tool/VideoPlayer.git

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			/*
			* Copyright (c) 2013-2018 Andreas Unterweger
			*
			* This file is part of FFmpeg.
			*
			* FFmpeg is free software; you can redistribute it and/or
			* modify it under the terms of the GNU Lesser General Public
			* License as published by the Free Software Foundation; either
			* version 2.1 of the License, or (at your option) any later version.
			*
			* FFmpeg is distributed in the hope that it will be useful,
			* but WITHOUT ANY WARRANTY; without even the implied warranty of
			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
			* Lesser General Public License for more details.
			*
			* You should have received a copy of the GNU Lesser General Public
			* License along with FFmpeg; if not, write to the Free Software
			* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
			*/

			/**
			* @file
			* Simple audio converter
			*
			* @example transcode_aac.c
			* Convert an input audio file to AAC in an MP4 container using FFmpeg.
			* Formats other than MP4 are supported based on the output file extension.
			* @author Andreas Unterweger (dustsigns@gmail.com)
			*/

			#include <stdio.h>

			#include "libavformat/avformat.h"
			#include "libavformat/avio.h"

			#include "libavcodec/avcodec.h"

			#include "libavutil/audio_fifo.h"
			#include "libavutil/avassert.h"
			#include "libavutil/avstring.h"
			#include "libavutil/frame.h"
			#include "libavutil/opt.h"

			#include "libswresample/swresample.h"

			/* The output bit rate in bit/s */
			#define OUTPUT_BIT_RATE 96000
			/* The number of output channels */
			#define OUTPUT_CHANNELS 2

			/**
			* Open an input file and the required decoder.
			* @param filename File to be opened
			* @param[out] input_format_context Format context of opened file
			* @param[out] input_codec_context Codec context of opened file
			* @return Error code (0 if successful)
			*/
			static int open_input_file(const char *filename,
			AVFormatContext **input_format_context,
			AVCodecContext **input_codec_context)
			{
			AVCodecContext *avctx;
			AVCodec *input_codec;
			int error;

			/* Open the input file to read from it. */
			if ((error = avformat_open_input(input_format_context, filename, NULL,
			NULL)) < 0) {
			fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
			filename, av_err2str(error));
			*input_format_context = NULL;
			return error;
			}

			/* Get information on the input file (number of streams etc.). */
			if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
			fprintf(stderr, "Could not open find stream info (error '%s')\n",
			av_err2str(error));
			avformat_close_input(input_format_context);
			return error;
			}

			/* Make sure that there is only one stream in the input file. */
			if ((*input_format_context)->nb_streams != 1) {
			fprintf(stderr, "Expected one audio input stream, but found %d\n",
			(*input_format_context)->nb_streams);
			avformat_close_input(input_format_context);
			return AVERROR_EXIT;
			}

			/* Find a decoder for the audio stream. */
			if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
			fprintf(stderr, "Could not find input codec\n");
			avformat_close_input(input_format_context);
			return AVERROR_EXIT;
			}

			/* Allocate a new decoding context. */
			avctx = avcodec_alloc_context3(input_codec);
			if (!avctx) {
			fprintf(stderr, "Could not allocate a decoding context\n");
			avformat_close_input(input_format_context);
			return AVERROR(ENOMEM);
			}

			/* Initialize the stream parameters with demuxer information. */
			error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
			if (error < 0) {
			avformat_close_input(input_format_context);
			avcodec_free_context(&avctx);
			return error;
			}

			/* Open the decoder for the audio stream to use it later. */
			if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
			fprintf(stderr, "Could not open input codec (error '%s')\n",
			av_err2str(error));
			avcodec_free_context(&avctx);
			avformat_close_input(input_format_context);
			return error;
			}

			/* Save the decoder context for easier access later. */
			*input_codec_context = avctx;

			return 0;
			}

			/**
			* Open an output file and the required encoder.
			* Also set some basic encoder parameters.
			* Some of these parameters are based on the input file's parameters.
			* @param filename File to be opened
			* @param input_codec_context Codec context of input file
			* @param[out] output_format_context Format context of output file
			* @param[out] output_codec_context Codec context of output file
			* @return Error code (0 if successful)
			*/
			static int open_output_file(const char *filename,
			AVCodecContext *input_codec_context,
			AVFormatContext **output_format_context,
			AVCodecContext **output_codec_context)
			{
			AVCodecContext *avctx = NULL;
			AVIOContext *output_io_context = NULL;
			AVStream *stream = NULL;
			AVCodec *output_codec = NULL;
			int error;

			/* Open the output file to write to it. */
			if ((error = avio_open(&output_io_context, filename,
			AVIO_FLAG_WRITE)) < 0) {
			fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
			filename, av_err2str(error));
			return error;
			}

			/* Create a new format context for the output container format. */
			if (!(*output_format_context = avformat_alloc_context())) {
			fprintf(stderr, "Could not allocate output format context\n");
			return AVERROR(ENOMEM);
			}

			/* Associate the output file (pointer) with the container format context. */
			(*output_format_context)->pb = output_io_context;

			/* Guess the desired container format based on the file extension. */
			if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
			NULL))) {
			fprintf(stderr, "Could not find output file format\n");
			goto cleanup;
			}

			if (!((*output_format_context)->url = av_strdup(filename))) {
			fprintf(stderr, "Could not allocate url.\n");
			error = AVERROR(ENOMEM);
			goto cleanup;
			}

			/* Find the encoder to be used by its name. */
			if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
			fprintf(stderr, "Could not find an AAC encoder.\n");
			goto cleanup;
			}

			/* Create a new audio stream in the output file container. */
			if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
			fprintf(stderr, "Could not create new stream\n");
			error = AVERROR(ENOMEM);
			goto cleanup;
			}

			avctx = avcodec_alloc_context3(output_codec);
			if (!avctx) {
			fprintf(stderr, "Could not allocate an encoding context\n");
			error = AVERROR(ENOMEM);
			goto cleanup;
			}

			/* Set the basic encoder parameters.
			* The input file's sample rate is used to avoid a sample rate conversion. */
			avctx->channels = OUTPUT_CHANNELS;
			avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
			avctx->sample_rate = input_codec_context->sample_rate;
			avctx->sample_fmt = output_codec->sample_fmts[0];
			avctx->bit_rate = OUTPUT_BIT_RATE;

			/* Allow the use of the experimental AAC encoder. */
			avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

			/* Set the sample rate for the container. */
			stream->time_base.den = input_codec_context->sample_rate;
			stream->time_base.num = 1;

			/* Some container formats (like MP4) require global headers to be present.
			* Mark the encoder so that it behaves accordingly. */
			if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
			avctx->flags \|= AV_CODEC_FLAG_GLOBAL_HEADER;

			/* Open the encoder for the audio stream to use it later. */
			if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
			fprintf(stderr, "Could not open output codec (error '%s')\n",
			av_err2str(error));
			goto cleanup;
			}

			error = avcodec_parameters_from_context(stream->codecpar, avctx);
			if (error < 0) {
			fprintf(stderr, "Could not initialize stream parameters\n");
			goto cleanup;
			}

			/* Save the encoder context for easier access later. */
			*output_codec_context = avctx;

			return 0;

			cleanup:
			avcodec_free_context(&avctx);
			avio_closep(&(*output_format_context)->pb);
			avformat_free_context(*output_format_context);
			*output_format_context = NULL;
			return error < 0 ? error : AVERROR_EXIT;
			}

			/**
			* Initialize one data packet for reading or writing.
			* @param packet Packet to be initialized
			*/
			static void init_packet(AVPacket *packet)
			{
			av_init_packet(packet);
			/* Set the packet data and size so that it is recognized as being empty. */
			packet->data = NULL;
			packet->size = 0;
			}

			/**
			* Initialize one audio frame for reading from the input file.
			* @param[out] frame Frame to be initialized
			* @return Error code (0 if successful)
			*/
			static int init_input_frame(AVFrame **frame)
			{
			if (!(*frame = av_frame_alloc())) {
			fprintf(stderr, "Could not allocate input frame\n");
			return AVERROR(ENOMEM);
			}
			return 0;
			}

			/**
			* Initialize the audio resampler based on the input and output codec settings.
			* If the input and output sample formats differ, a conversion is required
			* libswresample takes care of this, but requires initialization.
			* @param input_codec_context Codec context of the input file
			* @param output_codec_context Codec context of the output file
			* @param[out] resample_context Resample context for the required conversion
			* @return Error code (0 if successful)
			*/
			static int init_resampler(AVCodecContext *input_codec_context,
			AVCodecContext *output_codec_context,
			SwrContext **resample_context)
			{
			int error;

			/*
			* Create a resampler context for the conversion.
			* Set the conversion parameters.
			* Default channel layouts based on the number of channels
			* are assumed for simplicity (they are sometimes not detected
			* properly by the demuxer and/or decoder).
			*/
			*resample_context = swr_alloc_set_opts(NULL,
			av_get_default_channel_layout(output_codec_context->channels),
			output_codec_context->sample_fmt,
			output_codec_context->sample_rate,
			av_get_default_channel_layout(input_codec_context->channels),
			input_codec_context->sample_fmt,
			input_codec_context->sample_rate,
			0, NULL);
			if (!*resample_context) {
			fprintf(stderr, "Could not allocate resample context\n");
			return AVERROR(ENOMEM);
			}
			/*
			* Perform a sanity check so that the number of converted samples is
			* not greater than the number of samples to be converted.
			* If the sample rates differ, this case has to be handled differently
			*/
			av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);

			/* Open the resampler with the specified parameters. */
			if ((error = swr_init(*resample_context)) < 0) {
			fprintf(stderr, "Could not open resample context\n");
			swr_free(resample_context);
			return error;
			}
			return 0;
			}

			/**
			* Initialize a FIFO buffer for the audio samples to be encoded.
			* @param[out] fifo Sample buffer
			* @param output_codec_context Codec context of the output file
			* @return Error code (0 if successful)
			*/
			static int init_fifo(AVAudioFifo *fifo, AVCodecContext output_codec_context)
			{
			/* Create the FIFO buffer based on the specified output sample format. */
			if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
			output_codec_context->channels, 1))) {
			fprintf(stderr, "Could not allocate FIFO\n");
			return AVERROR(ENOMEM);
			}
			return 0;
			}

			/**
			* Write the header of the output file container.
			* @param output_format_context Format context of the output file
			* @return Error code (0 if successful)
			*/
			static int write_output_file_header(AVFormatContext *output_format_context)
			{
			int error;
			if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
			fprintf(stderr, "Could not write output file header (error '%s')\n",
			av_err2str(error));
			return error;
			}
			return 0;
			}

			/**
			* Decode one audio frame from the input file.
			* @param frame Audio frame to be decoded
			* @param input_format_context Format context of the input file
			* @param input_codec_context Codec context of the input file
			* @param[out] data_present Indicates whether data has been decoded
			* @param[out] finished Indicates whether the end of file has
			* been reached and all data has been
			* decoded. If this flag is false, there
			* is more data to be decoded, i.e., this
			* function has to be called again.
			* @return Error code (0 if successful)
			*/
			static int decode_audio_frame(AVFrame *frame,
			AVFormatContext *input_format_context,
			AVCodecContext *input_codec_context,
			int data_present, int finished)
			{
			/* Packet used for temporary storage. */
			AVPacket input_packet;
			int error;
			init_packet(&input_packet);

			/* Read one audio frame from the input file into a temporary packet. */
			if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
			/* If we are at the end of the file, flush the decoder below. */
			if (error == AVERROR_EOF)
			*finished = 1;
			else {
			fprintf(stderr, "Could not read frame (error '%s')\n",
			av_err2str(error));
			return error;
			}
			}

			/* Send the audio frame stored in the temporary packet to the decoder.
			* The input audio stream decoder is used to do this. */
			if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
			fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
			av_err2str(error));
			return error;
			}

			/* Receive one frame from the decoder. */
			error = avcodec_receive_frame(input_codec_context, frame);
			/* If the decoder asks for more data to be able to decode a frame,
			* return indicating that no data is present. */
			if (error == AVERROR(EAGAIN)) {
			error = 0;
			goto cleanup;
			/* If the end of the input file is reached, stop decoding. */
			} else if (error == AVERROR_EOF) {
			*finished = 1;
			error = 0;
			goto cleanup;
			} else if (error < 0) {
			fprintf(stderr, "Could not decode frame (error '%s')\n",
			av_err2str(error));
			goto cleanup;
			/* Default case: Return decoded data. */
			} else {
			*data_present = 1;
			goto cleanup;
			}

			cleanup:
			av_packet_unref(&input_packet);
			return error;
			}

			/**
			* Initialize a temporary storage for the specified number of audio samples.
			* The conversion requires temporary storage due to the different format.
			* The number of audio samples to be allocated is specified in frame_size.
			* @param[out] converted_input_samples Array of converted samples. The
			* dimensions are reference, channel
			* (for multi-channel audio), sample.
			* @param output_codec_context Codec context of the output file
			* @param frame_size Number of samples to be converted in
			* each round
			* @return Error code (0 if successful)
			*/
			static int init_converted_samples(uint8_t ***converted_input_samples,
			AVCodecContext *output_codec_context,
			int frame_size)
			{
			int error;

			/* Allocate as many pointers as there are audio channels.
			* Each pointer will later point to the audio samples of the corresponding
			* channels (although it may be NULL for interleaved formats).
			*/
			if (!(*converted_input_samples = calloc(output_codec_context->channels,
			sizeof(**converted_input_samples)))) {
			fprintf(stderr, "Could not allocate converted input sample pointers\n");
			return AVERROR(ENOMEM);
			}

			/* Allocate memory for the samples of all channels in one consecutive
			* block for convenience. */
			if ((error = av_samples_alloc(*converted_input_samples, NULL,
			output_codec_context->channels,
			frame_size,
			output_codec_context->sample_fmt, 0)) < 0) {
			fprintf(stderr,
			"Could not allocate converted input samples (error '%s')\n",
			av_err2str(error));
			av_freep(&(*converted_input_samples)[0]);
			free(*converted_input_samples);
			return error;
			}
			return 0;
			}

			/**
			* Convert the input audio samples into the output sample format.
			* The conversion happens on a per-frame basis, the size of which is
			* specified by frame_size.
			* @param input_data Samples to be decoded. The dimensions are
			* channel (for multi-channel audio), sample.
			* @param[out] converted_data Converted samples. The dimensions are channel
			* (for multi-channel audio), sample.
			* @param frame_size Number of samples to be converted
			* @param resample_context Resample context for the conversion
			* @return Error code (0 if successful)
			*/
			static int convert_samples(const uint8_t **input_data,
			uint8_t **converted_data, const int frame_size,
			SwrContext *resample_context)
			{
			int error;

			/* Convert the samples using the resampler. */
			if ((error = swr_convert(resample_context,
			converted_data, frame_size,
			input_data , frame_size)) < 0) {
			fprintf(stderr, "Could not convert input samples (error '%s')\n",
			av_err2str(error));
			return error;
			}

			return 0;
			}

			/**
			* Add converted input audio samples to the FIFO buffer for later processing.
			* @param fifo Buffer to add the samples to
			* @param converted_input_samples Samples to be added. The dimensions are channel
			* (for multi-channel audio), sample.
			* @param frame_size Number of samples to be converted
			* @return Error code (0 if successful)
			*/
			static int add_samples_to_fifo(AVAudioFifo *fifo,
			uint8_t **converted_input_samples,
			const int frame_size)
			{
			int error;

			/* Make the FIFO as large as it needs to be to hold both,
			* the old and the new samples. */
			if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
			fprintf(stderr, "Could not reallocate FIFO\n");
			return error;
			}

			/* Store the new samples in the FIFO buffer. */
			if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
			frame_size) < frame_size) {
			fprintf(stderr, "Could not write data to FIFO\n");
			return AVERROR_EXIT;
			}
			return 0;
			}

			/**
			* Read one audio frame from the input file, decode, convert and store
			* it in the FIFO buffer.
			* @param fifo Buffer used for temporary storage
			* @param input_format_context Format context of the input file
			* @param input_codec_context Codec context of the input file
			* @param output_codec_context Codec context of the output file
			* @param resampler_context Resample context for the conversion
			* @param[out] finished Indicates whether the end of file has
			* been reached and all data has been
			* decoded. If this flag is false,
			* there is more data to be decoded,
			* i.e., this function has to be called
			* again.
			* @return Error code (0 if successful)
			*/
			static int read_decode_convert_and_store(AVAudioFifo *fifo,
			AVFormatContext *input_format_context,
			AVCodecContext *input_codec_context,
			AVCodecContext *output_codec_context,
			SwrContext *resampler_context,
			int *finished)
			{
			/* Temporary storage of the input samples of the frame read from the file. */
			AVFrame *input_frame = NULL;
			/* Temporary storage for the converted input samples. */
			uint8_t **converted_input_samples = NULL;
			int data_present = 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;
			}