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transcode_aac.c
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1 /*
2  * Copyright (c) 2013-2018 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  if (!((*output_format_context)->url = av_strdup(filename))) {
175  fprintf(stderr, "Could not allocate url.\n");
176  error = AVERROR(ENOMEM);
177  goto cleanup;
178  }
179 
180  /* Find the encoder to be used by its name. */
181  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
182  fprintf(stderr, "Could not find an AAC encoder.\n");
183  goto cleanup;
184  }
185 
186  /* Create a new audio stream in the output file container. */
187  if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
188  fprintf(stderr, "Could not create new stream\n");
189  error = AVERROR(ENOMEM);
190  goto cleanup;
191  }
192 
193  avctx = avcodec_alloc_context3(output_codec);
194  if (!avctx) {
195  fprintf(stderr, "Could not allocate an encoding context\n");
196  error = AVERROR(ENOMEM);
197  goto cleanup;
198  }
199 
200  /* Set the basic encoder parameters.
201  * The input file's sample rate is used to avoid a sample rate conversion. */
202  avctx->channels = OUTPUT_CHANNELS;
204  avctx->sample_rate = input_codec_context->sample_rate;
205  avctx->sample_fmt = output_codec->sample_fmts[0];
206  avctx->bit_rate = OUTPUT_BIT_RATE;
207 
208  /* Allow the use of the experimental AAC encoder. */
210 
211  /* Set the sample rate for the container. */
212  stream->time_base.den = input_codec_context->sample_rate;
213  stream->time_base.num = 1;
214 
215  /* Some container formats (like MP4) require global headers to be present.
216  * Mark the encoder so that it behaves accordingly. */
217  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
219 
220  /* Open the encoder for the audio stream to use it later. */
221  if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
222  fprintf(stderr, "Could not open output codec (error '%s')\n",
223  av_err2str(error));
224  goto cleanup;
225  }
226 
227  error = avcodec_parameters_from_context(stream->codecpar, avctx);
228  if (error < 0) {
229  fprintf(stderr, "Could not initialize stream parameters\n");
230  goto cleanup;
231  }
232 
233  /* Save the encoder context for easier access later. */
234  *output_codec_context = avctx;
235 
236  return 0;
237 
238 cleanup:
239  avcodec_free_context(&avctx);
240  avio_closep(&(*output_format_context)->pb);
241  avformat_free_context(*output_format_context);
242  *output_format_context = NULL;
243  return error < 0 ? error : AVERROR_EXIT;
244 }
245 
246 /**
247  * Initialize one data packet for reading or writing.
248  * @param packet Packet to be initialized
249  */
250 static void init_packet(AVPacket *packet)
251 {
252  av_init_packet(packet);
253  /* Set the packet data and size so that it is recognized as being empty. */
254  packet->data = NULL;
255  packet->size = 0;
256 }
257 
258 /**
259  * Initialize one audio frame for reading from the input file.
260  * @param[out] frame Frame to be initialized
261  * @return Error code (0 if successful)
262  */
264 {
265  if (!(*frame = av_frame_alloc())) {
266  fprintf(stderr, "Could not allocate input frame\n");
267  return AVERROR(ENOMEM);
268  }
269  return 0;
270 }
271 
272 /**
273  * Initialize the audio resampler based on the input and output codec settings.
274  * If the input and output sample formats differ, a conversion is required
275  * libswresample takes care of this, but requires initialization.
276  * @param input_codec_context Codec context of the input file
277  * @param output_codec_context Codec context of the output file
278  * @param[out] resample_context Resample context for the required conversion
279  * @return Error code (0 if successful)
280  */
281 static int init_resampler(AVCodecContext *input_codec_context,
282  AVCodecContext *output_codec_context,
283  SwrContext **resample_context)
284 {
285  int error;
286 
287  /*
288  * Create a resampler context for the conversion.
289  * Set the conversion parameters.
290  * Default channel layouts based on the number of channels
291  * are assumed for simplicity (they are sometimes not detected
292  * properly by the demuxer and/or decoder).
293  */
294  *resample_context = swr_alloc_set_opts(NULL,
295  av_get_default_channel_layout(output_codec_context->channels),
296  output_codec_context->sample_fmt,
297  output_codec_context->sample_rate,
298  av_get_default_channel_layout(input_codec_context->channels),
299  input_codec_context->sample_fmt,
300  input_codec_context->sample_rate,
301  0, NULL);
302  if (!*resample_context) {
303  fprintf(stderr, "Could not allocate resample context\n");
304  return AVERROR(ENOMEM);
305  }
306  /*
307  * Perform a sanity check so that the number of converted samples is
308  * not greater than the number of samples to be converted.
309  * If the sample rates differ, this case has to be handled differently
310  */
311  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
312 
313  /* Open the resampler with the specified parameters. */
314  if ((error = swr_init(*resample_context)) < 0) {
315  fprintf(stderr, "Could not open resample context\n");
316  swr_free(resample_context);
317  return error;
318  }
319  return 0;
320 }
321 
322 /**
323  * Initialize a FIFO buffer for the audio samples to be encoded.
324  * @param[out] fifo Sample buffer
325  * @param output_codec_context Codec context of the output file
326  * @return Error code (0 if successful)
327  */
328 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
329 {
330  /* Create the FIFO buffer based on the specified output sample format. */
331  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
332  output_codec_context->channels, 1))) {
333  fprintf(stderr, "Could not allocate FIFO\n");
334  return AVERROR(ENOMEM);
335  }
336  return 0;
337 }
338 
339 /**
340  * Write the header of the output file container.
341  * @param output_format_context Format context of the output file
342  * @return Error code (0 if successful)
343  */
344 static int write_output_file_header(AVFormatContext *output_format_context)
345 {
346  int error;
347  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
348  fprintf(stderr, "Could not write output file header (error '%s')\n",
349  av_err2str(error));
350  return error;
351  }
352  return 0;
353 }
354 
355 /**
356  * Decode one audio frame from the input file.
357  * @param frame Audio frame to be decoded
358  * @param input_format_context Format context of the input file
359  * @param input_codec_context Codec context of the input file
360  * @param[out] data_present Indicates whether data has been decoded
361  * @param[out] finished Indicates whether the end of file has
362  * been reached and all data has been
363  * decoded. If this flag is false, there
364  * is more data to be decoded, i.e., this
365  * function has to be called again.
366  * @return Error code (0 if successful)
367  */
369  AVFormatContext *input_format_context,
370  AVCodecContext *input_codec_context,
371  int *data_present, int *finished)
372 {
373  /* Packet used for temporary storage. */
374  AVPacket input_packet;
375  int error;
376  init_packet(&input_packet);
377 
378  /* Read one audio frame from the input file into a temporary packet. */
379  if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
380  /* If we are at the end of the file, flush the decoder below. */
381  if (error == AVERROR_EOF)
382  *finished = 1;
383  else {
384  fprintf(stderr, "Could not read frame (error '%s')\n",
385  av_err2str(error));
386  return error;
387  }
388  }
389 
390  /* Send the audio frame stored in the temporary packet to the decoder.
391  * The input audio stream decoder is used to do this. */
392  if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
393  fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
394  av_err2str(error));
395  return error;
396  }
397 
398  /* Receive one frame from the decoder. */
399  error = avcodec_receive_frame(input_codec_context, frame);
400  /* If the decoder asks for more data to be able to decode a frame,
401  * return indicating that no data is present. */
402  if (error == AVERROR(EAGAIN)) {
403  error = 0;
404  goto cleanup;
405  /* If the end of the input file is reached, stop decoding. */
406  } else if (error == AVERROR_EOF) {
407  *finished = 1;
408  error = 0;
409  goto cleanup;
410  } else if (error < 0) {
411  fprintf(stderr, "Could not decode frame (error '%s')\n",
412  av_err2str(error));
413  goto cleanup;
414  /* Default case: Return decoded data. */
415  } else {
416  *data_present = 1;
417  goto cleanup;
418  }
419 
420 cleanup:
421  av_packet_unref(&input_packet);
422  return error;
423 }
424 
425 /**
426  * Initialize a temporary storage for the specified number of audio samples.
427  * The conversion requires temporary storage due to the different format.
428  * The number of audio samples to be allocated is specified in frame_size.
429  * @param[out] converted_input_samples Array of converted samples. The
430  * dimensions are reference, channel
431  * (for multi-channel audio), sample.
432  * @param output_codec_context Codec context of the output file
433  * @param frame_size Number of samples to be converted in
434  * each round
435  * @return Error code (0 if successful)
436  */
437 static int init_converted_samples(uint8_t ***converted_input_samples,
438  AVCodecContext *output_codec_context,
439  int frame_size)
440 {
441  int error;
442 
443  /* Allocate as many pointers as there are audio channels.
444  * Each pointer will later point to the audio samples of the corresponding
445  * channels (although it may be NULL for interleaved formats).
446  */
447  if (!(*converted_input_samples = calloc(output_codec_context->channels,
448  sizeof(**converted_input_samples)))) {
449  fprintf(stderr, "Could not allocate converted input sample pointers\n");
450  return AVERROR(ENOMEM);
451  }
452 
453  /* Allocate memory for the samples of all channels in one consecutive
454  * block for convenience. */
455  if ((error = av_samples_alloc(*converted_input_samples, NULL,
456  output_codec_context->channels,
457  frame_size,
458  output_codec_context->sample_fmt, 0)) < 0) {
459  fprintf(stderr,
460  "Could not allocate converted input samples (error '%s')\n",
461  av_err2str(error));
462  av_freep(&(*converted_input_samples)[0]);
463  free(*converted_input_samples);
464  return error;
465  }
466  return 0;
467 }
468 
469 /**
470  * Convert the input audio samples into the output sample format.
471  * The conversion happens on a per-frame basis, the size of which is
472  * specified by frame_size.
473  * @param input_data Samples to be decoded. The dimensions are
474  * channel (for multi-channel audio), sample.
475  * @param[out] converted_data Converted samples. The dimensions are channel
476  * (for multi-channel audio), sample.
477  * @param frame_size Number of samples to be converted
478  * @param resample_context Resample context for the conversion
479  * @return Error code (0 if successful)
480  */
481 static int convert_samples(const uint8_t **input_data,
482  uint8_t **converted_data, const int frame_size,
483  SwrContext *resample_context)
484 {
485  int error;
486 
487  /* Convert the samples using the resampler. */
488  if ((error = swr_convert(resample_context,
489  converted_data, frame_size,
490  input_data , frame_size)) < 0) {
491  fprintf(stderr, "Could not convert input samples (error '%s')\n",
492  av_err2str(error));
493  return error;
494  }
495 
496  return 0;
497 }
498 
499 /**
500  * Add converted input audio samples to the FIFO buffer for later processing.
501  * @param fifo Buffer to add the samples to
502  * @param converted_input_samples Samples to be added. The dimensions are channel
503  * (for multi-channel audio), sample.
504  * @param frame_size Number of samples to be converted
505  * @return Error code (0 if successful)
506  */
508  uint8_t **converted_input_samples,
509  const int frame_size)
510 {
511  int error;
512 
513  /* Make the FIFO as large as it needs to be to hold both,
514  * the old and the new samples. */
515  if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
516  fprintf(stderr, "Could not reallocate FIFO\n");
517  return error;
518  }
519 
520  /* Store the new samples in the FIFO buffer. */
521  if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
522  frame_size) < frame_size) {
523  fprintf(stderr, "Could not write data to FIFO\n");
524  return AVERROR_EXIT;
525  }
526  return 0;
527 }
528 
529 /**
530  * Read one audio frame from the input file, decode, convert and store
531  * it in the FIFO buffer.
532  * @param fifo Buffer used for temporary storage
533  * @param input_format_context Format context of the input file
534  * @param input_codec_context Codec context of the input file
535  * @param output_codec_context Codec context of the output file
536  * @param resampler_context Resample context for the conversion
537  * @param[out] finished Indicates whether the end of file has
538  * been reached and all data has been
539  * decoded. If this flag is false,
540  * there is more data to be decoded,
541  * i.e., this function has to be called
542  * again.
543  * @return Error code (0 if successful)
544  */
546  AVFormatContext *input_format_context,
547  AVCodecContext *input_codec_context,
548  AVCodecContext *output_codec_context,
549  SwrContext *resampler_context,
550  int *finished)
551 {
552  /* Temporary storage of the input samples of the frame read from the file. */
553  AVFrame *input_frame = NULL;
554  /* Temporary storage for the converted input samples. */
555  uint8_t **converted_input_samples = NULL;
556  int data_present = 0;
557  int ret = AVERROR_EXIT;
558 
559  /* Initialize temporary storage for one input frame. */
560  if (init_input_frame(&input_frame))
561  goto cleanup;
562  /* Decode one frame worth of audio samples. */
563  if (decode_audio_frame(input_frame, input_format_context,
564  input_codec_context, &data_present, finished))
565  goto cleanup;
566  /* If we are at the end of the file and there are no more samples
567  * in the decoder which are delayed, we are actually finished.
568  * This must not be treated as an error. */
569  if (*finished) {
570  ret = 0;
571  goto cleanup;
572  }
573  /* If there is decoded data, convert and store it. */
574  if (data_present) {
575  /* Initialize the temporary storage for the converted input samples. */
576  if (init_converted_samples(&converted_input_samples, output_codec_context,
577  input_frame->nb_samples))
578  goto cleanup;
579 
580  /* Convert the input samples to the desired output sample format.
581  * This requires a temporary storage provided by converted_input_samples. */
582  if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
583  input_frame->nb_samples, resampler_context))
584  goto cleanup;
585 
586  /* Add the converted input samples to the FIFO buffer for later processing. */
587  if (add_samples_to_fifo(fifo, converted_input_samples,
588  input_frame->nb_samples))
589  goto cleanup;
590  ret = 0;
591  }
592  ret = 0;
593 
594 cleanup:
595  if (converted_input_samples) {
596  av_freep(&converted_input_samples[0]);
597  free(converted_input_samples);
598  }
599  av_frame_free(&input_frame);
600 
601  return ret;
602 }
603 
604 /**
605  * Initialize one input frame for writing to the output file.
606  * The frame will be exactly frame_size samples large.
607  * @param[out] frame Frame to be initialized
608  * @param output_codec_context Codec context of the output file
609  * @param frame_size Size of the frame
610  * @return Error code (0 if successful)
611  */
613  AVCodecContext *output_codec_context,
614  int frame_size)
615 {
616  int error;
617 
618  /* Create a new frame to store the audio samples. */
619  if (!(*frame = av_frame_alloc())) {
620  fprintf(stderr, "Could not allocate output frame\n");
621  return AVERROR_EXIT;
622  }
623 
624  /* Set the frame's parameters, especially its size and format.
625  * av_frame_get_buffer needs this to allocate memory for the
626  * audio samples of the frame.
627  * Default channel layouts based on the number of channels
628  * are assumed for simplicity. */
629  (*frame)->nb_samples = frame_size;
630  (*frame)->channel_layout = output_codec_context->channel_layout;
631  (*frame)->format = output_codec_context->sample_fmt;
632  (*frame)->sample_rate = output_codec_context->sample_rate;
633 
634  /* Allocate the samples of the created frame. This call will make
635  * sure that the audio frame can hold as many samples as specified. */
636  if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
637  fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
638  av_err2str(error));
639  av_frame_free(frame);
640  return error;
641  }
642 
643  return 0;
644 }
645 
646 /* Global timestamp for the audio frames. */
647 static int64_t pts = 0;
648 
649 /**
650  * Encode one frame worth of audio to the output file.
651  * @param frame Samples to be encoded
652  * @param output_format_context Format context of the output file
653  * @param output_codec_context Codec context of the output file
654  * @param[out] data_present Indicates whether data has been
655  * encoded
656  * @return Error code (0 if successful)
657  */
659  AVFormatContext *output_format_context,
660  AVCodecContext *output_codec_context,
661  int *data_present)
662 {
663  /* Packet used for temporary storage. */
665  int error;
666  init_packet(&output_packet);
667 
668  /* Set a timestamp based on the sample rate for the container. */
669  if (frame) {
670  frame->pts = pts;
671  pts += frame->nb_samples;
672  }
673 
674  /* 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  /* The encoder signals that it has nothing more to encode. */
678  if (error == AVERROR_EOF) {
679  error = 0;
680  goto cleanup;
681  } else if (error < 0) {
682  fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
683  av_err2str(error));
684  return error;
685  }
686 
687  /* Receive one encoded frame from the encoder. */
688  error = avcodec_receive_packet(output_codec_context, &output_packet);
689  /* If the encoder asks for more data to be able to provide an
690  * encoded frame, return indicating that no data is present. */
691  if (error == AVERROR(EAGAIN)) {
692  error = 0;
693  goto cleanup;
694  /* If the last frame has been encoded, stop encoding. */
695  } else if (error == AVERROR_EOF) {
696  error = 0;
697  goto cleanup;
698  } else if (error < 0) {
699  fprintf(stderr, "Could not encode frame (error '%s')\n",
700  av_err2str(error));
701  goto cleanup;
702  /* Default case: Return encoded data. */
703  } else {
704  *data_present = 1;
705  }
706 
707  /* Write one audio frame from the temporary packet to the output file. */
708  if (*data_present &&
709  (error = av_write_frame(output_format_context, &output_packet)) < 0) {
710  fprintf(stderr, "Could not write frame (error '%s')\n",
711  av_err2str(error));
712  goto cleanup;
713  }
714 
715 cleanup:
716  av_packet_unref(&output_packet);
717  return error;
718 }
719 
720 /**
721  * Load one audio frame from the FIFO buffer, encode and write it to the
722  * output file.
723  * @param fifo Buffer used for temporary storage
724  * @param output_format_context Format context of the output file
725  * @param output_codec_context Codec context of the output file
726  * @return Error code (0 if successful)
727  */
729  AVFormatContext *output_format_context,
730  AVCodecContext *output_codec_context)
731 {
732  /* Temporary storage of the output samples of the frame written to the file. */
734  /* Use the maximum number of possible samples per frame.
735  * If there is less than the maximum possible frame size in the FIFO
736  * buffer use this number. Otherwise, use the maximum possible frame size. */
737  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
738  output_codec_context->frame_size);
739  int data_written;
740 
741  /* Initialize temporary storage for one output frame. */
742  if (init_output_frame(&output_frame, output_codec_context, frame_size))
743  return AVERROR_EXIT;
744 
745  /* Read as many samples from the FIFO buffer as required to fill the frame.
746  * The samples are stored in the frame temporarily. */
747  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
748  fprintf(stderr, "Could not read data from FIFO\n");
749  av_frame_free(&output_frame);
750  return AVERROR_EXIT;
751  }
752 
753  /* Encode one frame worth of audio samples. */
754  if (encode_audio_frame(output_frame, output_format_context,
755  output_codec_context, &data_written)) {
756  av_frame_free(&output_frame);
757  return AVERROR_EXIT;
758  }
759  av_frame_free(&output_frame);
760  return 0;
761 }
762 
763 /**
764  * Write the trailer of the output file container.
765  * @param output_format_context Format context of the output file
766  * @return Error code (0 if successful)
767  */
768 static int write_output_file_trailer(AVFormatContext *output_format_context)
769 {
770  int error;
771  if ((error = av_write_trailer(output_format_context)) < 0) {
772  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
773  av_err2str(error));
774  return error;
775  }
776  return 0;
777 }
778 
779 int main(int argc, char **argv)
780 {
781  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
782  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
783  SwrContext *resample_context = NULL;
784  AVAudioFifo *fifo = NULL;
785  int ret = AVERROR_EXIT;
786 
787  if (argc != 3) {
788  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
789  exit(1);
790  }
791 
792  /* Open the input file for reading. */
793  if (open_input_file(argv[1], &input_format_context,
794  &input_codec_context))
795  goto cleanup;
796  /* Open the output file for writing. */
797  if (open_output_file(argv[2], input_codec_context,
798  &output_format_context, &output_codec_context))
799  goto cleanup;
800  /* Initialize the resampler to be able to convert audio sample formats. */
801  if (init_resampler(input_codec_context, output_codec_context,
802  &resample_context))
803  goto cleanup;
804  /* Initialize the FIFO buffer to store audio samples to be encoded. */
805  if (init_fifo(&fifo, output_codec_context))
806  goto cleanup;
807  /* Write the header of the output file container. */
808  if (write_output_file_header(output_format_context))
809  goto cleanup;
810 
811  /* Loop as long as we have input samples to read or output samples
812  * to write; abort as soon as we have neither. */
813  while (1) {
814  /* Use the encoder's desired frame size for processing. */
815  const int output_frame_size = output_codec_context->frame_size;
816  int finished = 0;
817 
818  /* Make sure that there is one frame worth of samples in the FIFO
819  * buffer so that the encoder can do its work.
820  * Since the decoder's and the encoder's frame size may differ, we
821  * need to FIFO buffer to store as many frames worth of input samples
822  * that they make up at least one frame worth of output samples. */
823  while (av_audio_fifo_size(fifo) < output_frame_size) {
824  /* Decode one frame worth of audio samples, convert it to the
825  * output sample format and put it into the FIFO buffer. */
826  if (read_decode_convert_and_store(fifo, input_format_context,
827  input_codec_context,
828  output_codec_context,
829  resample_context, &finished))
830  goto cleanup;
831 
832  /* If we are at the end of the input file, we continue
833  * encoding the remaining audio samples to the output file. */
834  if (finished)
835  break;
836  }
837 
838  /* If we have enough samples for the encoder, we encode them.
839  * At the end of the file, we pass the remaining samples to
840  * the encoder. */
841  while (av_audio_fifo_size(fifo) >= output_frame_size ||
842  (finished && av_audio_fifo_size(fifo) > 0))
843  /* Take one frame worth of audio samples from the FIFO buffer,
844  * encode it and write it to the output file. */
845  if (load_encode_and_write(fifo, output_format_context,
846  output_codec_context))
847  goto cleanup;
848 
849  /* If we are at the end of the input file and have encoded
850  * all remaining samples, we can exit this loop and finish. */
851  if (finished) {
852  int data_written;
853  /* Flush the encoder as it may have delayed frames. */
854  do {
855  data_written = 0;
856  if (encode_audio_frame(NULL, output_format_context,
857  output_codec_context, &data_written))
858  goto cleanup;
859  } while (data_written);
860  break;
861  }
862  }
863 
864  /* Write the trailer of the output file container. */
865  if (write_output_file_trailer(output_format_context))
866  goto cleanup;
867  ret = 0;
868 
869 cleanup:
870  if (fifo)
871  av_audio_fifo_free(fifo);
872  swr_free(&resample_context);
873  if (output_codec_context)
874  avcodec_free_context(&output_codec_context);
875  if (output_format_context) {
876  avio_closep(&output_format_context->pb);
877  avformat_free_context(output_format_context);
878  }
879  if (input_codec_context)
880  avcodec_free_context(&input_codec_context);
881  if (input_format_context)
882  avformat_close_input(&input_format_context);
883 
884  return ret;
885 }
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:1154
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2582
#define NULL
Definition: coverity.c:32
static void output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof)
Definition: ffmpeg.c:846
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:218
AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: allcodecs.c:864
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:878
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1569
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
Definition: encode.c:420
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:1432
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:655
AVCodec.
Definition: avcodec.h:3409
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:1351
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:2182
uint8_t
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:189
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:311
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:4456
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:2078
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:1431
#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:202
The libswresample context.
int avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Return decoded output data from a decoder.
Definition: decode.c:709
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1599
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.
reference-counted frame API
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2225
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:508
#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:156
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:51
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:874
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition: decode.c:642
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2194
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:2117
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:251
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:171
#define OUTPUT_BIT_RATE
Definition: transcode_aac.c:47
int sample_rate
samples per second
Definition: avcodec.h:2174
main external API structure.
Definition: avcodec.h:1519
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: allcodecs.c:869
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:592
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
Definition: encode.c:390
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264dec.c:836
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:2021
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: utils.c:538
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4390
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:1767
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:322
static int64_t pts
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
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:883
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:3565
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:4428
Audio FIFO Buffer.
int channels
number of audio channels
Definition: avcodec.h:2175
int avformat_open_input(AVFormatContext **ps, const char *url, AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition: utils.c:537
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:1247
#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:1021
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: avcodec.h:3432
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:903
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:265
This structure stores compressed data.
Definition: avcodec.h:1408
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:1215
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2577
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