FFmpeg: doc/examples/transcode_aac.c Source File

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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>
33#include <libavformat/avformat.h>
34#include <libavformat/avio.h>
35
36#include <libavcodec/avcodec.h>
37
38#include <libavutil/audio_fifo.h>
39#include <libavutil/avassert.h>
40#include <libavutil/avstring.h>
41#include <libavutil/channel_layout.h>
42#include <libavutil/frame.h>
43#include <libavutil/opt.h>
44
45#include <libswresample/swresample.h>
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 */
59 static 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",
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. */
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",
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 */
146 static 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 const void *sample_fmts;
156 int error;
157
158 /* Open the output file to write to it. */
159 if ((error = avio_open(&output_io_context, filename,
160 AVIO_FLAG_WRITE)) < 0) {
161 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
162 filename, av_err2str(error));
163 return error;
164 }
165
166 /* Create a new format context for the output container format. */
167 if (!(*output_format_context = avformat_alloc_context())) {
168 fprintf(stderr, "Could not allocate output format context\n");
169 return AVERROR(ENOMEM);
170 }
171
172 /* Associate the output file (pointer) with the container format context. */
173 (*output_format_context)->pb = output_io_context;
174
175 /* Guess the desired container format based on the file extension. */
176 if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
177 NULL))) {
178 fprintf(stderr, "Could not find output file format\n");
179 goto cleanup;
180 }
181
182 if (!((*output_format_context)->url = av_strdup(filename))) {
183 fprintf(stderr, "Could not allocate url.\n");
184 error = AVERROR(ENOMEM);
185 goto cleanup;
186 }
187
188 /* Find the encoder to be used by its name. */
189 if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
190 fprintf(stderr, "Could not find an AAC encoder.\n");
191 goto cleanup;
192 }
193
194 /* Create a new audio stream in the output file container. */
195 if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
196 fprintf(stderr, "Could not create new stream\n");
197 error = AVERROR(ENOMEM);
198 goto cleanup;
199 }
200
201 avctx = avcodec_alloc_context3(output_codec);
202 if (!avctx) {
203 fprintf(stderr, "Could not allocate an encoding context\n");
204 error = AVERROR(ENOMEM);
205 goto cleanup;
206 }
207
208 /* Set the basic encoder parameters.
209 * The input file's sample rate is used to avoid a sample rate conversion. */
212 0, &sample_fmts, NULL);
213 av_assert0(error >= 0);
214 avctx->sample_fmt = sample_fmts ? *(const enum AVSampleFormat*)sample_fmts
215 : input_codec_context->sample_fmt;
216 avctx->sample_rate = input_codec_context->sample_rate;
217 avctx->bit_rate = OUTPUT_BIT_RATE;
218
219 /* Set the sample rate for the container. */
220 stream->time_base.den = input_codec_context->sample_rate;
221 stream->time_base.num = 1;
222
223 /* Some container formats (like MP4) require global headers to be present.
224 * Mark the encoder so that it behaves accordingly. */
225 if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
227
228 /* Open the encoder for the audio stream to use it later. */
229 if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
230 fprintf(stderr, "Could not open output codec (error '%s')\n",
232 goto cleanup;
233 }
234
236 if (error < 0) {
237 fprintf(stderr, "Could not initialize stream parameters\n");
238 goto cleanup;
239 }
240
241 /* Save the encoder context for easier access later. */
242 *output_codec_context = avctx;
243
244 return 0;
245
246cleanup:
247 avcodec_free_context(&avctx);
248 avio_closep(&(*output_format_context)->pb);
249 avformat_free_context(*output_format_context);
250 *output_format_context = NULL;
251 return error < 0 ? error : AVERROR_EXIT;
252}
253
254/**
255 * Initialize one data packet for reading or writing.
256 * @param[out] packet Packet to be initialized
257 * @return Error code (0 if successful)
258 */
259 static int init_packet(AVPacket **packet)
260{
261 if (!(*packet = av_packet_alloc())) {
262 fprintf(stderr, "Could not allocate packet\n");
263 return AVERROR(ENOMEM);
264 }
265 return 0;
266}
267
268/**
269 * Initialize one audio frame for reading from the input file.
270 * @param[out] frame Frame to be initialized
271 * @return Error code (0 if successful)
272 */
274{
275 if (!(*frame = av_frame_alloc())) {
276 fprintf(stderr, "Could not allocate input frame\n");
277 return AVERROR(ENOMEM);
278 }
279 return 0;
280}
281
282/**
283 * Initialize the audio resampler based on the input and output codec settings.
284 * If the input and output sample formats differ, a conversion is required
285 * libswresample takes care of this, but requires initialization.
286 * @param input_codec_context Codec context of the input file
287 * @param output_codec_context Codec context of the output file
288 * @param[out] resample_context Resample context for the required conversion
289 * @return Error code (0 if successful)
290 */
291 static int init_resampler(AVCodecContext *input_codec_context,
292 AVCodecContext *output_codec_context,
293 SwrContext **resample_context)
294{
295 int error;
296
297 /*
298 * Create a resampler context for the conversion.
299 * Set the conversion parameters.
300 */
301 error = swr_alloc_set_opts2(resample_context,
302 &output_codec_context->ch_layout,
303 output_codec_context->sample_fmt,
304 output_codec_context->sample_rate,
305 &input_codec_context->ch_layout,
306 input_codec_context->sample_fmt,
307 input_codec_context->sample_rate,
308 0, NULL);
309 if (error < 0) {
310 fprintf(stderr, "Could not allocate resample context\n");
311 return error;
312 }
313 /*
314 * Perform a sanity check so that the number of converted samples is
315 * not greater than the number of samples to be converted.
316 * If the sample rates differ, this case has to be handled differently
317 */
318 av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
319
320 /* Open the resampler with the specified parameters. */
321 if ((error = swr_init(*resample_context)) < 0) {
322 fprintf(stderr, "Could not open resample context\n");
323 swr_free(resample_context);
324 return error;
325 }
326 return 0;
327}
328
329/**
330 * Initialize a FIFO buffer for the audio samples to be encoded.
331 * @param[out] fifo Sample buffer
332 * @param output_codec_context Codec context of the output file
333 * @return Error code (0 if successful)
334 */
335 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
336{
337 /* Create the FIFO buffer based on the specified output sample format. */
338 if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
339 output_codec_context->ch_layout.nb_channels, 1))) {
340 fprintf(stderr, "Could not allocate FIFO\n");
341 return AVERROR(ENOMEM);
342 }
343 return 0;
344}
345
346/**
347 * Write the header of the output file container.
348 * @param output_format_context Format context of the output file
349 * @return Error code (0 if successful)
350 */
351 static int write_output_file_header(AVFormatContext *output_format_context)
352{
353 int error;
354 if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
355 fprintf(stderr, "Could not write output file header (error '%s')\n",
357 return error;
358 }
359 return 0;
360}
361
362/**
363 * Decode one audio frame from the input file.
364 * @param frame Audio frame to be decoded
365 * @param input_format_context Format context of the input file
366 * @param input_codec_context Codec context of the input file
367 * @param[out] data_present Indicates whether data has been decoded
368 * @param[out] finished Indicates whether the end of file has
369 * been reached and all data has been
370 * decoded. If this flag is false, there
371 * is more data to be decoded, i.e., this
372 * function has to be called again.
373 * @return Error code (0 if successful)
374 */
376 AVFormatContext *input_format_context,
377 AVCodecContext *input_codec_context,
378 int *data_present, int *finished)
379{
380 /* Packet used for temporary storage. */
381 AVPacket *input_packet;
382 int error;
383
384 error = init_packet(&input_packet);
385 if (error < 0)
386 return error;
387
388 *data_present = 0;
389 *finished = 0;
390 /* Read one audio frame from the input file into a temporary packet. */
391 if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
392 /* If we are at the end of the file, flush the decoder below. */
393 if (error == AVERROR_EOF)
394 *finished = 1;
395 else {
396 fprintf(stderr, "Could not read frame (error '%s')\n",
398 goto cleanup;
399 }
400 }
401
402 /* Send the audio frame stored in the temporary packet to the decoder.
403 * The input audio stream decoder is used to do this. */
404 if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
405 fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
407 goto cleanup;
408 }
409
410 /* Receive one frame from the decoder. */
411 error = avcodec_receive_frame(input_codec_context, frame);
412 /* If the decoder asks for more data to be able to decode a frame,
413 * return indicating that no data is present. */
414 if (error == AVERROR(EAGAIN)) {
415 error = 0;
416 goto cleanup;
417 /* If the end of the input file is reached, stop decoding. */
418 } else if (error == AVERROR_EOF) {
419 *finished = 1;
420 error = 0;
421 goto cleanup;
422 } else if (error < 0) {
423 fprintf(stderr, "Could not decode frame (error '%s')\n",
425 goto cleanup;
426 /* Default case: Return decoded data. */
427 } else {
428 *data_present = 1;
429 goto cleanup;
430 }
431
432cleanup:
433 av_packet_free(&input_packet);
434 return error;
435}
436
437/**
438 * Initialize a temporary storage for the specified number of audio samples.
439 * The conversion requires temporary storage due to the different format.
440 * The number of audio samples to be allocated is specified in frame_size.
441 * @param[out] converted_input_samples Array of converted samples. The
442 * dimensions are reference, channel
443 * (for multi-channel audio), sample.
444 * @param output_codec_context Codec context of the output file
445 * @param frame_size Number of samples to be converted in
446 * each round
447 * @return Error code (0 if successful)
448 */
449 static int init_converted_samples(uint8_t ***converted_input_samples,
450 AVCodecContext *output_codec_context,
451 int frame_size)
452{
453 int error;
454
455 /* Allocate as many pointers as there are audio channels.
456 * Each pointer will point to the audio samples of the corresponding
457 * channels (although it may be NULL for interleaved formats).
458 * Allocate memory for the samples of all channels in one consecutive
459 * block for convenience. */
460 if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
461 output_codec_context->ch_layout.nb_channels,
463 output_codec_context->sample_fmt, 0)) < 0) {
464 fprintf(stderr,
465 "Could not allocate converted input samples (error '%s')\n",
467 return error;
468 }
469 return 0;
470}
471
472/**
473 * Convert the input audio samples into the output sample format.
474 * The conversion happens on a per-frame basis, the size of which is
475 * specified by frame_size.
476 * @param input_data Samples to be decoded. The dimensions are
477 * channel (for multi-channel audio), sample.
478 * @param[out] converted_data Converted samples. The dimensions are channel
479 * (for multi-channel audio), sample.
480 * @param frame_size Number of samples to be converted
481 * @param resample_context Resample context for the conversion
482 * @return Error code (0 if successful)
483 */
484 static int convert_samples(const uint8_t **input_data,
485 uint8_t **converted_data, const int frame_size,
486 SwrContext *resample_context)
487{
488 int error;
489
490 /* Convert the samples using the resampler. */
491 if ((error = swr_convert(resample_context,
492 converted_data, frame_size,
493 input_data , frame_size)) < 0) {
494 fprintf(stderr, "Could not convert input samples (error '%s')\n",
496 return error;
497 }
498
499 return 0;
500}
501
502/**
503 * Add converted input audio samples to the FIFO buffer for later processing.
504 * @param fifo Buffer to add the samples to
505 * @param converted_input_samples Samples to be added. The dimensions are channel
506 * (for multi-channel audio), sample.
507 * @param frame_size Number of samples to be converted
508 * @return Error code (0 if successful)
509 */
511 uint8_t **converted_input_samples,
512 const int frame_size)
513{
514 int error;
515
516 /* Make the FIFO as large as it needs to be to hold both,
517 * the old and the new samples. */
518 if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
519 fprintf(stderr, "Could not reallocate FIFO\n");
520 return error;
521 }
522
523 /* Store the new samples in the FIFO buffer. */
524 if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
526 fprintf(stderr, "Could not write data to FIFO\n");
527 return AVERROR_EXIT;
528 }
529 return 0;
530}
531
532/**
533 * Read one audio frame from the input file, decode, convert and store
534 * it in the FIFO buffer.
535 * @param fifo Buffer used for temporary storage
536 * @param input_format_context Format context of the input file
537 * @param input_codec_context Codec context of the input file
538 * @param output_codec_context Codec context of the output file
539 * @param resampler_context Resample context for the conversion
540 * @param[out] finished Indicates whether the end of file has
541 * been reached and all data has been
542 * decoded. If this flag is false,
543 * there is more data to be decoded,
544 * i.e., this function has to be called
545 * again.
546 * @return Error code (0 if successful)
547 */
549 AVFormatContext *input_format_context,
550 AVCodecContext *input_codec_context,
551 AVCodecContext *output_codec_context,
552 SwrContext *resampler_context,
553 int *finished)
554{
555 /* Temporary storage of the input samples of the frame read from the file. */
556 AVFrame *input_frame = NULL;
557 /* Temporary storage for the converted input samples. */
558 uint8_t **converted_input_samples = NULL;
559 int data_present;
560 int ret = AVERROR_EXIT;
561
562 /* Initialize temporary storage for one input frame. */
563 if (init_input_frame(&input_frame))
564 goto cleanup;
565 /* Decode one frame worth of audio samples. */
566 if (decode_audio_frame(input_frame, input_format_context,
567 input_codec_context, &data_present, finished))
568 goto cleanup;
569 /* If we are at the end of the file and there are no more samples
570 * in the decoder which are delayed, we are actually finished.
571 * This must not be treated as an error. */
572 if (*finished) {
573 ret = 0;
574 goto cleanup;
575 }
576 /* If there is decoded data, convert and store it. */
577 if (data_present) {
578 /* Initialize the temporary storage for the converted input samples. */
579 if (init_converted_samples(&converted_input_samples, output_codec_context,
580 input_frame->nb_samples))
581 goto cleanup;
582
583 /* Convert the input samples to the desired output sample format.
584 * This requires a temporary storage provided by converted_input_samples. */
585 if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
586 input_frame->nb_samples, resampler_context))
587 goto cleanup;
588
589 /* Add the converted input samples to the FIFO buffer for later processing. */
590 if (add_samples_to_fifo(fifo, converted_input_samples,
591 input_frame->nb_samples))
592 goto cleanup;
593 ret = 0;
594 }
595 ret = 0;
596
597cleanup:
598 if (converted_input_samples)
599 av_freep(&converted_input_samples[0]);
600 av_freep(&converted_input_samples);
601 av_frame_free(&input_frame);
602
603 return ret;
604}
605
606/**
607 * Initialize one input frame for writing to the output file.
608 * The frame will be exactly frame_size samples large.
609 * @param[out] frame Frame to be initialized
610 * @param output_codec_context Codec context of the output file
611 * @param frame_size Size of the frame
612 * @return Error code (0 if successful)
613 */
615 AVCodecContext *output_codec_context,
616 int frame_size)
617{
618 int error;
619
620 /* Create a new frame to store the audio samples. */
621 if (!(*frame = av_frame_alloc())) {
622 fprintf(stderr, "Could not allocate output frame\n");
623 return AVERROR_EXIT;
624 }
625
626 /* Set the frame's parameters, especially its size and format.
627 * av_frame_get_buffer needs this to allocate memory for the
628 * audio samples of the frame.
629 * Default channel layouts based on the number of channels
630 * are assumed for simplicity. */
631 (*frame)->nb_samples = frame_size;
632 av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
633 (*frame)->format = output_codec_context->sample_fmt;
634 (*frame)->sample_rate = output_codec_context->sample_rate;
635
636 /* Allocate the samples of the created frame. This call will make
637 * sure that the audio frame can hold as many samples as specified. */
638 if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
639 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
642 return error;
643 }
644
645 return 0;
646}
647
648/* Global timestamp for the audio frames. */
649 static int64_t pts = 0;
650
651/**
652 * Encode one frame worth of audio to the output file.
653 * @param frame Samples to be encoded
654 * @param output_format_context Format context of the output file
655 * @param output_codec_context Codec context of the output file
656 * @param[out] data_present Indicates whether data has been
657 * encoded
658 * @return Error code (0 if successful)
659 */
661 AVFormatContext *output_format_context,
662 AVCodecContext *output_codec_context,
663 int *data_present)
664{
665 /* Packet used for temporary storage. */
667 int error;
668
670 if (error < 0)
671 return error;
672
673 /* Set a timestamp based on the sample rate for the container. */
674 if (frame) {
675 frame->pts = pts;
676 pts += frame->nb_samples;
677 }
678
679 *data_present = 0;
680 /* Send the audio frame stored in the temporary packet to the encoder.
681 * The output audio stream encoder is used to do this. */
682 error = avcodec_send_frame(output_codec_context, frame);
683 /* Check for errors, but proceed with fetching encoded samples if the
684 * encoder signals that it has nothing more to encode. */
685 if (error < 0 && error != AVERROR_EOF) {
686 fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
688 goto cleanup;
689 }
690
691 /* Receive one encoded frame from the encoder. */
692 error = avcodec_receive_packet(output_codec_context, output_packet);
693 /* If the encoder asks for more data to be able to provide an
694 * encoded frame, return indicating that no data is present. */
695 if (error == AVERROR(EAGAIN)) {
696 error = 0;
697 goto cleanup;
698 /* If the last frame has been encoded, stop encoding. */
699 } else if (error == AVERROR_EOF) {
700 error = 0;
701 goto cleanup;
702 } else if (error < 0) {
703 fprintf(stderr, "Could not encode frame (error '%s')\n",
705 goto cleanup;
706 /* Default case: Return encoded data. */
707 } else {
708 *data_present = 1;
709 }
710
711 /* Write one audio frame from the temporary packet to the output file. */
712 if (*data_present &&
713 (error = av_write_frame(output_format_context, output_packet)) < 0) {
714 fprintf(stderr, "Could not write frame (error '%s')\n",
716 goto cleanup;
717 }
718
719cleanup:
721 return error;
722}
723
724/**
725 * Load one audio frame from the FIFO buffer, encode and write it to the
726 * output file.
727 * @param fifo Buffer used for temporary storage
728 * @param output_format_context Format context of the output file
729 * @param output_codec_context Codec context of the output file
730 * @return Error code (0 if successful)
731 */
733 AVFormatContext *output_format_context,
734 AVCodecContext *output_codec_context)
735{
736 /* Temporary storage of the output samples of the frame written to the file. */
738 /* Use the maximum number of possible samples per frame.
739 * If there is less than the maximum possible frame size in the FIFO
740 * buffer use this number. Otherwise, use the maximum possible frame size. */
741 const int frame_size = FFMIN(av_audio_fifo_size(fifo),
742 output_codec_context->frame_size);
743 int data_written;
744
745 /* Initialize temporary storage for one output frame. */
746 if (init_output_frame(&output_frame, output_codec_context, frame_size))
747 return AVERROR_EXIT;
748
749 /* Read as many samples from the FIFO buffer as required to fill the frame.
750 * The samples are stored in the frame temporarily. */
751 if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
752 fprintf(stderr, "Could not read data from FIFO\n");
754 return AVERROR_EXIT;
755 }
756
757 /* Encode one frame worth of audio samples. */
758 if (encode_audio_frame(output_frame, output_format_context,
759 output_codec_context, &data_written)) {
761 return AVERROR_EXIT;
762 }
764 return 0;
765}
766
767/**
768 * Write the trailer of the output file container.
769 * @param output_format_context Format context of the output file
770 * @return Error code (0 if successful)
771 */
772 static int write_output_file_trailer(AVFormatContext *output_format_context)
773{
774 int error;
775 if ((error = av_write_trailer(output_format_context)) < 0) {
776 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
778 return error;
779 }
780 return 0;
781}
782
783 int main(int argc, char **argv)
784{
785 AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
786 AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
787 SwrContext *resample_context = NULL;
788 AVAudioFifo *fifo = NULL;
789 int ret = AVERROR_EXIT;
790
791 if (argc != 3) {
792 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
793 exit(1);
794 }
795
796 /* Open the input file for reading. */
797 if (open_input_file(argv[1], &input_format_context,
798 &input_codec_context))
799 goto cleanup;
800 /* Open the output file for writing. */
801 if (open_output_file(argv[2], input_codec_context,
802 &output_format_context, &output_codec_context))
803 goto cleanup;
804 /* Initialize the resampler to be able to convert audio sample formats. */
805 if (init_resampler(input_codec_context, output_codec_context,
806 &resample_context))
807 goto cleanup;
808 /* Initialize the FIFO buffer to store audio samples to be encoded. */
809 if (init_fifo(&fifo, output_codec_context))
810 goto cleanup;
811 /* Write the header of the output file container. */
812 if (write_output_file_header(output_format_context))
813 goto cleanup;
814
815 /* Loop as long as we have input samples to read or output samples
816 * to write; abort as soon as we have neither. */
817 while (1) {
818 /* Use the encoder's desired frame size for processing. */
819 const int output_frame_size = output_codec_context->frame_size;
820 int finished = 0;
821
822 /* Make sure that there is one frame worth of samples in the FIFO
823 * buffer so that the encoder can do its work.
824 * Since the decoder's and the encoder's frame size may differ, we
825 * need to FIFO buffer to store as many frames worth of input samples
826 * that they make up at least one frame worth of output samples. */
827 while (av_audio_fifo_size(fifo) < output_frame_size) {
828 /* Decode one frame worth of audio samples, convert it to the
829 * output sample format and put it into the FIFO buffer. */
830 if (read_decode_convert_and_store(fifo, input_format_context,
831 input_codec_context,
832 output_codec_context,
833 resample_context, &finished))
834 goto cleanup;
835
836 /* If we are at the end of the input file, we continue
837 * encoding the remaining audio samples to the output file. */
838 if (finished)
839 break;
840 }
841
842 /* If we have enough samples for the encoder, we encode them.
843 * At the end of the file, we pass the remaining samples to
844 * the encoder. */
845 while (av_audio_fifo_size(fifo) >= output_frame_size ||
846 (finished && av_audio_fifo_size(fifo) > 0))
847 /* Take one frame worth of audio samples from the FIFO buffer,
848 * encode it and write it to the output file. */
849 if (load_encode_and_write(fifo, output_format_context,
850 output_codec_context))
851 goto cleanup;
852
853 /* If we are at the end of the input file and have encoded
854 * all remaining samples, we can exit this loop and finish. */
855 if (finished) {
856 int data_written;
857 /* Flush the encoder as it may have delayed frames. */
858 do {
859 if (encode_audio_frame(NULL, output_format_context,
860 output_codec_context, &data_written))
861 goto cleanup;
862 } while (data_written);
863 break;
864 }
865 }
866
867 /* Write the trailer of the output file container. */
868 if (write_output_file_trailer(output_format_context))
869 goto cleanup;
870 ret = 0;
871
872cleanup:
873 if (fifo)
874 av_audio_fifo_free(fifo);
875 swr_free(&resample_context);
876 if (output_codec_context)
877 avcodec_free_context(&output_codec_context);
878 if (output_format_context) {
879 avio_closep(&output_format_context->pb);
880 avformat_free_context(output_format_context);
881 }
882 if (input_codec_context)
883 avcodec_free_context(&input_codec_context);
884 if (input_format_context)
885 avformat_close_input(&input_format_context);
886
887 return ret;
888}
static enum AVSampleFormat sample_fmts[]
Definition adpcmenc.c:933
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:42
Libavcodec external API header.
Main libavformat public API header.
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition avformat.h:497
int avio_open(AVIOContext **s, const char *filename, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
Definition avio.c:565
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
Definition avio.c:717
Buffered I/O operations.
#define AVIO_FLAG_WRITE
write-only
Definition avio.h:618
Public libavutil channel layout APIs header.
int avcodec_parameters_from_context(AVCodecParameters *par, const AVCodecContext *codec)
Definition codec_par.c:138
int avcodec_parameters_to_context(AVCodecContext *codec, const AVCodecParameters *par)
Definition codec_par.c:206
#define NULL
Definition coverity.c:32
long long int64_t
Definition coverity.c:34
static int open_input_file(const char *filename)
static AVFrame * frame
Definition demux_decode.c:54
int main
Definition dovi_rpuenc.c:38
static av_cold void cleanup(FlashSV2Context *s)
Definition flashsv2enc.c:130
reference-counted frame API
static const uint8_t frame_size[4]
Definition g723_1.h:222
int attribute_align_arg avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition avcodec.c:144
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition options.c:149
const AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition allcodecs.c:991
const AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition allcodecs.c:986
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition avcodec.h:318
void avcodec_free_context(AVCodecContext **avctx)
Free the codec context and everything associated with it and write NULL to the provided pointer.
Definition options.c:164
@ AV_CODEC_ID_AAC
Definition codec_id.h:455
int avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Alias for avcodec_receive_frame_flags(avctx, frame, 0).
Definition avcodec.c:720
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition decode.c:730
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
Definition encode.c:578
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
Definition encode.c:545
int avcodec_get_supported_config(const AVCodecContext *avctx, const AVCodec *codec, enum AVCodecConfig config, unsigned flags, const void **out, int *out_num)
Retrieve a list of all supported values for a given configuration type.
Definition avcodec.c:818
@ AV_CODEC_CONFIG_SAMPLE_FORMAT
AVSampleFormat, terminated by AV_SAMPLE_FMT_NONE.
Definition avcodec.h:2587
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition packet.c:74
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition packet.c:63
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.
Definition options.c:165
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition avformat.c:150
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition demux.c:1588
int avformat_open_input(AVFormatContext **ps, const char *url, const AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition demux.c:231
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
Definition demux.c:2606
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition demux.c:377
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.
Definition mux.c:467
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition mux.c:1238
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...
Definition format.c:79
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition mux.c:1176
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.
Definition audio_fifo.c:62
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition audio_fifo.c:99
int av_audio_fifo_write(AVAudioFifo *af, void *const *data, int nb_samples)
Write data to an AVAudioFifo.
Definition audio_fifo.c:119
int av_audio_fifo_read(AVAudioFifo *af, void *const *data, int nb_samples)
Read data from an AVAudioFifo.
Definition audio_fifo.c:175
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition audio_fifo.c:48
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition audio_fifo.c:222
#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.
Definition frame.c:206
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition frame.c:64
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition frame.c:52
AVSampleFormat
Audio sample formats.
Definition samplefmt.h:55
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...
Definition samplefmt.c:208
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.
Definition swresample.c:54
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition swresample.c:137
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *const *out_arg, int out_count, const uint8_t *const *in_arg, int in_count)
Convert audio.
Definition swresample.c:743
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition swresample.c:156
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition h264dec.c:912
#define FFMIN(a, b)
Definition macros.h:49
Memory handling functions.
static void input_data(MLPEncodeContext *ctx, MLPSubstream *s, uint8_t **const samples, int nb_samples)
Wrapper function for inputting data in two different bit-depths.
Definition mlpenc.c:1219
static int output_packet(AVFormatContext *ctx, int flush)
Definition mpegenc.c:1022
#define av_strdup(s)
Definition ops_static.c:55
AVOptions.
Context for an Audio FIFO Buffer.
Definition audio_fifo.c:37
int nb_channels
Number of channels in this layout.
main external API structure.
Definition avcodec.h:443
AVChannelLayout ch_layout
Audio channel layout.
Definition avcodec.h:1055
enum AVSampleFormat sample_fmt
audio sample format
Definition avcodec.h:1047
AVRational pkt_timebase
Timebase in which pkt_dts/pts and AVPacket.dts/pts are expressed.
Definition avcodec.h:554
int64_t bit_rate
the average bitrate
Definition avcodec.h:493
int sample_rate
samples per second
Definition avcodec.h:1040
int flags
AV_CODEC_FLAG_*.
Definition avcodec.h:500
int frame_size
Number of samples per channel in an audio frame.
Definition avcodec.h:1068
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition codec_par.h:57
AVCodec.
Definition codec.h:175
Format I/O context.
Definition avformat.h:1333
This structure describes decoded (raw) audio or video data.
Definition frame.h:472
int nb_samples
number of audio samples (per channel) described by this frame
Definition frame.h:552
uint8_t ** extended_data
pointers to the data planes/channels.
Definition frame.h:533
Bytestream IO Context.
Definition avio.h:160
This structure stores compressed data.
Definition packet.h:580
int num
Numerator.
Definition rational.h:59
int den
Denominator.
Definition rational.h:60
Stream structure.
Definition avformat.h:766
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition avformat.h:789
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition avformat.h:805
The libswresample context.
libswresample public header
#define av_freep(p)
static void error(const char *err)
static int open_output_file(const char *filename)
Definition transcode.c:129
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
Definition transcode_aac.c:48
#define OUTPUT_CHANNELS
Definition transcode_aac.c:50
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 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.
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