FFmpeg 7.1.1
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transcode_aac.c
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1/*
2 * Copyright (c) 2013-2022 Andreas Unterweger
3 *
4 * This file is part of FFmpeg.
5 *
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.
10 *
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.
15 *
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
19 */
20
21/**
22 * @file audio transcoding to MPEG/AAC API usage example
23 * @example transcode_aac.c
24 *
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)
28 */
29
30#include <stdio.h>
31
32#include <libavutil/mem.h>
34#include <libavformat/avio.h>
35
36#include <libavcodec/avcodec.h>
37
39#include <libavutil/avassert.h>
40#include <libavutil/avstring.h>
42#include <libavutil/frame.h>
43#include <libavutil/opt.h>
44
46
47/* The output bit rate in bit/s */
48#define OUTPUT_BIT_RATE 96000
49/* The number of output channels */
50#define OUTPUT_CHANNELS 2
51
52/**
53 * Open an input file and the required decoder.
54 * @param filename File to be opened
55 * @param[out] input_format_context Format context of opened file
56 * @param[out] input_codec_context Codec context of opened file
57 * @return Error code (0 if successful)
58 */
59static int open_input_file(const char *filename,
60 AVFormatContext **input_format_context,
61 AVCodecContext **input_codec_context)
62{
63 AVCodecContext *avctx;
64 const AVCodec *input_codec;
65 const AVStream *stream;
66 int error;
67
68 /* Open the input file to read from it. */
69 if ((error = avformat_open_input(input_format_context, filename, NULL,
70 NULL)) < 0) {
71 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
72 filename, av_err2str(error));
73 *input_format_context = NULL;
74 return error;
75 }
76
77 /* Get information on the input file (number of streams etc.). */
78 if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
79 fprintf(stderr, "Could not open find stream info (error '%s')\n",
80 av_err2str(error));
81 avformat_close_input(input_format_context);
82 return error;
83 }
84
85 /* Make sure that there is only one stream in the input file. */
86 if ((*input_format_context)->nb_streams != 1) {
87 fprintf(stderr, "Expected one audio input stream, but found %d\n",
88 (*input_format_context)->nb_streams);
89 avformat_close_input(input_format_context);
90 return AVERROR_EXIT;
91 }
92
93 stream = (*input_format_context)->streams[0];
94
95 /* Find a decoder for the audio stream. */
96 if (!(input_codec = avcodec_find_decoder(stream->codecpar->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 /* Allocate a new decoding context. */
103 avctx = avcodec_alloc_context3(input_codec);
104 if (!avctx) {
105 fprintf(stderr, "Could not allocate a decoding context\n");
106 avformat_close_input(input_format_context);
107 return AVERROR(ENOMEM);
108 }
109
110 /* Initialize the stream parameters with demuxer information. */
111 error = avcodec_parameters_to_context(avctx, stream->codecpar);
112 if (error < 0) {
113 avformat_close_input(input_format_context);
114 avcodec_free_context(&avctx);
115 return error;
116 }
117
118 /* Open the decoder for the audio stream to use it later. */
119 if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
120 fprintf(stderr, "Could not open input codec (error '%s')\n",
121 av_err2str(error));
122 avcodec_free_context(&avctx);
123 avformat_close_input(input_format_context);
124 return error;
125 }
126
127 /* Set the packet timebase for the decoder. */
128 avctx->pkt_timebase = stream->time_base;
129
130 /* Save the decoder context for easier access later. */
131 *input_codec_context = avctx;
132
133 return 0;
134}
135
136/**
137 * Open an output file and the required encoder.
138 * Also set some basic encoder parameters.
139 * Some of these parameters are based on the input file's parameters.
140 * @param filename File to be opened
141 * @param input_codec_context Codec context of input file
142 * @param[out] output_format_context Format context of output file
143 * @param[out] output_codec_context Codec context of output file
144 * @return Error code (0 if successful)
145 */
146static int open_output_file(const char *filename,
147 AVCodecContext *input_codec_context,
148 AVFormatContext **output_format_context,
149 AVCodecContext **output_codec_context)
150{
151 AVCodecContext *avctx = NULL;
152 AVIOContext *output_io_context = NULL;
153 AVStream *stream = NULL;
154 const AVCodec *output_codec = NULL;
155 int error;
156
157 /* Open the output file to write to it. */
158 if ((error = avio_open(&output_io_context, filename,
159 AVIO_FLAG_WRITE)) < 0) {
160 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
161 filename, av_err2str(error));
162 return error;
163 }
164
165 /* Create a new format context for the output container format. */
166 if (!(*output_format_context = avformat_alloc_context())) {
167 fprintf(stderr, "Could not allocate output format context\n");
168 return AVERROR(ENOMEM);
169 }
170
171 /* Associate the output file (pointer) with the container format context. */
172 (*output_format_context)->pb = output_io_context;
173
174 /* Guess the desired container format based on the file extension. */
175 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
176 NULL))) {
177 fprintf(stderr, "Could not find output file format\n");
178 goto cleanup;
179 }
180
181 if (!((*output_format_context)->url = av_strdup(filename))) {
182 fprintf(stderr, "Could not allocate url.\n");
183 error = AVERROR(ENOMEM);
184 goto cleanup;
185 }
186
187 /* Find the encoder to be used by its name. */
188 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
189 fprintf(stderr, "Could not find an AAC encoder.\n");
190 goto cleanup;
191 }
192
193 /* Create a new audio stream in the output file container. */
194 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
195 fprintf(stderr, "Could not create new stream\n");
196 error = AVERROR(ENOMEM);
197 goto cleanup;
198 }
199
200 avctx = avcodec_alloc_context3(output_codec);
201 if (!avctx) {
202 fprintf(stderr, "Could not allocate an encoding context\n");
203 error = AVERROR(ENOMEM);
204 goto cleanup;
205 }
206
207 /* Set the basic encoder parameters.
208 * The input file's sample rate is used to avoid a sample rate conversion. */
210 avctx->sample_rate = input_codec_context->sample_rate;
211 avctx->sample_fmt = output_codec->sample_fmts[0];
212 avctx->bit_rate = OUTPUT_BIT_RATE;
213
214 /* Set the sample rate for the container. */
215 stream->time_base.den = input_codec_context->sample_rate;
216 stream->time_base.num = 1;
217
218 /* Some container formats (like MP4) require global headers to be present.
219 * Mark the encoder so that it behaves accordingly. */
220 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
222
223 /* Open the encoder for the audio stream to use it later. */
224 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
225 fprintf(stderr, "Could not open output codec (error '%s')\n",
226 av_err2str(error));
227 goto cleanup;
228 }
229
230 error = avcodec_parameters_from_context(stream->codecpar, avctx);
231 if (error < 0) {
232 fprintf(stderr, "Could not initialize stream parameters\n");
233 goto cleanup;
234 }
235
236 /* Save the encoder context for easier access later. */
237 *output_codec_context = avctx;
238
239 return 0;
240
241cleanup:
242 avcodec_free_context(&avctx);
243 avio_closep(&(*output_format_context)->pb);
244 avformat_free_context(*output_format_context);
245 *output_format_context = NULL;
246 return error < 0 ? error : AVERROR_EXIT;
247}
248
249/**
250 * Initialize one data packet for reading or writing.
251 * @param[out] packet Packet to be initialized
252 * @return Error code (0 if successful)
253 */
254static int init_packet(AVPacket **packet)
255{
256 if (!(*packet = av_packet_alloc())) {
257 fprintf(stderr, "Could not allocate packet\n");
258 return AVERROR(ENOMEM);
259 }
260 return 0;
261}
262
263/**
264 * Initialize one audio frame for reading from the input file.
265 * @param[out] frame Frame to be initialized
266 * @return Error code (0 if successful)
267 */
269{
270 if (!(*frame = av_frame_alloc())) {
271 fprintf(stderr, "Could not allocate input frame\n");
272 return AVERROR(ENOMEM);
273 }
274 return 0;
275}
276
277/**
278 * Initialize the audio resampler based on the input and output codec settings.
279 * If the input and output sample formats differ, a conversion is required
280 * libswresample takes care of this, but requires initialization.
281 * @param input_codec_context Codec context of the input file
282 * @param output_codec_context Codec context of the output file
283 * @param[out] resample_context Resample context for the required conversion
284 * @return Error code (0 if successful)
285 */
286static int init_resampler(AVCodecContext *input_codec_context,
287 AVCodecContext *output_codec_context,
288 SwrContext **resample_context)
289{
290 int error;
291
292 /*
293 * Create a resampler context for the conversion.
294 * Set the conversion parameters.
295 */
296 error = swr_alloc_set_opts2(resample_context,
297 &output_codec_context->ch_layout,
298 output_codec_context->sample_fmt,
299 output_codec_context->sample_rate,
300 &input_codec_context->ch_layout,
301 input_codec_context->sample_fmt,
302 input_codec_context->sample_rate,
303 0, NULL);
304 if (error < 0) {
305 fprintf(stderr, "Could not allocate resample context\n");
306 return error;
307 }
308 /*
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
312 */
313 av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
314
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);
319 return error;
320 }
321 return 0;
322}
323
324/**
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)
329 */
330static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
331{
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->ch_layout.nb_channels, 1))) {
335 fprintf(stderr, "Could not allocate FIFO\n");
336 return AVERROR(ENOMEM);
337 }
338 return 0;
339}
340
341/**
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)
345 */
346static int write_output_file_header(AVFormatContext *output_format_context)
347{
348 int error;
349 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
350 fprintf(stderr, "Could not write output file header (error '%s')\n",
351 av_err2str(error));
352 return error;
353 }
354 return 0;
355}
356
357/**
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)
369 */
371 AVFormatContext *input_format_context,
372 AVCodecContext *input_codec_context,
373 int *data_present, int *finished)
374{
375 /* Packet used for temporary storage. */
376 AVPacket *input_packet;
377 int error;
378
379 error = init_packet(&input_packet);
380 if (error < 0)
381 return error;
382
383 *data_present = 0;
384 *finished = 0;
385 /* Read one audio frame from the input file into a temporary packet. */
386 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
387 /* If we are at the end of the file, flush the decoder below. */
388 if (error == AVERROR_EOF)
389 *finished = 1;
390 else {
391 fprintf(stderr, "Could not read frame (error '%s')\n",
392 av_err2str(error));
393 goto cleanup;
394 }
395 }
396
397 /* Send the audio frame stored in the temporary packet to the decoder.
398 * The input audio stream decoder is used to do this. */
399 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
400 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
401 av_err2str(error));
402 goto cleanup;
403 }
404
405 /* Receive one frame from the decoder. */
406 error = avcodec_receive_frame(input_codec_context, frame);
407 /* If the decoder asks for more data to be able to decode a frame,
408 * return indicating that no data is present. */
409 if (error == AVERROR(EAGAIN)) {
410 error = 0;
411 goto cleanup;
412 /* If the end of the input file is reached, stop decoding. */
413 } else if (error == AVERROR_EOF) {
414 *finished = 1;
415 error = 0;
416 goto cleanup;
417 } else if (error < 0) {
418 fprintf(stderr, "Could not decode frame (error '%s')\n",
419 av_err2str(error));
420 goto cleanup;
421 /* Default case: Return decoded data. */
422 } else {
423 *data_present = 1;
424 goto cleanup;
425 }
426
427cleanup:
428 av_packet_free(&input_packet);
429 return error;
430}
431
432/**
433 * Initialize a temporary storage for the specified number of audio samples.
434 * The conversion requires temporary storage due to the different format.
435 * The number of audio samples to be allocated is specified in frame_size.
436 * @param[out] converted_input_samples Array of converted samples. The
437 * dimensions are reference, channel
438 * (for multi-channel audio), sample.
439 * @param output_codec_context Codec context of the output file
440 * @param frame_size Number of samples to be converted in
441 * each round
442 * @return Error code (0 if successful)
443 */
444static int init_converted_samples(uint8_t ***converted_input_samples,
445 AVCodecContext *output_codec_context,
446 int frame_size)
447{
448 int error;
449
450 /* Allocate as many pointers as there are audio channels.
451 * Each pointer will point to the audio samples of the corresponding
452 * channels (although it may be NULL for interleaved formats).
453 * Allocate memory for the samples of all channels in one consecutive
454 * block for convenience. */
455 if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
456 output_codec_context->ch_layout.nb_channels,
457 frame_size,
458 output_codec_context->sample_fmt, 0)) < 0) {
459 fprintf(stderr,
460 "Could not allocate converted input samples (error '%s')\n",
461 av_err2str(error));
462 return error;
463 }
464 return 0;
465}
466
467/**
468 * Convert the input audio samples into the output sample format.
469 * The conversion happens on a per-frame basis, the size of which is
470 * specified by frame_size.
471 * @param input_data Samples to be decoded. The dimensions are
472 * channel (for multi-channel audio), sample.
473 * @param[out] converted_data Converted samples. The dimensions are channel
474 * (for multi-channel audio), sample.
475 * @param frame_size Number of samples to be converted
476 * @param resample_context Resample context for the conversion
477 * @return Error code (0 if successful)
478 */
479static int convert_samples(const uint8_t **input_data,
480 uint8_t **converted_data, const int frame_size,
481 SwrContext *resample_context)
482{
483 int error;
484
485 /* Convert the samples using the resampler. */
486 if ((error = swr_convert(resample_context,
487 converted_data, frame_size,
488 input_data , frame_size)) < 0) {
489 fprintf(stderr, "Could not convert input samples (error '%s')\n",
490 av_err2str(error));
491 return error;
492 }
493
494 return 0;
495}
496
497/**
498 * Add converted input audio samples to the FIFO buffer for later processing.
499 * @param fifo Buffer to add the samples to
500 * @param converted_input_samples Samples to be added. The dimensions are channel
501 * (for multi-channel audio), sample.
502 * @param frame_size Number of samples to be converted
503 * @return Error code (0 if successful)
504 */
506 uint8_t **converted_input_samples,
507 const int frame_size)
508{
509 int error;
510
511 /* Make the FIFO as large as it needs to be to hold both,
512 * the old and the new samples. */
513 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
514 fprintf(stderr, "Could not reallocate FIFO\n");
515 return error;
516 }
517
518 /* Store the new samples in the FIFO buffer. */
519 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
520 frame_size) < frame_size) {
521 fprintf(stderr, "Could not write data to FIFO\n");
522 return AVERROR_EXIT;
523 }
524 return 0;
525}
526
527/**
528 * Read one audio frame from the input file, decode, convert and store
529 * it in the FIFO buffer.
530 * @param fifo Buffer used for temporary storage
531 * @param input_format_context Format context of the input file
532 * @param input_codec_context Codec context of the input file
533 * @param output_codec_context Codec context of the output file
534 * @param resampler_context Resample context for the conversion
535 * @param[out] finished Indicates whether the end of file has
536 * been reached and all data has been
537 * decoded. If this flag is false,
538 * there is more data to be decoded,
539 * i.e., this function has to be called
540 * again.
541 * @return Error code (0 if successful)
542 */
544 AVFormatContext *input_format_context,
545 AVCodecContext *input_codec_context,
546 AVCodecContext *output_codec_context,
547 SwrContext *resampler_context,
548 int *finished)
549{
550 /* Temporary storage of the input samples of the frame read from the file. */
551 AVFrame *input_frame = NULL;
552 /* Temporary storage for the converted input samples. */
553 uint8_t **converted_input_samples = NULL;
554 int data_present;
555 int ret = AVERROR_EXIT;
556
557 /* Initialize temporary storage for one input frame. */
558 if (init_input_frame(&input_frame))
559 goto cleanup;
560 /* Decode one frame worth of audio samples. */
561 if (decode_audio_frame(input_frame, input_format_context,
562 input_codec_context, &data_present, finished))
563 goto cleanup;
564 /* If we are at the end of the file and there are no more samples
565 * in the decoder which are delayed, we are actually finished.
566 * This must not be treated as an error. */
567 if (*finished) {
568 ret = 0;
569 goto cleanup;
570 }
571 /* If there is decoded data, convert and store it. */
572 if (data_present) {
573 /* Initialize the temporary storage for the converted input samples. */
574 if (init_converted_samples(&converted_input_samples, output_codec_context,
575 input_frame->nb_samples))
576 goto cleanup;
577
578 /* Convert the input samples to the desired output sample format.
579 * This requires a temporary storage provided by converted_input_samples. */
580 if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
581 input_frame->nb_samples, resampler_context))
582 goto cleanup;
583
584 /* Add the converted input samples to the FIFO buffer for later processing. */
585 if (add_samples_to_fifo(fifo, converted_input_samples,
586 input_frame->nb_samples))
587 goto cleanup;
588 ret = 0;
589 }
590 ret = 0;
591
592cleanup:
593 if (converted_input_samples)
594 av_freep(&converted_input_samples[0]);
595 av_freep(&converted_input_samples);
596 av_frame_free(&input_frame);
597
598 return ret;
599}
600
601/**
602 * Initialize one input frame for writing to the output file.
603 * The frame will be exactly frame_size samples large.
604 * @param[out] frame Frame to be initialized
605 * @param output_codec_context Codec context of the output file
606 * @param frame_size Size of the frame
607 * @return Error code (0 if successful)
608 */
610 AVCodecContext *output_codec_context,
611 int frame_size)
612{
613 int error;
614
615 /* Create a new frame to store the audio samples. */
616 if (!(*frame = av_frame_alloc())) {
617 fprintf(stderr, "Could not allocate output frame\n");
618 return AVERROR_EXIT;
619 }
620
621 /* Set the frame's parameters, especially its size and format.
622 * av_frame_get_buffer needs this to allocate memory for the
623 * audio samples of the frame.
624 * Default channel layouts based on the number of channels
625 * are assumed for simplicity. */
626 (*frame)->nb_samples = frame_size;
627 av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
628 (*frame)->format = output_codec_context->sample_fmt;
629 (*frame)->sample_rate = output_codec_context->sample_rate;
630
631 /* Allocate the samples of the created frame. This call will make
632 * sure that the audio frame can hold as many samples as specified. */
633 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
634 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
635 av_err2str(error));
637 return error;
638 }
639
640 return 0;
641}
642
643/* Global timestamp for the audio frames. */
644static int64_t pts = 0;
645
646/**
647 * Encode one frame worth of audio to the output file.
648 * @param frame Samples to be encoded
649 * @param output_format_context Format context of the output file
650 * @param output_codec_context Codec context of the output file
651 * @param[out] data_present Indicates whether data has been
652 * encoded
653 * @return Error code (0 if successful)
654 */
656 AVFormatContext *output_format_context,
657 AVCodecContext *output_codec_context,
658 int *data_present)
659{
660 /* Packet used for temporary storage. */
661 AVPacket *output_packet;
662 int error;
663
664 error = init_packet(&output_packet);
665 if (error < 0)
666 return error;
667
668 /* Set a timestamp based on the sample rate for the container. */
669 if (frame) {
670 frame->pts = pts;
671 pts += frame->nb_samples;
672 }
673
674 *data_present = 0;
675 /* Send the audio frame stored in the temporary packet to the encoder.
676 * The output audio stream encoder is used to do this. */
677 error = avcodec_send_frame(output_codec_context, frame);
678 /* Check for errors, but proceed with fetching encoded samples if the
679 * encoder signals that it has nothing more to encode. */
680 if (error < 0 && error != AVERROR_EOF) {
681 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
682 av_err2str(error));
683 goto cleanup;
684 }
685
686 /* Receive one encoded frame from the encoder. */
687 error = avcodec_receive_packet(output_codec_context, output_packet);
688 /* If the encoder asks for more data to be able to provide an
689 * encoded frame, return indicating that no data is present. */
690 if (error == AVERROR(EAGAIN)) {
691 error = 0;
692 goto cleanup;
693 /* If the last frame has been encoded, stop encoding. */
694 } else if (error == AVERROR_EOF) {
695 error = 0;
696 goto cleanup;
697 } else if (error < 0) {
698 fprintf(stderr, "Could not encode frame (error '%s')\n",
699 av_err2str(error));
700 goto cleanup;
701 /* Default case: Return encoded data. */
702 } else {
703 *data_present = 1;
704 }
705
706 /* Write one audio frame from the temporary packet to the output file. */
707 if (*data_present &&
708 (error = av_write_frame(output_format_context, output_packet)) < 0) {
709 fprintf(stderr, "Could not write frame (error '%s')\n",
710 av_err2str(error));
711 goto cleanup;
712 }
713
714cleanup:
715 av_packet_free(&output_packet);
716 return error;
717}
718
719/**
720 * Load one audio frame from the FIFO buffer, encode and write it to the
721 * output file.
722 * @param fifo Buffer used for temporary storage
723 * @param output_format_context Format context of the output file
724 * @param output_codec_context Codec context of the output file
725 * @return Error code (0 if successful)
726 */
728 AVFormatContext *output_format_context,
729 AVCodecContext *output_codec_context)
730{
731 /* Temporary storage of the output samples of the frame written to the file. */
732 AVFrame *output_frame;
733 /* Use the maximum number of possible samples per frame.
734 * If there is less than the maximum possible frame size in the FIFO
735 * buffer use this number. Otherwise, use the maximum possible frame size. */
736 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
737 output_codec_context->frame_size);
738 int data_written;
739
740 /* Initialize temporary storage for one output frame. */
741 if (init_output_frame(&output_frame, output_codec_context, frame_size))
742 return AVERROR_EXIT;
743
744 /* Read as many samples from the FIFO buffer as required to fill the frame.
745 * The samples are stored in the frame temporarily. */
746 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
747 fprintf(stderr, "Could not read data from FIFO\n");
748 av_frame_free(&output_frame);
749 return AVERROR_EXIT;
750 }
751
752 /* Encode one frame worth of audio samples. */
753 if (encode_audio_frame(output_frame, output_format_context,
754 output_codec_context, &data_written)) {
755 av_frame_free(&output_frame);
756 return AVERROR_EXIT;
757 }
758 av_frame_free(&output_frame);
759 return 0;
760}
761
762/**
763 * Write the trailer of the output file container.
764 * @param output_format_context Format context of the output file
765 * @return Error code (0 if successful)
766 */
767static int write_output_file_trailer(AVFormatContext *output_format_context)
768{
769 int error;
770 if ((error = av_write_trailer(output_format_context)) < 0) {
771 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
772 av_err2str(error));
773 return error;
774 }
775 return 0;
776}
777
778int main(int argc, char **argv)
779{
780 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
781 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
782 SwrContext *resample_context = NULL;
783 AVAudioFifo *fifo = NULL;
784 int ret = AVERROR_EXIT;
785
786 if (argc != 3) {
787 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
788 exit(1);
789 }
790
791 /* Open the input file for reading. */
792 if (open_input_file(argv[1], &input_format_context,
793 &input_codec_context))
794 goto cleanup;
795 /* Open the output file for writing. */
796 if (open_output_file(argv[2], input_codec_context,
797 &output_format_context, &output_codec_context))
798 goto cleanup;
799 /* Initialize the resampler to be able to convert audio sample formats. */
800 if (init_resampler(input_codec_context, output_codec_context,
801 &resample_context))
802 goto cleanup;
803 /* Initialize the FIFO buffer to store audio samples to be encoded. */
804 if (init_fifo(&fifo, output_codec_context))
805 goto cleanup;
806 /* Write the header of the output file container. */
807 if (write_output_file_header(output_format_context))
808 goto cleanup;
809
810 /* Loop as long as we have input samples to read or output samples
811 * to write; abort as soon as we have neither. */
812 while (1) {
813 /* Use the encoder's desired frame size for processing. */
814 const int output_frame_size = output_codec_context->frame_size;
815 int finished = 0;
816
817 /* Make sure that there is one frame worth of samples in the FIFO
818 * buffer so that the encoder can do its work.
819 * Since the decoder's and the encoder's frame size may differ, we
820 * need to FIFO buffer to store as many frames worth of input samples
821 * that they make up at least one frame worth of output samples. */
822 while (av_audio_fifo_size(fifo) < output_frame_size) {
823 /* Decode one frame worth of audio samples, convert it to the
824 * output sample format and put it into the FIFO buffer. */
825 if (read_decode_convert_and_store(fifo, input_format_context,
826 input_codec_context,
827 output_codec_context,
828 resample_context, &finished))
829 goto cleanup;
830
831 /* If we are at the end of the input file, we continue
832 * encoding the remaining audio samples to the output file. */
833 if (finished)
834 break;
835 }
836
837 /* If we have enough samples for the encoder, we encode them.
838 * At the end of the file, we pass the remaining samples to
839 * the encoder. */
840 while (av_audio_fifo_size(fifo) >= output_frame_size ||
841 (finished && av_audio_fifo_size(fifo) > 0))
842 /* Take one frame worth of audio samples from the FIFO buffer,
843 * encode it and write it to the output file. */
844 if (load_encode_and_write(fifo, output_format_context,
845 output_codec_context))
846 goto cleanup;
847
848 /* If we are at the end of the input file and have encoded
849 * all remaining samples, we can exit this loop and finish. */
850 if (finished) {
851 int data_written;
852 /* Flush the encoder as it may have delayed frames. */
853 do {
854 if (encode_audio_frame(NULL, output_format_context,
855 output_codec_context, &data_written))
856 goto cleanup;
857 } while (data_written);
858 break;
859 }
860 }
861
862 /* Write the trailer of the output file container. */
863 if (write_output_file_trailer(output_format_context))
864 goto cleanup;
865 ret = 0;
866
867cleanup:
868 if (fifo)
869 av_audio_fifo_free(fifo);
870 swr_free(&resample_context);
871 if (output_codec_context)
872 avcodec_free_context(&output_codec_context);
873 if (output_format_context) {
874 avio_closep(&output_format_context->pb);
875 avformat_free_context(output_format_context);
876 }
877 if (input_codec_context)
878 avcodec_free_context(&input_codec_context);
879 if (input_format_context)
880 avformat_close_input(&input_format_context);
881
882 return ret;
883}
Audio FIFO Buffer.
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition avassert.h:40
Libavcodec external API header.
Main libavformat public API header.
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition avformat.h:478
Buffered I/O operations.
int avio_open(AVIOContext **s, const char *url, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
#define AVIO_FLAG_WRITE
write-only
Definition avio.h:618
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
int main(int argc, char *argv[])
Public libavutil channel layout APIs header.
static int open_input_file(const char *filename)
static AVFrame * frame
reference-counted frame API
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
int avcodec_parameters_from_context(struct AVCodecParameters *par, const AVCodecContext *codec)
Fill the parameters struct based on the values from the supplied codec context.
const AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
int avcodec_parameters_to_context(AVCodecContext *codec, const struct AVCodecParameters *par)
Fill the codec context based on the values from the supplied codec parameters.
const AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition avcodec.h:338
void avcodec_free_context(AVCodecContext **avctx)
Free the codec context and everything associated with it and write NULL to the provided pointer.
@ AV_CODEC_ID_AAC
Definition codec_id.h:442
int avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Return decoded output data from a decoder or encoder (when the AV_CODEC_FLAG_RECON_FRAME flag is used...
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
AVStream * avformat_new_stream(AVFormatContext *s, const struct AVCodec *c)
Add a new stream to a media file.
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
int avformat_open_input(AVFormatContext **ps, const char *url, const AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
av_warn_unused_result int avformat_write_header(AVFormatContext *s, AVDictionary **options)
Allocate the stream private data and write the stream header to an output media file.
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
const AVOutputFormat * av_guess_format(const char *short_name, const char *filename, const char *mime_type)
Return the output format in the list of registered output formats which best matches the provided par...
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
void av_channel_layout_default(AVChannelLayout *ch_layout, int nb_channels)
Get the default channel layout for a given number of channels.
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
av_warn_unused_result int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
struct AVAudioFifo AVAudioFifo
Context for an Audio FIFO Buffer.
Definition audio_fifo.h:48
int av_audio_fifo_write(AVAudioFifo *af, void *const *data, int nb_samples)
Write data to an AVAudioFifo.
int av_audio_fifo_read(AVAudioFifo *af, void *const *data, int nb_samples)
Read data from an AVAudioFifo.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition error.h:58
#define AVERROR_EOF
End of file.
Definition error.h:57
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition error.h:122
#define AVERROR(e)
Definition error.h:45
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
void av_freep(void *ptr)
Free a memory block which has been allocated with a function of av_malloc() or av_realloc() family,...
char * av_strdup(const char *s) av_malloc_attrib
Duplicate a string.
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a data pointers array, samples buffer for nb_samples samples, and fill data pointers and lin...
struct SwrContext SwrContext
The libswresample context.
Definition swresample.h:189
int swr_alloc_set_opts2(struct SwrContext **ps, const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
void swr_free(struct SwrContext **s)
Free the given SwrContext and set the pointer to NULL.
int swr_convert(struct SwrContext *s, uint8_t *const *out, int out_count, const uint8_t *const *in, int in_count)
Convert audio.
int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
#define FFMIN(a, b)
Definition macros.h:49
Memory handling functions.
AVOptions.
int nb_channels
Number of channels in this layout.
main external API structure.
Definition avcodec.h:451
AVChannelLayout ch_layout
Audio channel layout.
Definition avcodec.h:1071
enum AVSampleFormat sample_fmt
audio sample format
Definition avcodec.h:1063
AVRational pkt_timebase
Timebase in which pkt_dts/pts and AVPacket.dts/pts are expressed.
Definition avcodec.h:557
int64_t bit_rate
the average bitrate
Definition avcodec.h:501
int sample_rate
samples per second
Definition avcodec.h:1056
int flags
AV_CODEC_FLAG_*.
Definition avcodec.h:508
int frame_size
Number of samples per channel in an audio frame.
Definition avcodec.h:1083
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition codec_par.h:55
AVCodec.
Definition codec.h:187
attribute_deprecated enum AVSampleFormat * sample_fmts
Definition codec.h:219
Format I/O context.
Definition avformat.h:1287
This structure describes decoded (raw) audio or video data.
Definition frame.h:389
int nb_samples
number of audio samples (per channel) described by this frame
Definition frame.h:469
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition frame.h:501
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition frame.h:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition frame.h:450
Bytestream IO Context.
Definition avio.h:160
This structure stores compressed data.
Definition packet.h:516
int num
Numerator.
Definition rational.h:59
int den
Denominator.
Definition rational.h:60
Stream structure.
Definition avformat.h:748
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition avformat.h:771
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition avformat.h:787
libswresample public header
static int add_samples_to_fifo(AVAudioFifo *fifo, uint8_t **converted_input_samples, const int frame_size)
Add converted input audio samples to the FIFO buffer for later processing.
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, AVCodecContext *output_codec_context, int *data_present)
Encode one frame worth of audio to the output file.
#define OUTPUT_BIT_RATE
#define OUTPUT_CHANNELS
static int64_t pts
static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished)
Decode one audio frame from the input file.
static int open_output_file(const char *filename, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context)
Open an output file and the required encoder.
static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context)
Initialize the audio resampler based on the input and output codec settings.
static int init_packet(AVPacket **packet)
Initialize one data packet for reading or writing.
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
Initialize one input frame for writing to the output file.
static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resampler_context, int *finished)
Read one audio frame from the input file, decode, convert and store it in the FIFO buffer.
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context)
Load one audio frame from the FIFO buffer, encode and write it to the output file.
static int write_output_file_trailer(AVFormatContext *output_format_context)
Write the trailer of the output file container.
static int convert_samples(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, SwrContext *resample_context)
Convert the input audio samples into the output sample format.
static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size)
Initialize a temporary storage for the specified number of audio samples.