL;
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int data_present;
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int ret = AVERROR_EXIT;
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/** Initialize temporary storage for one input frame. */
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if (init_input_frame(&input_frame))
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goto cleanup;
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/** Decode one frame worth of audio samples. */
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if (decode_audio_frame(input_frame, input_format_context,
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input_codec_context, &data_present, finished))
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goto cleanup;
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/**
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* If we are at the end of the file and there are no more samples
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* in the decoder which are delayed, we are actually finished.
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* This must not be treated as an error.
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*/
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if (*finished && !data_present) {
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ret = 0;
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goto cleanup;
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}
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/** If there is decoded data, convert and store it */
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if (data_present) {
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/** Initialize the temporary storage for the converted input samples. */
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if (init_converted_samples(&converted_input_samples, output_codec_context,
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input_frame->nb_samples))
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goto cleanup;
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/**
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* Convert the input samples to the desired output sample format.
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* This requires a temporary storage provided by converted_input_samples.
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*/
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if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
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input_frame->nb_samples, resampler_context))
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goto cleanup;
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/** Add the converted input samples to the FIFO buffer for later processing. */
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if (add_samples_to_fifo(fifo, converted_input_samples,
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input_frame->nb_samples))
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goto cleanup;
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ret = 0;
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}
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ret = 0;
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cleanup:
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if (converted_input_samples) {
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av_freep(&converted_input_samples[0]);
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free(converted_input_samples);
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}
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av_frame_free(&input_frame);
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return ret;
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}
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/**
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* Initialize one input frame for writing to the output file.
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* The frame will be exactly frame_size samples large.
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*/
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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 allocate output frame samples (error '%s')\n",
|
|
get_error_text(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. */
|
|
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;
|
|
}
|
|
|
|
/**
|
|
* Encode the audio frame and store it in the temporary packet.
|
|
* The output audio stream encoder is used to do this.
|
|
*/
|
|
if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
|
|
frame, data_present)) < 0) {
|
|
fprintf(stderr, "Could not encode frame (error '%s')\n",
|
|
get_error_text(error));
|
|
av_free_packet(&output_packet);
|
|
return error;
|
|
}
|
|
|
|
/** Write one audio frame from the temporary packet to the output file. */
|
|
if (*data_present) {
|
|
if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
|
|
fprintf(stderr, "Could not write frame (error '%s')\n",
|
|
get_error_text(error));
|
|
av_free_packet(&output_packet);
|
|
return error;
|
|
}
|
|
|
|
av_free_packet(&output_packet);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Load one audio frame from the FIFO buffer, encode and write it to the
|
|
* output file.
|
|
*/
|
|
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. */
|
|
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",
|
|
get_error_text(error));
|
|
return error;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/** Convert an audio file to an AAC file in an MP4 container. */
|
|
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);
|
|
}
|
|
|
|
/** Register all codecs and formats so that they can be used. */
|
|
av_register_all();
|
|
/** 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_close(output_codec_context);
|
|
if (output_format_context) {
|
|
avio_closep(&output_format_context->pb);
|
|
avformat_free_context(output_format_context);
|
|
}
|
|
if (input_codec_context)
|
|
avcodec_close(input_codec_context);
|
|
if (input_format_context)
|
|
avformat_close_input(&input_format_context);
|
|
|
|
return ret;
|
|
}
|