1/*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19/**
20 * @file
21 * simple audio converter
22 *
23 * @example transcode_aac.c
24 * Convert an input audio file to AAC in an MP4 container using FFmpeg.
25 * @author Andreas Unterweger (dustsigns@gmail.com)
26 */
27
28#include <stdio.h>
29
30#include "libavformat/avformat.h"
31#include "libavformat/avio.h"
32
33#include "libavcodec/avcodec.h"
34
35#include "libavutil/audio_fifo.h"
36#include "libavutil/avassert.h"
37#include "libavutil/avstring.h"
38#include "libavutil/frame.h"
39#include "libavutil/opt.h"
40
41#include "libswresample/swresample.h"
42
43/** The output bit rate in kbit/s */
44#define OUTPUT_BIT_RATE 48000
45/** The number of output channels */
46#define OUTPUT_CHANNELS 2
47/** The audio sample output format */
48#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
49
50/**
51 * Convert an error code into a text message.
52 * @param error Error code to be converted
53 * @return Corresponding error text (not thread-safe)
54 */
55static char *const get_error_text(const int error)
56{
57    static char error_buffer[255];
58    av_strerror(error, error_buffer, sizeof(error_buffer));
59    return error_buffer;
60}
61
62/** Open an input file and the required decoder. */
63static int open_input_file(const char *filename,
64                           AVFormatContext **input_format_context,
65                           AVCodecContext **input_codec_context)
66{
67    AVCodec *input_codec;
68    int error;
69
70    /** Open the input file to read from it. */
71    if ((error = avformat_open_input(input_format_context, filename, NULL,
72                                     NULL)) < 0) {
73        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
74                filename, get_error_text(error));
75        *input_format_context = NULL;
76        return error;
77    }
78
79    /** Get information on the input file (number of streams etc.). */
80    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
81        fprintf(stderr, "Could not open find stream info (error '%s')\n",
82                get_error_text(error));
83        avformat_close_input(input_format_context);
84        return error;
85    }
86
87    /** Make sure that there is only one stream in the input file. */
88    if ((*input_format_context)->nb_streams != 1) {
89        fprintf(stderr, "Expected one audio input stream, but found %d\n",
90                (*input_format_context)->nb_streams);
91        avformat_close_input(input_format_context);
92        return AVERROR_EXIT;
93    }
94
95    /** Find a decoder for the audio stream. */
96    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
97        fprintf(stderr, "Could not find input codec\n");
98        avformat_close_input(input_format_context);
99        return AVERROR_EXIT;
100    }
101
102    /** Open the decoder for the audio stream to use it later. */
103    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
104                               input_codec, NULL)) < 0) {
105        fprintf(stderr, "Could not open input codec (error '%s')\n",
106                get_error_text(error));
107        avformat_close_input(input_format_context);
108        return error;
109    }
110
111    /** Save the decoder context for easier access later. */
112    *input_codec_context = (*input_format_context)->streams[0]->codec;
113
114    return 0;
115}
116
117/**
118 * Open an output file and the required encoder.
119 * Also set some basic encoder parameters.
120 * Some of these parameters are based on the input file's parameters.
121 */
122static int open_output_file(const char *filename,
123                            AVCodecContext *input_codec_context,
124                            AVFormatContext **output_format_context,
125                            AVCodecContext **output_codec_context)
126{
127    AVIOContext *output_io_context = NULL;
128    AVStream *stream               = NULL;
129    AVCodec *output_codec          = NULL;
130    int error;
131
132    /** Open the output file to write to it. */
133    if ((error = avio_open(&output_io_context, filename,
134                           AVIO_FLAG_WRITE)) < 0) {
135        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
136                filename, get_error_text(error));
137        return error;
138    }
139
140    /** Create a new format context for the output container format. */
141    if (!(*output_format_context = avformat_alloc_context())) {
142        fprintf(stderr, "Could not allocate output format context\n");
143        return AVERROR(ENOMEM);
144    }
145
146    /** Associate the output file (pointer) with the container format context. */
147    (*output_format_context)->pb = output_io_context;
148
149    /** Guess the desired container format based on the file extension. */
150    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
151                                                              NULL))) {
152        fprintf(stderr, "Could not find output file format\n");
153        goto cleanup;
154    }
155
156    av_strlcpy((*output_format_context)->filename, filename,
157               sizeof((*output_format_context)->filename));
158
159    /** Find the encoder to be used by its name. */
160    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
161        fprintf(stderr, "Could not find an AAC encoder.\n");
162        goto cleanup;
163    }
164
165    /** Create a new audio stream in the output file container. */
166    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
167        fprintf(stderr, "Could not create new stream\n");
168        error = AVERROR(ENOMEM);
169        goto cleanup;
170    }
171
172    /** Save the encoder context for easiert access later. */
173    *output_codec_context = stream->codec;
174
175    /**
176     * Set the basic encoder parameters.
177     * The input file's sample rate is used to avoid a sample rate conversion.
178     */
179    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
180    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
181    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
182    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
183    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
184
185    /**
186     * Some container formats (like MP4) require global headers to be present
187     * Mark the encoder so that it behaves accordingly.
188     */
189    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
190        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
191
192    /** Open the encoder for the audio stream to use it later. */
193    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
194        fprintf(stderr, "Could not open output codec (error '%s')\n",
195                get_error_text(error));
196        goto cleanup;
197    }
198
199    return 0;
200
201cleanup:
202    avio_close((*output_format_context)->pb);
203    avformat_free_context(*output_format_context);
204    *output_format_context = NULL;
205    return error < 0 ? error : AVERROR_EXIT;
206}
207
208/** Initialize one data packet for reading or writing. */
209static void init_packet(AVPacket *packet)
210{
211    av_init_packet(packet);
212    /** Set the packet data and size so that it is recognized as being empty. */
213    packet->data = NULL;
214    packet->size = 0;
215}
216
217/** Initialize one audio frame for reading from the input file */
218static int init_input_frame(AVFrame **frame)
219{
220    if (!(*frame = av_frame_alloc())) {
221        fprintf(stderr, "Could not allocate input frame\n");
222        return AVERROR(ENOMEM);
223    }
224    return 0;
225}
226
227/**
228 * Initialize the audio resampler based on the input and output codec settings.
229 * If the input and output sample formats differ, a conversion is required
230 * libswresample takes care of this, but requires initialization.
231 */
232static int init_resampler(AVCodecContext *input_codec_context,
233                          AVCodecContext *output_codec_context,
234                          SwrContext **resample_context)
235{
236        int error;
237
238        /**
239         * Create a resampler context for the conversion.
240         * Set the conversion parameters.
241         * Default channel layouts based on the number of channels
242         * are assumed for simplicity (they are sometimes not detected
243         * properly by the demuxer and/or decoder).
244         */
245        *resample_context = swr_alloc_set_opts(NULL,
246                                              av_get_default_channel_layout(output_codec_context->channels),
247                                              output_codec_context->sample_fmt,
248                                              output_codec_context->sample_rate,
249                                              av_get_default_channel_layout(input_codec_context->channels),
250                                              input_codec_context->sample_fmt,
251                                              input_codec_context->sample_rate,
252                                              0, NULL);
253        if (!*resample_context) {
254            fprintf(stderr, "Could not allocate resample context\n");
255            return AVERROR(ENOMEM);
256        }
257        /**
258        * Perform a sanity check so that the number of converted samples is
259        * not greater than the number of samples to be converted.
260        * If the sample rates differ, this case has to be handled differently
261        */
262        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
263
264        /** Open the resampler with the specified parameters. */
265        if ((error = swr_init(*resample_context)) < 0) {
266            fprintf(stderr, "Could not open resample context\n");
267            swr_free(resample_context);
268            return error;
269        }
270    return 0;
271}
272
273/** Initialize a FIFO buffer for the audio samples to be encoded. */
274static int init_fifo(AVAudioFifo **fifo)
275{
276    /** Create the FIFO buffer based on the specified output sample format. */
277    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
278        fprintf(stderr, "Could not allocate FIFO\n");
279        return AVERROR(ENOMEM);
280    }
281    return 0;
282}
283
284/** Write the header of the output file container. */
285static int write_output_file_header(AVFormatContext *output_format_context)
286{
287    int error;
288    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
289        fprintf(stderr, "Could not write output file header (error '%s')\n",
290                get_error_text(error));
291        return error;
292    }
293    return 0;
294}
295
296/** Decode one audio frame from the input file. */
297static int decode_audio_frame(AVFrame *frame,
298                              AVFormatContext *input_format_context,
299                              AVCodecContext *input_codec_context,
300                              int *data_present, int *finished)
301{
302    /** Packet used for temporary storage. */
303    AVPacket input_packet;
304    int error;
305    init_packet(&input_packet);
306
307    /** Read one audio frame from the input file into a temporary packet. */
308    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
309        /** If we are the the end of the file, flush the decoder below. */
310        if (error == AVERROR_EOF)
311            *finished = 1;
312        else {
313            fprintf(stderr, "Could not read frame (error '%s')\n",
314                    get_error_text(error));
315            return error;
316        }
317    }
318
319    /**
320     * Decode the audio frame stored in the temporary packet.
321     * The input audio stream decoder is used to do this.
322     * If we are at the end of the file, pass an empty packet to the decoder
323     * to flush it.
324     */
325    if ((error = avcodec_decode_audio4(input_codec_context, frame,
326                                       data_present, &input_packet)) < 0) {
327        fprintf(stderr, "Could not decode frame (error '%s')\n",
328                get_error_text(error));
329        av_free_packet(&input_packet);
330        return error;
331    }
332
333    /**
334     * If the decoder has not been flushed completely, we are not finished,
335     * so that this function has to be called again.
336     */
337    if (*finished && *data_present)
338        *finished = 0;
339    av_free_packet(&input_packet);
340    return 0;
341}
342
343/**
344 * Initialize a temporary storage for the specified number of audio samples.
345 * The conversion requires temporary storage due to the different format.
346 * The number of audio samples to be allocated is specified in frame_size.
347 */
348static int init_converted_samples(uint8_t ***converted_input_samples,
349                                  AVCodecContext *output_codec_context,
350                                  int frame_size)
351{
352    int error;
353
354    /**
355     * Allocate as many pointers as there are audio channels.
356     * Each pointer will later point to the audio samples of the corresponding
357     * channels (although it may be NULL for interleaved formats).
358     */
359    if (!(*converted_input_samples = calloc(output_codec_context->channels,
360                                            sizeof(**converted_input_samples)))) {
361        fprintf(stderr, "Could not allocate converted input sample pointers\n");
362        return AVERROR(ENOMEM);
363    }
364
365    /**
366     * Allocate memory for the samples of all channels in one consecutive
367     * block for convenience.
368     */
369    if ((error = av_samples_alloc(*converted_input_samples, NULL,
370                                  output_codec_context->channels,
371                                  frame_size,
372                                  output_codec_context->sample_fmt, 0)) < 0) {
373        fprintf(stderr,
374                "Could not allocate converted input samples (error '%s')\n",
375                get_error_text(error));
376        av_freep(&(*converted_input_samples)[0]);
377        free(*converted_input_samples);
378        return error;
379    }
380    return 0;
381}
382
383/**
384 * Convert the input audio samples into the output sample format.
385 * The conversion happens on a per-frame basis, the size of which is specified
386 * by frame_size.
387 */
388static int convert_samples(const uint8_t **input_data,
389                           uint8_t **converted_data, const int frame_size,
390                           SwrContext *resample_context)
391{
392    int error;
393
394    /** Convert the samples using the resampler. */
395    if ((error = swr_convert(resample_context,
396                             converted_data, frame_size,
397                             input_data    , frame_size)) < 0) {
398        fprintf(stderr, "Could not convert input samples (error '%s')\n",
399                get_error_text(error));
400        return error;
401    }
402
403    return 0;
404}
405
406/** Add converted input audio samples to the FIFO buffer for later processing. */
407static int add_samples_to_fifo(AVAudioFifo *fifo,
408                               uint8_t **converted_input_samples,
409                               const int frame_size)
410{
411    int error;
412
413    /**
414     * Make the FIFO as large as it needs to be to hold both,
415     * the old and the new samples.
416     */
417    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
418        fprintf(stderr, "Could not reallocate FIFO\n");
419        return error;
420    }
421
422    /** Store the new samples in the FIFO buffer. */
423    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
424                            frame_size) < frame_size) {
425        fprintf(stderr, "Could not write data to FIFO\n");
426        return AVERROR_EXIT;
427    }
428    return 0;
429}
430
431/**
432 * Read one audio frame from the input file, decodes, converts and stores
433 * it in the FIFO buffer.
434 */
435static int read_decode_convert_and_store(AVAudioFifo *fifo,
436                                         AVFormatContext *input_format_context,
437                                         AVCodecContext *input_codec_context,
438                                         AVCodecContext *output_codec_context,
439                                         SwrContext *resampler_context,
440                                         int *finished)
441{
442    /** Temporary storage of the input samples of the frame read from the file. */
443    AVFrame *input_frame = NULL;
444    /** Temporary storage for the converted input samples. */
445    uint8_t **converted_input_samples = NULL;
446    int data_present;
447    int ret = AVERROR_EXIT;
448
449    /** Initialize temporary storage for one input frame. */
450    if (init_input_frame(&input_frame))
451        goto cleanup;
452    /** Decode one frame worth of audio samples. */
453    if (decode_audio_frame(input_frame, input_format_context,
454                           input_codec_context, &data_present, finished))
455        goto cleanup;
456    /**
457     * If we are at the end of the file and there are no more samples
458     * in the decoder which are delayed, we are actually finished.
459     * This must not be treated as an error.
460     */
461    if (*finished && !data_present) {
462        ret = 0;
463        goto cleanup;
464    }
465    /** If there is decoded data, convert and store it */
466    if (data_present) {
467        /** Initialize the temporary storage for the converted input samples. */
468        if (init_converted_samples(&converted_input_samples, output_codec_context,
469                                   input_frame->nb_samples))
470            goto cleanup;
471
472        /**
473         * Convert the input samples to the desired output sample format.
474         * This requires a temporary storage provided by converted_input_samples.
475         */
476        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
477                            input_frame->nb_samples, resampler_context))
478            goto cleanup;
479
480        /** Add the converted input samples to the FIFO buffer for later processing. */
481        if (add_samples_to_fifo(fifo, converted_input_samples,
482                                input_frame->nb_samples))
483            goto cleanup;
484        ret = 0;
485    }
486    ret = 0;
487
488cleanup:
489    if (converted_input_samples) {
490        av_freep(&converted_input_samples[0]);
491        free(converted_input_samples);
492    }
493    av_frame_free(&input_frame);
494
495    return ret;
496}
497
498/**
499 * Initialize one input frame for writing to the output file.
500 * The frame will be exactly frame_size samples large.
501 */
502static int init_output_frame(AVFrame **frame,
503                             AVCodecContext *output_codec_context,
504                             int frame_size)
505{
506    int error;
507
508    /** Create a new frame to store the audio samples. */
509    if (!(*frame = av_frame_alloc())) {
510        fprintf(stderr, "Could not allocate output frame\n");
511        return AVERROR_EXIT;
512    }
513
514    /**
515     * Set the frame's parameters, especially its size and format.
516     * av_frame_get_buffer needs this to allocate memory for the
517     * audio samples of the frame.
518     * Default channel layouts based on the number of channels
519     * are assumed for simplicity.
520     */
521    (*frame)->nb_samples     = frame_size;
522    (*frame)->channel_layout = output_codec_context->channel_layout;
523    (*frame)->format         = output_codec_context->sample_fmt;
524    (*frame)->sample_rate    = output_codec_context->sample_rate;
525
526    /**
527     * Allocate the samples of the created frame. This call will make
528     * sure that the audio frame can hold as many samples as specified.
529     */
530    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
531        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
532                get_error_text(error));
533        av_frame_free(frame);
534        return error;
535    }
536
537    return 0;
538}
539
540/** Encode one frame worth of audio to the output file. */
541static int encode_audio_frame(AVFrame *frame,
542                              AVFormatContext *output_format_context,
543                              AVCodecContext *output_codec_context,
544                              int *data_present)
545{
546    /** Packet used for temporary storage. */
547    AVPacket output_packet;
548    int error;
549    init_packet(&output_packet);
550
551    /**
552     * Encode the audio frame and store it in the temporary packet.
553     * The output audio stream encoder is used to do this.
554     */
555    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
556                                       frame, data_present)) < 0) {
557        fprintf(stderr, "Could not encode frame (error '%s')\n",
558                get_error_text(error));
559        av_free_packet(&output_packet);
560        return error;
561    }
562
563    /** Write one audio frame from the temporary packet to the output file. */
564    if (*data_present) {
565        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
566            fprintf(stderr, "Could not write frame (error '%s')\n",
567                    get_error_text(error));
568            av_free_packet(&output_packet);
569            return error;
570        }
571
572        av_free_packet(&output_packet);
573    }
574
575    return 0;
576}
577
578/**
579 * Load one audio frame from the FIFO buffer, encode and write it to the
580 * output file.
581 */
582static int load_encode_and_write(AVAudioFifo *fifo,
583                                 AVFormatContext *output_format_context,
584                                 AVCodecContext *output_codec_context)
585{
586    /** Temporary storage of the output samples of the frame written to the file. */
587    AVFrame *output_frame;
588    /**
589     * Use the maximum number of possible samples per frame.
590     * If there is less than the maximum possible frame size in the FIFO
591     * buffer use this number. Otherwise, use the maximum possible frame size
592     */
593    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
594                                 output_codec_context->frame_size);
595    int data_written;
596
597    /** Initialize temporary storage for one output frame. */
598    if (init_output_frame(&output_frame, output_codec_context, frame_size))
599        return AVERROR_EXIT;
600
601    /**
602     * Read as many samples from the FIFO buffer as required to fill the frame.
603     * The samples are stored in the frame temporarily.
604     */
605    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
606        fprintf(stderr, "Could not read data from FIFO\n");
607        av_frame_free(&output_frame);
608        return AVERROR_EXIT;
609    }
610
611    /** Encode one frame worth of audio samples. */
612    if (encode_audio_frame(output_frame, output_format_context,
613                           output_codec_context, &data_written)) {
614        av_frame_free(&output_frame);
615        return AVERROR_EXIT;
616    }
617    av_frame_free(&output_frame);
618    return 0;
619}
620
621/** Write the trailer of the output file container. */
622static int write_output_file_trailer(AVFormatContext *output_format_context)
623{
624    int error;
625    if ((error = av_write_trailer(output_format_context)) < 0) {
626        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
627                get_error_text(error));
628        return error;
629    }
630    return 0;
631}
632
633/** Convert an audio file to an AAC file in an MP4 container. */
634int main(int argc, char **argv)
635{
636    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
637    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
638    SwrContext *resample_context = NULL;
639    AVAudioFifo *fifo = NULL;
640    int ret = AVERROR_EXIT;
641
642    if (argc < 3) {
643        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
644        exit(1);
645    }
646
647    /** Register all codecs and formats so that they can be used. */
648    av_register_all();
649    /** Open the input file for reading. */
650    if (open_input_file(argv[1], &input_format_context,
651                        &input_codec_context))
652        goto cleanup;
653    /** Open the output file for writing. */
654    if (open_output_file(argv[2], input_codec_context,
655                         &output_format_context, &output_codec_context))
656        goto cleanup;
657    /** Initialize the resampler to be able to convert audio sample formats. */
658    if (init_resampler(input_codec_context, output_codec_context,
659                       &resample_context))
660        goto cleanup;
661    /** Initialize the FIFO buffer to store audio samples to be encoded. */
662    if (init_fifo(&fifo))
663        goto cleanup;
664    /** Write the header of the output file container. */
665    if (write_output_file_header(output_format_context))
666        goto cleanup;
667
668    /**
669     * Loop as long as we have input samples to read or output samples
670     * to write; abort as soon as we have neither.
671     */
672    while (1) {
673        /** Use the encoder's desired frame size for processing. */
674        const int output_frame_size = output_codec_context->frame_size;
675        int finished                = 0;
676
677        /**
678         * Make sure that there is one frame worth of samples in the FIFO
679         * buffer so that the encoder can do its work.
680         * Since the decoder's and the encoder's frame size may differ, we
681         * need to FIFO buffer to store as many frames worth of input samples
682         * that they make up at least one frame worth of output samples.
683         */
684        while (av_audio_fifo_size(fifo) < output_frame_size) {
685            /**
686             * Decode one frame worth of audio samples, convert it to the
687             * output sample format and put it into the FIFO buffer.
688             */
689            if (read_decode_convert_and_store(fifo, input_format_context,
690                                              input_codec_context,
691                                              output_codec_context,
692                                              resample_context, &finished))
693                goto cleanup;
694
695            /**
696             * If we are at the end of the input file, we continue
697             * encoding the remaining audio samples to the output file.
698             */
699            if (finished)
700                break;
701        }
702
703        /**
704         * If we have enough samples for the encoder, we encode them.
705         * At the end of the file, we pass the remaining samples to
706         * the encoder.
707         */
708        while (av_audio_fifo_size(fifo) >= output_frame_size ||
709               (finished && av_audio_fifo_size(fifo) > 0))
710            /**
711             * Take one frame worth of audio samples from the FIFO buffer,
712             * encode it and write it to the output file.
713             */
714            if (load_encode_and_write(fifo, output_format_context,
715                                      output_codec_context))
716                goto cleanup;
717
718        /**
719         * If we are at the end of the input file and have encoded
720         * all remaining samples, we can exit this loop and finish.
721         */
722        if (finished) {
723            int data_written;
724            /** Flush the encoder as it may have delayed frames. */
725            do {
726                if (encode_audio_frame(NULL, output_format_context,
727                                       output_codec_context, &data_written))
728                    goto cleanup;
729            } while (data_written);
730            break;
731        }
732    }
733
734    /** Write the trailer of the output file container. */
735    if (write_output_file_trailer(output_format_context))
736        goto cleanup;
737    ret = 0;
738
739cleanup:
740    if (fifo)
741        av_audio_fifo_free(fifo);
742    swr_free(&resample_context);
743    if (output_codec_context)
744        avcodec_close(output_codec_context);
745    if (output_format_context) {
746        avio_close(output_format_context->pb);
747        avformat_free_context(output_format_context);
748    }
749    if (input_codec_context)
750        avcodec_close(input_codec_context);
751    if (input_format_context)
752        avformat_close_input(&input_format_context);
753
754    return ret;
755}
756