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transcode_aac.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 /**
20  * @file
21  * simple audio converter
22  *
23  * @example transcode_aac.c
24  * Convert an input audio file to AAC in an MP4 container using FFmpeg.
25  * @author Andreas Unterweger (dustsigns@gmail.com)
26  */
27 
28 #include <stdio.h>
29 
30 #include "libavformat/avformat.h"
31 #include "libavformat/avio.h"
32 
33 #include "libavcodec/avcodec.h"
34 
35 #include "libavutil/audio_fifo.h"
36 #include "libavutil/avassert.h"
37 #include "libavutil/avstring.h"
38 #include "libavutil/frame.h"
39 #include "libavutil/opt.h"
40 
42 
43 /** The output bit rate in kbit/s */
44 #define OUTPUT_BIT_RATE 96000
45 /** The number of output channels */
46 #define OUTPUT_CHANNELS 2
47 
48 /**
49  * Convert an error code into a text message.
50  * @param error Error code to be converted
51  * @return Corresponding error text (not thread-safe)
52  */
53 static const char *get_error_text(const int error)
54 {
55  static char error_buffer[255];
56  av_strerror(error, error_buffer, sizeof(error_buffer));
57  return error_buffer;
58 }
59 
60 /** Open an input file and the required decoder. */
61 static int open_input_file(const char *filename,
62  AVFormatContext **input_format_context,
63  AVCodecContext **input_codec_context)
64 {
65  AVCodec *input_codec;
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, get_error_text(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  get_error_text(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  /** Find a decoder for the audio stream. */
94  if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
95  fprintf(stderr, "Could not find input codec\n");
96  avformat_close_input(input_format_context);
97  return AVERROR_EXIT;
98  }
99 
100  /** Open the decoder for the audio stream to use it later. */
101  if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
102  input_codec, NULL)) < 0) {
103  fprintf(stderr, "Could not open input codec (error '%s')\n",
104  get_error_text(error));
105  avformat_close_input(input_format_context);
106  return error;
107  }
108 
109  /** Save the decoder context for easier access later. */
110  *input_codec_context = (*input_format_context)->streams[0]->codec;
111 
112  return 0;
113 }
114 
115 /**
116  * Open an output file and the required encoder.
117  * Also set some basic encoder parameters.
118  * Some of these parameters are based on the input file's parameters.
119  */
120 static int open_output_file(const char *filename,
121  AVCodecContext *input_codec_context,
122  AVFormatContext **output_format_context,
123  AVCodecContext **output_codec_context)
124 {
125  AVIOContext *output_io_context = NULL;
126  AVStream *stream = NULL;
127  AVCodec *output_codec = NULL;
128  int error;
129 
130  /** Open the output file to write to it. */
131  if ((error = avio_open(&output_io_context, filename,
132  AVIO_FLAG_WRITE)) < 0) {
133  fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
134  filename, get_error_text(error));
135  return error;
136  }
137 
138  /** Create a new format context for the output container format. */
139  if (!(*output_format_context = avformat_alloc_context())) {
140  fprintf(stderr, "Could not allocate output format context\n");
141  return AVERROR(ENOMEM);
142  }
143 
144  /** Associate the output file (pointer) with the container format context. */
145  (*output_format_context)->pb = output_io_context;
146 
147  /** Guess the desired container format based on the file extension. */
148  if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
149  NULL))) {
150  fprintf(stderr, "Could not find output file format\n");
151  goto cleanup;
152  }
153 
154  av_strlcpy((*output_format_context)->filename, filename,
155  sizeof((*output_format_context)->filename));
156 
157  /** Find the encoder to be used by its name. */
158  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
159  fprintf(stderr, "Could not find an AAC encoder.\n");
160  goto cleanup;
161  }
162 
163  /** Create a new audio stream in the output file container. */
164  if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
165  fprintf(stderr, "Could not create new stream\n");
166  error = AVERROR(ENOMEM);
167  goto cleanup;
168  }
169 
170  /** Save the encoder context for easier access later. */
171  *output_codec_context = stream->codec;
172 
173  /**
174  * Set the basic encoder parameters.
175  * The input file's sample rate is used to avoid a sample rate conversion.
176  */
177  (*output_codec_context)->channels = OUTPUT_CHANNELS;
178  (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
179  (*output_codec_context)->sample_rate = input_codec_context->sample_rate;
180  (*output_codec_context)->sample_fmt = output_codec->sample_fmts[0];
181  (*output_codec_context)->bit_rate = OUTPUT_BIT_RATE;
182 
183  /** Allow the use of the experimental AAC encoder */
184  (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
185 
186  /** Set the sample rate for the container. */
187  stream->time_base.den = input_codec_context->sample_rate;
188  stream->time_base.num = 1;
189 
190  /**
191  * Some container formats (like MP4) require global headers to be present
192  * Mark the encoder so that it behaves accordingly.
193  */
194  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
195  (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
196 
197  /** Open the encoder for the audio stream to use it later. */
198  if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
199  fprintf(stderr, "Could not open output codec (error '%s')\n",
200  get_error_text(error));
201  goto cleanup;
202  }
203 
204  return 0;
205 
206 cleanup:
207  avio_closep(&(*output_format_context)->pb);
208  avformat_free_context(*output_format_context);
209  *output_format_context = NULL;
210  return error < 0 ? error : AVERROR_EXIT;
211 }
212 
213 /** Initialize one data packet for reading or writing. */
214 static void init_packet(AVPacket *packet)
215 {
216  av_init_packet(packet);
217  /** Set the packet data and size so that it is recognized as being empty. */
218  packet->data = NULL;
219  packet->size = 0;
220 }
221 
222 /** Initialize one audio frame for reading from the input file */
224 {
225  if (!(*frame = av_frame_alloc())) {
226  fprintf(stderr, "Could not allocate input frame\n");
227  return AVERROR(ENOMEM);
228  }
229  return 0;
230 }
231 
232 /**
233  * Initialize the audio resampler based on the input and output codec settings.
234  * If the input and output sample formats differ, a conversion is required
235  * libswresample takes care of this, but requires initialization.
236  */
237 static int init_resampler(AVCodecContext *input_codec_context,
238  AVCodecContext *output_codec_context,
239  SwrContext **resample_context)
240 {
241  int error;
242 
243  /**
244  * Create a resampler context for the conversion.
245  * Set the conversion parameters.
246  * Default channel layouts based on the number of channels
247  * are assumed for simplicity (they are sometimes not detected
248  * properly by the demuxer and/or decoder).
249  */
250  *resample_context = swr_alloc_set_opts(NULL,
251  av_get_default_channel_layout(output_codec_context->channels),
252  output_codec_context->sample_fmt,
253  output_codec_context->sample_rate,
254  av_get_default_channel_layout(input_codec_context->channels),
255  input_codec_context->sample_fmt,
256  input_codec_context->sample_rate,
257  0, NULL);
258  if (!*resample_context) {
259  fprintf(stderr, "Could not allocate resample context\n");
260  return AVERROR(ENOMEM);
261  }
262  /**
263  * Perform a sanity check so that the number of converted samples is
264  * not greater than the number of samples to be converted.
265  * If the sample rates differ, this case has to be handled differently
266  */
267  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
268 
269  /** Open the resampler with the specified parameters. */
270  if ((error = swr_init(*resample_context)) < 0) {
271  fprintf(stderr, "Could not open resample context\n");
272  swr_free(resample_context);
273  return error;
274  }
275  return 0;
276 }
277 
278 /** Initialize a FIFO buffer for the audio samples to be encoded. */
279 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
280 {
281  /** Create the FIFO buffer based on the specified output sample format. */
282  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
283  output_codec_context->channels, 1))) {
284  fprintf(stderr, "Could not allocate FIFO\n");
285  return AVERROR(ENOMEM);
286  }
287  return 0;
288 }
289 
290 /** Write the header of the output file container. */
291 static int write_output_file_header(AVFormatContext *output_format_context)
292 {
293  int error;
294  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
295  fprintf(stderr, "Could not write output file header (error '%s')\n",
296  get_error_text(error));
297  return error;
298  }
299  return 0;
300 }
301 
302 /** Decode one audio frame from the input file. */
304  AVFormatContext *input_format_context,
305  AVCodecContext *input_codec_context,
306  int *data_present, int *finished)
307 {
308  /** Packet used for temporary storage. */
309  AVPacket input_packet;
310  int error;
311  init_packet(&input_packet);
312 
313  /** Read one audio frame from the input file into a temporary packet. */
314  if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
315  /** If we are at the end of the file, flush the decoder below. */
316  if (error == AVERROR_EOF)
317  *finished = 1;
318  else {
319  fprintf(stderr, "Could not read frame (error '%s')\n",
320  get_error_text(error));
321  return error;
322  }
323  }
324 
325  /**
326  * Decode the audio frame stored in the temporary packet.
327  * The input audio stream decoder is used to do this.
328  * If we are at the end of the file, pass an empty packet to the decoder
329  * to flush it.
330  */
331  if ((error = avcodec_decode_audio4(input_codec_context, frame,
332  data_present, &input_packet)) < 0) {
333  fprintf(stderr, "Could not decode frame (error '%s')\n",
334  get_error_text(error));
335  av_packet_unref(&input_packet);
336  return error;
337  }
338 
339  /**
340  * If the decoder has not been flushed completely, we are not finished,
341  * so that this function has to be called again.
342  */
343  if (*finished && *data_present)
344  *finished = 0;
345  av_packet_unref(&input_packet);
346  return 0;
347 }
348 
349 /**
350  * Initialize a temporary storage for the specified number of audio samples.
351  * The conversion requires temporary storage due to the different format.
352  * The number of audio samples to be allocated is specified in frame_size.
353  */
354 static int init_converted_samples(uint8_t ***converted_input_samples,
355  AVCodecContext *output_codec_context,
356  int frame_size)
357 {
358  int error;
359 
360  /**
361  * Allocate as many pointers as there are audio channels.
362  * Each pointer will later point to the audio samples of the corresponding
363  * channels (although it may be NULL for interleaved formats).
364  */
365  if (!(*converted_input_samples = calloc(output_codec_context->channels,
366  sizeof(**converted_input_samples)))) {
367  fprintf(stderr, "Could not allocate converted input sample pointers\n");
368  return AVERROR(ENOMEM);
369  }
370 
371  /**
372  * Allocate memory for the samples of all channels in one consecutive
373  * block for convenience.
374  */
375  if ((error = av_samples_alloc(*converted_input_samples, NULL,
376  output_codec_context->channels,
377  frame_size,
378  output_codec_context->sample_fmt, 0)) < 0) {
379  fprintf(stderr,
380  "Could not allocate converted input samples (error '%s')\n",
381  get_error_text(error));
382  av_freep(&(*converted_input_samples)[0]);
383  free(*converted_input_samples);
384  return error;
385  }
386  return 0;
387 }
388 
389 /**
390  * Convert the input audio samples into the output sample format.
391  * The conversion happens on a per-frame basis, the size of which is specified
392  * by frame_size.
393  */
394 static int convert_samples(const uint8_t **input_data,
395  uint8_t **converted_data, const int frame_size,
396  SwrContext *resample_context)
397 {
398  int error;
399 
400  /** Convert the samples using the resampler. */
401  if ((error = swr_convert(resample_context,
402  converted_data, frame_size,
403  input_data , frame_size)) < 0) {
404  fprintf(stderr, "Could not convert input samples (error '%s')\n",
405  get_error_text(error));
406  return error;
407  }
408 
409  return 0;
410 }
411 
412 /** Add converted input audio samples to the FIFO buffer for later processing. */
414  uint8_t **converted_input_samples,
415  const int frame_size)
416 {
417  int error;
418 
419  /**
420  * Make the FIFO as large as it needs to be to hold both,
421  * the old and the new samples.
422  */
423  if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
424  fprintf(stderr, "Could not reallocate FIFO\n");
425  return error;
426  }
427 
428  /** Store the new samples in the FIFO buffer. */
429  if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
430  frame_size) < frame_size) {
431  fprintf(stderr, "Could not write data to FIFO\n");
432  return AVERROR_EXIT;
433  }
434  return 0;
435 }
436 
437 /**
438  * Read one audio frame from the input file, decodes, converts and stores
439  * it in the FIFO buffer.
440  */
442  AVFormatContext *input_format_context,
443  AVCodecContext *input_codec_context,
444  AVCodecContext *output_codec_context,
445  SwrContext *resampler_context,
446  int *finished)
447 {
448  /** Temporary storage of the input samples of the frame read from the file. */
449  AVFrame *input_frame = NULL;
450  /** Temporary storage for the converted input samples. */
451  uint8_t **converted_input_samples = NULL;
452  int data_present;
453  int ret = AVERROR_EXIT;
454 
455  /** Initialize temporary storage for one input frame. */
456  if (init_input_frame(&input_frame))
457  goto cleanup;
458  /** Decode one frame worth of audio samples. */
459  if (decode_audio_frame(input_frame, input_format_context,
460  input_codec_context, &data_present, finished))
461  goto cleanup;
462  /**
463  * If we are at the end of the file and there are no more samples
464  * in the decoder which are delayed, we are actually finished.
465  * This must not be treated as an error.
466  */
467  if (*finished && !data_present) {
468  ret = 0;
469  goto cleanup;
470  }
471  /** If there is decoded data, convert and store it */
472  if (data_present) {
473  /** Initialize the temporary storage for the converted input samples. */
474  if (init_converted_samples(&converted_input_samples, output_codec_context,
475  input_frame->nb_samples))
476  goto cleanup;
477 
478  /**
479  * Convert the input samples to the desired output sample format.
480  * This requires a temporary storage provided by converted_input_samples.
481  */
482  if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
483  input_frame->nb_samples, resampler_context))
484  goto cleanup;
485 
486  /** Add the converted input samples to the FIFO buffer for later processing. */
487  if (add_samples_to_fifo(fifo, converted_input_samples,
488  input_frame->nb_samples))
489  goto cleanup;
490  ret = 0;
491  }
492  ret = 0;
493 
494 cleanup:
495  if (converted_input_samples) {
496  av_freep(&converted_input_samples[0]);
497  free(converted_input_samples);
498  }
499  av_frame_free(&input_frame);
500 
501  return ret;
502 }
503 
504 /**
505  * Initialize one input frame for writing to the output file.
506  * The frame will be exactly frame_size samples large.
507  */
509  AVCodecContext *output_codec_context,
510  int frame_size)
511 {
512  int error;
513 
514  /** Create a new frame to store the audio samples. */
515  if (!(*frame = av_frame_alloc())) {
516  fprintf(stderr, "Could not allocate output frame\n");
517  return AVERROR_EXIT;
518  }
519 
520  /**
521  * Set the frame's parameters, especially its size and format.
522  * av_frame_get_buffer needs this to allocate memory for the
523  * audio samples of the frame.
524  * Default channel layouts based on the number of channels
525  * are assumed for simplicity.
526  */
527  (*frame)->nb_samples = frame_size;
528  (*frame)->channel_layout = output_codec_context->channel_layout;
529  (*frame)->format = output_codec_context->sample_fmt;
530  (*frame)->sample_rate = output_codec_context->sample_rate;
531 
532  /**
533  * Allocate the samples of the created frame. This call will make
534  * sure that the audio frame can hold as many samples as specified.
535  */
536  if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
537  fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
538  get_error_text(error));
539  av_frame_free(frame);
540  return error;
541  }
542 
543  return 0;
544 }
545 
546 /** Global timestamp for the audio frames */
547 static int64_t pts = 0;
548 
549 /** Encode one frame worth of audio to the output file. */
551  AVFormatContext *output_format_context,
552  AVCodecContext *output_codec_context,
553  int *data_present)
554 {
555  /** Packet used for temporary storage. */
557  int error;
558  init_packet(&output_packet);
559 
560  /** Set a timestamp based on the sample rate for the container. */
561  if (frame) {
562  frame->pts = pts;
563  pts += frame->nb_samples;
564  }
565 
566  /**
567  * Encode the audio frame and store it in the temporary packet.
568  * The output audio stream encoder is used to do this.
569  */
570  if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
571  frame, data_present)) < 0) {
572  fprintf(stderr, "Could not encode frame (error '%s')\n",
573  get_error_text(error));
574  av_packet_unref(&output_packet);
575  return error;
576  }
577 
578  /** Write one audio frame from the temporary packet to the output file. */
579  if (*data_present) {
580  if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
581  fprintf(stderr, "Could not write frame (error '%s')\n",
582  get_error_text(error));
583  av_packet_unref(&output_packet);
584  return error;
585  }
586 
587  av_packet_unref(&output_packet);
588  }
589 
590  return 0;
591 }
592 
593 /**
594  * Load one audio frame from the FIFO buffer, encode and write it to the
595  * output file.
596  */
598  AVFormatContext *output_format_context,
599  AVCodecContext *output_codec_context)
600 {
601  /** Temporary storage of the output samples of the frame written to the file. */
603  /**
604  * Use the maximum number of possible samples per frame.
605  * If there is less than the maximum possible frame size in the FIFO
606  * buffer use this number. Otherwise, use the maximum possible frame size
607  */
608  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
609  output_codec_context->frame_size);
610  int data_written;
611 
612  /** Initialize temporary storage for one output frame. */
613  if (init_output_frame(&output_frame, output_codec_context, frame_size))
614  return AVERROR_EXIT;
615 
616  /**
617  * Read as many samples from the FIFO buffer as required to fill the frame.
618  * The samples are stored in the frame temporarily.
619  */
620  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
621  fprintf(stderr, "Could not read data from FIFO\n");
622  av_frame_free(&output_frame);
623  return AVERROR_EXIT;
624  }
625 
626  /** Encode one frame worth of audio samples. */
627  if (encode_audio_frame(output_frame, output_format_context,
628  output_codec_context, &data_written)) {
629  av_frame_free(&output_frame);
630  return AVERROR_EXIT;
631  }
632  av_frame_free(&output_frame);
633  return 0;
634 }
635 
636 /** Write the trailer of the output file container. */
637 static int write_output_file_trailer(AVFormatContext *output_format_context)
638 {
639  int error;
640  if ((error = av_write_trailer(output_format_context)) < 0) {
641  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
642  get_error_text(error));
643  return error;
644  }
645  return 0;
646 }
647 
648 /** Convert an audio file to an AAC file in an MP4 container. */
649 int main(int argc, char **argv)
650 {
651  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
652  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
653  SwrContext *resample_context = NULL;
654  AVAudioFifo *fifo = NULL;
655  int ret = AVERROR_EXIT;
656 
657  if (argc < 3) {
658  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
659  exit(1);
660  }
661 
662  /** Register all codecs and formats so that they can be used. */
663  av_register_all();
664  /** Open the input file for reading. */
665  if (open_input_file(argv[1], &input_format_context,
666  &input_codec_context))
667  goto cleanup;
668  /** Open the output file for writing. */
669  if (open_output_file(argv[2], input_codec_context,
670  &output_format_context, &output_codec_context))
671  goto cleanup;
672  /** Initialize the resampler to be able to convert audio sample formats. */
673  if (init_resampler(input_codec_context, output_codec_context,
674  &resample_context))
675  goto cleanup;
676  /** Initialize the FIFO buffer to store audio samples to be encoded. */
677  if (init_fifo(&fifo, output_codec_context))
678  goto cleanup;
679  /** Write the header of the output file container. */
680  if (write_output_file_header(output_format_context))
681  goto cleanup;
682 
683  /**
684  * Loop as long as we have input samples to read or output samples
685  * to write; abort as soon as we have neither.
686  */
687  while (1) {
688  /** Use the encoder's desired frame size for processing. */
689  const int output_frame_size = output_codec_context->frame_size;
690  int finished = 0;
691 
692  /**
693  * Make sure that there is one frame worth of samples in the FIFO
694  * buffer so that the encoder can do its work.
695  * Since the decoder's and the encoder's frame size may differ, we
696  * need to FIFO buffer to store as many frames worth of input samples
697  * that they make up at least one frame worth of output samples.
698  */
699  while (av_audio_fifo_size(fifo) < output_frame_size) {
700  /**
701  * Decode one frame worth of audio samples, convert it to the
702  * output sample format and put it into the FIFO buffer.
703  */
704  if (read_decode_convert_and_store(fifo, input_format_context,
705  input_codec_context,
706  output_codec_context,
707  resample_context, &finished))
708  goto cleanup;
709 
710  /**
711  * If we are at the end of the input file, we continue
712  * encoding the remaining audio samples to the output file.
713  */
714  if (finished)
715  break;
716  }
717 
718  /**
719  * If we have enough samples for the encoder, we encode them.
720  * At the end of the file, we pass the remaining samples to
721  * the encoder.
722  */
723  while (av_audio_fifo_size(fifo) >= output_frame_size ||
724  (finished && av_audio_fifo_size(fifo) > 0))
725  /**
726  * Take one frame worth of audio samples from the FIFO buffer,
727  * encode it and write it to the output file.
728  */
729  if (load_encode_and_write(fifo, output_format_context,
730  output_codec_context))
731  goto cleanup;
732 
733  /**
734  * If we are at the end of the input file and have encoded
735  * all remaining samples, we can exit this loop and finish.
736  */
737  if (finished) {
738  int data_written;
739  /** Flush the encoder as it may have delayed frames. */
740  do {
741  if (encode_audio_frame(NULL, output_format_context,
742  output_codec_context, &data_written))
743  goto cleanup;
744  } while (data_written);
745  break;
746  }
747  }
748 
749  /** Write the trailer of the output file container. */
750  if (write_output_file_trailer(output_format_context))
751  goto cleanup;
752  ret = 0;
753 
754 cleanup:
755  if (fifo)
756  av_audio_fifo_free(fifo);
757  swr_free(&resample_context);
758  if (output_codec_context)
759  avcodec_close(output_codec_context);
760  if (output_format_context) {
761  avio_closep(&output_format_context->pb);
762  avformat_free_context(output_format_context);
763  }
764  if (input_codec_context)
765  avcodec_close(input_codec_context);
766  if (input_format_context)
767  avformat_close_input(&input_format_context);
768 
769  return ret;
770 }
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:926
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2746
#define NULL
Definition: coverity.c:32
const struct AVCodec * codec
Definition: avcodec.h:1541
Bytestream IO Context.
Definition: avio.h:111
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:60
Buffered I/O operations.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:158
This structure describes decoded (raw) audio or video data.
Definition: frame.h:181
AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: utils.c:2610
int main(int argc, char **argv)
Convert an audio file to an AAC file in an MP4 container.
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition: mux.c:777
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
int num
numerator
Definition: rational.h:44
int size
Definition: avcodec.h:1468
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:538
int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Encode a frame of audio.
Definition: utils.c:1707
AVCodec.
Definition: avcodec.h:3392
static void init_packet(AVPacket *packet)
Initialize one data packet for reading or writing.
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 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:61
Format I/O context.
Definition: avformat.h:1314
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.
static int output_packet(AVFormatContext *ctx, int flush)
Definition: mpegenc.c:961
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2295
uint8_t
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:141
AVOptions.
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, decodes, converts and stores it in the FIFO buffer...
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:262
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.
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:3805
static AVFrame * frame
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
Definition: options.c:132
uint8_t * data
Definition: avcodec.h:1467
#define AVERROR_EOF
End of file.
Definition: error.h:55
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
int avcodec_close(AVCodecContext *avctx)
Close a given AVCodecContext and free all the data associated with it (but not the AVCodecContext its...
Definition: utils.c:2529
libswresample public header
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:154
The libswresample context.
simple assert() macros that are a bit more flexible than ISO C assert().
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.
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
Definition: avstring.c:83
reference-counted frame API
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2338
AVCodecContext * codec
Codec context associated with this stream.
Definition: avformat.h:896
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:205
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:97
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:451
#define FFMIN(a, b)
Definition: common.h:96
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 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.
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition: avformat.h:484
struct SwrContext * swr_alloc_set_opts(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, int64_t 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:59
int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, const AVPacket *avpkt)
Decode the audio frame of size avpkt->size from avpkt->data into frame.
Definition: utils.c:2185
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:94
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 output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264.c:1703
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
Stream structure.
Definition: avformat.h:877
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2307
int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align)
Allocate a samples buffer for nb_samples samples, and fill data pointers and linesize accordingly...
Definition: samplefmt.c:171
int frame_size
Definition: mxfenc.c:1821
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
Libavcodec external API header.
#define OUTPUT_BIT_RATE
The output bit rate in kbit/s.
Definition: transcode_aac.c:44
int sample_rate
samples per second
Definition: avcodec.h:2287
main external API structure.
Definition: avcodec.h:1532
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:2629
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:545
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:140
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: utils.c:1170
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:3741
static const char * get_error_text(const int error)
Convert an error code into a text message.
Definition: transcode_aac.c:53
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
Definition: swresample.c:695
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition: utils.c:1509
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:271
static int64_t pts
Global timestamp for the audio frames.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:192
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:113
#define OUTPUT_CHANNELS
The number of output channels.
Definition: transcode_aac.c:46
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition: avcodec.h:783
int av_strerror(int errnum, char *errbuf, size_t errbuf_size)
Put a description of the AVERROR code errnum in errbuf.
Definition: error.c:105
Main libavformat public API header.
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
Definition: utils.c:3139
static int write_output_file_trailer(AVFormatContext *output_format_context)
Write the trailer of the output file container.
void av_init_packet(AVPacket *pkt)
Initialize optional fields of a packet with default values.
Definition: avpacket.c:33
int den
denominator
Definition: rational.h:45
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:3777
Audio FIFO Buffer.
int channels
number of audio channels
Definition: avcodec.h:2288
int avformat_open_input(AVFormatContext **ps, const char *url, AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition: utils.c:422
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1083
#define av_freep(p)
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.
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: avcodec.h:3415
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
Initialize one input frame for writing to the output file.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
Definition: avformat.h:919
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:225
This structure stores compressed data.
Definition: avcodec.h:1444
void av_register_all(void)
Initialize libavformat and register all the muxers, demuxers and protocols.
Definition: allformats.c:51
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:981
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:235
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:155
static av_cold void cleanup(FlashSV2Context *s)
Definition: flashsv2enc.c:127