FFmpeg
transcode_aac.c
Go to the documentation of this file.
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"
42 #include "libavutil/frame.h"
43 #include "libavutil/opt.h"
44 
46 
47 /* The output bit rate in bit/s */
48 #define OUTPUT_BIT_RATE 96000
49 /* The number of output channels */
50 #define OUTPUT_CHANNELS 2
51 
52 /**
53  * Open an input file and the required decoder.
54  * @param filename File to be opened
55  * @param[out] input_format_context Format context of opened file
56  * @param[out] input_codec_context Codec context of opened file
57  * @return Error code (0 if successful)
58  */
59 static int open_input_file(const char *filename,
60  AVFormatContext **input_format_context,
61  AVCodecContext **input_codec_context)
62 {
63  AVCodecContext *avctx;
64  const AVCodec *input_codec;
65  int error;
66 
67  /* Open the input file to read from it. */
68  if ((error = avformat_open_input(input_format_context, filename, NULL,
69  NULL)) < 0) {
70  fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
71  filename, av_err2str(error));
72  *input_format_context = NULL;
73  return error;
74  }
75 
76  /* Get information on the input file (number of streams etc.). */
77  if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
78  fprintf(stderr, "Could not open find stream info (error '%s')\n",
79  av_err2str(error));
80  avformat_close_input(input_format_context);
81  return error;
82  }
83 
84  /* Make sure that there is only one stream in the input file. */
85  if ((*input_format_context)->nb_streams != 1) {
86  fprintf(stderr, "Expected one audio input stream, but found %d\n",
87  (*input_format_context)->nb_streams);
88  avformat_close_input(input_format_context);
89  return AVERROR_EXIT;
90  }
91 
92  /* Find a decoder for the audio stream. */
93  if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
94  fprintf(stderr, "Could not find input codec\n");
95  avformat_close_input(input_format_context);
96  return AVERROR_EXIT;
97  }
98 
99  /* Allocate a new decoding context. */
100  avctx = avcodec_alloc_context3(input_codec);
101  if (!avctx) {
102  fprintf(stderr, "Could not allocate a decoding context\n");
103  avformat_close_input(input_format_context);
104  return AVERROR(ENOMEM);
105  }
106 
107  /* Initialize the stream parameters with demuxer information. */
108  error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
109  if (error < 0) {
110  avformat_close_input(input_format_context);
111  avcodec_free_context(&avctx);
112  return error;
113  }
114 
115  /* Open the decoder for the audio stream to use it later. */
116  if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
117  fprintf(stderr, "Could not open input codec (error '%s')\n",
118  av_err2str(error));
119  avcodec_free_context(&avctx);
120  avformat_close_input(input_format_context);
121  return error;
122  }
123 
124  /* Save the decoder context for easier access later. */
125  *input_codec_context = avctx;
126 
127  return 0;
128 }
129 
130 /**
131  * Open an output file and the required encoder.
132  * Also set some basic encoder parameters.
133  * Some of these parameters are based on the input file's parameters.
134  * @param filename File to be opened
135  * @param input_codec_context Codec context of input file
136  * @param[out] output_format_context Format context of output file
137  * @param[out] output_codec_context Codec context of output file
138  * @return Error code (0 if successful)
139  */
140 static int open_output_file(const char *filename,
141  AVCodecContext *input_codec_context,
142  AVFormatContext **output_format_context,
143  AVCodecContext **output_codec_context)
144 {
145  AVCodecContext *avctx = NULL;
146  AVIOContext *output_io_context = NULL;
147  AVStream *stream = NULL;
148  const AVCodec *output_codec = NULL;
149  int error;
150 
151  /* Open the output file to write to it. */
152  if ((error = avio_open(&output_io_context, filename,
153  AVIO_FLAG_WRITE)) < 0) {
154  fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
155  filename, av_err2str(error));
156  return error;
157  }
158 
159  /* Create a new format context for the output container format. */
160  if (!(*output_format_context = avformat_alloc_context())) {
161  fprintf(stderr, "Could not allocate output format context\n");
162  return AVERROR(ENOMEM);
163  }
164 
165  /* Associate the output file (pointer) with the container format context. */
166  (*output_format_context)->pb = output_io_context;
167 
168  /* Guess the desired container format based on the file extension. */
169  if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
170  NULL))) {
171  fprintf(stderr, "Could not find output file format\n");
172  goto cleanup;
173  }
174 
175  if (!((*output_format_context)->url = av_strdup(filename))) {
176  fprintf(stderr, "Could not allocate url.\n");
177  error = AVERROR(ENOMEM);
178  goto cleanup;
179  }
180 
181  /* Find the encoder to be used by its name. */
182  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
183  fprintf(stderr, "Could not find an AAC encoder.\n");
184  goto cleanup;
185  }
186 
187  /* Create a new audio stream in the output file container. */
188  if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
189  fprintf(stderr, "Could not create new stream\n");
190  error = AVERROR(ENOMEM);
191  goto cleanup;
192  }
193 
194  avctx = avcodec_alloc_context3(output_codec);
195  if (!avctx) {
196  fprintf(stderr, "Could not allocate an encoding context\n");
197  error = AVERROR(ENOMEM);
198  goto cleanup;
199  }
200 
201  /* Set the basic encoder parameters.
202  * The input file's sample rate is used to avoid a sample rate conversion. */
203  avctx->channels = OUTPUT_CHANNELS;
205  avctx->sample_rate = input_codec_context->sample_rate;
206  avctx->sample_fmt = output_codec->sample_fmts[0];
207  avctx->bit_rate = OUTPUT_BIT_RATE;
208 
209  /* Allow the use of the experimental AAC encoder. */
211 
212  /* Set the sample rate for the container. */
213  stream->time_base.den = input_codec_context->sample_rate;
214  stream->time_base.num = 1;
215 
216  /* Some container formats (like MP4) require global headers to be present.
217  * Mark the encoder so that it behaves accordingly. */
218  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
220 
221  /* Open the encoder for the audio stream to use it later. */
222  if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
223  fprintf(stderr, "Could not open output codec (error '%s')\n",
224  av_err2str(error));
225  goto cleanup;
226  }
227 
229  if (error < 0) {
230  fprintf(stderr, "Could not initialize stream parameters\n");
231  goto cleanup;
232  }
233 
234  /* Save the encoder context for easier access later. */
235  *output_codec_context = avctx;
236 
237  return 0;
238 
239 cleanup:
240  avcodec_free_context(&avctx);
241  avio_closep(&(*output_format_context)->pb);
242  avformat_free_context(*output_format_context);
243  *output_format_context = NULL;
244  return error < 0 ? error : AVERROR_EXIT;
245 }
246 
247 /**
248  * Initialize one data packet for reading or writing.
249  * @param[out] packet Packet to be initialized
250  * @return Error code (0 if successful)
251  */
252 static int init_packet(AVPacket **packet)
253 {
254  if (!(*packet = av_packet_alloc())) {
255  fprintf(stderr, "Could not allocate packet\n");
256  return AVERROR(ENOMEM);
257  }
258  return 0;
259 }
260 
261 /**
262  * Initialize one audio frame for reading from the input file.
263  * @param[out] frame Frame to be initialized
264  * @return Error code (0 if successful)
265  */
267 {
268  if (!(*frame = av_frame_alloc())) {
269  fprintf(stderr, "Could not allocate input frame\n");
270  return AVERROR(ENOMEM);
271  }
272  return 0;
273 }
274 
275 /**
276  * Initialize the audio resampler based on the input and output codec settings.
277  * If the input and output sample formats differ, a conversion is required
278  * libswresample takes care of this, but requires initialization.
279  * @param input_codec_context Codec context of the input file
280  * @param output_codec_context Codec context of the output file
281  * @param[out] resample_context Resample context for the required conversion
282  * @return Error code (0 if successful)
283  */
284 static int init_resampler(AVCodecContext *input_codec_context,
285  AVCodecContext *output_codec_context,
286  SwrContext **resample_context)
287 {
288  int error;
289 
290  /*
291  * Create a resampler context for the conversion.
292  * Set the conversion parameters.
293  * Default channel layouts based on the number of channels
294  * are assumed for simplicity (they are sometimes not detected
295  * properly by the demuxer and/or decoder).
296  */
297  *resample_context = swr_alloc_set_opts(NULL,
298  av_get_default_channel_layout(output_codec_context->channels),
299  output_codec_context->sample_fmt,
300  output_codec_context->sample_rate,
301  av_get_default_channel_layout(input_codec_context->channels),
302  input_codec_context->sample_fmt,
303  input_codec_context->sample_rate,
304  0, NULL);
305  if (!*resample_context) {
306  fprintf(stderr, "Could not allocate resample context\n");
307  return AVERROR(ENOMEM);
308  }
309  /*
310  * Perform a sanity check so that the number of converted samples is
311  * not greater than the number of samples to be converted.
312  * If the sample rates differ, this case has to be handled differently
313  */
314  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
315 
316  /* Open the resampler with the specified parameters. */
317  if ((error = swr_init(*resample_context)) < 0) {
318  fprintf(stderr, "Could not open resample context\n");
319  swr_free(resample_context);
320  return error;
321  }
322  return 0;
323 }
324 
325 /**
326  * Initialize a FIFO buffer for the audio samples to be encoded.
327  * @param[out] fifo Sample buffer
328  * @param output_codec_context Codec context of the output file
329  * @return Error code (0 if successful)
330  */
331 static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
332 {
333  /* Create the FIFO buffer based on the specified output sample format. */
334  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
335  output_codec_context->channels, 1))) {
336  fprintf(stderr, "Could not allocate FIFO\n");
337  return AVERROR(ENOMEM);
338  }
339  return 0;
340 }
341 
342 /**
343  * Write the header of the output file container.
344  * @param output_format_context Format context of the output file
345  * @return Error code (0 if successful)
346  */
347 static int write_output_file_header(AVFormatContext *output_format_context)
348 {
349  int error;
350  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
351  fprintf(stderr, "Could not write output file header (error '%s')\n",
352  av_err2str(error));
353  return error;
354  }
355  return 0;
356 }
357 
358 /**
359  * Decode one audio frame from the input file.
360  * @param frame Audio frame to be decoded
361  * @param input_format_context Format context of the input file
362  * @param input_codec_context Codec context of the input file
363  * @param[out] data_present Indicates whether data has been decoded
364  * @param[out] finished Indicates whether the end of file has
365  * been reached and all data has been
366  * decoded. If this flag is false, there
367  * is more data to be decoded, i.e., this
368  * function has to be called again.
369  * @return Error code (0 if successful)
370  */
372  AVFormatContext *input_format_context,
373  AVCodecContext *input_codec_context,
374  int *data_present, int *finished)
375 {
376  /* Packet used for temporary storage. */
377  AVPacket *input_packet;
378  int error;
379 
380  error = init_packet(&input_packet);
381  if (error < 0)
382  return error;
383 
384  /* Read one audio frame from the input file into a temporary packet. */
385  if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
386  /* If we are at the end of the file, flush the decoder below. */
387  if (error == AVERROR_EOF)
388  *finished = 1;
389  else {
390  fprintf(stderr, "Could not read frame (error '%s')\n",
391  av_err2str(error));
392  goto cleanup;
393  }
394  }
395 
396  /* Send the audio frame stored in the temporary packet to the decoder.
397  * The input audio stream decoder is used to do this. */
398  if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
399  fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
400  av_err2str(error));
401  goto cleanup;
402  }
403 
404  /* Receive one frame from the decoder. */
405  error = avcodec_receive_frame(input_codec_context, frame);
406  /* If the decoder asks for more data to be able to decode a frame,
407  * return indicating that no data is present. */
408  if (error == AVERROR(EAGAIN)) {
409  error = 0;
410  goto cleanup;
411  /* If the end of the input file is reached, stop decoding. */
412  } else if (error == AVERROR_EOF) {
413  *finished = 1;
414  error = 0;
415  goto cleanup;
416  } else if (error < 0) {
417  fprintf(stderr, "Could not decode frame (error '%s')\n",
418  av_err2str(error));
419  goto cleanup;
420  /* Default case: Return decoded data. */
421  } else {
422  *data_present = 1;
423  goto cleanup;
424  }
425 
426 cleanup:
427  av_packet_free(&input_packet);
428  return error;
429 }
430 
431 /**
432  * Initialize a temporary storage for the specified number of audio samples.
433  * The conversion requires temporary storage due to the different format.
434  * The number of audio samples to be allocated is specified in frame_size.
435  * @param[out] converted_input_samples Array of converted samples. The
436  * dimensions are reference, channel
437  * (for multi-channel audio), sample.
438  * @param output_codec_context Codec context of the output file
439  * @param frame_size Number of samples to be converted in
440  * each round
441  * @return Error code (0 if successful)
442  */
443 static int init_converted_samples(uint8_t ***converted_input_samples,
444  AVCodecContext *output_codec_context,
445  int frame_size)
446 {
447  int error;
448 
449  /* Allocate as many pointers as there are audio channels.
450  * Each pointer will later point to the audio samples of the corresponding
451  * channels (although it may be NULL for interleaved formats).
452  */
453  if (!(*converted_input_samples = calloc(output_codec_context->channels,
454  sizeof(**converted_input_samples)))) {
455  fprintf(stderr, "Could not allocate converted input sample pointers\n");
456  return AVERROR(ENOMEM);
457  }
458 
459  /* Allocate memory for the samples of all channels in one consecutive
460  * block for convenience. */
461  if ((error = av_samples_alloc(*converted_input_samples, NULL,
462  output_codec_context->channels,
463  frame_size,
464  output_codec_context->sample_fmt, 0)) < 0) {
465  fprintf(stderr,
466  "Could not allocate converted input samples (error '%s')\n",
467  av_err2str(error));
468  av_freep(&(*converted_input_samples)[0]);
469  free(*converted_input_samples);
470  return error;
471  }
472  return 0;
473 }
474 
475 /**
476  * Convert the input audio samples into the output sample format.
477  * The conversion happens on a per-frame basis, the size of which is
478  * specified by frame_size.
479  * @param input_data Samples to be decoded. The dimensions are
480  * channel (for multi-channel audio), sample.
481  * @param[out] converted_data Converted samples. The dimensions are channel
482  * (for multi-channel audio), sample.
483  * @param frame_size Number of samples to be converted
484  * @param resample_context Resample context for the conversion
485  * @return Error code (0 if successful)
486  */
487 static int convert_samples(const uint8_t **input_data,
488  uint8_t **converted_data, const int frame_size,
489  SwrContext *resample_context)
490 {
491  int error;
492 
493  /* Convert the samples using the resampler. */
494  if ((error = swr_convert(resample_context,
495  converted_data, frame_size,
496  input_data , frame_size)) < 0) {
497  fprintf(stderr, "Could not convert input samples (error '%s')\n",
498  av_err2str(error));
499  return error;
500  }
501 
502  return 0;
503 }
504 
505 /**
506  * Add converted input audio samples to the FIFO buffer for later processing.
507  * @param fifo Buffer to add the samples to
508  * @param converted_input_samples Samples to be added. The dimensions are channel
509  * (for multi-channel audio), sample.
510  * @param frame_size Number of samples to be converted
511  * @return Error code (0 if successful)
512  */
514  uint8_t **converted_input_samples,
515  const int frame_size)
516 {
517  int error;
518 
519  /* Make the FIFO as large as it needs to be to hold both,
520  * the old and the new samples. */
521  if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
522  fprintf(stderr, "Could not reallocate FIFO\n");
523  return error;
524  }
525 
526  /* Store the new samples in the FIFO buffer. */
527  if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
528  frame_size) < frame_size) {
529  fprintf(stderr, "Could not write data to FIFO\n");
530  return AVERROR_EXIT;
531  }
532  return 0;
533 }
534 
535 /**
536  * Read one audio frame from the input file, decode, convert and store
537  * it in the FIFO buffer.
538  * @param fifo Buffer used for temporary storage
539  * @param input_format_context Format context of the input file
540  * @param input_codec_context Codec context of the input file
541  * @param output_codec_context Codec context of the output file
542  * @param resampler_context Resample context for the conversion
543  * @param[out] finished Indicates whether the end of file has
544  * been reached and all data has been
545  * decoded. If this flag is false,
546  * there is more data to be decoded,
547  * i.e., this function has to be called
548  * again.
549  * @return Error code (0 if successful)
550  */
552  AVFormatContext *input_format_context,
553  AVCodecContext *input_codec_context,
554  AVCodecContext *output_codec_context,
555  SwrContext *resampler_context,
556  int *finished)
557 {
558  /* Temporary storage of the input samples of the frame read from the file. */
559  AVFrame *input_frame = NULL;
560  /* Temporary storage for the converted input samples. */
561  uint8_t **converted_input_samples = NULL;
562  int data_present = 0;
563  int ret = AVERROR_EXIT;
564 
565  /* Initialize temporary storage for one input frame. */
566  if (init_input_frame(&input_frame))
567  goto cleanup;
568  /* Decode one frame worth of audio samples. */
569  if (decode_audio_frame(input_frame, input_format_context,
570  input_codec_context, &data_present, finished))
571  goto cleanup;
572  /* If we are at the end of the file and there are no more samples
573  * in the decoder which are delayed, we are actually finished.
574  * This must not be treated as an error. */
575  if (*finished) {
576  ret = 0;
577  goto cleanup;
578  }
579  /* If there is decoded data, convert and store it. */
580  if (data_present) {
581  /* Initialize the temporary storage for the converted input samples. */
582  if (init_converted_samples(&converted_input_samples, output_codec_context,
583  input_frame->nb_samples))
584  goto cleanup;
585 
586  /* Convert the input samples to the desired output sample format.
587  * This requires a temporary storage provided by converted_input_samples. */
588  if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
589  input_frame->nb_samples, resampler_context))
590  goto cleanup;
591 
592  /* Add the converted input samples to the FIFO buffer for later processing. */
593  if (add_samples_to_fifo(fifo, converted_input_samples,
594  input_frame->nb_samples))
595  goto cleanup;
596  ret = 0;
597  }
598  ret = 0;
599 
600 cleanup:
601  if (converted_input_samples) {
602  av_freep(&converted_input_samples[0]);
603  free(converted_input_samples);
604  }
605  av_frame_free(&input_frame);
606 
607  return ret;
608 }
609 
610 /**
611  * Initialize one input frame for writing to the output file.
612  * The frame will be exactly frame_size samples large.
613  * @param[out] frame Frame to be initialized
614  * @param output_codec_context Codec context of the output file
615  * @param frame_size Size of the frame
616  * @return Error code (0 if successful)
617  */
619  AVCodecContext *output_codec_context,
620  int frame_size)
621 {
622  int error;
623 
624  /* Create a new frame to store the audio samples. */
625  if (!(*frame = av_frame_alloc())) {
626  fprintf(stderr, "Could not allocate output frame\n");
627  return AVERROR_EXIT;
628  }
629 
630  /* Set the frame's parameters, especially its size and format.
631  * av_frame_get_buffer needs this to allocate memory for the
632  * audio samples of the frame.
633  * Default channel layouts based on the number of channels
634  * are assumed for simplicity. */
635  (*frame)->nb_samples = frame_size;
636  (*frame)->channel_layout = output_codec_context->channel_layout;
637  (*frame)->format = output_codec_context->sample_fmt;
638  (*frame)->sample_rate = output_codec_context->sample_rate;
639 
640  /* Allocate the samples of the created frame. This call will make
641  * sure that the audio frame can hold as many samples as specified. */
642  if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
643  fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
644  av_err2str(error));
646  return error;
647  }
648 
649  return 0;
650 }
651 
652 /* Global timestamp for the audio frames. */
653 static int64_t pts = 0;
654 
655 /**
656  * Encode one frame worth of audio to the output file.
657  * @param frame Samples to be encoded
658  * @param output_format_context Format context of the output file
659  * @param output_codec_context Codec context of the output file
660  * @param[out] data_present Indicates whether data has been
661  * encoded
662  * @return Error code (0 if successful)
663  */
665  AVFormatContext *output_format_context,
666  AVCodecContext *output_codec_context,
667  int *data_present)
668 {
669  /* Packet used for temporary storage. */
671  int error;
672 
674  if (error < 0)
675  return error;
676 
677  /* Set a timestamp based on the sample rate for the container. */
678  if (frame) {
679  frame->pts = pts;
680  pts += frame->nb_samples;
681  }
682 
683  /* Send the audio frame stored in the temporary packet to the encoder.
684  * The output audio stream encoder is used to do this. */
685  error = avcodec_send_frame(output_codec_context, frame);
686  /* The encoder signals that it has nothing more to encode. */
687  if (error == AVERROR_EOF) {
688  error = 0;
689  goto cleanup;
690  } else if (error < 0) {
691  fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
692  av_err2str(error));
693  goto cleanup;
694  }
695 
696  /* Receive one encoded frame from the encoder. */
697  error = avcodec_receive_packet(output_codec_context, output_packet);
698  /* If the encoder asks for more data to be able to provide an
699  * encoded frame, return indicating that no data is present. */
700  if (error == AVERROR(EAGAIN)) {
701  error = 0;
702  goto cleanup;
703  /* If the last frame has been encoded, stop encoding. */
704  } else if (error == AVERROR_EOF) {
705  error = 0;
706  goto cleanup;
707  } else if (error < 0) {
708  fprintf(stderr, "Could not encode frame (error '%s')\n",
709  av_err2str(error));
710  goto cleanup;
711  /* Default case: Return encoded data. */
712  } else {
713  *data_present = 1;
714  }
715 
716  /* Write one audio frame from the temporary packet to the output file. */
717  if (*data_present &&
718  (error = av_write_frame(output_format_context, output_packet)) < 0) {
719  fprintf(stderr, "Could not write frame (error '%s')\n",
720  av_err2str(error));
721  goto cleanup;
722  }
723 
724 cleanup:
726  return error;
727 }
728 
729 /**
730  * Load one audio frame from the FIFO buffer, encode and write it to the
731  * output file.
732  * @param fifo Buffer used for temporary storage
733  * @param output_format_context Format context of the output file
734  * @param output_codec_context Codec context of the output file
735  * @return Error code (0 if successful)
736  */
738  AVFormatContext *output_format_context,
739  AVCodecContext *output_codec_context)
740 {
741  /* Temporary storage of the output samples of the frame written to the file. */
743  /* Use the maximum number of possible samples per frame.
744  * If there is less than the maximum possible frame size in the FIFO
745  * buffer use this number. Otherwise, use the maximum possible frame size. */
746  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
747  output_codec_context->frame_size);
748  int data_written;
749 
750  /* Initialize temporary storage for one output frame. */
751  if (init_output_frame(&output_frame, output_codec_context, frame_size))
752  return AVERROR_EXIT;
753 
754  /* Read as many samples from the FIFO buffer as required to fill the frame.
755  * The samples are stored in the frame temporarily. */
756  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
757  fprintf(stderr, "Could not read data from FIFO\n");
759  return AVERROR_EXIT;
760  }
761 
762  /* Encode one frame worth of audio samples. */
763  if (encode_audio_frame(output_frame, output_format_context,
764  output_codec_context, &data_written)) {
766  return AVERROR_EXIT;
767  }
769  return 0;
770 }
771 
772 /**
773  * Write the trailer of the output file container.
774  * @param output_format_context Format context of the output file
775  * @return Error code (0 if successful)
776  */
777 static int write_output_file_trailer(AVFormatContext *output_format_context)
778 {
779  int error;
780  if ((error = av_write_trailer(output_format_context)) < 0) {
781  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
782  av_err2str(error));
783  return error;
784  }
785  return 0;
786 }
787 
788 int main(int argc, char **argv)
789 {
790  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
791  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
792  SwrContext *resample_context = NULL;
793  AVAudioFifo *fifo = NULL;
794  int ret = AVERROR_EXIT;
795 
796  if (argc != 3) {
797  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
798  exit(1);
799  }
800 
801  /* Open the input file for reading. */
802  if (open_input_file(argv[1], &input_format_context,
803  &input_codec_context))
804  goto cleanup;
805  /* Open the output file for writing. */
806  if (open_output_file(argv[2], input_codec_context,
807  &output_format_context, &output_codec_context))
808  goto cleanup;
809  /* Initialize the resampler to be able to convert audio sample formats. */
810  if (init_resampler(input_codec_context, output_codec_context,
811  &resample_context))
812  goto cleanup;
813  /* Initialize the FIFO buffer to store audio samples to be encoded. */
814  if (init_fifo(&fifo, output_codec_context))
815  goto cleanup;
816  /* Write the header of the output file container. */
817  if (write_output_file_header(output_format_context))
818  goto cleanup;
819 
820  /* Loop as long as we have input samples to read or output samples
821  * to write; abort as soon as we have neither. */
822  while (1) {
823  /* Use the encoder's desired frame size for processing. */
824  const int output_frame_size = output_codec_context->frame_size;
825  int finished = 0;
826 
827  /* Make sure that there is one frame worth of samples in the FIFO
828  * buffer so that the encoder can do its work.
829  * Since the decoder's and the encoder's frame size may differ, we
830  * need to FIFO buffer to store as many frames worth of input samples
831  * that they make up at least one frame worth of output samples. */
832  while (av_audio_fifo_size(fifo) < output_frame_size) {
833  /* Decode one frame worth of audio samples, convert it to the
834  * output sample format and put it into the FIFO buffer. */
835  if (read_decode_convert_and_store(fifo, input_format_context,
836  input_codec_context,
837  output_codec_context,
838  resample_context, &finished))
839  goto cleanup;
840 
841  /* If we are at the end of the input file, we continue
842  * encoding the remaining audio samples to the output file. */
843  if (finished)
844  break;
845  }
846 
847  /* If we have enough samples for the encoder, we encode them.
848  * At the end of the file, we pass the remaining samples to
849  * the encoder. */
850  while (av_audio_fifo_size(fifo) >= output_frame_size ||
851  (finished && av_audio_fifo_size(fifo) > 0))
852  /* Take one frame worth of audio samples from the FIFO buffer,
853  * encode it and write it to the output file. */
854  if (load_encode_and_write(fifo, output_format_context,
855  output_codec_context))
856  goto cleanup;
857 
858  /* If we are at the end of the input file and have encoded
859  * all remaining samples, we can exit this loop and finish. */
860  if (finished) {
861  int data_written;
862  /* Flush the encoder as it may have delayed frames. */
863  do {
864  data_written = 0;
865  if (encode_audio_frame(NULL, output_format_context,
866  output_codec_context, &data_written))
867  goto cleanup;
868  } while (data_written);
869  break;
870  }
871  }
872 
873  /* Write the trailer of the output file container. */
874  if (write_output_file_trailer(output_format_context))
875  goto cleanup;
876  ret = 0;
877 
878 cleanup:
879  if (fifo)
880  av_audio_fifo_free(fifo);
881  swr_free(&resample_context);
882  if (output_codec_context)
883  avcodec_free_context(&output_codec_context);
884  if (output_format_context) {
885  avio_closep(&output_format_context->pb);
886  avformat_free_context(output_format_context);
887  }
888  if (input_codec_context)
889  avcodec_free_context(&input_codec_context);
890  if (input_format_context)
891  avformat_close_input(&input_format_context);
892 
893  return ret;
894 }
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:31
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1008
AVCodec
AVCodec.
Definition: codec.h:197
load_encode_and_write
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context)
Load one audio frame from the FIFO buffer, encode and write it to the output file.
Definition: transcode_aac.c:737
avcodec_receive_packet
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
Definition: encode.c:392
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4374
open_input_file
static int open_input_file(const char *filename, AVFormatContext **input_format_context, AVCodecContext **input_codec_context)
Open an input file and the required decoder.
Definition: transcode_aac.c:59
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1039
av_frame_get_buffer
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:246
FF_COMPLIANCE_EXPERIMENTAL
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:1281
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:988
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
avcodec_parameters_from_context
int avcodec_parameters_from_context(AVCodecParameters *par, const AVCodecContext *codec)
Fill the parameters struct based on the values from the supplied codec context.
Definition: codec_par.c:90
av_audio_fifo_realloc
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:96
init_fifo
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
Definition: transcode_aac.c:331
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:112
avcodec_find_encoder
const AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: allcodecs.c:910
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
cleanup
static av_cold void cleanup(FlashSV2Context *s)
Definition: flashsv2enc.c:128
write_output_file_header
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
Definition: transcode_aac.c:347
open_output_file
static int open_output_file(const char *filename, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context)
Open an output file and the required encoder.
Definition: transcode_aac.c:140
av_read_frame
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition: utils.c:1668
av_packet_free
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:70
AV_CODEC_FLAG_GLOBAL_HEADER
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition: avcodec.h:264
avformat_close_input
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: utils.c:4346
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
AVCodec::sample_fmts
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:220
av_samples_alloc
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:180
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:459
pts
static int64_t pts
Definition: transcode_aac.c:653
AVRational::num
int num
Numerator.
Definition: rational.h:59
av_frame_alloc
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:99
avassert.h
swr_init
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
avformat_open_input
int avformat_open_input(AVFormatContext **ps, const char *url, const AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition: utils.c:482
avcodec_alloc_context3
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition: options.c:141
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
add_samples_to_fifo
static int add_samples_to_fifo(AVAudioFifo *fifo, uint8_t **converted_input_samples, const int frame_size)
Add converted input audio samples to the FIFO buffer for later processing.
Definition: transcode_aac.c:513
frame_size
int frame_size
Definition: mxfenc.c:2206
decode_audio_frame
static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished)
Decode one audio frame from the input file.
Definition: transcode_aac.c:371
avcodec_receive_frame
int avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Return decoded output data from a decoder.
Definition: decode.c:639
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AVIO_FLAG_WRITE
#define AVIO_FLAG_WRITE
write-only
Definition: avio.h:656
swr_convert
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:714
SwrContext
The libswresample context.
Definition: swresample_internal.h:95
swr_alloc_set_opts
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
avformat_write_header
av_warn_unused_result int avformat_write_header(AVFormatContext *s, AVDictionary **options)
Allocate the stream private data and write the stream header to an output media file.
Definition: mux.c:469
AVFormatContext
Format I/O context.
Definition: avformat.h:1107
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:994
AVStream::time_base
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
Definition: avformat.h:870
NULL
#define NULL
Definition: coverity.c:32
avcodec_free_context
void avcodec_free_context(AVCodecContext **avctx)
Free the codec context and everything associated with it and write NULL to the provided pointer.
Definition: options.c:156
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
read_decode_convert_and_store
static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resampler_context, int *finished)
Read one audio frame from the input file, decode, convert and store it in the FIFO buffer.
Definition: transcode_aac.c:551
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:429
OUTPUT_BIT_RATE
#define OUTPUT_BIT_RATE
Definition: transcode_aac.c:48
avcodec_open2
int attribute_align_arg avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: avcodec.c:137
av_write_frame
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition: mux.c:1175
init_output_frame
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
Initialize one input frame for writing to the output file.
Definition: transcode_aac.c:618
swresample.h
avcodec_find_decoder
const AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: allcodecs.c:915
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:424
init_input_frame
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
Definition: transcode_aac.c:266
avformat_find_stream_info
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
Definition: utils.c:3546
AVIOContext
Bytestream IO Context.
Definition: avio.h:161
avformat_alloc_context
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
Definition: options.c:166
av_err2str
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
Definition: error.h:119
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:996
encode_audio_frame
static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, AVCodecContext *output_codec_context, int *data_present)
Encode one frame worth of audio to the output file.
Definition: transcode_aac.c:664
main
int main(int argc, char **argv)
Definition: transcode_aac.c:788
avio.h
swr_free
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
init_packet
static int init_packet(AVPacket **packet)
Initialize one data packet for reading or writing.
Definition: transcode_aac.c:252
frame.h
OUTPUT_CHANNELS
#define OUTPUT_CHANNELS
Definition: transcode_aac.c:50
FFMIN
#define FFMIN(a, b)
Definition: common.h:105
output_frame
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264dec.c:824
av_packet_alloc
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:59
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
init_resampler
static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context)
Initialize the audio resampler based on the input and output codec settings.
Definition: transcode_aac.c:284
input_data
static void input_data(MLPEncodeContext *ctx, void *samples)
Wrapper function for inputting data in two different bit-depths.
Definition: mlpenc.c:1280
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:989
avcodec_send_packet
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition: decode.c:576
avio_closep
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
Definition: aviobuf.c:1175
av_audio_fifo_read
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
av_write_trailer
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
Definition: mux.c:1237
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:369
AVFMT_GLOBALHEADER
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition: avformat.h:466
convert_samples
static int convert_samples(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, SwrContext *resample_context)
Convert the input audio samples into the output sample format.
Definition: transcode_aac.c:487
avcodec_parameters_to_context
int avcodec_parameters_to_context(AVCodecContext *codec, const AVCodecParameters *par)
Fill the codec context based on the values from the supplied codec parameters.
Definition: codec_par.c:147
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:350
init_converted_samples
static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size)
Initialize a temporary storage for the specified number of audio samples.
Definition: transcode_aac.c:443
audio_fifo.h
avcodec_send_frame
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
Definition: encode.c:361
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVStream
Stream structure.
Definition: avformat.h:847
output_packet
static void output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int eof)
Definition: ffmpeg.c:897
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1276
avformat.h
AVCodecContext
main external API structure.
Definition: avcodec.h:379
channel_layout.h
AVRational::den
int den
Denominator.
Definition: rational.h:60
avformat_free_context
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: utils.c:4301
avio_open
int avio_open(AVIOContext **s, const char *url, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
Definition: aviobuf.c:1120
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:259
av_guess_format
const AVOutputFormat * av_guess_format(const char *short_name, const char *filename, const char *mime_type)
Return the output format in the list of registered output formats which best matches the provided par...
Definition: format.c:51
av_get_default_channel_layout
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
Definition: channel_layout.c:231
AVPacket
This structure stores compressed data.
Definition: packet.h:350
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
AVERROR_EXIT
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:56
avstring.h
write_output_file_trailer
static int write_output_file_trailer(AVFormatContext *output_format_context)
Write the trailer of the output file container.
Definition: transcode_aac.c:777