2 * Copyright (c) 2013-2022 Andreas Unterweger
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 * @file audio transcoding to MPEG/AAC API usage example
23 * @example transcode_aac.c
25 * Convert an input audio file to AAC in an MP4 container. Formats other than
26 * MP4 are supported based on the output file extension.
27 * @author Andreas Unterweger (dustsigns@gmail.com)
32 #include "libavformat/avformat.h"
33 #include "libavformat/avio.h"
35 #include "libavcodec/avcodec.h"
37 #include "libavutil/audio_fifo.h"
38 #include "libavutil/avassert.h"
39 #include "libavutil/avstring.h"
40 #include "libavutil/channel_layout.h"
41 #include "libavutil/frame.h"
42 #include "libavutil/opt.h"
44 #include "libswresample/swresample.h"
46 /* The output bit rate in bit/s */
47 #define OUTPUT_BIT_RATE 96000
48 /* The number of output channels */
49 #define OUTPUT_CHANNELS 2
52 * Open an input file and the required decoder.
53 * @param filename File to be opened
54 * @param[out] input_format_context Format context of opened file
55 * @param[out] input_codec_context Codec context of opened file
56 * @return Error code (0 if successful)
58 static int open_input_file(const char *filename,
59 AVFormatContext **input_format_context,
60 AVCodecContext **input_codec_context)
62 AVCodecContext *avctx;
63 const AVCodec *input_codec;
64 const AVStream *stream;
67 /* Open the input file to read from it. */
68 if ((error = avformat_open_input(input_format_context, filename, NULL,
70 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
71 filename, av_err2str(error));
72 *input_format_context = NULL;
76 /* Get information on the input file (number of streams etc.). */
77 if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
78 fprintf(stderr, "Could not open find stream info (error '%s')\n",
80 avformat_close_input(input_format_context);
84 /* Make sure that there is only one stream in the input file. */
85 if ((*input_format_context)->nb_streams != 1) {
86 fprintf(stderr, "Expected one audio input stream, but found %d\n",
87 (*input_format_context)->nb_streams);
88 avformat_close_input(input_format_context);
92 stream = (*input_format_context)->streams[0];
94 /* Find a decoder for the audio stream. */
95 if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
96 fprintf(stderr, "Could not find input codec\n");
97 avformat_close_input(input_format_context);
101 /* Allocate a new decoding context. */
102 avctx = avcodec_alloc_context3(input_codec);
104 fprintf(stderr, "Could not allocate a decoding context\n");
105 avformat_close_input(input_format_context);
106 return AVERROR(ENOMEM);
109 /* Initialize the stream parameters with demuxer information. */
110 error = avcodec_parameters_to_context(avctx, stream->codecpar);
112 avformat_close_input(input_format_context);
113 avcodec_free_context(&avctx);
117 /* Open the decoder for the audio stream to use it later. */
118 if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
119 fprintf(stderr, "Could not open input codec (error '%s')\n",
121 avcodec_free_context(&avctx);
122 avformat_close_input(input_format_context);
126 /* Set the packet timebase for the decoder. */
127 avctx->pkt_timebase = stream->time_base;
129 /* Save the decoder context for easier access later. */
130 *input_codec_context = avctx;
136 * Open an output file and the required encoder.
137 * Also set some basic encoder parameters.
138 * Some of these parameters are based on the input file's parameters.
139 * @param filename File to be opened
140 * @param input_codec_context Codec context of input file
141 * @param[out] output_format_context Format context of output file
142 * @param[out] output_codec_context Codec context of output file
143 * @return Error code (0 if successful)
145 static int open_output_file(const char *filename,
146 AVCodecContext *input_codec_context,
147 AVFormatContext **output_format_context,
148 AVCodecContext **output_codec_context)
150 AVCodecContext *avctx = NULL;
151 AVIOContext *output_io_context = NULL;
152 AVStream *stream = NULL;
153 const AVCodec *output_codec = NULL;
156 /* Open the output file to write to it. */
157 if ((error = avio_open(&output_io_context, filename,
158 AVIO_FLAG_WRITE)) < 0) {
159 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
160 filename, av_err2str(error));
164 /* Create a new format context for the output container format. */
165 if (!(*output_format_context = avformat_alloc_context())) {
166 fprintf(stderr, "Could not allocate output format context\n");
167 return AVERROR(ENOMEM);
170 /* Associate the output file (pointer) with the container format context. */
171 (*output_format_context)->pb = output_io_context;
173 /* Guess the desired container format based on the file extension. */
174 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
176 fprintf(stderr, "Could not find output file format\n");
180 if (!((*output_format_context)->url = av_strdup(filename))) {
181 fprintf(stderr, "Could not allocate url.\n");
182 error = AVERROR(ENOMEM);
186 /* Find the encoder to be used by its name. */
187 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
188 fprintf(stderr, "Could not find an AAC encoder.\n");
192 /* Create a new audio stream in the output file container. */
193 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
194 fprintf(stderr, "Could not create new stream\n");
195 error = AVERROR(ENOMEM);
199 avctx = avcodec_alloc_context3(output_codec);
201 fprintf(stderr, "Could not allocate an encoding context\n");
202 error = AVERROR(ENOMEM);
206 /* Set the basic encoder parameters.
207 * The input file's sample rate is used to avoid a sample rate conversion. */
208 av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
209 avctx->sample_rate = input_codec_context->sample_rate;
210 avctx->sample_fmt = output_codec->sample_fmts[0];
211 avctx->bit_rate = OUTPUT_BIT_RATE;
213 /* Set the sample rate for the container. */
214 stream->time_base.den = input_codec_context->sample_rate;
215 stream->time_base.num = 1;
217 /* Some container formats (like MP4) require global headers to be present.
218 * Mark the encoder so that it behaves accordingly. */
219 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
220 avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
222 /* Open the encoder for the audio stream to use it later. */
223 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
224 fprintf(stderr, "Could not open output codec (error '%s')\n",
229 error = avcodec_parameters_from_context(stream->codecpar, avctx);
231 fprintf(stderr, "Could not initialize stream parameters\n");
235 /* Save the encoder context for easier access later. */
236 *output_codec_context = avctx;
241 avcodec_free_context(&avctx);
242 avio_closep(&(*output_format_context)->pb);
243 avformat_free_context(*output_format_context);
244 *output_format_context = NULL;
245 return error < 0 ? error : AVERROR_EXIT;
249 * Initialize one data packet for reading or writing.
250 * @param[out] packet Packet to be initialized
251 * @return Error code (0 if successful)
253 static int init_packet(AVPacket **packet)
255 if (!(*packet = av_packet_alloc())) {
256 fprintf(stderr, "Could not allocate packet\n");
257 return AVERROR(ENOMEM);
263 * Initialize one audio frame for reading from the input file.
264 * @param[out] frame Frame to be initialized
265 * @return Error code (0 if successful)
267 static int init_input_frame(AVFrame **frame)
269 if (!(*frame = av_frame_alloc())) {
270 fprintf(stderr, "Could not allocate input frame\n");
271 return AVERROR(ENOMEM);
277 * Initialize the audio resampler based on the input and output codec settings.
278 * If the input and output sample formats differ, a conversion is required
279 * libswresample takes care of this, but requires initialization.
280 * @param input_codec_context Codec context of the input file
281 * @param output_codec_context Codec context of the output file
282 * @param[out] resample_context Resample context for the required conversion
283 * @return Error code (0 if successful)
285 static int init_resampler(AVCodecContext *input_codec_context,
286 AVCodecContext *output_codec_context,
287 SwrContext **resample_context)
292 * Create a resampler context for the conversion.
293 * Set the conversion parameters.
295 error = swr_alloc_set_opts2(resample_context,
296 &output_codec_context->ch_layout,
297 output_codec_context->sample_fmt,
298 output_codec_context->sample_rate,
299 &input_codec_context->ch_layout,
300 input_codec_context->sample_fmt,
301 input_codec_context->sample_rate,
304 fprintf(stderr, "Could not allocate resample context\n");
308 * Perform a sanity check so that the number of converted samples is
309 * not greater than the number of samples to be converted.
310 * If the sample rates differ, this case has to be handled differently
312 av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
314 /* Open the resampler with the specified parameters. */
315 if ((error = swr_init(*resample_context)) < 0) {
316 fprintf(stderr, "Could not open resample context\n");
317 swr_free(resample_context);
324 * Initialize a FIFO buffer for the audio samples to be encoded.
325 * @param[out] fifo Sample buffer
326 * @param output_codec_context Codec context of the output file
327 * @return Error code (0 if successful)
329 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
331 /* Create the FIFO buffer based on the specified output sample format. */
332 if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
333 output_codec_context->ch_layout.nb_channels, 1))) {
334 fprintf(stderr, "Could not allocate FIFO\n");
335 return AVERROR(ENOMEM);
341 * Write the header of the output file container.
342 * @param output_format_context Format context of the output file
343 * @return Error code (0 if successful)
345 static int write_output_file_header(AVFormatContext *output_format_context)
348 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
349 fprintf(stderr, "Could not write output file header (error '%s')\n",
357 * Decode one audio frame from the input file.
358 * @param frame Audio frame to be decoded
359 * @param input_format_context Format context of the input file
360 * @param input_codec_context Codec context of the input file
361 * @param[out] data_present Indicates whether data has been decoded
362 * @param[out] finished Indicates whether the end of file has
363 * been reached and all data has been
364 * decoded. If this flag is false, there
365 * is more data to be decoded, i.e., this
366 * function has to be called again.
367 * @return Error code (0 if successful)
369 static int decode_audio_frame(AVFrame *frame,
370 AVFormatContext *input_format_context,
371 AVCodecContext *input_codec_context,
372 int *data_present, int *finished)
374 /* Packet used for temporary storage. */
375 AVPacket *input_packet;
378 error = init_packet(&input_packet);
384 /* Read one audio frame from the input file into a temporary packet. */
385 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
386 /* If we are at the end of the file, flush the decoder below. */
387 if (error == AVERROR_EOF)
390 fprintf(stderr, "Could not read frame (error '%s')\n",
396 /* Send the audio frame stored in the temporary packet to the decoder.
397 * The input audio stream decoder is used to do this. */
398 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
399 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
404 /* Receive one frame from the decoder. */
405 error = avcodec_receive_frame(input_codec_context, frame);
406 /* If the decoder asks for more data to be able to decode a frame,
407 * return indicating that no data is present. */
408 if (error == AVERROR(EAGAIN)) {
411 /* If the end of the input file is reached, stop decoding. */
412 } else if (error == AVERROR_EOF) {
416 } else if (error < 0) {
417 fprintf(stderr, "Could not decode frame (error '%s')\n",
420 /* Default case: Return decoded data. */
427 av_packet_free(&input_packet);
432 * Initialize a temporary storage for the specified number of audio samples.
433 * The conversion requires temporary storage due to the different format.
434 * The number of audio samples to be allocated is specified in frame_size.
435 * @param[out] converted_input_samples Array of converted samples. The
436 * dimensions are reference, channel
437 * (for multi-channel audio), sample.
438 * @param output_codec_context Codec context of the output file
439 * @param frame_size Number of samples to be converted in
441 * @return Error code (0 if successful)
443 static int init_converted_samples(uint8_t ***converted_input_samples,
444 AVCodecContext *output_codec_context,
449 /* Allocate as many pointers as there are audio channels.
450 * Each pointer will point to the audio samples of the corresponding
451 * channels (although it may be NULL for interleaved formats).
452 * Allocate memory for the samples of all channels in one consecutive
453 * block for convenience. */
454 if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
455 output_codec_context->ch_layout.nb_channels,
457 output_codec_context->sample_fmt, 0)) < 0) {
459 "Could not allocate converted input samples (error '%s')\n",
467 * Convert the input audio samples into the output sample format.
468 * The conversion happens on a per-frame basis, the size of which is
469 * specified by frame_size.
470 * @param input_data Samples to be decoded. The dimensions are
471 * channel (for multi-channel audio), sample.
472 * @param[out] converted_data Converted samples. The dimensions are channel
473 * (for multi-channel audio), sample.
474 * @param frame_size Number of samples to be converted
475 * @param resample_context Resample context for the conversion
476 * @return Error code (0 if successful)
478 static int convert_samples(const uint8_t **input_data,
479 uint8_t **converted_data, const int frame_size,
480 SwrContext *resample_context)
484 /* Convert the samples using the resampler. */
485 if ((error = swr_convert(resample_context,
486 converted_data, frame_size,
487 input_data , frame_size)) < 0) {
488 fprintf(stderr, "Could not convert input samples (error '%s')\n",
497 * Add converted input audio samples to the FIFO buffer for later processing.
498 * @param fifo Buffer to add the samples to
499 * @param converted_input_samples Samples to be added. The dimensions are channel
500 * (for multi-channel audio), sample.
501 * @param frame_size Number of samples to be converted
502 * @return Error code (0 if successful)
504 static int add_samples_to_fifo(AVAudioFifo *fifo,
505 uint8_t **converted_input_samples,
506 const int frame_size)
510 /* Make the FIFO as large as it needs to be to hold both,
511 * the old and the new samples. */
512 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
513 fprintf(stderr, "Could not reallocate FIFO\n");
517 /* Store the new samples in the FIFO buffer. */
518 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
519 frame_size) < frame_size) {
520 fprintf(stderr, "Could not write data to FIFO\n");
527 * Read one audio frame from the input file, decode, convert and store
528 * it in the FIFO buffer.
529 * @param fifo Buffer used for temporary storage
530 * @param input_format_context Format context of the input file
531 * @param input_codec_context Codec context of the input file
532 * @param output_codec_context Codec context of the output file
533 * @param resampler_context Resample context for the conversion
534 * @param[out] finished Indicates whether the end of file has
535 * been reached and all data has been
536 * decoded. If this flag is false,
537 * there is more data to be decoded,
538 * i.e., this function has to be called
540 * @return Error code (0 if successful)
542 static int read_decode_convert_and_store(AVAudioFifo *fifo,
543 AVFormatContext *input_format_context,
544 AVCodecContext *input_codec_context,
545 AVCodecContext *output_codec_context,
546 SwrContext *resampler_context,
549 /* Temporary storage of the input samples of the frame read from the file. */
550 AVFrame *input_frame = NULL;
551 /* Temporary storage for the converted input samples. */
552 uint8_t **converted_input_samples = NULL;
554 int ret = AVERROR_EXIT;
556 /* Initialize temporary storage for one input frame. */
557 if (init_input_frame(&input_frame))
559 /* Decode one frame worth of audio samples. */
560 if (decode_audio_frame(input_frame, input_format_context,
561 input_codec_context, &data_present, finished))
563 /* If we are at the end of the file and there are no more samples
564 * in the decoder which are delayed, we are actually finished.
565 * This must not be treated as an error. */
570 /* If there is decoded data, convert and store it. */
572 /* Initialize the temporary storage for the converted input samples. */
573 if (init_converted_samples(&converted_input_samples, output_codec_context,
574 input_frame->nb_samples))
577 /* Convert the input samples to the desired output sample format.
578 * This requires a temporary storage provided by converted_input_samples. */
579 if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
580 input_frame->nb_samples, resampler_context))
583 /* Add the converted input samples to the FIFO buffer for later processing. */
584 if (add_samples_to_fifo(fifo, converted_input_samples,
585 input_frame->nb_samples))
592 if (converted_input_samples)
593 av_freep(&converted_input_samples[0]);
594 av_freep(&converted_input_samples);
595 av_frame_free(&input_frame);
601 * Initialize one input frame for writing to the output file.
602 * The frame will be exactly frame_size samples large.
603 * @param[out] frame Frame to be initialized
604 * @param output_codec_context Codec context of the output file
605 * @param frame_size Size of the frame
606 * @return Error code (0 if successful)
608 static int init_output_frame(AVFrame **frame,
609 AVCodecContext *output_codec_context,
614 /* Create a new frame to store the audio samples. */
615 if (!(*frame = av_frame_alloc())) {
616 fprintf(stderr, "Could not allocate output frame\n");
620 /* Set the frame's parameters, especially its size and format.
621 * av_frame_get_buffer needs this to allocate memory for the
622 * audio samples of the frame.
623 * Default channel layouts based on the number of channels
624 * are assumed for simplicity. */
625 (*frame)->nb_samples = frame_size;
626 av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
627 (*frame)->format = output_codec_context->sample_fmt;
628 (*frame)->sample_rate = output_codec_context->sample_rate;
630 /* Allocate the samples of the created frame. This call will make
631 * sure that the audio frame can hold as many samples as specified. */
632 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
633 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
635 av_frame_free(frame);
642 /* Global timestamp for the audio frames. */
643 static int64_t pts = 0;
646 * Encode one frame worth of audio to the output file.
647 * @param frame Samples to be encoded
648 * @param output_format_context Format context of the output file
649 * @param output_codec_context Codec context of the output file
650 * @param[out] data_present Indicates whether data has been
652 * @return Error code (0 if successful)
654 static int encode_audio_frame(AVFrame *frame,
655 AVFormatContext *output_format_context,
656 AVCodecContext *output_codec_context,
659 /* Packet used for temporary storage. */
660 AVPacket *output_packet;
663 error = init_packet(&output_packet);
667 /* Set a timestamp based on the sample rate for the container. */
670 pts += frame->nb_samples;
674 /* Send the audio frame stored in the temporary packet to the encoder.
675 * The output audio stream encoder is used to do this. */
676 error = avcodec_send_frame(output_codec_context, frame);
677 /* Check for errors, but proceed with fetching encoded samples if the
678 * encoder signals that it has nothing more to encode. */
679 if (error < 0 && error != AVERROR_EOF) {
680 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
685 /* Receive one encoded frame from the encoder. */
686 error = avcodec_receive_packet(output_codec_context, output_packet);
687 /* If the encoder asks for more data to be able to provide an
688 * encoded frame, return indicating that no data is present. */
689 if (error == AVERROR(EAGAIN)) {
692 /* If the last frame has been encoded, stop encoding. */
693 } else if (error == AVERROR_EOF) {
696 } else if (error < 0) {
697 fprintf(stderr, "Could not encode frame (error '%s')\n",
700 /* Default case: Return encoded data. */
705 /* Write one audio frame from the temporary packet to the output file. */
707 (error = av_write_frame(output_format_context, output_packet)) < 0) {
708 fprintf(stderr, "Could not write frame (error '%s')\n",
714 av_packet_free(&output_packet);
719 * Load one audio frame from the FIFO buffer, encode and write it to the
721 * @param fifo Buffer used for temporary storage
722 * @param output_format_context Format context of the output file
723 * @param output_codec_context Codec context of the output file
724 * @return Error code (0 if successful)
726 static int load_encode_and_write(AVAudioFifo *fifo,
727 AVFormatContext *output_format_context,
728 AVCodecContext *output_codec_context)
730 /* Temporary storage of the output samples of the frame written to the file. */
731 AVFrame *output_frame;
732 /* Use the maximum number of possible samples per frame.
733 * If there is less than the maximum possible frame size in the FIFO
734 * buffer use this number. Otherwise, use the maximum possible frame size. */
735 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
736 output_codec_context->frame_size);
739 /* Initialize temporary storage for one output frame. */
740 if (init_output_frame(&output_frame, output_codec_context, frame_size))
743 /* Read as many samples from the FIFO buffer as required to fill the frame.
744 * The samples are stored in the frame temporarily. */
745 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
746 fprintf(stderr, "Could not read data from FIFO\n");
747 av_frame_free(&output_frame);
751 /* Encode one frame worth of audio samples. */
752 if (encode_audio_frame(output_frame, output_format_context,
753 output_codec_context, &data_written)) {
754 av_frame_free(&output_frame);
757 av_frame_free(&output_frame);
762 * Write the trailer of the output file container.
763 * @param output_format_context Format context of the output file
764 * @return Error code (0 if successful)
766 static int write_output_file_trailer(AVFormatContext *output_format_context)
769 if ((error = av_write_trailer(output_format_context)) < 0) {
770 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
777 int main(int argc, char **argv)
779 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
780 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
781 SwrContext *resample_context = NULL;
782 AVAudioFifo *fifo = NULL;
783 int ret = AVERROR_EXIT;
786 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
790 /* Open the input file for reading. */
791 if (open_input_file(argv[1], &input_format_context,
792 &input_codec_context))
794 /* Open the output file for writing. */
795 if (open_output_file(argv[2], input_codec_context,
796 &output_format_context, &output_codec_context))
798 /* Initialize the resampler to be able to convert audio sample formats. */
799 if (init_resampler(input_codec_context, output_codec_context,
802 /* Initialize the FIFO buffer to store audio samples to be encoded. */
803 if (init_fifo(&fifo, output_codec_context))
805 /* Write the header of the output file container. */
806 if (write_output_file_header(output_format_context))
809 /* Loop as long as we have input samples to read or output samples
810 * to write; abort as soon as we have neither. */
812 /* Use the encoder's desired frame size for processing. */
813 const int output_frame_size = output_codec_context->frame_size;
816 /* Make sure that there is one frame worth of samples in the FIFO
817 * buffer so that the encoder can do its work.
818 * Since the decoder's and the encoder's frame size may differ, we
819 * need to FIFO buffer to store as many frames worth of input samples
820 * that they make up at least one frame worth of output samples. */
821 while (av_audio_fifo_size(fifo) < output_frame_size) {
822 /* Decode one frame worth of audio samples, convert it to the
823 * output sample format and put it into the FIFO buffer. */
824 if (read_decode_convert_and_store(fifo, input_format_context,
826 output_codec_context,
827 resample_context, &finished))
830 /* If we are at the end of the input file, we continue
831 * encoding the remaining audio samples to the output file. */
836 /* If we have enough samples for the encoder, we encode them.
837 * At the end of the file, we pass the remaining samples to
839 while (av_audio_fifo_size(fifo) >= output_frame_size ||
840 (finished && av_audio_fifo_size(fifo) > 0))
841 /* Take one frame worth of audio samples from the FIFO buffer,
842 * encode it and write it to the output file. */
843 if (load_encode_and_write(fifo, output_format_context,
844 output_codec_context))
847 /* If we are at the end of the input file and have encoded
848 * all remaining samples, we can exit this loop and finish. */
851 /* Flush the encoder as it may have delayed frames. */
853 if (encode_audio_frame(NULL, output_format_context,
854 output_codec_context, &data_written))
856 } while (data_written);
861 /* Write the trailer of the output file container. */
862 if (write_output_file_trailer(output_format_context))
868 av_audio_fifo_free(fifo);
869 swr_free(&resample_context);
870 if (output_codec_context)
871 avcodec_free_context(&output_codec_context);
872 if (output_format_context) {
873 avio_closep(&output_format_context->pb);
874 avformat_free_context(output_format_context);
876 if (input_codec_context)
877 avcodec_free_context(&input_codec_context);
878 if (input_format_context)
879 avformat_close_input(&input_format_context);