1 /*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19 /**
20 * @file
21 * simple audio converter
22 *
23 * @example transcode_aac.c
24 * Convert an input audio file to AAC in an MP4 container using FFmpeg.
25 * @author Andreas Unterweger (dustsigns@gmail.com)
26 */
27
28 #include <stdio.h>
29
32
34
40
42
43 /** The output bit rate in kbit/s */
44 #define OUTPUT_BIT_RATE 96000
45 /** The number of output channels */
46 #define OUTPUT_CHANNELS 2
47
48 /**
49 * Convert an error code into a text message.
50 * @param error Error code to be converted
51 * @return Corresponding error text (not thread-safe)
52 */
54 {
55 static char error_buffer[255];
56 av_strerror(error, error_buffer,
sizeof(error_buffer));
57 return error_buffer;
58 }
59
60 /** Open an input file and the required decoder. */
64 {
66 int error;
67
68 /** Open the input file to read from it. */
71 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
73 *input_format_context =
NULL;
74 return error;
75 }
76
77 /** Get information on the input file (number of streams etc.). */
79 fprintf(stderr, "Could not open find stream info (error '%s')\n",
82 return error;
83 }
84
85 /** Make sure that there is only one stream in the input file. */
86 if ((*input_format_context)->nb_streams != 1) {
87 fprintf(stderr, "Expected one audio input stream, but found %d\n",
88 (*input_format_context)->nb_streams);
91 }
92
93 /** Find a decoder for the audio stream. */
94 if (!(input_codec =
avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
95 fprintf(stderr, "Could not find input codec\n");
98 }
99
100 /** Open the decoder for the audio stream to use it later. */
101 if ((error =
avcodec_open2((*input_format_context)->streams[0]->codec,
102 input_codec,
NULL)) < 0) {
103 fprintf(stderr, "Could not open input codec (error '%s')\n",
106 return error;
107 }
108
109 /** Save the decoder context for easier access later. */
110 *input_codec_context = (*input_format_context)->streams[0]->
codec;
111
112 return 0;
113 }
114
115 /**
116 * Open an output file and the required encoder.
117 * Also set some basic encoder parameters.
118 * Some of these parameters are based on the input file's parameters.
119 */
124 {
128 int error;
129
130 /** Open the output file to write to it. */
131 if ((error =
avio_open(&output_io_context, filename,
133 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
135 return error;
136 }
137
138 /** Create a new format context for the output container format. */
140 fprintf(stderr, "Could not allocate output format context\n");
142 }
143
144 /** Associate the output file (pointer) with the container format context. */
145 (*output_format_context)->pb = output_io_context;
146
147 /** Guess the desired container format based on the file extension. */
150 fprintf(stderr, "Could not find output file format\n");
152 }
153
154 av_strlcpy((*output_format_context)->filename, filename,
155 sizeof((*output_format_context)->filename));
156
157 /** Find the encoder to be used by its name. */
159 fprintf(stderr, "Could not find an AAC encoder.\n");
161 }
162
163 /** Create a new audio stream in the output file container. */
165 fprintf(stderr, "Could not create new stream\n");
168 }
169
170 /** Save the encoder context for easier access later. */
171 *output_codec_context = stream->
codec;
172
173 /**
174 * Set the basic encoder parameters.
175 * The input file's sample rate is used to avoid a sample rate conversion.
176 */
179 (*output_codec_context)->sample_rate = input_codec_context->
sample_rate;
180 (*output_codec_context)->sample_fmt = output_codec->
sample_fmts[0];
182
183 /** Allow the use of the experimental AAC encoder */
185
186 /** Set the sample rate for the container. */
189
190 /**
191 * Some container formats (like MP4) require global headers to be present
192 * Mark the encoder so that it behaves accordingly.
193 */
196
197 /** Open the encoder for the audio stream to use it later. */
199 fprintf(stderr, "Could not open output codec (error '%s')\n",
202 }
203
204 return 0;
205
209 *output_format_context =
NULL;
211 }
212
213 /** Initialize one data packet for reading or writing. */
215 {
217 /** Set the packet data and size so that it is recognized as being empty. */
220 }
221
222 /** Initialize one audio frame for reading from the input file */
224 {
226 fprintf(stderr, "Could not allocate input frame\n");
228 }
229 return 0;
230 }
231
232 /**
233 * Initialize the audio resampler based on the input and output codec settings.
234 * If the input and output sample formats differ, a conversion is required
235 * libswresample takes care of this, but requires initialization.
236 */
240 {
241 int error;
242
243 /**
244 * Create a resampler context for the conversion.
245 * Set the conversion parameters.
246 * Default channel layouts based on the number of channels
247 * are assumed for simplicity (they are sometimes not detected
248 * properly by the demuxer and/or decoder).
249 */
258 if (!*resample_context) {
259 fprintf(stderr, "Could not allocate resample context\n");
261 }
262 /**
263 * Perform a sanity check so that the number of converted samples is
264 * not greater than the number of samples to be converted.
265 * If the sample rates differ, this case has to be handled differently
266 */
268
269 /** Open the resampler with the specified parameters. */
270 if ((error =
swr_init(*resample_context)) < 0) {
271 fprintf(stderr, "Could not open resample context\n");
273 return error;
274 }
275 return 0;
276 }
277
278 /** Initialize a FIFO buffer for the audio samples to be encoded. */
280 {
281 /** Create the FIFO buffer based on the specified output sample format. */
283 output_codec_context->
channels, 1))) {
284 fprintf(stderr, "Could not allocate FIFO\n");
286 }
287 return 0;
288 }
289
290 /** Write the header of the output file container. */
292 {
293 int error;
295 fprintf(stderr, "Could not write output file header (error '%s')\n",
297 return error;
298 }
299 return 0;
300 }
301
302 /** Decode one audio frame from the input file. */
306 int *data_present, int *finished)
307 {
308 /** Packet used for temporary storage. */
310 int error;
312
313 /** Read one audio frame from the input file into a temporary packet. */
314 if ((error =
av_read_frame(input_format_context, &input_packet)) < 0) {
315 /** If we are at the end of the file, flush the decoder below. */
317 *finished = 1;
318 else {
319 fprintf(stderr, "Could not read frame (error '%s')\n",
321 return error;
322 }
323 }
324
325 /**
326 * Decode the audio frame stored in the temporary packet.
327 * The input audio stream decoder is used to do this.
328 * If we are at the end of the file, pass an empty packet to the decoder
329 * to flush it.
330 */
332 data_present, &input_packet)) < 0) {
333 fprintf(stderr, "Could not decode frame (error '%s')\n",
336 return error;
337 }
338
339 /**
340 * If the decoder has not been flushed completely, we are not finished,
341 * so that this function has to be called again.
342 */
343 if (*finished && *data_present)
344 *finished = 0;
346 return 0;
347 }
348
349 /**
350 * Initialize a temporary storage for the specified number of audio samples.
351 * The conversion requires temporary storage due to the different format.
352 * The number of audio samples to be allocated is specified in frame_size.
353 */
357 {
358 int error;
359
360 /**
361 * Allocate as many pointers as there are audio channels.
362 * Each pointer will later point to the audio samples of the corresponding
363 * channels (although it may be NULL for interleaved formats).
364 */
365 if (!(*converted_input_samples = calloc(output_codec_context->
channels,
366 sizeof(**converted_input_samples)))) {
367 fprintf(stderr, "Could not allocate converted input sample pointers\n");
369 }
370
371 /**
372 * Allocate memory for the samples of all channels in one consecutive
373 * block for convenience.
374 */
377 frame_size,
379 fprintf(stderr,
380 "Could not allocate converted input samples (error '%s')\n",
382 av_freep(&(*converted_input_samples)[0]);
383 free(*converted_input_samples);
384 return error;
385 }
386 return 0;
387 }
388
389 /**
390 * Convert the input audio samples into the output sample format.
391 * The conversion happens on a per-frame basis, the size of which is specified
392 * by frame_size.
393 */
397 {
398 int error;
399
400 /** Convert the samples using the resampler. */
402 converted_data, frame_size,
403 input_data , frame_size)) < 0) {
404 fprintf(stderr, "Could not convert input samples (error '%s')\n",
406 return error;
407 }
408
409 return 0;
410 }
411
412 /** Add converted input audio samples to the FIFO buffer for later processing. */
414 uint8_t **converted_input_samples,
416 {
417 int error;
418
419 /**
420 * Make the FIFO as large as it needs to be to hold both,
421 * the old and the new samples.
422 */
424 fprintf(stderr, "Could not reallocate FIFO\n");
425 return error;
426 }
427
428 /** Store the new samples in the FIFO buffer. */
430 frame_size) < frame_size) {
431 fprintf(stderr, "Could not write data to FIFO\n");
433 }
434 return 0;
435 }
436
437 /**
438 * Read one audio frame from the input file, decodes, converts and stores
439 * it in the FIFO buffer.
440 */
446 int *finished)
447 {
448 /** Temporary storage of the input samples of the frame read from the file. */
450 /** Temporary storage for the converted input samples. */
452 int data_present;
454
455 /** Initialize temporary storage for one input frame. */
458 /** Decode one frame worth of audio samples. */
460 input_codec_context, &data_present, finished))
462 /**
463 * If we are at the end of the file and there are no more samples
464 * in the decoder which are delayed, we are actually finished.
465 * This must not be treated as an error.
466 */
467 if (*finished && !data_present) {
468 ret = 0;
470 }
471 /** If there is decoded data, convert and store it */
472 if (data_present) {
473 /** Initialize the temporary storage for the converted input samples. */
477
478 /**
479 * Convert the input samples to the desired output sample format.
480 * This requires a temporary storage provided by converted_input_samples.
481 */
485
486 /** Add the converted input samples to the FIFO buffer for later processing. */
490 ret = 0;
491 }
492 ret = 0;
493
495 if (converted_input_samples) {
496 av_freep(&converted_input_samples[0]);
497 free(converted_input_samples);
498 }
500
501 return ret;
502 }
503
504 /**
505 * Initialize one input frame for writing to the output file.
506 * The frame will be exactly frame_size samples large.
507 */
511 {
512 int error;
513
514 /** Create a new frame to store the audio samples. */
516 fprintf(stderr, "Could not allocate output frame\n");
518 }
519
520 /**
521 * Set the frame's parameters, especially its size and format.
522 * av_frame_get_buffer needs this to allocate memory for the
523 * audio samples of the frame.
524 * Default channel layouts based on the number of channels
525 * are assumed for simplicity.
526 */
529 (*frame)->format = output_codec_context->
sample_fmt;
530 (*frame)->sample_rate = output_codec_context->
sample_rate;
531
532 /**
533 * Allocate the samples of the created frame. This call will make
534 * sure that the audio frame can hold as many samples as specified.
535 */
537 fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
540 return error;
541 }
542
543 return 0;
544 }
545
546 /** Global timestamp for the audio frames */
548
549 /** Encode one frame worth of audio to the output file. */
553 int *data_present)
554 {
555 /** Packet used for temporary storage. */
557 int error;
559
560 /** Set a timestamp based on the sample rate for the container. */
561 if (frame) {
564 }
565
566 /**
567 * Encode the audio frame and store it in the temporary packet.
568 * The output audio stream encoder is used to do this.
569 */
571 frame, data_present)) < 0) {
572 fprintf(stderr, "Could not encode frame (error '%s')\n",
575 return error;
576 }
577
578 /** Write one audio frame from the temporary packet to the output file. */
579 if (*data_present) {
580 if ((error =
av_write_frame(output_format_context, &output_packet)) < 0) {
581 fprintf(stderr, "Could not write frame (error '%s')\n",
584 return error;
585 }
586
588 }
589
590 return 0;
591 }
592
593 /**
594 * Load one audio frame from the FIFO buffer, encode and write it to the
595 * output file.
596 */
600 {
601 /** Temporary storage of the output samples of the frame written to the file. */
603 /**
604 * Use the maximum number of possible samples per frame.
605 * If there is less than the maximum possible frame size in the FIFO
606 * buffer use this number. Otherwise, use the maximum possible frame size
607 */
610 int data_written;
611
612 /** Initialize temporary storage for one output frame. */
615
616 /**
617 * Read as many samples from the FIFO buffer as required to fill the frame.
618 * The samples are stored in the frame temporarily.
619 */
621 fprintf(stderr, "Could not read data from FIFO\n");
624 }
625
626 /** Encode one frame worth of audio samples. */
628 output_codec_context, &data_written)) {
631 }
633 return 0;
634 }
635
636 /** Write the trailer of the output file container. */
638 {
639 int error;
641 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
643 return error;
644 }
645 return 0;
646 }
647
648 /** Convert an audio file to an AAC file in an MP4 container. */
649 int main(
int argc,
char **argv)
650 {
656
657 if (argc < 3) {
658 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
659 exit(1);
660 }
661
662 /** Register all codecs and formats so that they can be used. */
664 /** Open the input file for reading. */
666 &input_codec_context))
668 /** Open the output file for writing. */
670 &output_format_context, &output_codec_context))
672 /** Initialize the resampler to be able to convert audio sample formats. */
674 &resample_context))
676 /** Initialize the FIFO buffer to store audio samples to be encoded. */
677 if (
init_fifo(&fifo, output_codec_context))
679 /** Write the header of the output file container. */
682
683 /**
684 * Loop as long as we have input samples to read or output samples
685 * to write; abort as soon as we have neither.
686 */
687 while (1) {
688 /** Use the encoder's desired frame size for processing. */
689 const int output_frame_size = output_codec_context->frame_size;
690 int finished = 0;
691
692 /**
693 * Make sure that there is one frame worth of samples in the FIFO
694 * buffer so that the encoder can do its work.
695 * Since the decoder's and the encoder's frame size may differ, we
696 * need to FIFO buffer to store as many frames worth of input samples
697 * that they make up at least one frame worth of output samples.
698 */
700 /**
701 * Decode one frame worth of audio samples, convert it to the
702 * output sample format and put it into the FIFO buffer.
703 */
705 input_codec_context,
706 output_codec_context,
707 resample_context, &finished))
709
710 /**
711 * If we are at the end of the input file, we continue
712 * encoding the remaining audio samples to the output file.
713 */
714 if (finished)
715 break;
716 }
717
718 /**
719 * If we have enough samples for the encoder, we encode them.
720 * At the end of the file, we pass the remaining samples to
721 * the encoder.
722 */
725 /**
726 * Take one frame worth of audio samples from the FIFO buffer,
727 * encode it and write it to the output file.
728 */
730 output_codec_context))
732
733 /**
734 * If we are at the end of the input file and have encoded
735 * all remaining samples, we can exit this loop and finish.
736 */
737 if (finished) {
738 int data_written;
739 /** Flush the encoder as it may have delayed frames. */
740 do {
742 output_codec_context, &data_written))
744 } while (data_written);
745 break;
746 }
747 }
748
749 /** Write the trailer of the output file container. */
752 ret = 0;
753
755 if (fifo)
758 if (output_codec_context)
760 if (output_format_context) {
763 }
764 if (input_codec_context)
766 if (input_format_context)
768
769 return ret;
770 }
int avio_open(AVIOContext **s, const char *url, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
const struct AVCodec * codec
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
void av_free_packet(AVPacket *pkt)
Free a packet.
This structure describes decoded (raw) audio or video data.
AVCodec * avcodec_find_encoder(enum AVCodecID id)
Find a registered encoder with a matching codec ID.
int main(int argc, char **argv)
Convert an audio file to an AAC file in an MP4 container.
int avformat_write_header(AVFormatContext *s, AVDictionary **options)
Allocate the stream private data and write the stream header to an output media file.
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
int avformat_open_input(AVFormatContext **ps, const char *filename, AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
#define AVIO_FLAG_WRITE
write-only
int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Encode a frame of audio.
static void init_packet(AVPacket *packet)
Initialize one data packet for reading or writing.
static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context)
Initialize the audio resampler based on the input and output codec settings.
static int open_input_file(const char *filename, AVFormatContext **input_format_context, AVCodecContext **input_codec_context)
Open an input file and the required decoder.
static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, AVCodecContext *output_codec_context, int *data_present)
Encode one frame worth of audio to the output file.
static int output_packet(AVFormatContext *ctx, int flush)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
enum AVSampleFormat sample_fmt
audio sample format
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resampler_context, int *finished)
Read one audio frame from the input file, decodes, converts and stores it in the FIFO buffer...
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static int convert_samples(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, SwrContext *resample_context)
Convert the input audio samples into the output sample format.
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
#define AVERROR_EOF
End of file.
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
int avcodec_close(AVCodecContext *avctx)
Close a given AVCodecContext and free all the data associated with it (but not the AVCodecContext its...
libswresample public header
static int write_output_file_header(AVFormatContext *output_format_context)
Write the header of the output file container.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
The libswresample context.
simple assert() macros that are a bit more flexible than ISO C assert().
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context)
Load one audio frame from the FIFO buffer, encode and write it to the output file.
Libavcodec external API header.
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
reference-counted frame API
uint64_t channel_layout
Audio channel layout.
AVCodecContext * codec
Codec context associated with this stream.
Context for an Audio FIFO Buffer.
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
static int open_output_file(const char *filename, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context)
Open an output file and the required encoder.
static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished)
Decode one audio frame from the input file.
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.
int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, const AVPacket *avpkt)
Decode the audio frame of size avpkt->size from avpkt->data into frame.
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...
static int add_samples_to_fifo(AVAudioFifo *fifo, uint8_t **converted_input_samples, const int frame_size)
Add converted input audio samples to the FIFO buffer for later processing.
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
int frame_size
Number of samples per channel in an audio frame.
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...
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
Initialize a FIFO buffer for the audio samples to be encoded.
#define OUTPUT_BIT_RATE
The output bit rate in kbit/s.
int sample_rate
samples per second
main external API structure.
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
void avformat_free_context(AVFormatContext *s)
Free an AVFormatContext and all its streams.
static const char * get_error_text(const int error)
Convert an error code into a text message.
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)
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
Return the next frame of a stream.
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
static int64_t pts
Global timestamp for the audio frames.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
#define OUTPUT_CHANNELS
The number of output channels.
#define AV_CODEC_FLAG_GLOBAL_HEADER
Place global headers in extradata instead of every keyframe.
int av_strerror(int errnum, char *errbuf, size_t errbuf_size)
Put a description of the AVERROR code errnum in errbuf.
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
int avformat_find_stream_info(AVFormatContext *ic, AVDictionary **options)
Read packets of a media file to get stream information.
static int write_output_file_trailer(AVFormatContext *output_format_context)
Write the trailer of the output file container.
void av_init_packet(AVPacket *pkt)
Initialize optional fields of a packet with default values.
void avformat_close_input(AVFormatContext **s)
Close an opened input AVFormatContext.
int channels
number of audio channels
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
int av_write_trailer(AVFormatContext *s)
Write the stream trailer to an output media file and free the file private data.
static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size)
Initialize a temporary storage for the specified number of audio samples.
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, int frame_size)
Initialize one input frame for writing to the output file.
AVRational time_base
This is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented...
uint8_t ** extended_data
pointers to the data planes/channels.
This structure stores compressed data.
void av_register_all(void)
Initialize libavformat and register all the muxers, demuxers and protocols.
int avio_closep(AVIOContext **s)
Close the resource accessed by the AVIOContext *s, free it and set the pointer pointing to it to NULL...
int nb_samples
number of audio samples (per channel) described by this frame
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
static av_cold void cleanup(FlashSV2Context *s)