FFmpeg-examples-transcode_aac

源码来源于官网FFmpeg-example：https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/transcoding.c

该example官网例子还使用avcodec_encode_audio4 API，以下使用了新的API avcodec_send_packet/avcodec_receive_frame等

Version:FFmpeg 3.3.2

#include "stdafx.h"
#include "transcode_audio_et.h"

/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 25000
/** The number of output channels */
#define OUTPUT_CHANNELS 1

/** Open an input file and the required decoder. */
static int open_input_file(const char *filename,
	AVFormatContext **input_format_context,
	AVCodecContext **input_codec_context)
{
	AVCodecContext *avctx;
	AVCodec *input_codec;
	int error;

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

	/** Get information on the input file (number of streams etc.). */
	if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
		fprintf(stderr, "Could not open find stream info (error '%s')\n",
			av_err2str(error));
		avformat_close_input(input_format_context);
		return error;
	}
	//为什么确保只有一个流
	/** Make sure that there is only one stream in the input file. */
	if ((*input_format_context)->nb_streams != 1) {
		fprintf(stderr, "Expected one audio input stream, but found %d\n",
			(*input_format_context)->nb_streams);
		avformat_close_input(input_format_context);
		return AVERROR_EXIT;
	}

	/** Find a decoder for the audio stream. */
	if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
		fprintf(stderr, "Could not find input codec\n");
		avformat_close_input(input_format_context);
		return AVERROR_EXIT;
	}
	/** allocate a new decoding context */
	avctx = avcodec_alloc_context3(input_codec);
	if (!avctx) {
		fprintf(stderr, "Could not allocate a decoding context\n");
		avformat_close_input(input_format_context);
		return AVERROR(ENOMEM);
	}

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

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

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

	return 0;
}

/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
*/
static int open_output_file(const char *filename,
	AVCodecContext *input_codec_context,
	AVFormatContext **output_format_context,
	AVCodecContext **output_codec_context)
{
	AVCodecContext *avctx = NULL;
	AVIOContext *output_io_context = NULL;
	AVStream *stream = NULL;
	AVCodec *output_codec = NULL;
	int error;
	//out_format_context->pb?
	/** Open the output file to write to it. */
	if ((error = avio_open(&output_io_context, filename,
		AVIO_FLAG_WRITE)) < 0) {
		fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
			filename, av_err2str(error));
		return error;
	}

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

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

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

	av_strlcpy((*output_format_context)->filename, filename,
		sizeof((*output_format_context)->filename));

	/** Find the encoder to be used by its name. */
	if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_MP3))) {
		fprintf(stderr, "Could not find an AAC encoder.\n");
		goto cleanup;
	}

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

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

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

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

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

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

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

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

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

	return 0;

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

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

/** Initialize one audio frame for reading from the input file */
static int init_input_frame(AVFrame **frame)
{
	if (!(*frame = av_frame_alloc())) {
		fprintf(stderr, "Could not allocate input frame\n");
		return AVERROR(ENOMEM);
	}
	return 0;
}

/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
*/
static int init_resampler(AVCodecContext *input_codec_context,
	AVCodecContext *output_codec_context,
	SwrContext **resample_context)
{
	int error;

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

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

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

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

/** Decode one audio frame from the input file. */
static int decode_audio_frame(AVFrame *frame,
	AVFormatContext *input_format_context,
	AVCodecContext *input_codec_context,
	int *data_present, int *finished)
{
	/** Packet used for temporary storage. */
	AVPacket input_packet;
	int error;
	init_packet(&input_packet);
	int recv_err, send_err;
	/* */
	*data_present = 1;

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

	/**
	* Decode the audio frame stored in the temporary packet.
	* The input audio stream decoder is used to do this.
	* If we are at the end of the file, pass an empty packet to the decoder
	* to flush it.
	*/

	/* if ((error = avcodec_decode_audio4(input_codec_context, frame,
	data_present, &input_packet)) < 0) {
	fprintf(stderr, "Could not decode frame (error '%s')\n",
	av_err2str(error));
	av_packet_unref(&input_packet);
	return error;
	}*/
	if ((recv_err = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
		fprintf(stderr, "avcodec_send_packet failed (error '%s')\n",
			av_err2str(recv_err));
	}
	if ((send_err = avcodec_receive_frame(input_codec_context, frame)) < 0) {
		fprintf(stderr, "avcodec_receive_frame failed (error '%')\n");
		av_err2str(send_err);
	}
	if (send_err == AVERROR_EOF) {
		*data_present = 0;
	}
	if (recv_err || send_err) {
		av_packet_unref(&input_packet);
		return 0;
	}


	/**
	* If the decoder has not been flushed completely, we are not finished,
	* so that this function has to be called again.
	*/
	if (*finished && *data_present)
		*finished = 0;
	av_packet_unref(&input_packet);
	return 0;
}

/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
	AVCodecContext *output_codec_context,
	int frame_size)
{
	int error;

	/**
	* Allocate as many pointers as there are audio channels.
	* Each pointer will later point to the audio samples of the corresponding
	* channels (although it may be NULL for interleaved formats).
	*/
	if (!(*converted_input_samples = (uint8_t **)calloc(output_codec_context->channels,
		sizeof(**converted_input_samples)))) {
		fprintf(stderr, "Could not allocate converted input sample pointers\n");
		return AVERROR(ENOMEM);
	}

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

/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is specified
* by frame_size.
*/
static int convert_samples(const uint8_t **input_data,
	uint8_t **converted_data, const int frame_size,
	SwrContext *resample_context)
{
	int error;

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

	return 0;
}

/** Add converted input audio samples to the FIFO buffer for later processing. */
static int add_samples_to_fifo(AVAudioFifo *fifo,
	uint8_t **converted_input_samples,
	const int frame_size)
{
	int error;

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

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

/**
* Read one audio frame from the input file, decodes, converts and stores
* it in the FIFO buffer.
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
	AVFormatContext *input_format_context,
	AVCodecContext *input_codec_context,
	AVCodecContext *output_codec_context,
	SwrContext *resampler_context,
	int *finished)
{
	/** Temporary storage of the input samples of the frame read from the file. */
	AVFrame *input_frame = NULL;
	/** Temporary storage for the converted input samples. */
	uint8_t **converted_input_samples = NULL;
	int data_present;
	int ret = AVERROR_EXIT;

	/** Initialize temporary storage for one input frame. */
	if (init_input_frame(&input_frame))
		goto cleanup;
	/** Decode one frame worth of audio samples. */
	if (decode_audio_frame(input_frame, input_format_context,
		input_codec_context, &data_present, finished))
		goto cleanup;
	/**
	* If we are at the end of the file and there are no more samples
	* in the decoder which are delayed, we are actually finished.
	* This must not be treated as an error.
	*/
	if (*finished && !data_present) {
		ret = 0;
		goto cleanup;
	}
	/** If there is decoded data, convert and store it */
	if (data_present) {
		/** Initialize the temporary storage for the converted input samples. */
		if (init_converted_samples(&converted_input_samples, output_codec_context,
			input_frame->nb_samples))
			goto cleanup;

		/**
		* Convert the input samples to the desired output sample format.
		* This requires a temporary storage provided by converted_input_samples.
		*/
		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.
*/
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. */
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);

	int send_err, recv_err;

	/** 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",
	//            av_err2str(error));
	//    av_packet_unref(&output_packet);
	//    return error;
	//}

	if ((send_err = avcodec_send_frame(output_codec_context, frame)) < 0) {
		fprintf(stderr, "avcodec_send_frame failed (error '%s')\n",
			av_err2str(send_err));
	}
	if ((recv_err = avcodec_receive_packet(output_codec_context, &output_packet)) < 0) {
		fprintf(stderr, "avcodec_receive_packet failed (error '%s')\n",
			av_err2str(recv_err));
	}
	if (send_err || recv_err) {
		if (send_err == AVERROR_EOF || recv_err == AVERROR_EOF) {
			*data_present = 0;
		}
		av_packet_unref(&output_packet);
		return 0;
	}
	/** 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",
				av_err2str(error));
			av_packet_unref(&output_packet);
			return error;
		}

		av_packet_unref(&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",
			av_err2str(error));
		return error;
	}
	return 0;
}

/** Convert an audio file to an AAC file in an MP4 container. */
int transcode_audio(const char *inAudio, const char *outAudio)
{
	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(inAudio, &input_format_context,
		&input_codec_context))
		goto cleanup;
	/** Open the output file for writing. */
	if (open_output_file(outAudio, 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_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;
}