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transcode_aac.c
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1 /*
2  * Copyright (c) 2013-2017 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
23  * Simple audio converter
24  *
25  * @example transcode_aac.c
26  * Convert an input audio file to AAC in an MP4 container using FFmpeg.
27  * Formats other than MP4 are supported based on the output file extension.
28  * @author Andreas Unterweger (dustsigns@gmail.com)
29  */
30 
31 #include <stdio.h>
32 
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/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  AVCodec *input_codec;
64  int error;
65 
66  /* Open the input file to read from it. */
67  if ((error = avformat_open_input(input_format_context, filename, NULL,
68  NULL)) < 0) {
69  fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
70  filename, av_err2str(error));
71  *input_format_context = NULL;
72  return error;
73  }
74 
75  /* Get information on the input file (number of streams etc.). */
76  if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
77  fprintf(stderr, "Could not open find stream info (error '%s')\n",
78  av_err2str(error));
79  avformat_close_input(input_format_context);
80  return error;
81  }
82 
83  /* Make sure that there is only one stream in the input file. */
84  if ((*input_format_context)->nb_streams != 1) {
85  fprintf(stderr, "Expected one audio input stream, but found %d\n",
86  (*input_format_context)->nb_streams);
87  avformat_close_input(input_format_context);
88  return AVERROR_EXIT;
89  }
90 
91  /* Find a decoder for the audio stream. */
92  if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
93  fprintf(stderr, "Could not find input codec\n");
94  avformat_close_input(input_format_context);
95  return AVERROR_EXIT;
96  }
97 
98  /* Allocate a new decoding context. */
99  avctx = avcodec_alloc_context3(input_codec);
100  if (!avctx) {
101  fprintf(stderr, "Could not allocate a decoding context\n");
102  avformat_close_input(input_format_context);
103  return AVERROR(ENOMEM);
104  }
105 
106  /* Initialize the stream parameters with demuxer information. */
107  error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
108  if (error < 0) {
109  avformat_close_input(input_format_context);
110  avcodec_free_context(&avctx);
111  return error;
112  }
113 
114  /* Open the decoder for the audio stream to use it later. */
115  if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
116  fprintf(stderr, "Could not open input codec (error '%s')\n",
117  av_err2str(error));
118  avcodec_free_context(&avctx);
119  avformat_close_input(input_format_context);
120  return error;
121  }
122 
123  /* Save the decoder context for easier access later. */
124  *input_codec_context = avctx;
125 
126  return 0;
127 }
128 
129 /**
130  * Open an output file and the required encoder.
131  * Also set some basic encoder parameters.
132  * Some of these parameters are based on the input file's parameters.
133  * @param filename File to be opened
134  * @param input_codec_context Codec context of input file
135  * @param[out] output_format_context Format context of output file
136  * @param[out] output_codec_context Codec context of output file
137  * @return Error code (0 if successful)
138  */
139 static int open_output_file(const char *filename,
140  AVCodecContext *input_codec_context,
141  AVFormatContext **output_format_context,
142  AVCodecContext **output_codec_context)
143 {
144  AVCodecContext *avctx = NULL;
145  AVIOContext *output_io_context = NULL;
146  AVStream *stream = NULL;
147  AVCodec *output_codec = NULL;
148  int error;
149 
150  /* Open the output file to write to it. */
151  if ((error = avio_open(&output_io_context, filename,
152  AVIO_FLAG_WRITE)) < 0) {
153  fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
154  filename, av_err2str(error));
155  return error;
156  }
157 
158  /* Create a new format context for the output container format. */
159  if (!(*output_format_context = avformat_alloc_context())) {
160  fprintf(stderr, "Could not allocate output format context\n");
161  return AVERROR(ENOMEM);
162  }
163 
164  /* Associate the output file (pointer) with the container format context. */
165  (*output_format_context)->pb = output_io_context;
166 
167  /* Guess the desired container format based on the file extension. */
168  if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
169  NULL))) {
170  fprintf(stderr, "Could not find output file format\n");
171  goto cleanup;
172  }
173 
174  av_strlcpy((*output_format_context)->filename, filename,
175  sizeof((*output_format_context)->filename));
176 
177  /* Find the encoder to be used by its name. */
178  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
179  fprintf(stderr, "Could not find an AAC encoder.\n");
180  goto cleanup;
181  }
182 
183  /* Create a new audio stream in the output file container. */
184  if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
185  fprintf(stderr, "Could not create new stream\n");
186  error = AVERROR(ENOMEM);
187  goto cleanup;
188  }
189 
190  avctx = avcodec_alloc_context3(output_codec);
191  if (!avctx) {
192  fprintf(stderr, "Could not allocate an encoding context\n");
193  error = AVERROR(ENOMEM);
194  goto cleanup;
195  }
196 
197  /* Set the basic encoder parameters.
198  * The input file's sample rate is used to avoid a sample rate conversion. */
199  avctx->channels = OUTPUT_CHANNELS;
201  avctx->sample_rate = input_codec_context->sample_rate;
202  avctx->sample_fmt = output_codec->sample_fmts[0];
203  avctx->bit_rate = OUTPUT_BIT_RATE;
204 
205  /* Allow the use of the experimental AAC encoder. */
207 
208  /* Set the sample rate for the container. */
209  stream->time_base.den = input_codec_context->sample_rate;
210  stream->time_base.num = 1;
211 
212  /* Some container formats (like MP4) require global headers to be present.
213  * Mark the encoder so that it behaves accordingly. */
214  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
216 
217  /* Open the encoder for the audio stream to use it later. */
218  if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
219  fprintf(stderr, "Could not open output codec (error '%s')\n",
220  av_err2str(error));
221  goto cleanup;
222  }
223 
224  error = avcodec_parameters_from_context(stream->codecpar, avctx);
225  if (error < 0) {
226  fprintf(stderr, "Could not initialize stream parameters\n");
227  goto cleanup;
228  }
229 
230  /* Save the encoder context for easier access later. */
231  *output_codec_context = avctx;
232 
233  return 0;
234 
235 cleanup:
236  avcodec_free_context(&avctx);
237  avio_closep(&(*output_format_context)->pb);
238  avformat_free_context(*output_format_context);
239  *output_format_context = NULL;
240  return error < 0 ? error : AVERROR_EXIT;
241 }
242 
243 /**
244  * Initialize one data packet for reading or writing.
245  * @param packet Packet to be initialized
246  */
247 static void init_packet(AVPacket *packet)
248 {
249  av_init_packet(packet);
250  /* Set the packet data and size so that it is recognized as being empty. */
251  packet->data = NULL;
252  packet->size = 0;
253 }
254 
255 /**
256  * Initialize one audio frame for reading from the input file.
257  * @param[out] frame Frame to be initialized
258  * @return Error code (0 if successful)
259  */
261 {
262  if (!(*frame = av_frame_alloc())) {
263  fprintf(stderr, "Could not allocate input frame\n");
264  return AVERROR(ENOMEM);
265  }
266  return 0;
267 }
268 
269 /**
270  * Initialize the audio resampler based on the input and output codec settings.
271  * If the input and output sample formats differ, a conversion is required
272  * libswresample takes care of this, but requires initialization.
273  * @param input_codec_context Codec context of the input file
274  * @param output_codec_context Codec context of the output file
275  * @param[out] resample_context Resample context for the required conversion
276  * @return Error code (0 if successful)
277  */
278 static int init_resampler(AVCodecContext *input_codec_context,
279  AVCodecContext *output_codec_context,
280  SwrContext **resample_context)
281 {
282  int error;
283 
284  /*
285  * Create a resampler context for the conversion.
286  * Set the conversion parameters.
287  * Default channel layouts based on the number of channels
288  * are assumed for simplicity (they are sometimes not detected
289  * properly by the demuxer and/or decoder).
290  */
291  *resample_context = swr_alloc_set_opts(NULL,
292  av_get_default_channel_layout(output_codec_context->channels),
293  output_codec_context->sample_fmt,
294  output_codec_context->sample_rate,
295  av_get_default_channel_layout(input_codec_context->channels),
296  input_codec_context->sample_fmt,
297  input_codec_context->sample_rate,
298  0, NULL);
299  if (!*resample_context) {
300  fprintf(stderr, "Could not allocate resample context\n");
301  return AVERROR(ENOMEM);
302  }
303  /*
304  * Perform a sanity check so that the number of converted samples is
305  * not greater than the number of samples to be converted.
306  * If the sample rates differ, this case has to be handled differently
307  */
308  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
309 
310  /* Open the resampler with the specified parameters. */
311  if ((error = swr_init(*resample_context)) < 0) {
312  fprintf(stderr, "Could not open resample context\n");
313  swr_free(resample_context);
314  return error;
315  }
316  return 0;
317 }
318 
319 /**
320  * Initialize a FIFO buffer for the audio samples to be encoded.
321  * @param[out] fifo Sample buffer
322  * @param output_codec_context Codec context of the output file
323  * @return Error code (0 if successful)
324  */
325 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
326 {
327  /* Create the FIFO buffer based on the specified output sample format. */
328  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
329  output_codec_context->channels, 1))) {
330  fprintf(stderr, "Could not allocate FIFO\n");
331  return AVERROR(ENOMEM);
332  }
333  return 0;
334 }
335 
336 /**
337  * Write the header of the output file container.
338  * @param output_format_context Format context of the output file
339  * @return Error code (0 if successful)
340  */
341 static int write_output_file_header(AVFormatContext *output_format_context)
342 {
343  int error;
344  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
345  fprintf(stderr, "Could not write output file header (error '%s')\n",
346  av_err2str(error));
347  return error;
348  }
349  return 0;
350 }
351 
352 /**
353  * Decode one audio frame from the input file.
354  * @param frame Audio frame to be decoded
355  * @param input_format_context Format context of the input file
356  * @param input_codec_context Codec context of the input file
357  * @param[out] data_present Indicates whether data has been decoded
358  * @param[out] finished Indicates whether the end of file has
359  * been reached and all data has been
360  * decoded. If this flag is false, there
361  * is more data to be decoded, i.e., this
362  * function has to be called again.
363  * @return Error code (0 if successful)
364  */
366  AVFormatContext *input_format_context,
367  AVCodecContext *input_codec_context,
368  int *data_present, int *finished)
369 {
370  /* Packet used for temporary storage. */
371  AVPacket input_packet;
372  int error;
373  init_packet(&input_packet);
374 
375  /* Read one audio frame from the input file into a temporary packet. */
376  if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
377  /* If we are at the end of the file, flush the decoder below. */
378  if (error == AVERROR_EOF)
379  *finished = 1;
380  else {
381  fprintf(stderr, "Could not read frame (error '%s')\n",
382  av_err2str(error));
383  return error;
384  }
385  }
386 
387  /* Decode the audio frame stored in the temporary packet.
388  * The input audio stream decoder is used to do this.
389  * If we are at the end of the file, pass an empty packet to the decoder
390  * to flush it. */
391  if ((error = avcodec_decode_audio4(input_codec_context, frame,
392  data_present, &input_packet)) < 0) {
393  fprintf(stderr, "Could not decode frame (error '%s')\n",
394  av_err2str(error));
395  av_packet_unref(&input_packet);
396  return error;
397  }
398 
399  /* If the decoder has not been flushed completely, we are not finished,
400  * so that this function has to be called again. */
401  if (*finished && *data_present)
402  *finished = 0;
403  av_packet_unref(&input_packet);
404  return 0;
405 }
406 
407 /**
408  * Initialize a temporary storage for the specified number of audio samples.
409  * The conversion requires temporary storage due to the different format.
410  * The number of audio samples to be allocated is specified in frame_size.
411  * @param[out] converted_input_samples Array of converted samples. The
412  * dimensions are reference, channel
413  * (for multi-channel audio), sample.
414  * @param output_codec_context Codec context of the output file
415  * @param frame_size Number of samples to be converted in
416  * each round
417  * @return Error code (0 if successful)
418  */
419 static int init_converted_samples(uint8_t ***converted_input_samples,
420  AVCodecContext *output_codec_context,
421  int frame_size)
422 {
423  int error;
424 
425  /* Allocate as many pointers as there are audio channels.
426  * Each pointer will later point to the audio samples of the corresponding
427  * channels (although it may be NULL for interleaved formats).
428  */
429  if (!(*converted_input_samples = calloc(output_codec_context->channels,
430  sizeof(**converted_input_samples)))) {
431  fprintf(stderr, "Could not allocate converted input sample pointers\n");
432  return AVERROR(ENOMEM);
433  }
434 
435  /* Allocate memory for the samples of all channels in one consecutive
436  * block for convenience. */
437  if ((error = av_samples_alloc(*converted_input_samples, NULL,
438  output_codec_context->channels,
439  frame_size,
440  output_codec_context->sample_fmt, 0)) < 0) {
441  fprintf(stderr,
442  "Could not allocate converted input samples (error '%s')\n",
443  av_err2str(error));
444  av_freep(&(*converted_input_samples)[0]);
445  free(*converted_input_samples);
446  return error;
447  }
448  return 0;
449 }
450 
451 /**
452  * Convert the input audio samples into the output sample format.
453  * The conversion happens on a per-frame basis, the size of which is
454  * specified by frame_size.
455  * @param input_data Samples to be decoded. The dimensions are
456  * channel (for multi-channel audio), sample.
457  * @param[out] converted_data Converted samples. The dimensions are channel
458  * (for multi-channel audio), sample.
459  * @param frame_size Number of samples to be converted
460  * @param resample_context Resample context for the conversion
461  * @return Error code (0 if successful)
462  */
463 static int convert_samples(const uint8_t **input_data,
464  uint8_t **converted_data, const int frame_size,
465  SwrContext *resample_context)
466 {
467  int error;
468 
469  /* Convert the samples using the resampler. */
470  if ((error = swr_convert(resample_context,
471  converted_data, frame_size,
472  input_data , frame_size)) < 0) {
473  fprintf(stderr, "Could not convert input samples (error '%s')\n",
474  av_err2str(error));
475  return error;
476  }
477 
478  return 0;
479 }
480 
481 /**
482  * Add converted input audio samples to the FIFO buffer for later processing.
483  * @param fifo Buffer to add the samples to
484  * @param converted_input_samples Samples to be added. The dimensions are channel
485  * (for multi-channel audio), sample.
486  * @param frame_size Number of samples to be converted
487  * @return Error code (0 if successful)
488  */
490  uint8_t **converted_input_samples,
491  const int frame_size)
492 {
493  int error;
494 
495  /* Make the FIFO as large as it needs to be to hold both,
496  * the old and the new samples. */
497  if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
498  fprintf(stderr, "Could not reallocate FIFO\n");
499  return error;
500  }
501 
502  /* Store the new samples in the FIFO buffer. */
503  if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
504  frame_size) < frame_size) {
505  fprintf(stderr, "Could not write data to FIFO\n");
506  return AVERROR_EXIT;
507  }
508  return 0;
509 }
510 
511 /**
512  * Read one audio frame from the input file, decode, convert and store
513  * it in the FIFO buffer.
514  * @param fifo Buffer used for temporary storage
515  * @param input_format_context Format context of the input file
516  * @param input_codec_context Codec context of the input file
517  * @param output_codec_context Codec context of the output file
518  * @param resampler_context Resample context for the conversion
519  * @param[out] finished Indicates whether the end of file has
520  * been reached and all data has been
521  * decoded. If this flag is false,
522  * there is more data to be decoded,
523  * i.e., this function has to be called
524  * again.
525  * @return Error code (0 if successful)
526  */
528  AVFormatContext *input_format_context,
529  AVCodecContext *input_codec_context,
530  AVCodecContext *output_codec_context,
531  SwrContext *resampler_context,
532  int *finished)
533 {
534  /* Temporary storage of the input samples of the frame read from the file. */
535  AVFrame *input_frame = NULL;
536  /* Temporary storage for the converted input samples. */
537  uint8_t **converted_input_samples = NULL;
538  int data_present;
539  int ret = AVERROR_EXIT;
540 
541  /* Initialize temporary storage for one input frame. */
542  if (init_input_frame(&input_frame))
543  goto cleanup;
544  /* Decode one frame worth of audio samples. */
545  if (decode_audio_frame(input_frame, input_format_context,
546  input_codec_context, &data_present, finished))
547  goto cleanup;
548  /* If we are at the end of the file and there are no more samples
549  * in the decoder which are delayed, we are actually finished.
550  * This must not be treated as an error. */
551  if (*finished && !data_present) {
552  ret = 0;
553  goto cleanup;
554  }
555  /* If there is decoded data, convert and store it. */
556  if (data_present) {
557  /* Initialize the temporary storage for the converted input samples. */
558  if (init_converted_samples(&converted_input_samples, output_codec_context,
559  input_frame->nb_samples))
560  goto cleanup;
561 
562  /* Convert the input samples to the desired output sample format.
563  * This requires a temporary storage provided by converted_input_samples. */
564  if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
565  input_frame->nb_samples, resampler_context))
566  goto cleanup;
567 
568  /* Add the converted input samples to the FIFO buffer for later processing. */
569  if (add_samples_to_fifo(fifo, converted_input_samples,
570  input_frame->nb_samples))
571  goto cleanup;
572  ret = 0;
573  }
574  ret = 0;
575 
576 cleanup:
577  if (converted_input_samples) {
578  av_freep(&converted_input_samples[0]);
579  free(converted_input_samples);
580  }
581  av_frame_free(&input_frame);
582 
583  return ret;
584 }
585 
586 /**
587  * Initialize one input frame for writing to the output file.
588  * The frame will be exactly frame_size samples large.
589  * @param[out] frame Frame to be initialized
590  * @param output_codec_context Codec context of the output file
591  * @param frame_size Size of the frame
592  * @return Error code (0 if successful)
593  */
595  AVCodecContext *output_codec_context,
596  int frame_size)
597 {
598  int error;
599 
600  /* Create a new frame to store the audio samples. */
601  if (!(*frame = av_frame_alloc())) {
602  fprintf(stderr, "Could not allocate output frame\n");
603  return AVERROR_EXIT;
604  }
605 
606  /* Set the frame's parameters, especially its size and format.
607  * av_frame_get_buffer needs this to allocate memory for the
608  * audio samples of the frame.
609  * Default channel layouts based on the number of channels
610  * are assumed for simplicity. */
611  (*frame)->nb_samples = frame_size;
612  (*frame)->channel_layout = output_codec_context->channel_layout;
613  (*frame)->format = output_codec_context->sample_fmt;
614  (*frame)->sample_rate = output_codec_context->sample_rate;
615 
616  /* Allocate the samples of the created frame. This call will make
617  * sure that the audio frame can hold as many samples as specified. */
618  if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
619  fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
620  av_err2str(error));
621  av_frame_free(frame);
622  return error;
623  }
624 
625  return 0;
626 }
627 
628 /* Global timestamp for the audio frames. */
629 static int64_t pts = 0;
630 
631 /**
632  * Encode one frame worth of audio to the output file.
633  * @param frame Samples to be encoded
634  * @param output_format_context Format context of the output file
635  * @param output_codec_context Codec context of the output file
636  * @param[out] data_present Indicates whether data has been
637  * decoded
638  * @return Error code (0 if successful)
639  */
641  AVFormatContext *output_format_context,
642  AVCodecContext *output_codec_context,
643  int *data_present)
644 {
645  /* Packet used for temporary storage. */
647  int error;
648  init_packet(&output_packet);
649 
650  /* Set a timestamp based on the sample rate for the container. */
651  if (frame) {
652  frame->pts = pts;
653  pts += frame->nb_samples;
654  }
655 
656  /* Encode the audio frame and store it in the temporary packet.
657  * The output audio stream encoder is used to do this. */
658  if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
659  frame, data_present)) < 0) {
660  fprintf(stderr, "Could not encode frame (error '%s')\n",
661  av_err2str(error));
662  av_packet_unref(&output_packet);
663  return error;
664  }
665 
666  /* Write one audio frame from the temporary packet to the output file. */
667  if (*data_present) {
668  if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
669  fprintf(stderr, "Could not write frame (error '%s')\n",
670  av_err2str(error));
671  av_packet_unref(&output_packet);
672  return error;
673  }
674 
675  av_packet_unref(&output_packet);
676  }
677 
678  return 0;
679 }
680 
681 /**
682  * Load one audio frame from the FIFO buffer, encode and write it to the
683  * output file.
684  * @param fifo Buffer used for temporary storage
685  * @param output_format_context Format context of the output file
686  * @param output_codec_context Codec context of the output file
687  * @return Error code (0 if successful)
688  */
690  AVFormatContext *output_format_context,
691  AVCodecContext *output_codec_context)
692 {
693  /* Temporary storage of the output samples of the frame written to the file. */
695  /* Use the maximum number of possible samples per frame.
696  * If there is less than the maximum possible frame size in the FIFO
697  * buffer use this number. Otherwise, use the maximum possible frame size. */
698  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
699  output_codec_context->frame_size);
700  int data_written;
701 
702  /* Initialize temporary storage for one output frame. */
703  if (init_output_frame(&output_frame, output_codec_context, frame_size))
704  return AVERROR_EXIT;
705 
706  /* Read as many samples from the FIFO buffer as required to fill the frame.
707  * The samples are stored in the frame temporarily. */
708  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
709  fprintf(stderr, "Could not read data from FIFO\n");
710  av_frame_free(&output_frame);
711  return AVERROR_EXIT;
712  }
713 
714  /* Encode one frame worth of audio samples. */
715  if (encode_audio_frame(output_frame, output_format_context,
716  output_codec_context, &data_written)) {
717  av_frame_free(&output_frame);
718  return AVERROR_EXIT;
719  }
720  av_frame_free(&output_frame);
721  return 0;
722 }
723 
724 /**
725  * Write the trailer of the output file container.
726  * @param output_format_context Format context of the output file
727  * @return Error code (0 if successful)
728  */
729 static int write_output_file_trailer(AVFormatContext *output_format_context)
730 {
731  int error;
732  if ((error = av_write_trailer(output_format_context)) < 0) {
733  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
734  av_err2str(error));
735  return error;
736  }
737  return 0;
738 }
739 
740 int main(int argc, char **argv)
741 {
742  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
743  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
744  SwrContext *resample_context = NULL;
745  AVAudioFifo *fifo = NULL;
746  int ret = AVERROR_EXIT;
747 
748  if (argc != 3) {
749  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
750  exit(1);
751  }
752 
753  /* Register all codecs and formats so that they can be used. */
754  av_register_all();
755  /* Open the input file for reading. */
756  if (open_input_file(argv[1], &input_format_context,
757  &input_codec_context))
758  goto cleanup;
759  /* Open the output file for writing. */
760  if (open_output_file(argv[2], input_codec_context,
761  &output_format_context, &output_codec_context))
762  goto cleanup;
763  /* Initialize the resampler to be able to convert audio sample formats. */
764  if (init_resampler(input_codec_context, output_codec_context,
765  &resample_context))
766  goto cleanup;
767  /* Initialize the FIFO buffer to store audio samples to be encoded. */
768  if (init_fifo(&fifo, output_codec_context))
769  goto cleanup;
770  /* Write the header of the output file container. */
771  if (write_output_file_header(output_format_context))
772  goto cleanup;
773 
774  /* Loop as long as we have input samples to read or output samples
775  * to write; abort as soon as we have neither. */
776  while (1) {
777  /* Use the encoder's desired frame size for processing. */
778  const int output_frame_size = output_codec_context->frame_size;
779  int finished = 0;
780 
781  /* Make sure that there is one frame worth of samples in the FIFO
782  * buffer so that the encoder can do its work.
783  * Since the decoder's and the encoder's frame size may differ, we
784  * need to FIFO buffer to store as many frames worth of input samples
785  * that they make up at least one frame worth of output samples. */
786  while (av_audio_fifo_size(fifo) < output_frame_size) {
787  /* Decode one frame worth of audio samples, convert it to the
788  * output sample format and put it into the FIFO buffer. */
789  if (read_decode_convert_and_store(fifo, input_format_context,
790  input_codec_context,
791  output_codec_context,
792  resample_context, &finished))
793  goto cleanup;
794 
795  /* If we are at the end of the input file, we continue
796  * encoding the remaining audio samples to the output file. */
797  if (finished)
798  break;
799  }
800 
801  /* If we have enough samples for the encoder, we encode them.
802  * At the end of the file, we pass the remaining samples to
803  * the encoder. */
804  while (av_audio_fifo_size(fifo) >= output_frame_size ||
805  (finished && av_audio_fifo_size(fifo) > 0))
806  /* Take one frame worth of audio samples from the FIFO buffer,
807  * encode it and write it to the output file. */
808  if (load_encode_and_write(fifo, output_format_context,
809  output_codec_context))
810  goto cleanup;
811 
812  /* If we are at the end of the input file and have encoded
813  * all remaining samples, we can exit this loop and finish. */
814  if (finished) {
815  int data_written;
816  /* Flush the encoder as it may have delayed frames. */
817  do {
818  if (encode_audio_frame(NULL, output_format_context,
819  output_codec_context, &data_written))
820  goto cleanup;
821  } while (data_written);
822  break;
823  }
824  }
825 
826  /* Write the trailer of the output file container. */
827  if (write_output_file_trailer(output_format_context))
828  goto cleanup;
829  ret = 0;
830 
831 cleanup:
832  if (fifo)
833  av_audio_fifo_free(fifo);
834  swr_free(&resample_context);
835  if (output_codec_context)
836  avcodec_free_context(&output_codec_context);
837  if (output_format_context) {
838  avio_closep(&output_format_context->pb);
839  avformat_free_context(output_format_context);
840  }
841  if (input_codec_context)
842  avcodec_free_context(&input_codec_context);
843  if (input_format_context)
844  avformat_close_input(&input_format_context);
845 
846  return ret;
847 }
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:1098
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2551
#define NULL
Definition: coverity.c:32
static void output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof)
Definition: ffmpeg.c:838
Bytestream IO Context.
Definition: avio.h:161
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
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:181
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: utils.c:1232
int main(int argc, char **argv)
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition: mux.c:859
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1538
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
int num
Numerator.
Definition: rational.h:59
int size
Definition: avcodec.h:1401
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:655
attribute_deprecated int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Encode a frame of audio.
Definition: encode.c:118
attribute_deprecated 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: decode.c:833
AVCodec.
Definition: avcodec.h:3351
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:58
Format I/O context.
Definition: avformat.h:1325
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 av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2151
uint8_t
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:150
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, decode, convert and store 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:294
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:4378
int avcodec_parameters_to_context(AVCodecContext *codec, const AVCodecParameters *par)
Fill the codec context based on the values from the supplied codec parameters.
Definition: utils.c:2306
static AVFrame * frame
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
Definition: options.c:144
static void input_data(MLPEncodeContext *ctx, void *samples)
Wrapper function for inputting data in two different bit-depths.
Definition: mlpenc.c:1275
uint8_t * data
Definition: avcodec.h:1400
#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.
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:163
The libswresample context.
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1568
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:2194
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:228
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:96
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:489
#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.
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition: options.c:157
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:468
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: error.h:119
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
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:98
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.
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
static void error(const char *err)
Stream structure.
Definition: avformat.h:872
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2163
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:173
int frame_size
Definition: mxfenc.c:1947
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.
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:172
#define OUTPUT_BIT_RATE
Definition: transcode_aac.c:47
int sample_rate
samples per second
Definition: avcodec.h:2143
main external API structure.
Definition: avcodec.h:1488
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: utils.c:1251
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:590
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264dec.c:837
int avcodec_parameters_from_context(AVCodecParameters *par, const AVCodecContext *codec)
Fill the parameters struct based on the values from the supplied codec context.
Definition: utils.c:2249
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: utils.c:614
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4313
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:706
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition: utils.c:1712
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:283
static int64_t pts
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:215
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
#define OUTPUT_CHANNELS
Definition: transcode_aac.c:49
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition: avcodec.h:879
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:3497
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:60
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4350
Audio FIFO Buffer.
int channels
number of audio channels
Definition: avcodec.h:2144
int avformat_open_input(AVFormatContext **ps, const char *url, AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition: utils.c:515
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:1228
#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.
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1014
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: avcodec.h:3374
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:901
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
This structure stores compressed data.
Definition: avcodec.h:1377
void av_register_all(void)
Initialize libavformat and register all the muxers, demuxers and protocols.
Definition: allformats.c:395
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:1159
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2546
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
static av_cold void cleanup(FlashSV2Context *s)
Definition: flashsv2enc.c:127