2 * Copyright (c) 2013-2018 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;
63 const AVCodec *input_codec;
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 const 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 if (!((*output_format_context)->url = av_strdup(filename))) {
175 fprintf(stderr, "Could not allocate url.\n");
176 error = AVERROR(ENOMEM);
180 /* Find the encoder to be used by its name. */
181 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
182 fprintf(stderr, "Could not find an AAC encoder.\n");
186 /* Create a new audio stream in the output file container. */
187 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
188 fprintf(stderr, "Could not create new stream\n");
189 error = AVERROR(ENOMEM);
193 avctx = avcodec_alloc_context3(output_codec);
195 fprintf(stderr, "Could not allocate an encoding context\n");
196 error = AVERROR(ENOMEM);
200 /* Set the basic encoder parameters.
201 * The input file's sample rate is used to avoid a sample rate conversion. */
202 avctx->channels = OUTPUT_CHANNELS;
203 avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
204 avctx->sample_rate = input_codec_context->sample_rate;
205 avctx->sample_fmt = output_codec->sample_fmts[0];
206 avctx->bit_rate = OUTPUT_BIT_RATE;
208 /* Allow the use of the experimental AAC encoder. */
209 avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
211 /* Set the sample rate for the container. */
212 stream->time_base.den = input_codec_context->sample_rate;
213 stream->time_base.num = 1;
215 /* Some container formats (like MP4) require global headers to be present.
216 * Mark the encoder so that it behaves accordingly. */
217 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
218 avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
220 /* Open the encoder for the audio stream to use it later. */
221 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
222 fprintf(stderr, "Could not open output codec (error '%s')\n",
227 error = avcodec_parameters_from_context(stream->codecpar, avctx);
229 fprintf(stderr, "Could not initialize stream parameters\n");
233 /* Save the encoder context for easier access later. */
234 *output_codec_context = avctx;
239 avcodec_free_context(&avctx);
240 avio_closep(&(*output_format_context)->pb);
241 avformat_free_context(*output_format_context);
242 *output_format_context = NULL;
243 return error < 0 ? error : AVERROR_EXIT;
247 * Initialize one data packet for reading or writing.
248 * @param[out] packet Packet to be initialized
249 * @return Error code (0 if successful)
251 static int init_packet(AVPacket **packet)
253 if (!(*packet = av_packet_alloc())) {
254 fprintf(stderr, "Could not allocate packet\n");
255 return AVERROR(ENOMEM);
261 * Initialize one audio frame for reading from the input file.
262 * @param[out] frame Frame to be initialized
263 * @return Error code (0 if successful)
265 static int init_input_frame(AVFrame **frame)
267 if (!(*frame = av_frame_alloc())) {
268 fprintf(stderr, "Could not allocate input frame\n");
269 return AVERROR(ENOMEM);
275 * Initialize the audio resampler based on the input and output codec settings.
276 * If the input and output sample formats differ, a conversion is required
277 * libswresample takes care of this, but requires initialization.
278 * @param input_codec_context Codec context of the input file
279 * @param output_codec_context Codec context of the output file
280 * @param[out] resample_context Resample context for the required conversion
281 * @return Error code (0 if successful)
283 static int init_resampler(AVCodecContext *input_codec_context,
284 AVCodecContext *output_codec_context,
285 SwrContext **resample_context)
290 * Create a resampler context for the conversion.
291 * Set the conversion parameters.
292 * Default channel layouts based on the number of channels
293 * are assumed for simplicity (they are sometimes not detected
294 * properly by the demuxer and/or decoder).
296 *resample_context = swr_alloc_set_opts(NULL,
297 av_get_default_channel_layout(output_codec_context->channels),
298 output_codec_context->sample_fmt,
299 output_codec_context->sample_rate,
300 av_get_default_channel_layout(input_codec_context->channels),
301 input_codec_context->sample_fmt,
302 input_codec_context->sample_rate,
304 if (!*resample_context) {
305 fprintf(stderr, "Could not allocate resample context\n");
306 return AVERROR(ENOMEM);
309 * Perform a sanity check so that the number of converted samples is
310 * not greater than the number of samples to be converted.
311 * If the sample rates differ, this case has to be handled differently
313 av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
315 /* Open the resampler with the specified parameters. */
316 if ((error = swr_init(*resample_context)) < 0) {
317 fprintf(stderr, "Could not open resample context\n");
318 swr_free(resample_context);
325 * Initialize a FIFO buffer for the audio samples to be encoded.
326 * @param[out] fifo Sample buffer
327 * @param output_codec_context Codec context of the output file
328 * @return Error code (0 if successful)
330 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
332 /* Create the FIFO buffer based on the specified output sample format. */
333 if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
334 output_codec_context->channels, 1))) {
335 fprintf(stderr, "Could not allocate FIFO\n");
336 return AVERROR(ENOMEM);
342 * Write the header of the output file container.
343 * @param output_format_context Format context of the output file
344 * @return Error code (0 if successful)
346 static int write_output_file_header(AVFormatContext *output_format_context)
349 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
350 fprintf(stderr, "Could not write output file header (error '%s')\n",
358 * Decode one audio frame from the input file.
359 * @param frame Audio frame to be decoded
360 * @param input_format_context Format context of the input file
361 * @param input_codec_context Codec context of the input file
362 * @param[out] data_present Indicates whether data has been decoded
363 * @param[out] finished Indicates whether the end of file has
364 * been reached and all data has been
365 * decoded. If this flag is false, there
366 * is more data to be decoded, i.e., this
367 * function has to be called again.
368 * @return Error code (0 if successful)
370 static int decode_audio_frame(AVFrame *frame,
371 AVFormatContext *input_format_context,
372 AVCodecContext *input_codec_context,
373 int *data_present, int *finished)
375 /* Packet used for temporary storage. */
376 AVPacket *input_packet;
379 error = init_packet(&input_packet);
383 /* Read one audio frame from the input file into a temporary packet. */
384 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
385 /* If we are at the end of the file, flush the decoder below. */
386 if (error == AVERROR_EOF)
389 fprintf(stderr, "Could not read frame (error '%s')\n",
395 /* Send the audio frame stored in the temporary packet to the decoder.
396 * The input audio stream decoder is used to do this. */
397 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
398 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
403 /* Receive one frame from the decoder. */
404 error = avcodec_receive_frame(input_codec_context, frame);
405 /* If the decoder asks for more data to be able to decode a frame,
406 * return indicating that no data is present. */
407 if (error == AVERROR(EAGAIN)) {
410 /* If the end of the input file is reached, stop decoding. */
411 } else if (error == AVERROR_EOF) {
415 } else if (error < 0) {
416 fprintf(stderr, "Could not decode frame (error '%s')\n",
419 /* Default case: Return decoded data. */
426 av_packet_free(&input_packet);
431 * Initialize a temporary storage for the specified number of audio samples.
432 * The conversion requires temporary storage due to the different format.
433 * The number of audio samples to be allocated is specified in frame_size.
434 * @param[out] converted_input_samples Array of converted samples. The
435 * dimensions are reference, channel
436 * (for multi-channel audio), sample.
437 * @param output_codec_context Codec context of the output file
438 * @param frame_size Number of samples to be converted in
440 * @return Error code (0 if successful)
442 static int init_converted_samples(uint8_t ***converted_input_samples,
443 AVCodecContext *output_codec_context,
448 /* Allocate as many pointers as there are audio channels.
449 * Each pointer will later point to the audio samples of the corresponding
450 * channels (although it may be NULL for interleaved formats).
452 if (!(*converted_input_samples = calloc(output_codec_context->channels,
453 sizeof(**converted_input_samples)))) {
454 fprintf(stderr, "Could not allocate converted input sample pointers\n");
455 return AVERROR(ENOMEM);
458 /* Allocate memory for the samples of all channels in one consecutive
459 * block for convenience. */
460 if ((error = av_samples_alloc(*converted_input_samples, NULL,
461 output_codec_context->channels,
463 output_codec_context->sample_fmt, 0)) < 0) {
465 "Could not allocate converted input samples (error '%s')\n",
467 av_freep(&(*converted_input_samples)[0]);
468 free(*converted_input_samples);
475 * Convert the input audio samples into the output sample format.
476 * The conversion happens on a per-frame basis, the size of which is
477 * specified by frame_size.
478 * @param input_data Samples to be decoded. The dimensions are
479 * channel (for multi-channel audio), sample.
480 * @param[out] converted_data Converted samples. The dimensions are channel
481 * (for multi-channel audio), sample.
482 * @param frame_size Number of samples to be converted
483 * @param resample_context Resample context for the conversion
484 * @return Error code (0 if successful)
486 static int convert_samples(const uint8_t **input_data,
487 uint8_t **converted_data, const int frame_size,
488 SwrContext *resample_context)
492 /* Convert the samples using the resampler. */
493 if ((error = swr_convert(resample_context,
494 converted_data, frame_size,
495 input_data , frame_size)) < 0) {
496 fprintf(stderr, "Could not convert input samples (error '%s')\n",
505 * Add converted input audio samples to the FIFO buffer for later processing.
506 * @param fifo Buffer to add the samples to
507 * @param converted_input_samples Samples to be added. The dimensions are channel
508 * (for multi-channel audio), sample.
509 * @param frame_size Number of samples to be converted
510 * @return Error code (0 if successful)
512 static int add_samples_to_fifo(AVAudioFifo *fifo,
513 uint8_t **converted_input_samples,
514 const int frame_size)
518 /* Make the FIFO as large as it needs to be to hold both,
519 * the old and the new samples. */
520 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
521 fprintf(stderr, "Could not reallocate FIFO\n");
525 /* Store the new samples in the FIFO buffer. */
526 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
527 frame_size) < frame_size) {
528 fprintf(stderr, "Could not write data to FIFO\n");
535 * Read one audio frame from the input file, decode, convert and store
536 * it in the FIFO buffer.
537 * @param fifo Buffer used for temporary storage
538 * @param input_format_context Format context of the input file
539 * @param input_codec_context Codec context of the input file
540 * @param output_codec_context Codec context of the output file
541 * @param resampler_context Resample context for the conversion
542 * @param[out] finished Indicates whether the end of file has
543 * been reached and all data has been
544 * decoded. If this flag is false,
545 * there is more data to be decoded,
546 * i.e., this function has to be called
548 * @return Error code (0 if successful)
550 static int read_decode_convert_and_store(AVAudioFifo *fifo,
551 AVFormatContext *input_format_context,
552 AVCodecContext *input_codec_context,
553 AVCodecContext *output_codec_context,
554 SwrContext *resampler_context,
557 /* Temporary storage of the input samples of the frame read from the file. */
558 AVFrame *input_frame = NULL;
559 /* Temporary storage for the converted input samples. */
560 uint8_t **converted_input_samples = NULL;
561 int data_present = 0;
562 int ret = AVERROR_EXIT;
564 /* Initialize temporary storage for one input frame. */
565 if (init_input_frame(&input_frame))
567 /* Decode one frame worth of audio samples. */
568 if (decode_audio_frame(input_frame, input_format_context,
569 input_codec_context, &data_present, finished))
571 /* If we are at the end of the file and there are no more samples
572 * in the decoder which are delayed, we are actually finished.
573 * This must not be treated as an error. */
578 /* If there is decoded data, convert and store it. */
580 /* Initialize the temporary storage for the converted input samples. */
581 if (init_converted_samples(&converted_input_samples, output_codec_context,
582 input_frame->nb_samples))
585 /* Convert the input samples to the desired output sample format.
586 * This requires a temporary storage provided by converted_input_samples. */
587 if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
588 input_frame->nb_samples, resampler_context))
591 /* Add the converted input samples to the FIFO buffer for later processing. */
592 if (add_samples_to_fifo(fifo, converted_input_samples,
593 input_frame->nb_samples))
600 if (converted_input_samples) {
601 av_freep(&converted_input_samples[0]);
602 free(converted_input_samples);
604 av_frame_free(&input_frame);
610 * Initialize one input frame for writing to the output file.
611 * The frame will be exactly frame_size samples large.
612 * @param[out] frame Frame to be initialized
613 * @param output_codec_context Codec context of the output file
614 * @param frame_size Size of the frame
615 * @return Error code (0 if successful)
617 static int init_output_frame(AVFrame **frame,
618 AVCodecContext *output_codec_context,
623 /* Create a new frame to store the audio samples. */
624 if (!(*frame = av_frame_alloc())) {
625 fprintf(stderr, "Could not allocate output frame\n");
629 /* Set the frame's parameters, especially its size and format.
630 * av_frame_get_buffer needs this to allocate memory for the
631 * audio samples of the frame.
632 * Default channel layouts based on the number of channels
633 * are assumed for simplicity. */
634 (*frame)->nb_samples = frame_size;
635 (*frame)->channel_layout = output_codec_context->channel_layout;
636 (*frame)->format = output_codec_context->sample_fmt;
637 (*frame)->sample_rate = output_codec_context->sample_rate;
639 /* Allocate the samples of the created frame. This call will make
640 * sure that the audio frame can hold as many samples as specified. */
641 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
642 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
644 av_frame_free(frame);
651 /* Global timestamp for the audio frames. */
652 static int64_t pts = 0;
655 * Encode one frame worth of audio to the output file.
656 * @param frame Samples to be encoded
657 * @param output_format_context Format context of the output file
658 * @param output_codec_context Codec context of the output file
659 * @param[out] data_present Indicates whether data has been
661 * @return Error code (0 if successful)
663 static int encode_audio_frame(AVFrame *frame,
664 AVFormatContext *output_format_context,
665 AVCodecContext *output_codec_context,
668 /* Packet used for temporary storage. */
669 AVPacket *output_packet;
672 error = init_packet(&output_packet);
676 /* Set a timestamp based on the sample rate for the container. */
679 pts += frame->nb_samples;
682 /* Send the audio frame stored in the temporary packet to the encoder.
683 * The output audio stream encoder is used to do this. */
684 error = avcodec_send_frame(output_codec_context, frame);
685 /* The encoder signals that it has nothing more to encode. */
686 if (error == AVERROR_EOF) {
689 } else if (error < 0) {
690 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
695 /* Receive one encoded frame from the encoder. */
696 error = avcodec_receive_packet(output_codec_context, output_packet);
697 /* If the encoder asks for more data to be able to provide an
698 * encoded frame, return indicating that no data is present. */
699 if (error == AVERROR(EAGAIN)) {
702 /* If the last frame has been encoded, stop encoding. */
703 } else if (error == AVERROR_EOF) {
706 } else if (error < 0) {
707 fprintf(stderr, "Could not encode frame (error '%s')\n",
710 /* Default case: Return encoded data. */
715 /* Write one audio frame from the temporary packet to the output file. */
717 (error = av_write_frame(output_format_context, output_packet)) < 0) {
718 fprintf(stderr, "Could not write frame (error '%s')\n",
724 av_packet_free(&output_packet);
729 * Load one audio frame from the FIFO buffer, encode and write it to the
731 * @param fifo Buffer used for temporary storage
732 * @param output_format_context Format context of the output file
733 * @param output_codec_context Codec context of the output file
734 * @return Error code (0 if successful)
736 static int load_encode_and_write(AVAudioFifo *fifo,
737 AVFormatContext *output_format_context,
738 AVCodecContext *output_codec_context)
740 /* Temporary storage of the output samples of the frame written to the file. */
741 AVFrame *output_frame;
742 /* Use the maximum number of possible samples per frame.
743 * If there is less than the maximum possible frame size in the FIFO
744 * buffer use this number. Otherwise, use the maximum possible frame size. */
745 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
746 output_codec_context->frame_size);
749 /* Initialize temporary storage for one output frame. */
750 if (init_output_frame(&output_frame, output_codec_context, frame_size))
753 /* Read as many samples from the FIFO buffer as required to fill the frame.
754 * The samples are stored in the frame temporarily. */
755 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
756 fprintf(stderr, "Could not read data from FIFO\n");
757 av_frame_free(&output_frame);
761 /* Encode one frame worth of audio samples. */
762 if (encode_audio_frame(output_frame, output_format_context,
763 output_codec_context, &data_written)) {
764 av_frame_free(&output_frame);
767 av_frame_free(&output_frame);
772 * Write the trailer of the output file container.
773 * @param output_format_context Format context of the output file
774 * @return Error code (0 if successful)
776 static int write_output_file_trailer(AVFormatContext *output_format_context)
779 if ((error = av_write_trailer(output_format_context)) < 0) {
780 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
787 int main(int argc, char **argv)
789 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
790 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
791 SwrContext *resample_context = NULL;
792 AVAudioFifo *fifo = NULL;
793 int ret = AVERROR_EXIT;
796 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
800 /* Open the input file for reading. */
801 if (open_input_file(argv[1], &input_format_context,
802 &input_codec_context))
804 /* Open the output file for writing. */
805 if (open_output_file(argv[2], input_codec_context,
806 &output_format_context, &output_codec_context))
808 /* Initialize the resampler to be able to convert audio sample formats. */
809 if (init_resampler(input_codec_context, output_codec_context,
812 /* Initialize the FIFO buffer to store audio samples to be encoded. */
813 if (init_fifo(&fifo, output_codec_context))
815 /* Write the header of the output file container. */
816 if (write_output_file_header(output_format_context))
819 /* Loop as long as we have input samples to read or output samples
820 * to write; abort as soon as we have neither. */
822 /* Use the encoder's desired frame size for processing. */
823 const int output_frame_size = output_codec_context->frame_size;
826 /* Make sure that there is one frame worth of samples in the FIFO
827 * buffer so that the encoder can do its work.
828 * Since the decoder's and the encoder's frame size may differ, we
829 * need to FIFO buffer to store as many frames worth of input samples
830 * that they make up at least one frame worth of output samples. */
831 while (av_audio_fifo_size(fifo) < output_frame_size) {
832 /* Decode one frame worth of audio samples, convert it to the
833 * output sample format and put it into the FIFO buffer. */
834 if (read_decode_convert_and_store(fifo, input_format_context,
836 output_codec_context,
837 resample_context, &finished))
840 /* If we are at the end of the input file, we continue
841 * encoding the remaining audio samples to the output file. */
846 /* If we have enough samples for the encoder, we encode them.
847 * At the end of the file, we pass the remaining samples to
849 while (av_audio_fifo_size(fifo) >= output_frame_size ||
850 (finished && av_audio_fifo_size(fifo) > 0))
851 /* Take one frame worth of audio samples from the FIFO buffer,
852 * encode it and write it to the output file. */
853 if (load_encode_and_write(fifo, output_format_context,
854 output_codec_context))
857 /* If we are at the end of the input file and have encoded
858 * all remaining samples, we can exit this loop and finish. */
861 /* Flush the encoder as it may have delayed frames. */
864 if (encode_audio_frame(NULL, output_format_context,
865 output_codec_context, &data_written))
867 } while (data_written);
872 /* Write the trailer of the output file container. */
873 if (write_output_file_trailer(output_format_context))
879 av_audio_fifo_free(fifo);
880 swr_free(&resample_context);
881 if (output_codec_context)
882 avcodec_free_context(&output_codec_context);
883 if (output_format_context) {
884 avio_closep(&output_format_context->pb);
885 avformat_free_context(output_format_context);
887 if (input_codec_context)
888 avcodec_free_context(&input_codec_context);
889 if (input_format_context)
890 avformat_close_input(&input_format_context);