2 * Copyright (c) 2013-2017 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
23 * Simple audio converter
25 * @example transcode_aac.c
26 * Convert an input audio file to AAC in an MP4 container using FFmpeg.
27 * Formats other than MP4 are supported based on the output file extension.
28 * @author Andreas Unterweger (dustsigns@gmail.com)
33 #include "libavformat/avformat.h"
34 #include "libavformat/avio.h"
36 #include "libavcodec/avcodec.h"
38 #include "libavutil/audio_fifo.h"
39 #include "libavutil/avassert.h"
40 #include "libavutil/avstring.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;
66 /* Open the input file to read from it. */
67 if ((error = avformat_open_input(input_format_context, filename, NULL,
69 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
70 filename, av_err2str(error));
71 *input_format_context = NULL;
75 /* Get information on the input file (number of streams etc.). */
76 if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
77 fprintf(stderr, "Could not open find stream info (error '%s')\n",
79 avformat_close_input(input_format_context);
83 /* Make sure that there is only one stream in the input file. */
84 if ((*input_format_context)->nb_streams != 1) {
85 fprintf(stderr, "Expected one audio input stream, but found %d\n",
86 (*input_format_context)->nb_streams);
87 avformat_close_input(input_format_context);
91 /* Find a decoder for the audio stream. */
92 if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
93 fprintf(stderr, "Could not find input codec\n");
94 avformat_close_input(input_format_context);
98 /* Allocate a new decoding context. */
99 avctx = avcodec_alloc_context3(input_codec);
101 fprintf(stderr, "Could not allocate a decoding context\n");
102 avformat_close_input(input_format_context);
103 return AVERROR(ENOMEM);
106 /* Initialize the stream parameters with demuxer information. */
107 error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
109 avformat_close_input(input_format_context);
110 avcodec_free_context(&avctx);
114 /* Open the decoder for the audio stream to use it later. */
115 if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
116 fprintf(stderr, "Could not open input codec (error '%s')\n",
118 avcodec_free_context(&avctx);
119 avformat_close_input(input_format_context);
123 /* Save the decoder context for easier access later. */
124 *input_codec_context = avctx;
130 * Open an output file and the required encoder.
131 * Also set some basic encoder parameters.
132 * Some of these parameters are based on the input file's parameters.
133 * @param filename File to be opened
134 * @param input_codec_context Codec context of input file
135 * @param[out] output_format_context Format context of output file
136 * @param[out] output_codec_context Codec context of output file
137 * @return Error code (0 if successful)
139 static int open_output_file(const char *filename,
140 AVCodecContext *input_codec_context,
141 AVFormatContext **output_format_context,
142 AVCodecContext **output_codec_context)
144 AVCodecContext *avctx = NULL;
145 AVIOContext *output_io_context = NULL;
146 AVStream *stream = NULL;
147 AVCodec *output_codec = NULL;
150 /* Open the output file to write to it. */
151 if ((error = avio_open(&output_io_context, filename,
152 AVIO_FLAG_WRITE)) < 0) {
153 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
154 filename, av_err2str(error));
158 /* Create a new format context for the output container format. */
159 if (!(*output_format_context = avformat_alloc_context())) {
160 fprintf(stderr, "Could not allocate output format context\n");
161 return AVERROR(ENOMEM);
164 /* Associate the output file (pointer) with the container format context. */
165 (*output_format_context)->pb = output_io_context;
167 /* Guess the desired container format based on the file extension. */
168 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
170 fprintf(stderr, "Could not find output file format\n");
174 av_strlcpy((*output_format_context)->filename, filename,
175 sizeof((*output_format_context)->filename));
177 /* Find the encoder to be used by its name. */
178 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
179 fprintf(stderr, "Could not find an AAC encoder.\n");
183 /* Create a new audio stream in the output file container. */
184 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
185 fprintf(stderr, "Could not create new stream\n");
186 error = AVERROR(ENOMEM);
190 avctx = avcodec_alloc_context3(output_codec);
192 fprintf(stderr, "Could not allocate an encoding context\n");
193 error = AVERROR(ENOMEM);
197 /* Set the basic encoder parameters.
198 * The input file's sample rate is used to avoid a sample rate conversion. */
199 avctx->channels = OUTPUT_CHANNELS;
200 avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
201 avctx->sample_rate = input_codec_context->sample_rate;
202 avctx->sample_fmt = output_codec->sample_fmts[0];
203 avctx->bit_rate = OUTPUT_BIT_RATE;
205 /* Allow the use of the experimental AAC encoder. */
206 avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
208 /* Set the sample rate for the container. */
209 stream->time_base.den = input_codec_context->sample_rate;
210 stream->time_base.num = 1;
212 /* Some container formats (like MP4) require global headers to be present.
213 * Mark the encoder so that it behaves accordingly. */
214 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
215 avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
217 /* Open the encoder for the audio stream to use it later. */
218 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
219 fprintf(stderr, "Could not open output codec (error '%s')\n",
224 error = avcodec_parameters_from_context(stream->codecpar, avctx);
226 fprintf(stderr, "Could not initialize stream parameters\n");
230 /* Save the encoder context for easier access later. */
231 *output_codec_context = avctx;
236 avcodec_free_context(&avctx);
237 avio_closep(&(*output_format_context)->pb);
238 avformat_free_context(*output_format_context);
239 *output_format_context = NULL;
240 return error < 0 ? error : AVERROR_EXIT;
244 * Initialize one data packet for reading or writing.
245 * @param packet Packet to be initialized
247 static void init_packet(AVPacket *packet)
249 av_init_packet(packet);
250 /* Set the packet data and size so that it is recognized as being empty. */
256 * Initialize one audio frame for reading from the input file.
257 * @param[out] frame Frame to be initialized
258 * @return Error code (0 if successful)
260 static int init_input_frame(AVFrame **frame)
262 if (!(*frame = av_frame_alloc())) {
263 fprintf(stderr, "Could not allocate input frame\n");
264 return AVERROR(ENOMEM);
270 * Initialize the audio resampler based on the input and output codec settings.
271 * If the input and output sample formats differ, a conversion is required
272 * libswresample takes care of this, but requires initialization.
273 * @param input_codec_context Codec context of the input file
274 * @param output_codec_context Codec context of the output file
275 * @param[out] resample_context Resample context for the required conversion
276 * @return Error code (0 if successful)
278 static int init_resampler(AVCodecContext *input_codec_context,
279 AVCodecContext *output_codec_context,
280 SwrContext **resample_context)
285 * Create a resampler context for the conversion.
286 * Set the conversion parameters.
287 * Default channel layouts based on the number of channels
288 * are assumed for simplicity (they are sometimes not detected
289 * properly by the demuxer and/or decoder).
291 *resample_context = swr_alloc_set_opts(NULL,
292 av_get_default_channel_layout(output_codec_context->channels),
293 output_codec_context->sample_fmt,
294 output_codec_context->sample_rate,
295 av_get_default_channel_layout(input_codec_context->channels),
296 input_codec_context->sample_fmt,
297 input_codec_context->sample_rate,
299 if (!*resample_context) {
300 fprintf(stderr, "Could not allocate resample context\n");
301 return AVERROR(ENOMEM);
304 * Perform a sanity check so that the number of converted samples is
305 * not greater than the number of samples to be converted.
306 * If the sample rates differ, this case has to be handled differently
308 av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
310 /* Open the resampler with the specified parameters. */
311 if ((error = swr_init(*resample_context)) < 0) {
312 fprintf(stderr, "Could not open resample context\n");
313 swr_free(resample_context);
320 * Initialize a FIFO buffer for the audio samples to be encoded.
321 * @param[out] fifo Sample buffer
322 * @param output_codec_context Codec context of the output file
323 * @return Error code (0 if successful)
325 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
327 /* Create the FIFO buffer based on the specified output sample format. */
328 if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
329 output_codec_context->channels, 1))) {
330 fprintf(stderr, "Could not allocate FIFO\n");
331 return AVERROR(ENOMEM);
337 * Write the header of the output file container.
338 * @param output_format_context Format context of the output file
339 * @return Error code (0 if successful)
341 static int write_output_file_header(AVFormatContext *output_format_context)
344 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
345 fprintf(stderr, "Could not write output file header (error '%s')\n",
353 * Decode one audio frame from the input file.
354 * @param frame Audio frame to be decoded
355 * @param input_format_context Format context of the input file
356 * @param input_codec_context Codec context of the input file
357 * @param[out] data_present Indicates whether data has been decoded
358 * @param[out] finished Indicates whether the end of file has
359 * been reached and all data has been
360 * decoded. If this flag is false, there
361 * is more data to be decoded, i.e., this
362 * function has to be called again.
363 * @return Error code (0 if successful)
365 static int decode_audio_frame(AVFrame *frame,
366 AVFormatContext *input_format_context,
367 AVCodecContext *input_codec_context,
368 int *data_present, int *finished)
370 /* Packet used for temporary storage. */
371 AVPacket input_packet;
373 init_packet(&input_packet);
375 /* Read one audio frame from the input file into a temporary packet. */
376 if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
377 /* If we are at the end of the file, flush the decoder below. */
378 if (error == AVERROR_EOF)
381 fprintf(stderr, "Could not read frame (error '%s')\n",
387 /* Decode the audio frame stored in the temporary packet.
388 * The input audio stream decoder is used to do this.
389 * If we are at the end of the file, pass an empty packet to the decoder
391 if ((error = avcodec_decode_audio4(input_codec_context, frame,
392 data_present, &input_packet)) < 0) {
393 fprintf(stderr, "Could not decode frame (error '%s')\n",
395 av_packet_unref(&input_packet);
399 /* If the decoder has not been flushed completely, we are not finished,
400 * so that this function has to be called again. */
401 if (*finished && *data_present)
403 av_packet_unref(&input_packet);
408 * Initialize a temporary storage for the specified number of audio samples.
409 * The conversion requires temporary storage due to the different format.
410 * The number of audio samples to be allocated is specified in frame_size.
411 * @param[out] converted_input_samples Array of converted samples. The
412 * dimensions are reference, channel
413 * (for multi-channel audio), sample.
414 * @param output_codec_context Codec context of the output file
415 * @param frame_size Number of samples to be converted in
417 * @return Error code (0 if successful)
419 static int init_converted_samples(uint8_t ***converted_input_samples,
420 AVCodecContext *output_codec_context,
425 /* Allocate as many pointers as there are audio channels.
426 * Each pointer will later point to the audio samples of the corresponding
427 * channels (although it may be NULL for interleaved formats).
429 if (!(*converted_input_samples = calloc(output_codec_context->channels,
430 sizeof(**converted_input_samples)))) {
431 fprintf(stderr, "Could not allocate converted input sample pointers\n");
432 return AVERROR(ENOMEM);
435 /* Allocate memory for the samples of all channels in one consecutive
436 * block for convenience. */
437 if ((error = av_samples_alloc(*converted_input_samples, NULL,
438 output_codec_context->channels,
440 output_codec_context->sample_fmt, 0)) < 0) {
442 "Could not allocate converted input samples (error '%s')\n",
444 av_freep(&(*converted_input_samples)[0]);
445 free(*converted_input_samples);
452 * Convert the input audio samples into the output sample format.
453 * The conversion happens on a per-frame basis, the size of which is
454 * specified by frame_size.
455 * @param input_data Samples to be decoded. The dimensions are
456 * channel (for multi-channel audio), sample.
457 * @param[out] converted_data Converted samples. The dimensions are channel
458 * (for multi-channel audio), sample.
459 * @param frame_size Number of samples to be converted
460 * @param resample_context Resample context for the conversion
461 * @return Error code (0 if successful)
463 static int convert_samples(const uint8_t **input_data,
464 uint8_t **converted_data, const int frame_size,
465 SwrContext *resample_context)
469 /* Convert the samples using the resampler. */
470 if ((error = swr_convert(resample_context,
471 converted_data, frame_size,
472 input_data , frame_size)) < 0) {
473 fprintf(stderr, "Could not convert input samples (error '%s')\n",
482 * Add converted input audio samples to the FIFO buffer for later processing.
483 * @param fifo Buffer to add the samples to
484 * @param converted_input_samples Samples to be added. The dimensions are channel
485 * (for multi-channel audio), sample.
486 * @param frame_size Number of samples to be converted
487 * @return Error code (0 if successful)
489 static int add_samples_to_fifo(AVAudioFifo *fifo,
490 uint8_t **converted_input_samples,
491 const int frame_size)
495 /* Make the FIFO as large as it needs to be to hold both,
496 * the old and the new samples. */
497 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
498 fprintf(stderr, "Could not reallocate FIFO\n");
502 /* Store the new samples in the FIFO buffer. */
503 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
504 frame_size) < frame_size) {
505 fprintf(stderr, "Could not write data to FIFO\n");
512 * Read one audio frame from the input file, decode, convert and store
513 * it in the FIFO buffer.
514 * @param fifo Buffer used for temporary storage
515 * @param input_format_context Format context of the input file
516 * @param input_codec_context Codec context of the input file
517 * @param output_codec_context Codec context of the output file
518 * @param resampler_context Resample context for the conversion
519 * @param[out] finished Indicates whether the end of file has
520 * been reached and all data has been
521 * decoded. If this flag is false,
522 * there is more data to be decoded,
523 * i.e., this function has to be called
525 * @return Error code (0 if successful)
527 static int read_decode_convert_and_store(AVAudioFifo *fifo,
528 AVFormatContext *input_format_context,
529 AVCodecContext *input_codec_context,
530 AVCodecContext *output_codec_context,
531 SwrContext *resampler_context,
534 /* Temporary storage of the input samples of the frame read from the file. */
535 AVFrame *input_frame = NULL;
536 /* Temporary storage for the converted input samples. */
537 uint8_t **converted_input_samples = NULL;
539 int ret = AVERROR_EXIT;
541 /* Initialize temporary storage for one input frame. */
542 if (init_input_frame(&input_frame))
544 /* Decode one frame worth of audio samples. */
545 if (decode_audio_frame(input_frame, input_format_context,
546 input_codec_context, &data_present, finished))
548 /* If we are at the end of the file and there are no more samples
549 * in the decoder which are delayed, we are actually finished.
550 * This must not be treated as an error. */
551 if (*finished && !data_present) {
555 /* If there is decoded data, convert and store it. */
557 /* Initialize the temporary storage for the converted input samples. */
558 if (init_converted_samples(&converted_input_samples, output_codec_context,
559 input_frame->nb_samples))
562 /* Convert the input samples to the desired output sample format.
563 * This requires a temporary storage provided by converted_input_samples. */
564 if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
565 input_frame->nb_samples, resampler_context))
568 /* Add the converted input samples to the FIFO buffer for later processing. */
569 if (add_samples_to_fifo(fifo, converted_input_samples,
570 input_frame->nb_samples))
577 if (converted_input_samples) {
578 av_freep(&converted_input_samples[0]);
579 free(converted_input_samples);
581 av_frame_free(&input_frame);
587 * Initialize one input frame for writing to the output file.
588 * The frame will be exactly frame_size samples large.
589 * @param[out] frame Frame to be initialized
590 * @param output_codec_context Codec context of the output file
591 * @param frame_size Size of the frame
592 * @return Error code (0 if successful)
594 static int init_output_frame(AVFrame **frame,
595 AVCodecContext *output_codec_context,
600 /* Create a new frame to store the audio samples. */
601 if (!(*frame = av_frame_alloc())) {
602 fprintf(stderr, "Could not allocate output frame\n");
606 /* Set the frame's parameters, especially its size and format.
607 * av_frame_get_buffer needs this to allocate memory for the
608 * audio samples of the frame.
609 * Default channel layouts based on the number of channels
610 * are assumed for simplicity. */
611 (*frame)->nb_samples = frame_size;
612 (*frame)->channel_layout = output_codec_context->channel_layout;
613 (*frame)->format = output_codec_context->sample_fmt;
614 (*frame)->sample_rate = output_codec_context->sample_rate;
616 /* Allocate the samples of the created frame. This call will make
617 * sure that the audio frame can hold as many samples as specified. */
618 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
619 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
621 av_frame_free(frame);
628 /* Global timestamp for the audio frames. */
629 static int64_t pts = 0;
632 * Encode one frame worth of audio to the output file.
633 * @param frame Samples to be encoded
634 * @param output_format_context Format context of the output file
635 * @param output_codec_context Codec context of the output file
636 * @param[out] data_present Indicates whether data has been
638 * @return Error code (0 if successful)
640 static int encode_audio_frame(AVFrame *frame,
641 AVFormatContext *output_format_context,
642 AVCodecContext *output_codec_context,
645 /* Packet used for temporary storage. */
646 AVPacket output_packet;
648 init_packet(&output_packet);
650 /* Set a timestamp based on the sample rate for the container. */
653 pts += frame->nb_samples;
656 /* Encode the audio frame and store it in the temporary packet.
657 * The output audio stream encoder is used to do this. */
658 if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
659 frame, data_present)) < 0) {
660 fprintf(stderr, "Could not encode frame (error '%s')\n",
662 av_packet_unref(&output_packet);
666 /* Write one audio frame from the temporary packet to the output file. */
668 if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
669 fprintf(stderr, "Could not write frame (error '%s')\n",
671 av_packet_unref(&output_packet);
675 av_packet_unref(&output_packet);
682 * Load one audio frame from the FIFO buffer, encode and write it to the
684 * @param fifo Buffer used for temporary storage
685 * @param output_format_context Format context of the output file
686 * @param output_codec_context Codec context of the output file
687 * @return Error code (0 if successful)
689 static int load_encode_and_write(AVAudioFifo *fifo,
690 AVFormatContext *output_format_context,
691 AVCodecContext *output_codec_context)
693 /* Temporary storage of the output samples of the frame written to the file. */
694 AVFrame *output_frame;
695 /* Use the maximum number of possible samples per frame.
696 * If there is less than the maximum possible frame size in the FIFO
697 * buffer use this number. Otherwise, use the maximum possible frame size. */
698 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
699 output_codec_context->frame_size);
702 /* Initialize temporary storage for one output frame. */
703 if (init_output_frame(&output_frame, output_codec_context, frame_size))
706 /* Read as many samples from the FIFO buffer as required to fill the frame.
707 * The samples are stored in the frame temporarily. */
708 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
709 fprintf(stderr, "Could not read data from FIFO\n");
710 av_frame_free(&output_frame);
714 /* Encode one frame worth of audio samples. */
715 if (encode_audio_frame(output_frame, output_format_context,
716 output_codec_context, &data_written)) {
717 av_frame_free(&output_frame);
720 av_frame_free(&output_frame);
725 * Write the trailer of the output file container.
726 * @param output_format_context Format context of the output file
727 * @return Error code (0 if successful)
729 static int write_output_file_trailer(AVFormatContext *output_format_context)
732 if ((error = av_write_trailer(output_format_context)) < 0) {
733 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
740 int main(int argc, char **argv)
742 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
743 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
744 SwrContext *resample_context = NULL;
745 AVAudioFifo *fifo = NULL;
746 int ret = AVERROR_EXIT;
749 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
753 /* Register all codecs and formats so that they can be used. */
755 /* Open the input file for reading. */
756 if (open_input_file(argv[1], &input_format_context,
757 &input_codec_context))
759 /* Open the output file for writing. */
760 if (open_output_file(argv[2], input_codec_context,
761 &output_format_context, &output_codec_context))
763 /* Initialize the resampler to be able to convert audio sample formats. */
764 if (init_resampler(input_codec_context, output_codec_context,
767 /* Initialize the FIFO buffer to store audio samples to be encoded. */
768 if (init_fifo(&fifo, output_codec_context))
770 /* Write the header of the output file container. */
771 if (write_output_file_header(output_format_context))
774 /* Loop as long as we have input samples to read or output samples
775 * to write; abort as soon as we have neither. */
777 /* Use the encoder's desired frame size for processing. */
778 const int output_frame_size = output_codec_context->frame_size;
781 /* Make sure that there is one frame worth of samples in the FIFO
782 * buffer so that the encoder can do its work.
783 * Since the decoder's and the encoder's frame size may differ, we
784 * need to FIFO buffer to store as many frames worth of input samples
785 * that they make up at least one frame worth of output samples. */
786 while (av_audio_fifo_size(fifo) < output_frame_size) {
787 /* Decode one frame worth of audio samples, convert it to the
788 * output sample format and put it into the FIFO buffer. */
789 if (read_decode_convert_and_store(fifo, input_format_context,
791 output_codec_context,
792 resample_context, &finished))
795 /* If we are at the end of the input file, we continue
796 * encoding the remaining audio samples to the output file. */
801 /* If we have enough samples for the encoder, we encode them.
802 * At the end of the file, we pass the remaining samples to
804 while (av_audio_fifo_size(fifo) >= output_frame_size ||
805 (finished && av_audio_fifo_size(fifo) > 0))
806 /* Take one frame worth of audio samples from the FIFO buffer,
807 * encode it and write it to the output file. */
808 if (load_encode_and_write(fifo, output_format_context,
809 output_codec_context))
812 /* If we are at the end of the input file and have encoded
813 * all remaining samples, we can exit this loop and finish. */
816 /* Flush the encoder as it may have delayed frames. */
818 if (encode_audio_frame(NULL, output_format_context,
819 output_codec_context, &data_written))
821 } while (data_written);
826 /* Write the trailer of the output file container. */
827 if (write_output_file_trailer(output_format_context))
833 av_audio_fifo_free(fifo);
834 swr_free(&resample_context);
835 if (output_codec_context)
836 avcodec_free_context(&output_codec_context);
837 if (output_format_context) {
838 avio_closep(&output_format_context->pb);
839 avformat_free_context(output_format_context);
841 if (input_codec_context)
842 avcodec_free_context(&input_codec_context);
843 if (input_format_context)
844 avformat_close_input(&input_format_context);