FFmpeg
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 "libavformat/avformat.h"
33 #include "libavformat/avio.h"
34 
35 #include "libavcodec/avcodec.h"
36 
37 #include "libavutil/audio_fifo.h"
38 #include "libavutil/avassert.h"
39 #include "libavutil/avstring.h"
41 #include "libavutil/frame.h"
42 #include "libavutil/opt.h"
43 
45 
46 /* The output bit rate in bit/s */
47 #define OUTPUT_BIT_RATE 96000
48 /* The number of output channels */
49 #define OUTPUT_CHANNELS 2
50 
51 /**
52  * Open an input file and the required decoder.
53  * @param filename File to be opened
54  * @param[out] input_format_context Format context of opened file
55  * @param[out] input_codec_context Codec context of opened file
56  * @return Error code (0 if successful)
57  */
58 static int open_input_file(const char *filename,
59  AVFormatContext **input_format_context,
60  AVCodecContext **input_codec_context)
61 {
62  AVCodecContext *avctx;
63  const AVCodec *input_codec;
64  const AVStream *stream;
65  int error;
66 
67  /* Open the input file to read from it. */
68  if ((error = avformat_open_input(input_format_context, filename, NULL,
69  NULL)) < 0) {
70  fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
71  filename, av_err2str(error));
72  *input_format_context = NULL;
73  return error;
74  }
75 
76  /* Get information on the input file (number of streams etc.). */
77  if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
78  fprintf(stderr, "Could not open find stream info (error '%s')\n",
79  av_err2str(error));
80  avformat_close_input(input_format_context);
81  return error;
82  }
83 
84  /* Make sure that there is only one stream in the input file. */
85  if ((*input_format_context)->nb_streams != 1) {
86  fprintf(stderr, "Expected one audio input stream, but found %d\n",
87  (*input_format_context)->nb_streams);
88  avformat_close_input(input_format_context);
89  return AVERROR_EXIT;
90  }
91 
92  stream = (*input_format_context)->streams[0];
93 
94  /* Find a decoder for the audio stream. */
95  if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
96  fprintf(stderr, "Could not find input codec\n");
97  avformat_close_input(input_format_context);
98  return AVERROR_EXIT;
99  }
100 
101  /* Allocate a new decoding context. */
102  avctx = avcodec_alloc_context3(input_codec);
103  if (!avctx) {
104  fprintf(stderr, "Could not allocate a decoding context\n");
105  avformat_close_input(input_format_context);
106  return AVERROR(ENOMEM);
107  }
108 
109  /* Initialize the stream parameters with demuxer information. */
110  error = avcodec_parameters_to_context(avctx, stream->codecpar);
111  if (error < 0) {
112  avformat_close_input(input_format_context);
113  avcodec_free_context(&avctx);
114  return error;
115  }
116 
117  /* Open the decoder for the audio stream to use it later. */
118  if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
119  fprintf(stderr, "Could not open input codec (error '%s')\n",
120  av_err2str(error));
121  avcodec_free_context(&avctx);
122  avformat_close_input(input_format_context);
123  return error;
124  }
125 
126  /* Set the packet timebase for the decoder. */
127  avctx->pkt_timebase = stream->time_base;
128 
129  /* Save the decoder context for easier access later. */
130  *input_codec_context = avctx;
131 
132  return 0;
133 }
134 
135 /**
136  * Open an output file and the required encoder.
137  * Also set some basic encoder parameters.
138  * Some of these parameters are based on the input file's parameters.
139  * @param filename File to be opened
140  * @param input_codec_context Codec context of input file
141  * @param[out] output_format_context Format context of output file
142  * @param[out] output_codec_context Codec context of output file
143  * @return Error code (0 if successful)
144  */
145 static int open_output_file(const char *filename,
146  AVCodecContext *input_codec_context,
147  AVFormatContext **output_format_context,
148  AVCodecContext **output_codec_context)
149 {
150  AVCodecContext *avctx = NULL;
151  AVIOContext *output_io_context = NULL;
152  AVStream *stream = NULL;
153  const AVCodec *output_codec = NULL;
154  int error;
155 
156  /* Open the output file to write to it. */
157  if ((error = avio_open(&output_io_context, filename,
158  AVIO_FLAG_WRITE)) < 0) {
159  fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
160  filename, av_err2str(error));
161  return error;
162  }
163 
164  /* Create a new format context for the output container format. */
165  if (!(*output_format_context = avformat_alloc_context())) {
166  fprintf(stderr, "Could not allocate output format context\n");
167  return AVERROR(ENOMEM);
168  }
169 
170  /* Associate the output file (pointer) with the container format context. */
171  (*output_format_context)->pb = output_io_context;
172 
173  /* Guess the desired container format based on the file extension. */
174  if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
175  NULL))) {
176  fprintf(stderr, "Could not find output file format\n");
177  goto cleanup;
178  }
179 
180  if (!((*output_format_context)->url = av_strdup(filename))) {
181  fprintf(stderr, "Could not allocate url.\n");
182  error = AVERROR(ENOMEM);
183  goto cleanup;
184  }
185 
186  /* Find the encoder to be used by its name. */
187  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
188  fprintf(stderr, "Could not find an AAC encoder.\n");
189  goto cleanup;
190  }
191 
192  /* Create a new audio stream in the output file container. */
193  if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
194  fprintf(stderr, "Could not create new stream\n");
195  error = AVERROR(ENOMEM);
196  goto cleanup;
197  }
198 
199  avctx = avcodec_alloc_context3(output_codec);
200  if (!avctx) {
201  fprintf(stderr, "Could not allocate an encoding context\n");
202  error = AVERROR(ENOMEM);
203  goto cleanup;
204  }
205 
206  /* Set the basic encoder parameters.
207  * The input file's sample rate is used to avoid a sample rate conversion. */
209  avctx->sample_rate = input_codec_context->sample_rate;
210  avctx->sample_fmt = output_codec->sample_fmts[0];
211  avctx->bit_rate = OUTPUT_BIT_RATE;
212 
213  /* Set the sample rate for the container. */
214  stream->time_base.den = input_codec_context->sample_rate;
215  stream->time_base.num = 1;
216 
217  /* Some container formats (like MP4) require global headers to be present.
218  * Mark the encoder so that it behaves accordingly. */
219  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
221 
222  /* Open the encoder for the audio stream to use it later. */
223  if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
224  fprintf(stderr, "Could not open output codec (error '%s')\n",
225  av_err2str(error));
226  goto cleanup;
227  }
228 
230  if (error < 0) {
231  fprintf(stderr, "Could not initialize stream parameters\n");
232  goto cleanup;
233  }
234 
235  /* Save the encoder context for easier access later. */
236  *output_codec_context = avctx;
237 
238  return 0;
239 
240 cleanup:
241  avcodec_free_context(&avctx);
242  avio_closep(&(*output_format_context)->pb);
243  avformat_free_context(*output_format_context);
244  *output_format_context = NULL;
245  return error < 0 ? error : AVERROR_EXIT;
246 }
247 
248 /**
249  * Initialize one data packet for reading or writing.
250  * @param[out] packet Packet to be initialized
251  * @return Error code (0 if successful)
252  */
254 {
255  if (!(*packet = av_packet_alloc())) {
256  fprintf(stderr, "Could not allocate packet\n");
257  return AVERROR(ENOMEM);
258  }
259  return 0;
260 }
261 
262 /**
263  * Initialize one audio frame for reading from the input file.
264  * @param[out] frame Frame to be initialized
265  * @return Error code (0 if successful)
266  */
268 {
269  if (!(*frame = av_frame_alloc())) {
270  fprintf(stderr, "Could not allocate input frame\n");
271  return AVERROR(ENOMEM);
272  }
273  return 0;
274 }
275 
276 /**
277  * Initialize the audio resampler based on the input and output codec settings.
278  * If the input and output sample formats differ, a conversion is required
279  * libswresample takes care of this, but requires initialization.
280  * @param input_codec_context Codec context of the input file
281  * @param output_codec_context Codec context of the output file
282  * @param[out] resample_context Resample context for the required conversion
283  * @return Error code (0 if successful)
284  */
285 static int init_resampler(AVCodecContext *input_codec_context,
286  AVCodecContext *output_codec_context,
287  SwrContext **resample_context)
288 {
289  int error;
290 
291  /*
292  * Create a resampler context for the conversion.
293  * Set the conversion parameters.
294  */
295  error = swr_alloc_set_opts2(resample_context,
296  &output_codec_context->ch_layout,
297  output_codec_context->sample_fmt,
298  output_codec_context->sample_rate,
299  &input_codec_context->ch_layout,
300  input_codec_context->sample_fmt,
301  input_codec_context->sample_rate,
302  0, NULL);
303  if (error < 0) {
304  fprintf(stderr, "Could not allocate resample context\n");
305  return error;
306  }
307  /*
308  * Perform a sanity check so that the number of converted samples is
309  * not greater than the number of samples to be converted.
310  * If the sample rates differ, this case has to be handled differently
311  */
312  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
313 
314  /* Open the resampler with the specified parameters. */
315  if ((error = swr_init(*resample_context)) < 0) {
316  fprintf(stderr, "Could not open resample context\n");
317  swr_free(resample_context);
318  return error;
319  }
320  return 0;
321 }
322 
323 /**
324  * Initialize a FIFO buffer for the audio samples to be encoded.
325  * @param[out] fifo Sample buffer
326  * @param output_codec_context Codec context of the output file
327  * @return Error code (0 if successful)
328  */
329 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
330 {
331  /* Create the FIFO buffer based on the specified output sample format. */
332  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
333  output_codec_context->ch_layout.nb_channels, 1))) {
334  fprintf(stderr, "Could not allocate FIFO\n");
335  return AVERROR(ENOMEM);
336  }
337  return 0;
338 }
339 
340 /**
341  * Write the header of the output file container.
342  * @param output_format_context Format context of the output file
343  * @return Error code (0 if successful)
344  */
345 static int write_output_file_header(AVFormatContext *output_format_context)
346 {
347  int error;
348  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
349  fprintf(stderr, "Could not write output file header (error '%s')\n",
350  av_err2str(error));
351  return error;
352  }
353  return 0;
354 }
355 
356 /**
357  * Decode one audio frame from the input file.
358  * @param frame Audio frame to be decoded
359  * @param input_format_context Format context of the input file
360  * @param input_codec_context Codec context of the input file
361  * @param[out] data_present Indicates whether data has been decoded
362  * @param[out] finished Indicates whether the end of file has
363  * been reached and all data has been
364  * decoded. If this flag is false, there
365  * is more data to be decoded, i.e., this
366  * function has to be called again.
367  * @return Error code (0 if successful)
368  */
370  AVFormatContext *input_format_context,
371  AVCodecContext *input_codec_context,
372  int *data_present, int *finished)
373 {
374  /* Packet used for temporary storage. */
375  AVPacket *input_packet;
376  int error;
377 
378  error = init_packet(&input_packet);
379  if (error < 0)
380  return error;
381 
382  *data_present = 0;
383  *finished = 0;
384  /* Read one audio frame from the input file into a temporary packet. */
385  if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
386  /* If we are at the end of the file, flush the decoder below. */
387  if (error == AVERROR_EOF)
388  *finished = 1;
389  else {
390  fprintf(stderr, "Could not read frame (error '%s')\n",
391  av_err2str(error));
392  goto cleanup;
393  }
394  }
395 
396  /* Send the audio frame stored in the temporary packet to the decoder.
397  * The input audio stream decoder is used to do this. */
398  if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
399  fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
400  av_err2str(error));
401  goto cleanup;
402  }
403 
404  /* Receive one frame from the decoder. */
405  error = avcodec_receive_frame(input_codec_context, frame);
406  /* If the decoder asks for more data to be able to decode a frame,
407  * return indicating that no data is present. */
408  if (error == AVERROR(EAGAIN)) {
409  error = 0;
410  goto cleanup;
411  /* If the end of the input file is reached, stop decoding. */
412  } else if (error == AVERROR_EOF) {
413  *finished = 1;
414  error = 0;
415  goto cleanup;
416  } else if (error < 0) {
417  fprintf(stderr, "Could not decode frame (error '%s')\n",
418  av_err2str(error));
419  goto cleanup;
420  /* Default case: Return decoded data. */
421  } else {
422  *data_present = 1;
423  goto cleanup;
424  }
425 
426 cleanup:
427  av_packet_free(&input_packet);
428  return error;
429 }
430 
431 /**
432  * Initialize a temporary storage for the specified number of audio samples.
433  * The conversion requires temporary storage due to the different format.
434  * The number of audio samples to be allocated is specified in frame_size.
435  * @param[out] converted_input_samples Array of converted samples. The
436  * dimensions are reference, channel
437  * (for multi-channel audio), sample.
438  * @param output_codec_context Codec context of the output file
439  * @param frame_size Number of samples to be converted in
440  * each round
441  * @return Error code (0 if successful)
442  */
443 static int init_converted_samples(uint8_t ***converted_input_samples,
444  AVCodecContext *output_codec_context,
445  int frame_size)
446 {
447  int error;
448 
449  /* Allocate as many pointers as there are audio channels.
450  * Each pointer will point to the audio samples of the corresponding
451  * channels (although it may be NULL for interleaved formats).
452  * Allocate memory for the samples of all channels in one consecutive
453  * block for convenience. */
454  if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL,
455  output_codec_context->ch_layout.nb_channels,
456  frame_size,
457  output_codec_context->sample_fmt, 0)) < 0) {
458  fprintf(stderr,
459  "Could not allocate converted input samples (error '%s')\n",
460  av_err2str(error));
461  return error;
462  }
463  return 0;
464 }
465 
466 /**
467  * Convert the input audio samples into the output sample format.
468  * The conversion happens on a per-frame basis, the size of which is
469  * specified by frame_size.
470  * @param input_data Samples to be decoded. The dimensions are
471  * channel (for multi-channel audio), sample.
472  * @param[out] converted_data Converted samples. The dimensions are channel
473  * (for multi-channel audio), sample.
474  * @param frame_size Number of samples to be converted
475  * @param resample_context Resample context for the conversion
476  * @return Error code (0 if successful)
477  */
478 static int convert_samples(const uint8_t **input_data,
479  uint8_t **converted_data, const int frame_size,
480  SwrContext *resample_context)
481 {
482  int error;
483 
484  /* Convert the samples using the resampler. */
485  if ((error = swr_convert(resample_context,
486  converted_data, frame_size,
487  input_data , frame_size)) < 0) {
488  fprintf(stderr, "Could not convert input samples (error '%s')\n",
489  av_err2str(error));
490  return error;
491  }
492 
493  return 0;
494 }
495 
496 /**
497  * Add converted input audio samples to the FIFO buffer for later processing.
498  * @param fifo Buffer to add the samples to
499  * @param converted_input_samples Samples to be added. The dimensions are channel
500  * (for multi-channel audio), sample.
501  * @param frame_size Number of samples to be converted
502  * @return Error code (0 if successful)
503  */
505  uint8_t **converted_input_samples,
506  const int frame_size)
507 {
508  int error;
509 
510  /* Make the FIFO as large as it needs to be to hold both,
511  * the old and the new samples. */
512  if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
513  fprintf(stderr, "Could not reallocate FIFO\n");
514  return error;
515  }
516 
517  /* Store the new samples in the FIFO buffer. */
518  if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
519  frame_size) < frame_size) {
520  fprintf(stderr, "Could not write data to FIFO\n");
521  return AVERROR_EXIT;
522  }
523  return 0;
524 }
525 
526 /**
527  * Read one audio frame from the input file, decode, convert and store
528  * it in the FIFO buffer.
529  * @param fifo Buffer used for temporary storage
530  * @param input_format_context Format context of the input file
531  * @param input_codec_context Codec context of the input file
532  * @param output_codec_context Codec context of the output file
533  * @param resampler_context Resample context for the conversion
534  * @param[out] finished Indicates whether the end of file has
535  * been reached and all data has been
536  * decoded. If this flag is false,
537  * there is more data to be decoded,
538  * i.e., this function has to be called
539  * again.
540  * @return Error code (0 if successful)
541  */
543  AVFormatContext *input_format_context,
544  AVCodecContext *input_codec_context,
545  AVCodecContext *output_codec_context,
546  SwrContext *resampler_context,
547  int *finished)
548 {
549  /* Temporary storage of the input samples of the frame read from the file. */
550  AVFrame *input_frame = NULL;
551  /* Temporary storage for the converted input samples. */
552  uint8_t **converted_input_samples = NULL;
553  int data_present;
554  int ret = AVERROR_EXIT;
555 
556  /* Initialize temporary storage for one input frame. */
557  if (init_input_frame(&input_frame))
558  goto cleanup;
559  /* Decode one frame worth of audio samples. */
560  if (decode_audio_frame(input_frame, input_format_context,
561  input_codec_context, &data_present, finished))
562  goto cleanup;
563  /* If we are at the end of the file and there are no more samples
564  * in the decoder which are delayed, we are actually finished.
565  * This must not be treated as an error. */
566  if (*finished) {
567  ret = 0;
568  goto cleanup;
569  }
570  /* If there is decoded data, convert and store it. */
571  if (data_present) {
572  /* Initialize the temporary storage for the converted input samples. */
573  if (init_converted_samples(&converted_input_samples, output_codec_context,
574  input_frame->nb_samples))
575  goto cleanup;
576 
577  /* Convert the input samples to the desired output sample format.
578  * This requires a temporary storage provided by converted_input_samples. */
579  if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
580  input_frame->nb_samples, resampler_context))
581  goto cleanup;
582 
583  /* Add the converted input samples to the FIFO buffer for later processing. */
584  if (add_samples_to_fifo(fifo, converted_input_samples,
585  input_frame->nb_samples))
586  goto cleanup;
587  ret = 0;
588  }
589  ret = 0;
590 
591 cleanup:
592  if (converted_input_samples)
593  av_freep(&converted_input_samples[0]);
594  av_freep(&converted_input_samples);
595  av_frame_free(&input_frame);
596 
597  return ret;
598 }
599 
600 /**
601  * Initialize one input frame for writing to the output file.
602  * The frame will be exactly frame_size samples large.
603  * @param[out] frame Frame to be initialized
604  * @param output_codec_context Codec context of the output file
605  * @param frame_size Size of the frame
606  * @return Error code (0 if successful)
607  */
609  AVCodecContext *output_codec_context,
610  int frame_size)
611 {
612  int error;
613 
614  /* Create a new frame to store the audio samples. */
615  if (!(*frame = av_frame_alloc())) {
616  fprintf(stderr, "Could not allocate output frame\n");
617  return AVERROR_EXIT;
618  }
619 
620  /* Set the frame's parameters, especially its size and format.
621  * av_frame_get_buffer needs this to allocate memory for the
622  * audio samples of the frame.
623  * Default channel layouts based on the number of channels
624  * are assumed for simplicity. */
625  (*frame)->nb_samples = frame_size;
626  av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
627  (*frame)->format = output_codec_context->sample_fmt;
628  (*frame)->sample_rate = output_codec_context->sample_rate;
629 
630  /* Allocate the samples of the created frame. This call will make
631  * sure that the audio frame can hold as many samples as specified. */
632  if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
633  fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
634  av_err2str(error));
636  return error;
637  }
638 
639  return 0;
640 }
641 
642 /* Global timestamp for the audio frames. */
643 static int64_t pts = 0;
644 
645 /**
646  * Encode one frame worth of audio to the output file.
647  * @param frame Samples to be encoded
648  * @param output_format_context Format context of the output file
649  * @param output_codec_context Codec context of the output file
650  * @param[out] data_present Indicates whether data has been
651  * encoded
652  * @return Error code (0 if successful)
653  */
655  AVFormatContext *output_format_context,
656  AVCodecContext *output_codec_context,
657  int *data_present)
658 {
659  /* Packet used for temporary storage. */
661  int error;
662 
664  if (error < 0)
665  return error;
666 
667  /* Set a timestamp based on the sample rate for the container. */
668  if (frame) {
669  frame->pts = pts;
670  pts += frame->nb_samples;
671  }
672 
673  *data_present = 0;
674  /* Send the audio frame stored in the temporary packet to the encoder.
675  * The output audio stream encoder is used to do this. */
676  error = avcodec_send_frame(output_codec_context, frame);
677  /* Check for errors, but proceed with fetching encoded samples if the
678  * encoder signals that it has nothing more to encode. */
679  if (error < 0 && error != AVERROR_EOF) {
680  fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
681  av_err2str(error));
682  goto cleanup;
683  }
684 
685  /* Receive one encoded frame from the encoder. */
686  error = avcodec_receive_packet(output_codec_context, output_packet);
687  /* If the encoder asks for more data to be able to provide an
688  * encoded frame, return indicating that no data is present. */
689  if (error == AVERROR(EAGAIN)) {
690  error = 0;
691  goto cleanup;
692  /* If the last frame has been encoded, stop encoding. */
693  } else if (error == AVERROR_EOF) {
694  error = 0;
695  goto cleanup;
696  } else if (error < 0) {
697  fprintf(stderr, "Could not encode frame (error '%s')\n",
698  av_err2str(error));
699  goto cleanup;
700  /* Default case: Return encoded data. */
701  } else {
702  *data_present = 1;
703  }
704 
705  /* Write one audio frame from the temporary packet to the output file. */
706  if (*data_present &&
707  (error = av_write_frame(output_format_context, output_packet)) < 0) {
708  fprintf(stderr, "Could not write frame (error '%s')\n",
709  av_err2str(error));
710  goto cleanup;
711  }
712 
713 cleanup:
715  return error;
716 }
717 
718 /**
719  * Load one audio frame from the FIFO buffer, encode and write it to the
720  * output file.
721  * @param fifo Buffer used for temporary storage
722  * @param output_format_context Format context of the output file
723  * @param output_codec_context Codec context of the output file
724  * @return Error code (0 if successful)
725  */
727  AVFormatContext *output_format_context,
728  AVCodecContext *output_codec_context)
729 {
730  /* Temporary storage of the output samples of the frame written to the file. */
732  /* Use the maximum number of possible samples per frame.
733  * If there is less than the maximum possible frame size in the FIFO
734  * buffer use this number. Otherwise, use the maximum possible frame size. */
735  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
736  output_codec_context->frame_size);
737  int data_written;
738 
739  /* Initialize temporary storage for one output frame. */
740  if (init_output_frame(&output_frame, output_codec_context, frame_size))
741  return AVERROR_EXIT;
742 
743  /* Read as many samples from the FIFO buffer as required to fill the frame.
744  * The samples are stored in the frame temporarily. */
745  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
746  fprintf(stderr, "Could not read data from FIFO\n");
748  return AVERROR_EXIT;
749  }
750 
751  /* Encode one frame worth of audio samples. */
752  if (encode_audio_frame(output_frame, output_format_context,
753  output_codec_context, &data_written)) {
755  return AVERROR_EXIT;
756  }
758  return 0;
759 }
760 
761 /**
762  * Write the trailer of the output file container.
763  * @param output_format_context Format context of the output file
764  * @return Error code (0 if successful)
765  */
766 static int write_output_file_trailer(AVFormatContext *output_format_context)
767 {
768  int error;
769  if ((error = av_write_trailer(output_format_context)) < 0) {
770  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
771  av_err2str(error));
772  return error;
773  }
774  return 0;
775 }
776 
777 int main(int argc, char **argv)
778 {
779  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
780  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
781  SwrContext *resample_context = NULL;
782  AVAudioFifo *fifo = NULL;
783  int ret = AVERROR_EXIT;
784 
785  if (argc != 3) {
786  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
787  exit(1);
788  }
789 
790  /* Open the input file for reading. */
791  if (open_input_file(argv[1], &input_format_context,
792  &input_codec_context))
793  goto cleanup;
794  /* Open the output file for writing. */
795  if (open_output_file(argv[2], input_codec_context,
796  &output_format_context, &output_codec_context))
797  goto cleanup;
798  /* Initialize the resampler to be able to convert audio sample formats. */
799  if (init_resampler(input_codec_context, output_codec_context,
800  &resample_context))
801  goto cleanup;
802  /* Initialize the FIFO buffer to store audio samples to be encoded. */
803  if (init_fifo(&fifo, output_codec_context))
804  goto cleanup;
805  /* Write the header of the output file container. */
806  if (write_output_file_header(output_format_context))
807  goto cleanup;
808 
809  /* Loop as long as we have input samples to read or output samples
810  * to write; abort as soon as we have neither. */
811  while (1) {
812  /* Use the encoder's desired frame size for processing. */
813  const int output_frame_size = output_codec_context->frame_size;
814  int finished = 0;
815 
816  /* Make sure that there is one frame worth of samples in the FIFO
817  * buffer so that the encoder can do its work.
818  * Since the decoder's and the encoder's frame size may differ, we
819  * need to FIFO buffer to store as many frames worth of input samples
820  * that they make up at least one frame worth of output samples. */
821  while (av_audio_fifo_size(fifo) < output_frame_size) {
822  /* Decode one frame worth of audio samples, convert it to the
823  * output sample format and put it into the FIFO buffer. */
824  if (read_decode_convert_and_store(fifo, input_format_context,
825  input_codec_context,
826  output_codec_context,
827  resample_context, &finished))
828  goto cleanup;
829 
830  /* If we are at the end of the input file, we continue
831  * encoding the remaining audio samples to the output file. */
832  if (finished)
833  break;
834  }
835 
836  /* If we have enough samples for the encoder, we encode them.
837  * At the end of the file, we pass the remaining samples to
838  * the encoder. */
839  while (av_audio_fifo_size(fifo) >= output_frame_size ||
840  (finished && av_audio_fifo_size(fifo) > 0))
841  /* Take one frame worth of audio samples from the FIFO buffer,
842  * encode it and write it to the output file. */
843  if (load_encode_and_write(fifo, output_format_context,
844  output_codec_context))
845  goto cleanup;
846 
847  /* If we are at the end of the input file and have encoded
848  * all remaining samples, we can exit this loop and finish. */
849  if (finished) {
850  int data_written;
851  /* Flush the encoder as it may have delayed frames. */
852  do {
853  if (encode_audio_frame(NULL, output_format_context,
854  output_codec_context, &data_written))
855  goto cleanup;
856  } while (data_written);
857  break;
858  }
859  }
860 
861  /* Write the trailer of the output file container. */
862  if (write_output_file_trailer(output_format_context))
863  goto cleanup;
864  ret = 0;
865 
866 cleanup:
867  if (fifo)
868  av_audio_fifo_free(fifo);
869  swr_free(&resample_context);
870  if (output_codec_context)
871  avcodec_free_context(&output_codec_context);
872  if (output_format_context) {
873  avio_closep(&output_format_context->pb);
874  avformat_free_context(output_format_context);
875  }
876  if (input_codec_context)
877  avcodec_free_context(&input_codec_context);
878  if (input_format_context)
879  avformat_close_input(&input_format_context);
880 
881  return ret;
882 }
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:31
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:48
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1096
AVCodec
AVCodec.
Definition: codec.h:187
load_encode_and_write
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.
Definition: transcode_aac.c:726
avcodec_receive_packet
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
Definition: encode.c:556
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
open_input_file
static int open_input_file(const char *filename, AVFormatContext **input_format_context, AVCodecContext **input_codec_context)
Open an input file and the required decoder.
Definition: transcode_aac.c:58
av_frame_get_buffer
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:243
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1068
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const struct AVCodec *c)
Add a new stream to a media file.
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void *const *data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:119
av_audio_fifo_realloc
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:99
init_fifo
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
Definition: transcode_aac.c:329
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:100
avcodec_find_encoder
const AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: allcodecs.c:976
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:340
cleanup
static av_cold void cleanup(FlashSV2Context *s)
Definition: flashsv2enc.c:130
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:452
write_output_file_header
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
Definition: transcode_aac.c:345
open_output_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.
Definition: transcode_aac.c:145
av_read_frame
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition: demux.c:1558
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:323
av_packet_free
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:74
AV_CODEC_FLAG_GLOBAL_HEADER
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition: avcodec.h:338
avformat_close_input
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: demux.c:374
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:37
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2111
AVCodec::sample_fmts
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:211
output_packet
static int output_packet(AVFormatContext *ctx, int flush)
Definition: mpegenc.c:1009
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:525
pts
static int64_t pts
Definition: transcode_aac.c:643
AVRational::num
int num
Numerator.
Definition: rational.h:59
av_frame_alloc
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:88
avassert.h
swr_init
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:193
avformat_open_input
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:226
avcodec_alloc_context3
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition: options.c:154
add_samples_to_fifo
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.
Definition: transcode_aac.c:504
frame_size
int frame_size
Definition: mxfenc.c:2422
decode_audio_frame
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.
Definition: transcode_aac.c:369
avcodec_receive_frame
int attribute_align_arg 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...
Definition: avcodec.c:715
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
AVIO_FLAG_WRITE
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:637
SwrContext
The libswresample context.
Definition: swresample_internal.h:95
avformat_write_header
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:472
frame
static AVFrame * frame
Definition: demux_decode.c:54
AVFormatContext
Format I/O context.
Definition: avformat.h:1363
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:864
avcodec_parameters_to_context
int avcodec_parameters_to_context(AVCodecContext *codec, const struct AVCodecParameters *par)
Fill the codec context based on the values from the supplied codec parameters.
AVStream::time_base
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition: avformat.h:880
NULL
#define NULL
Definition: coverity.c:32
avcodec_free_context
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:169
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:62
read_decode_convert_and_store
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.
Definition: transcode_aac.c:542
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:495
OUTPUT_BIT_RATE
#define OUTPUT_BIT_RATE
Definition: transcode_aac.c:47
avcodec_open2
int attribute_align_arg avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: avcodec.c:128
av_write_frame
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition: mux.c:1218
init_output_frame
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
Initialize one input frame for writing to the output file.
Definition: transcode_aac.c:608
swresample.h
avcodec_find_decoder
const AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: allcodecs.c:981
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:445
init_input_frame
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
Definition: transcode_aac.c:267
avformat_find_stream_info
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
Definition: demux.c:2532
AVIOContext
Bytestream IO Context.
Definition: avio.h:166
swr_alloc_set_opts2
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:85
avformat_alloc_context
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
Definition: options.c:167
av_err2str
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: error.h:121
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1084
AVCodecContext::pkt_timebase
AVRational pkt_timebase
Timebase in which pkt_dts/pts and AVPacket.dts/pts are expressed.
Definition: avcodec.h:1821
av_audio_fifo_read
int av_audio_fifo_read(AVAudioFifo *af, void *const *data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:175
encode_audio_frame
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.
Definition: transcode_aac.c:654
main
int main(int argc, char **argv)
Definition: transcode_aac.c:777
avio.h
swr_free
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:174
init_packet
static int init_packet(AVPacket **packet)
Initialize one data packet for reading or writing.
Definition: transcode_aac.c:253
swr_convert
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t **out_arg, int out_count, const uint8_t **in_arg, int in_count)
Convert audio.
Definition: swresample.c:837
frame.h
OUTPUT_CHANNELS
#define OUTPUT_CHANNELS
Definition: transcode_aac.c:49
output_frame
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264dec.c:874
av_packet_alloc
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:63
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:222
init_resampler
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.
Definition: transcode_aac.c:285
av_channel_layout_default
void av_channel_layout_default(AVChannelLayout *ch_layout, int nb_channels)
Get the default channel layout for a given number of channels.
Definition: channel_layout.c:1033
avcodec_send_packet
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition: decode.c:709
avio_closep
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: aviobuf.c:1368
av_write_trailer
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1286
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:420
AVFMT_GLOBALHEADER
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition: avformat.h:479
convert_samples
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.
Definition: transcode_aac.c:478
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:401
packet
enum AVPacketSideDataType packet
Definition: decode.c:1425
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
init_converted_samples
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.
Definition: transcode_aac.c:443
audio_fifo.h
avcodec_send_frame
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
Definition: encode.c:518
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVStream
Stream structure.
Definition: avformat.h:841
avformat.h
AVCodecContext
main external API structure.
Definition: avcodec.h:445
channel_layout.h
AVRational::den
int den
Denominator.
Definition: rational.h:60
avformat_free_context
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: avformat.c:141
avio_open
int avio_open(AVIOContext **s, const char *url, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
Definition: aviobuf.c:1303
av_channel_layout_copy
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
Definition: channel_layout.c:669
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:270
av_guess_format
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
av_samples_alloc_array_and_samples
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:207
avcodec_parameters_from_context
int avcodec_parameters_from_context(struct AVCodecParameters *par, const AVCodecContext *codec)
Fill the parameters struct based on the values from the supplied codec context.
Definition: codec_par.c:137
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:55
AVPacket
This structure stores compressed data.
Definition: packet.h:499
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
AVERROR_EXIT
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:58
avstring.h
input_data
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:1224
write_output_file_trailer
static int write_output_file_trailer(AVFormatContext *output_format_context)
Write the trailer of the output file container.
Definition: transcode_aac.c:766