FFmpeg: doc/examples/transcode_aac.c Source File

FFmpeg
transcode_aac.c
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1 /*
2  * Copyright (c) 2013-2022 Andreas Unterweger
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 /**
22  * @file audio transcoding to MPEG/AAC API usage example
23  * @example transcode_aac.c
24  *
25  * Convert an input audio file to AAC in an MP4 container. Formats other than
26  * MP4 are supported based on the output file extension.
27  * @author Andreas Unterweger (dustsigns@gmail.com)
28  */
29 
30 #include <stdio.h>
31 
32 #include "libavformat/avformat.h"
33 #include "libavformat/avio.h"
34 
35 #include "libavcodec/avcodec.h"
36 
37 #include "libavutil/audio_fifo.h"
38 #include "libavutil/avassert.h"
39 #include "libavutil/avstring.h"
40 #include "libavutil/channel_layout.h"
41 #include "libavutil/frame.h"
42 #include "libavutil/opt.h"
43 
44 #include "libswresample/swresample.h"
45 
46 /* The output bit rate in bit/s */
47  #define OUTPUT_BIT_RATE 96000
48 /* The number of output channels */
49  #define OUTPUT_CHANNELS 2
50 
51 /**
52  * Open an input file and the required decoder.
53  * @param filename File to be opened
54  * @param[out] input_format_context Format context of opened file
55  * @param[out] input_codec_context Codec context of opened file
56  * @return Error code (0 if successful)
57  */
58  static int open_input_file(const char *filename,
59  AVFormatContext **input_format_context,
60  AVCodecContext **input_codec_context)
61 {
62  AVCodecContext *avctx;
63  const AVCodec *input_codec;
64  const AVStream *stream;
65  int error;
66 
67  /* Open the input file to read from it. */
68  if ((error = avformat_open_input(input_format_context, filename, NULL,
69  NULL)) < 0) {
70  fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
71  filename, av_err2str(error));
72  *input_format_context = NULL;
73  return error;
74  }
75 
76  /* Get information on the input file (number of streams etc.). */
77  if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
78  fprintf(stderr, "Could not open find stream info (error '%s')\n",
79  av_err2str(error));
80  avformat_close_input(input_format_context);
81  return error;
82  }
83 
84  /* Make sure that there is only one stream in the input file. */
85  if ((*input_format_context)->nb_streams != 1) {
86  fprintf(stderr, "Expected one audio input stream, but found %d\n",
87  (*input_format_context)->nb_streams);
88  avformat_close_input(input_format_context);
89  return AVERROR_EXIT;
90  }
91 
92  stream = (*input_format_context)->streams[0];
93 
94  /* Find a decoder for the audio stream. */
95  if (!(input_codec = avcodec_find_decoder(stream->codecpar->codec_id))) {
96  fprintf(stderr, "Could not find input codec\n");
97  avformat_close_input(input_format_context);
98  return AVERROR_EXIT;
99  }
100 
101  /* Allocate a new decoding context. */
102  avctx = avcodec_alloc_context3(input_codec);
103  if (!avctx) {
104  fprintf(stderr, "Could not allocate a decoding context\n");
105  avformat_close_input(input_format_context);
106  return AVERROR(ENOMEM);
107  }
108 
109  /* Initialize the stream parameters with demuxer information. */
110  error = avcodec_parameters_to_context(avctx, stream->codecpar);
111  if (error < 0) {
112  avformat_close_input(input_format_context);
113  avcodec_free_context(&avctx);
114  return error;
115  }
116 
117  /* Open the decoder for the audio stream to use it later. */
118  if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
119  fprintf(stderr, "Could not open input codec (error '%s')\n",
120  av_err2str(error));
121  avcodec_free_context(&avctx);
122  avformat_close_input(input_format_context);
123  return error;
124  }
125 
126  /* Set the packet timebase for the decoder. */
127  avctx->pkt_timebase = stream->time_base;
128 
129  /* Save the decoder context for easier access later. */
130  *input_codec_context = avctx;
131 
132  return 0;
133 }
134 
135 /**
136  * Open an output file and the required encoder.
137  * Also set some basic encoder parameters.
138  * Some of these parameters are based on the input file's parameters.
139  * @param filename File to be opened
140  * @param input_codec_context Codec context of input file
141  * @param[out] output_format_context Format context of output file
142  * @param[out] output_codec_context Codec context of output file
143  * @return Error code (0 if successful)
144  */
145  static int open_output_file(const char *filename,
146  AVCodecContext *input_codec_context,
147  AVFormatContext **output_format_context,
148  AVCodecContext **output_codec_context)
149 {
150  AVCodecContext *avctx = NULL;
151  AVIOContext *output_io_context = NULL;
152  AVStream *stream = NULL;
153  const AVCodec *output_codec = NULL;
154  int error;
155 
156  /* Open the output file to write to it. */
157  if ((error = avio_open(&output_io_context, filename,
158  AVIO_FLAG_WRITE)) < 0) {
159  fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
160  filename, av_err2str(error));
161  return error;
162  }
163 
164  /* Create a new format context for the output container format. */
165  if (!(*output_format_context = avformat_alloc_context())) {
166  fprintf(stderr, "Could not allocate output format context\n");
167  return AVERROR(ENOMEM);
168  }
169 
170  /* Associate the output file (pointer) with the container format context. */
171  (*output_format_context)->pb = output_io_context;
172 
173  /* Guess the desired container format based on the file extension. */
174  if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
175  NULL))) {
176  fprintf(stderr, "Could not find output file format\n");
177  goto cleanup;
178  }
179 
180  if (!((*output_format_context)->url = av_strdup(filename))) {
181  fprintf(stderr, "Could not allocate url.\n");
182  error = AVERROR(ENOMEM);
183  goto cleanup;
184  }
185 
186  /* Find the encoder to be used by its name. */
187  if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
188  fprintf(stderr, "Could not find an AAC encoder.\n");
189  goto cleanup;
190  }
191 
192  /* Create a new audio stream in the output file container. */
193  if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
194  fprintf(stderr, "Could not create new stream\n");
195  error = AVERROR(ENOMEM);
196  goto cleanup;
197  }
198 
199  avctx = avcodec_alloc_context3(output_codec);
200  if (!avctx) {
201  fprintf(stderr, "Could not allocate an encoding context\n");
202  error = AVERROR(ENOMEM);
203  goto cleanup;
204  }
205 
206  /* Set the basic encoder parameters.
207  * The input file's sample rate is used to avoid a sample rate conversion. */
208  av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
209  avctx->sample_rate = input_codec_context->sample_rate;
210  avctx->sample_fmt = output_codec->sample_fmts[0];
211  avctx->bit_rate = OUTPUT_BIT_RATE;
212 
213  /* Set the sample rate for the container. */
214  stream->time_base.den = input_codec_context->sample_rate;
215  stream->time_base.num = 1;
216 
217  /* Some container formats (like MP4) require global headers to be present.
218  * Mark the encoder so that it behaves accordingly. */
219  if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
220  avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
221 
222  /* Open the encoder for the audio stream to use it later. */
223  if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
224  fprintf(stderr, "Could not open output codec (error '%s')\n",
225  av_err2str(error));
226  goto cleanup;
227  }
228 
229  error = avcodec_parameters_from_context(stream->codecpar, avctx);
230  if (error < 0) {
231  fprintf(stderr, "Could not initialize stream parameters\n");
232  goto cleanup;
233  }
234 
235  /* Save the encoder context for easier access later. */
236  *output_codec_context = avctx;
237 
238  return 0;
239 
240 cleanup:
241  avcodec_free_context(&avctx);
242  avio_closep(&(*output_format_context)->pb);
243  avformat_free_context(*output_format_context);
244  *output_format_context = NULL;
245  return error < 0 ? error : AVERROR_EXIT;
246 }
247 
248 /**
249  * Initialize one data packet for reading or writing.
250  * @param[out] packet Packet to be initialized
251  * @return Error code (0 if successful)
252  */
253  static int init_packet(AVPacket **packet)
254 {
255  if (!(*packet = av_packet_alloc())) {
256  fprintf(stderr, "Could not allocate packet\n");
257  return AVERROR(ENOMEM);
258  }
259  return 0;
260 }
261 
262 /**
263  * Initialize one audio frame for reading from the input file.
264  * @param[out] frame Frame to be initialized
265  * @return Error code (0 if successful)
266  */
267  static int init_input_frame(AVFrame **frame)
268 {
269  if (!(*frame = av_frame_alloc())) {
270  fprintf(stderr, "Could not allocate input frame\n");
271  return AVERROR(ENOMEM);
272  }
273  return 0;
274 }
275 
276 /**
277  * Initialize the audio resampler based on the input and output codec settings.
278  * If the input and output sample formats differ, a conversion is required
279  * libswresample takes care of this, but requires initialization.
280  * @param input_codec_context Codec context of the input file
281  * @param output_codec_context Codec context of the output file
282  * @param[out] resample_context Resample context for the required conversion
283  * @return Error code (0 if successful)
284  */
285  static int init_resampler(AVCodecContext *input_codec_context,
286  AVCodecContext *output_codec_context,
287  SwrContext **resample_context)
288 {
289  int error;
290 
291  /*
292  * Create a resampler context for the conversion.
293  * Set the conversion parameters.
294  */
295  error = swr_alloc_set_opts2(resample_context,
296  &output_codec_context->ch_layout,
297  output_codec_context->sample_fmt,
298  output_codec_context->sample_rate,
299  &input_codec_context->ch_layout,
300  input_codec_context->sample_fmt,
301  input_codec_context->sample_rate,
302  0, NULL);
303  if (error < 0) {
304  fprintf(stderr, "Could not allocate resample context\n");
305  return error;
306  }
307  /*
308  * Perform a sanity check so that the number of converted samples is
309  * not greater than the number of samples to be converted.
310  * If the sample rates differ, this case has to be handled differently
311  */
312  av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
313 
314  /* Open the resampler with the specified parameters. */
315  if ((error = swr_init(*resample_context)) < 0) {
316  fprintf(stderr, "Could not open resample context\n");
317  swr_free(resample_context);
318  return error;
319  }
320  return 0;
321 }
322 
323 /**
324  * Initialize a FIFO buffer for the audio samples to be encoded.
325  * @param[out] fifo Sample buffer
326  * @param output_codec_context Codec context of the output file
327  * @return Error code (0 if successful)
328  */
329  static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
330 {
331  /* Create the FIFO buffer based on the specified output sample format. */
332  if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
333  output_codec_context->ch_layout.nb_channels, 1))) {
334  fprintf(stderr, "Could not allocate FIFO\n");
335  return AVERROR(ENOMEM);
336  }
337  return 0;
338 }
339 
340 /**
341  * Write the header of the output file container.
342  * @param output_format_context Format context of the output file
343  * @return Error code (0 if successful)
344  */
345  static int write_output_file_header(AVFormatContext *output_format_context)
346 {
347  int error;
348  if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
349  fprintf(stderr, "Could not write output file header (error '%s')\n",
350  av_err2str(error));
351  return error;
352  }
353  return 0;
354 }
355 
356 /**
357  * Decode one audio frame from the input file.
358  * @param frame Audio frame to be decoded
359  * @param input_format_context Format context of the input file
360  * @param input_codec_context Codec context of the input file
361  * @param[out] data_present Indicates whether data has been decoded
362  * @param[out] finished Indicates whether the end of file has
363  * been reached and all data has been
364  * decoded. If this flag is false, there
365  * is more data to be decoded, i.e., this
366  * function has to be called again.
367  * @return Error code (0 if successful)
368  */
369  static int decode_audio_frame(AVFrame *frame,
370  AVFormatContext *input_format_context,
371  AVCodecContext *input_codec_context,
372  int *data_present, int *finished)
373 {
374  /* Packet used for temporary storage. */
375  AVPacket *input_packet;
376  int error;
377 
378  error = init_packet(&input_packet);
379  if (error < 0)
380  return error;
381 
382  *data_present = 0;
383  *finished = 0;
384  /* Read one audio frame from the input file into a temporary packet. */
385  if ((error = av_read_frame(input_format_context, input_packet)) < 0) {
386  /* If we are at the end of the file, flush the decoder below. */
387  if (error == AVERROR_EOF)
388  *finished = 1;
389  else {
390  fprintf(stderr, "Could not read frame (error '%s')\n",
391  av_err2str(error));
392  goto cleanup;
393  }
394  }
395 
396  /* Send the audio frame stored in the temporary packet to the decoder.
397  * The input audio stream decoder is used to do this. */
398  if ((error = avcodec_send_packet(input_codec_context, input_packet)) < 0) {
399  fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
400  av_err2str(error));
401  goto cleanup;
402  }
403 
404  /* Receive one frame from the decoder. */
405  error = avcodec_receive_frame(input_codec_context, frame);
406  /* If the decoder asks for more data to be able to decode a frame,
407  * return indicating that no data is present. */
408  if (error == AVERROR(EAGAIN)) {
409  error = 0;
410  goto cleanup;
411  /* If the end of the input file is reached, stop decoding. */
412  } else if (error == AVERROR_EOF) {
413  *finished = 1;
414  error = 0;
415  goto cleanup;
416  } else if (error < 0) {
417  fprintf(stderr, "Could not decode frame (error '%s')\n",
418  av_err2str(error));
419  goto cleanup;
420  /* Default case: Return decoded data. */
421  } else {
422  *data_present = 1;
423  goto cleanup;
424  }
425 
426 cleanup:
427  av_packet_free(&input_packet);
428  return error;
429 }
430 
431 /**
432  * Initialize a temporary storage for the specified number of audio samples.
433  * The conversion requires temporary storage due to the different format.
434  * The number of audio samples to be allocated is specified in frame_size.
435  * @param[out] converted_input_samples Array of converted samples. The
436  * dimensions are reference, channel
437  * (for multi-channel audio), sample.
438  * @param output_codec_context Codec context of the output file
439  * @param frame_size Number of samples to be converted in
440  * each round
441  * @return Error code (0 if successful)
442  */
443  static int init_converted_samples(uint8_t ***converted_input_samples,
444  AVCodecContext *output_codec_context,
445  int frame_size)
446 {
447  int error;
448 
449  /* Allocate as many pointers as there are audio channels.
450  * Each pointer will 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->ch_layout.nb_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->ch_layout.nb_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  */
513  static int add_samples_to_fifo(AVAudioFifo *fifo,
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  */
551  static int read_decode_convert_and_store(AVAudioFifo *fifo,
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;
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  */
618  static int init_output_frame(AVFrame **frame,
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  av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_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));
645  av_frame_free(frame);
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  */
664  static int encode_audio_frame(AVFrame *frame,
665  AVFormatContext *output_format_context,
666  AVCodecContext *output_codec_context,
667  int *data_present)
668 {
669  /* Packet used for temporary storage. */
670  AVPacket *output_packet;
671  int error;
672 
673  error = init_packet(&output_packet);
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  *data_present = 0;
684  /* Send the audio frame stored in the temporary packet to the encoder.
685  * The output audio stream encoder is used to do this. */
686  error = avcodec_send_frame(output_codec_context, frame);
687  /* Check for errors, but proceed with fetching encoded samples if the
688  * encoder signals that it has nothing more to encode. */
689  if (error < 0 && error != AVERROR_EOF) {
690  fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
691  av_err2str(error));
692  goto cleanup;
693  }
694 
695  /* Receive one encoded frame from the encoder. */
696  error = avcodec_receive_packet(output_codec_context, output_packet);
697  /* If the encoder asks for more data to be able to provide an
698  * encoded frame, return indicating that no data is present. */
699  if (error == AVERROR(EAGAIN)) {
700  error = 0;
701  goto cleanup;
702  /* If the last frame has been encoded, stop encoding. */
703  } else if (error == AVERROR_EOF) {
704  error = 0;
705  goto cleanup;
706  } else if (error < 0) {
707  fprintf(stderr, "Could not encode frame (error '%s')\n",
708  av_err2str(error));
709  goto cleanup;
710  /* Default case: Return encoded data. */
711  } else {
712  *data_present = 1;
713  }
714 
715  /* Write one audio frame from the temporary packet to the output file. */
716  if (*data_present &&
717  (error = av_write_frame(output_format_context, output_packet)) < 0) {
718  fprintf(stderr, "Could not write frame (error '%s')\n",
719  av_err2str(error));
720  goto cleanup;
721  }
722 
723 cleanup:
724  av_packet_free(&output_packet);
725  return error;
726 }
727 
728 /**
729  * Load one audio frame from the FIFO buffer, encode and write it to the
730  * output file.
731  * @param fifo Buffer used for temporary storage
732  * @param output_format_context Format context of the output file
733  * @param output_codec_context Codec context of the output file
734  * @return Error code (0 if successful)
735  */
736  static int load_encode_and_write(AVAudioFifo *fifo,
737  AVFormatContext *output_format_context,
738  AVCodecContext *output_codec_context)
739 {
740  /* Temporary storage of the output samples of the frame written to the file. */
741  AVFrame *output_frame;
742  /* Use the maximum number of possible samples per frame.
743  * If there is less than the maximum possible frame size in the FIFO
744  * buffer use this number. Otherwise, use the maximum possible frame size. */
745  const int frame_size = FFMIN(av_audio_fifo_size(fifo),
746  output_codec_context->frame_size);
747  int data_written;
748 
749  /* Initialize temporary storage for one output frame. */
750  if (init_output_frame(&output_frame, output_codec_context, frame_size))
751  return AVERROR_EXIT;
752 
753  /* Read as many samples from the FIFO buffer as required to fill the frame.
754  * The samples are stored in the frame temporarily. */
755  if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
756  fprintf(stderr, "Could not read data from FIFO\n");
757  av_frame_free(&output_frame);
758  return AVERROR_EXIT;
759  }
760 
761  /* Encode one frame worth of audio samples. */
762  if (encode_audio_frame(output_frame, output_format_context,
763  output_codec_context, &data_written)) {
764  av_frame_free(&output_frame);
765  return AVERROR_EXIT;
766  }
767  av_frame_free(&output_frame);
768  return 0;
769 }
770 
771 /**
772  * Write the trailer of the output file container.
773  * @param output_format_context Format context of the output file
774  * @return Error code (0 if successful)
775  */
776  static int write_output_file_trailer(AVFormatContext *output_format_context)
777 {
778  int error;
779  if ((error = av_write_trailer(output_format_context)) < 0) {
780  fprintf(stderr, "Could not write output file trailer (error '%s')\n",
781  av_err2str(error));
782  return error;
783  }
784  return 0;
785 }
786 
787  int main(int argc, char **argv)
788 {
789  AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
790  AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
791  SwrContext *resample_context = NULL;
792  AVAudioFifo *fifo = NULL;
793  int ret = AVERROR_EXIT;
794 
795  if (argc != 3) {
796  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
797  exit(1);
798  }
799 
800  /* Open the input file for reading. */
801  if (open_input_file(argv[1], &input_format_context,
802  &input_codec_context))
803  goto cleanup;
804  /* Open the output file for writing. */
805  if (open_output_file(argv[2], input_codec_context,
806  &output_format_context, &output_codec_context))
807  goto cleanup;
808  /* Initialize the resampler to be able to convert audio sample formats. */
809  if (init_resampler(input_codec_context, output_codec_context,
810  &resample_context))
811  goto cleanup;
812  /* Initialize the FIFO buffer to store audio samples to be encoded. */
813  if (init_fifo(&fifo, output_codec_context))
814  goto cleanup;
815  /* Write the header of the output file container. */
816  if (write_output_file_header(output_format_context))
817  goto cleanup;
818 
819  /* Loop as long as we have input samples to read or output samples
820  * to write; abort as soon as we have neither. */
821  while (1) {
822  /* Use the encoder's desired frame size for processing. */
823  const int output_frame_size = output_codec_context->frame_size;
824  int finished = 0;
825 
826  /* Make sure that there is one frame worth of samples in the FIFO
827  * buffer so that the encoder can do its work.
828  * Since the decoder's and the encoder's frame size may differ, we
829  * need to FIFO buffer to store as many frames worth of input samples
830  * that they make up at least one frame worth of output samples. */
831  while (av_audio_fifo_size(fifo) < output_frame_size) {
832  /* Decode one frame worth of audio samples, convert it to the
833  * output sample format and put it into the FIFO buffer. */
834  if (read_decode_convert_and_store(fifo, input_format_context,
835  input_codec_context,
836  output_codec_context,
837  resample_context, &finished))
838  goto cleanup;
839 
840  /* If we are at the end of the input file, we continue
841  * encoding the remaining audio samples to the output file. */
842  if (finished)
843  break;
844  }
845 
846  /* If we have enough samples for the encoder, we encode them.
847  * At the end of the file, we pass the remaining samples to
848  * the encoder. */
849  while (av_audio_fifo_size(fifo) >= output_frame_size ||
850  (finished && av_audio_fifo_size(fifo) > 0))
851  /* Take one frame worth of audio samples from the FIFO buffer,
852  * encode it and write it to the output file. */
853  if (load_encode_and_write(fifo, output_format_context,
854  output_codec_context))
855  goto cleanup;
856 
857  /* If we are at the end of the input file and have encoded
858  * all remaining samples, we can exit this loop and finish. */
859  if (finished) {
860  int data_written;
861  /* Flush the encoder as it may have delayed frames. */
862  do {
863  if (encode_audio_frame(NULL, output_format_context,
864  output_codec_context, &data_written))
865  goto cleanup;
866  } while (data_written);
867  break;
868  }
869  }
870 
871  /* Write the trailer of the output file container. */
872  if (write_output_file_trailer(output_format_context))
873  goto cleanup;
874  ret = 0;
875 
876 cleanup:
877  if (fifo)
878  av_audio_fifo_free(fifo);
879  swr_free(&resample_context);
880  if (output_codec_context)
881  avcodec_free_context(&output_codec_context);
882  if (output_format_context) {
883  avio_closep(&output_format_context->pb);
884  avformat_free_context(output_format_context);
885  }
886  if (input_codec_context)
887  avcodec_free_context(&input_codec_context);
888  if (input_format_context)
889  avformat_close_input(&input_format_context);
890 
891  return ret;
892 }
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:31
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:48
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1062
AVCodec
AVCodec.
Definition: codec.h:184
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:736
avcodec_receive_packet
int avcodec_receive_packet(AVCodecContext *avctx, AVPacket *avpkt)
Read encoded data from the encoder.
Definition: encode.c:521
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: options.c:243
open_input_file
static int open_input_file(const char *filename, AVFormatContext **input_format_context, AVCodecContext **input_codec_context)
Open an input file and the required decoder.
Definition: transcode_aac.c:58
av_frame_get_buffer
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
Definition: frame.c:242
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1034
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
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:99
av_audio_fifo_realloc
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
Definition: audio_fifo.c:99
init_fifo
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
Definition: transcode_aac.c:329
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:99
avcodec_find_encoder
const AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
Definition: allcodecs.c:959
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:330
cleanup
static av_cold void cleanup(FlashSV2Context *s)
Definition: flashsv2enc.c:130
write_output_file_header
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
Definition: transcode_aac.c:345
open_output_file
static int open_output_file(const char *filename, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context)
Open an output file and the required encoder.
Definition: transcode_aac.c:145
av_read_frame
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
Definition: demux.c:1439
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:311
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:73
AV_CODEC_FLAG_GLOBAL_HEADER
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
Definition: avcodec.h:317
avformat_close_input
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
Definition: demux.c:369
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:37
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2054
AVCodec::sample_fmts
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:208
output_packet
static int output_packet(AVFormatContext *ctx, int flush)
Definition: mpegenc.c:999
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:182
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:506
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:87
avassert.h
swr_init
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:193
avformat_open_input
int avformat_open_input(AVFormatContext **ps, const char *url, const AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
Definition: demux.c:221
avcodec_alloc_context3
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition: options.c:153
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:119
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:2205
decode_audio_frame
static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished)
Decode one audio frame from the input file.
Definition: transcode_aac.c:369
avcodec_receive_frame
int attribute_align_arg avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Return decoded output data from a decoder or encoder (when the AV_CODEC_FLAG_RECON_FRAME flag is used...
Definition: avcodec.c:709
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:624
SwrContext
The libswresample context.
Definition: swresample_internal.h:95
avformat_write_header
av_warn_unused_result int avformat_write_header(AVFormatContext *s, AVDictionary **options)
Allocate the stream private data and write the stream header to an output media file.
Definition: mux.c:451
AVFormatContext
Format I/O context.
Definition: avformat.h:1104
input_data
static void input_data(MLPEncodeContext *ctx, const void *samples, int nb_samples)
Wrapper function for inputting data in two different bit-depths.
Definition: mlpenc.c:1235
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:861
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:877
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:168
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:62
read_decode_convert_and_store
static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resampler_context, int *finished)
Read one audio frame from the input file, decode, convert and store it in the FIFO buffer.
Definition: transcode_aac.c:551
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:476
OUTPUT_BIT_RATE
#define OUTPUT_BIT_RATE
Definition: transcode_aac.c:47
avcodec_open2
int attribute_align_arg avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: avcodec.c:115
av_write_frame
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
Definition: mux.c:1194
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:964
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:440
init_input_frame
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
Definition: transcode_aac.c:267
avformat_find_stream_info
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
Definition: demux.c:2425
AVIOContext
Bytestream IO Context.
Definition: avio.h:166
swr_alloc_set_opts2
int swr_alloc_set_opts2(struct SwrContext **ps, const AVChannelLayout *out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, const AVChannelLayout *in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, int log_offset, void *log_ctx)
Allocate SwrContext if needed and set/reset common parameters.
Definition: swresample.c:85
avformat_alloc_context
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
Definition: options.c: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:121
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1050
AVCodecContext::pkt_timebase
AVRational pkt_timebase
Timebase in which pkt_dts/pts and AVPacket.dts/pts are.
Definition: avcodec.h:1764
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:787
avio.h
swr_free
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:174
init_packet
static int init_packet(AVPacket **packet)
Initialize one data packet for reading or writing.
Definition: transcode_aac.c:253
swr_convert
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t **out_arg, int out_count, const uint8_t **in_arg, int in_count)
Convert audio.
Definition: swresample.c:830
frame.h
OUTPUT_CHANNELS
#define OUTPUT_CHANNELS
Definition: transcode_aac.c:49
output_frame
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
Definition: h264dec.c:838
av_packet_alloc
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:62
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:221
init_resampler
static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context)
Initialize the audio resampler based on the input and output codec settings.
Definition: transcode_aac.c:285
av_channel_layout_default
void av_channel_layout_default(AVChannelLayout *ch_layout, int nb_channels)
Get the default channel layout for a given number of channels.
Definition: channel_layout.c:962
avcodec_send_packet
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition: decode.c:598
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:1280
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:174
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:1256
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:410
AVFMT_GLOBALHEADER
#define AVFMT_GLOBALHEADER
Format wants global header.
Definition: avformat.h:478
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:182
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:391
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
init_converted_samples
static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size)
Initialize a temporary storage for the specified number of audio samples.
Definition: transcode_aac.c:443
audio_fifo.h
avcodec_send_frame
int avcodec_send_frame(AVCodecContext *avctx, const AVFrame *frame)
Supply a raw video or audio frame to the encoder.
Definition: encode.c:483
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVStream
Stream structure.
Definition: avformat.h:838
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
avformat.h
AVCodecContext
main external API structure.
Definition: avcodec.h:426
channel_layout.h
AVRational::den
int den
Denominator.
Definition: rational.h:60
avformat_free_context
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
Definition: avformat.c:96
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:1215
av_channel_layout_copy
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
Definition: channel_layout.c:639
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:270
av_guess_format
const AVOutputFormat * av_guess_format(const char *short_name, const char *filename, const char *mime_type)
Return the output format in the list of registered output formats which best matches the provided par...
Definition: format.c:53
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:62
AVPacket
This structure stores compressed data.
Definition: packet.h:351
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
AVERROR_EXIT
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
Definition: error.h:58
avstring.h
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:776
Generated on Tue Feb 28 2023 21:33:03 for FFmpeg by   doxygen 1.8.17

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