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 /** Open an input file and the required decoder. */
52 {
56
57 /** Open the input file to read from it. */
60 fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
62 *input_format_context =
NULL;
64 }
65
66 /** Get information on the input file (number of streams etc.). */
68 fprintf(stderr, "Could not open find stream info (error '%s')\n",
72 }
73
74 /** Make sure that there is only one stream in the input file. */
75 if ((*input_format_context)->nb_streams != 1) {
76 fprintf(stderr, "Expected one audio input stream, but found %d\n",
77 (*input_format_context)->nb_streams);
80 }
81
82 /** Find a decoder for the audio stream. */
83 if (!(input_codec =
avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
84 fprintf(stderr, "Could not find input codec\n");
87 }
88
89 /** allocate a new decoding context */
91 if (!avctx) {
92 fprintf(stderr, "Could not allocate a decoding context\n");
95 }
96
97 /** initialize the stream parameters with demuxer information */
99 if (error < 0) {
103 }
104
105 /** Open the decoder for the audio stream to use it later. */
107 fprintf(stderr, "Could not open input codec (error '%s')\n",
112 }
113
114 /** Save the decoder context for easier access later. */
115 *input_codec_context = avctx;
116
117 return 0;
118 }
119
120 /**
121 * Open an output file and the required encoder.
122 * Also set some basic encoder parameters.
123 * Some of these parameters are based on the input file's parameters.
124 */
129 {
135
136 /** Open the output file to write to it. */
137 if ((error =
avio_open(&output_io_context, filename,
139 fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
142 }
143
144 /** Create a new format context for the output container format. */
146 fprintf(stderr, "Could not allocate output format context\n");
148 }
149
150 /** Associate the output file (pointer) with the container format context. */
151 (*output_format_context)->pb = output_io_context;
152
153 /** Guess the desired container format based on the file extension. */
156 fprintf(stderr, "Could not find output file format\n");
158 }
159
160 av_strlcpy((*output_format_context)->filename, filename,
161 sizeof((*output_format_context)->filename));
162
163 /** Find the encoder to be used by its name. */
165 fprintf(stderr, "Could not find an AAC encoder.\n");
167 }
168
169 /** Create a new audio stream in the output file container. */
171 fprintf(stderr, "Could not create new stream\n");
174 }
175
177 if (!avctx) {
178 fprintf(stderr, "Could not allocate an encoding context\n");
181 }
182
183 /**
184 * Set the basic encoder parameters.
185 * The input file's sample rate is used to avoid a sample rate conversion.
186 */
192
193 /** Allow the use of the experimental AAC encoder */
195
196 /** Set the sample rate for the container. */
199
200 /**
201 * Some container formats (like MP4) require global headers to be present
202 * Mark the encoder so that it behaves accordingly.
203 */
206
207 /** Open the encoder for the audio stream to use it later. */
209 fprintf(stderr, "Could not open output codec (error '%s')\n",
212 }
213
215 if (error < 0) {
216 fprintf(stderr, "Could not initialize stream parameters\n");
218 }
219
220 /** Save the encoder context for easier access later. */
221 *output_codec_context = avctx;
222
223 return 0;
224
229 *output_format_context =
NULL;
231 }
232
233 /** Initialize one data packet for reading or writing. */
235 {
237 /** Set the packet data and size so that it is recognized as being empty. */
240 }
241
242 /** Initialize one audio frame for reading from the input file */
244 {
246 fprintf(stderr, "Could not allocate input frame\n");
248 }
249 return 0;
250 }
251
252 /**
253 * Initialize the audio resampler based on the input and output codec settings.
254 * If the input and output sample formats differ, a conversion is required
255 * libswresample takes care of this, but requires initialization.
256 */
260 {
262
263 /**
264 * Create a resampler context for the conversion.
265 * Set the conversion parameters.
266 * Default channel layouts based on the number of channels
267 * are assumed for simplicity (they are sometimes not detected
268 * properly by the demuxer and/or decoder).
269 */
278 if (!*resample_context) {
279 fprintf(stderr, "Could not allocate resample context\n");
281 }
282 /**
283 * Perform a sanity check so that the number of converted samples is
284 * not greater than the number of samples to be converted.
285 * If the sample rates differ, this case has to be handled differently
286 */
288
289 /** Open the resampler with the specified parameters. */
290 if ((error =
swr_init(*resample_context)) < 0) {
291 fprintf(stderr, "Could not open resample context\n");
294 }
295 return 0;
296 }
297
298 /** Initialize a FIFO buffer for the audio samples to be encoded. */
300 {
301 /** Create the FIFO buffer based on the specified output sample format. */
303 output_codec_context->
channels, 1))) {
304 fprintf(stderr, "Could not allocate FIFO\n");
306 }
307 return 0;
308 }
309
310 /** Write the header of the output file container. */
312 {
315 fprintf(stderr, "Could not write output file header (error '%s')\n",
318 }
319 return 0;
320 }
321
322 /** Decode one audio frame from the input file. */
326 int *data_present, int *finished)
327 {
328 /** Packet used for temporary storage. */
332
333 /** Read one audio frame from the input file into a temporary packet. */
334 if ((error =
av_read_frame(input_format_context, &input_packet)) < 0) {
335 /** If we are at the end of the file, flush the decoder below. */
337 *finished = 1;
338 else {
339 fprintf(stderr, "Could not read frame (error '%s')\n",
342 }
343 }
344
345 /**
346 * Decode the audio frame stored in the temporary packet.
347 * The input audio stream decoder is used to do this.
348 * If we are at the end of the file, pass an empty packet to the decoder
349 * to flush it.
350 */
352 data_present, &input_packet)) < 0) {
353 fprintf(stderr, "Could not decode frame (error '%s')\n",
357 }
358
359 /**
360 * If the decoder has not been flushed completely, we are not finished,
361 * so that this function has to be called again.
362 */
363 if (*finished && *data_present)
364 *finished = 0;
366 return 0;
367 }
368
369 /**
370 * Initialize a temporary storage for the specified number of audio samples.
371 * The conversion requires temporary storage due to the different format.
372 * The number of audio samples to be allocated is specified in frame_size.
373 */
377 {
379
380 /**
381 * Allocate as many pointers as there are audio channels.
382 * Each pointer will later point to the audio samples of the corresponding
383 * channels (although it may be NULL for interleaved formats).
384 */
385 if (!(*converted_input_samples = calloc(output_codec_context->
channels,
386 sizeof(**converted_input_samples)))) {
387 fprintf(stderr, "Could not allocate converted input sample pointers\n");
389 }
390
391 /**
392 * Allocate memory for the samples of all channels in one consecutive
393 * block for convenience.
394 */
397 frame_size,
399 fprintf(stderr,
400 "Could not allocate converted input samples (error '%s')\n",
402 av_freep(&(*converted_input_samples)[0]);
403 free(*converted_input_samples);
405 }
406 return 0;
407 }
408
409 /**
410 * Convert the input audio samples into the output sample format.
411 * The conversion happens on a per-frame basis, the size of which is specified
412 * by frame_size.
413 */
417 {
419
420 /** Convert the samples using the resampler. */
422 converted_data, frame_size,
423 input_data , frame_size)) < 0) {
424 fprintf(stderr, "Could not convert input samples (error '%s')\n",
427 }
428
429 return 0;
430 }
431
432 /** Add converted input audio samples to the FIFO buffer for later processing. */
434 uint8_t **converted_input_samples,
436 {
438
439 /**
440 * Make the FIFO as large as it needs to be to hold both,
441 * the old and the new samples.
442 */
444 fprintf(stderr, "Could not reallocate FIFO\n");
446 }
447
448 /** Store the new samples in the FIFO buffer. */
450 frame_size) < frame_size) {
451 fprintf(stderr, "Could not write data to FIFO\n");
453 }
454 return 0;
455 }
456
457 /**
458 * Read one audio frame from the input file, decodes, converts and stores
459 * it in the FIFO buffer.
460 */
466 int *finished)
467 {
468 /** Temporary storage of the input samples of the frame read from the file. */
470 /** Temporary storage for the converted input samples. */
472 int data_present;
474
475 /** Initialize temporary storage for one input frame. */
478 /** Decode one frame worth of audio samples. */
480 input_codec_context, &data_present, finished))
482 /**
483 * If we are at the end of the file and there are no more samples
484 * in the decoder which are delayed, we are actually finished.
485 * This must not be treated as an error.
486 */
487 if (*finished && !data_present) {
488 ret = 0;
490 }
491 /** If there is decoded data, convert and store it */
492 if (data_present) {
493 /** Initialize the temporary storage for the converted input samples. */
497
498 /**
499 * Convert the input samples to the desired output sample format.
500 * This requires a temporary storage provided by converted_input_samples.
501 */
505
506 /** Add the converted input samples to the FIFO buffer for later processing. */
510 ret = 0;
511 }
512 ret = 0;
513
515 if (converted_input_samples) {
516 av_freep(&converted_input_samples[0]);
517 free(converted_input_samples);
518 }
520
521 return ret;
522 }
523
524 /**
525 * Initialize one input frame for writing to the output file.
526 * The frame will be exactly frame_size samples large.
527 */
531 {
533
534 /** Create a new frame to store the audio samples. */
536 fprintf(stderr, "Could not allocate output frame\n");
538 }
539
540 /**
541 * Set the frame's parameters, especially its size and format.
542 * av_frame_get_buffer needs this to allocate memory for the
543 * audio samples of the frame.
544 * Default channel layouts based on the number of channels
545 * are assumed for simplicity.
546 */
549 (*frame)->format = output_codec_context->
sample_fmt;
550 (*frame)->sample_rate = output_codec_context->
sample_rate;
551
552 /**
553 * Allocate the samples of the created frame. This call will make
554 * sure that the audio frame can hold as many samples as specified.
555 */
557 fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
561 }
562
563 return 0;
564 }
565
566 /** Global timestamp for the audio frames */
568
569 /** Encode one frame worth of audio to the output file. */
573 int *data_present)
574 {
575 /** Packet used for temporary storage. */
579
580 /** Set a timestamp based on the sample rate for the container. */
581 if (frame) {
584 }
585
586 /**
587 * Encode the audio frame and store it in the temporary packet.
588 * The output audio stream encoder is used to do this.
589 */
591 frame, data_present)) < 0) {
592 fprintf(stderr, "Could not encode frame (error '%s')\n",
596 }
597
598 /** Write one audio frame from the temporary packet to the output file. */
599 if (*data_present) {
600 if ((error =
av_write_frame(output_format_context, &output_packet)) < 0) {
601 fprintf(stderr, "Could not write frame (error '%s')\n",
605 }
606
608 }
609
610 return 0;
611 }
612
613 /**
614 * Load one audio frame from the FIFO buffer, encode and write it to the
615 * output file.
616 */
620 {
621 /** Temporary storage of the output samples of the frame written to the file. */
623 /**
624 * Use the maximum number of possible samples per frame.
625 * If there is less than the maximum possible frame size in the FIFO
626 * buffer use this number. Otherwise, use the maximum possible frame size
627 */
630 int data_written;
631
632 /** Initialize temporary storage for one output frame. */
635
636 /**
637 * Read as many samples from the FIFO buffer as required to fill the frame.
638 * The samples are stored in the frame temporarily.
639 */
641 fprintf(stderr, "Could not read data from FIFO\n");
644 }
645
646 /** Encode one frame worth of audio samples. */
648 output_codec_context, &data_written)) {
651 }
653 return 0;
654 }
655
656 /** Write the trailer of the output file container. */
658 {
661 fprintf(stderr, "Could not write output file trailer (error '%s')\n",
664 }
665 return 0;
666 }
667
668 /** Convert an audio file to an AAC file in an MP4 container. */
669 int main(
int argc,
char **argv)
670 {
676
677 if (argc < 3) {
678 fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
679 exit(1);
680 }
681
682 /** Register all codecs and formats so that they can be used. */
684 /** Open the input file for reading. */
686 &input_codec_context))
688 /** Open the output file for writing. */
690 &output_format_context, &output_codec_context))
692 /** Initialize the resampler to be able to convert audio sample formats. */
694 &resample_context))
696 /** Initialize the FIFO buffer to store audio samples to be encoded. */
697 if (
init_fifo(&fifo, output_codec_context))
699 /** Write the header of the output file container. */
702
703 /**
704 * Loop as long as we have input samples to read or output samples
705 * to write; abort as soon as we have neither.
706 */
707 while (1) {
708 /** Use the encoder's desired frame size for processing. */
709 const int output_frame_size = output_codec_context->frame_size;
710 int finished = 0;
711
712 /**
713 * Make sure that there is one frame worth of samples in the FIFO
714 * buffer so that the encoder can do its work.
715 * Since the decoder's and the encoder's frame size may differ, we
716 * need to FIFO buffer to store as many frames worth of input samples
717 * that they make up at least one frame worth of output samples.
718 */
720 /**
721 * Decode one frame worth of audio samples, convert it to the
722 * output sample format and put it into the FIFO buffer.
723 */
725 input_codec_context,
726 output_codec_context,
727 resample_context, &finished))
729
730 /**
731 * If we are at the end of the input file, we continue
732 * encoding the remaining audio samples to the output file.
733 */
734 if (finished)
735 break;
736 }
737
738 /**
739 * If we have enough samples for the encoder, we encode them.
740 * At the end of the file, we pass the remaining samples to
741 * the encoder.
742 */
745 /**
746 * Take one frame worth of audio samples from the FIFO buffer,
747 * encode it and write it to the output file.
748 */
750 output_codec_context))
752
753 /**
754 * If we are at the end of the input file and have encoded
755 * all remaining samples, we can exit this loop and finish.
756 */
757 if (finished) {
758 int data_written;
759 /** Flush the encoder as it may have delayed frames. */
760 do {
762 output_codec_context, &data_written))
764 } while (data_written);
765 break;
766 }
767 }
768
769 /** Write the trailer of the output file container. */
772 ret = 0;
773
775 if (fifo)
778 if (output_codec_context)
780 if (output_format_context) {
783 }
784 if (input_codec_context)
786 if (input_format_context)
788
789 return ret;
790 }
int avio_open(AVIOContext **s, const char *url, int flags)
Create and initialize a AVIOContext for accessing the resource indicated by url.
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
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.
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 av_write_frame(AVFormatContext *s, AVPacket *pkt)
Write a packet to an output media file.
int64_t bit_rate
the average bitrate
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
#define AVIO_FLAG_WRITE
write-only
attribute_deprecated int avcodec_encode_audio2(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Encode a frame of audio.
attribute_deprecated int avcodec_decode_audio4(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, const AVPacket *avpkt)
Decode the audio frame of size avpkt->size from avpkt->data into frame.
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.
#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.
int avcodec_parameters_to_context(AVCodecContext *codec, const AVCodecParameters *par)
Fill the codec context based on the values from the supplied codec parameters.
AVFormatContext * avformat_alloc_context(void)
Allocate an AVFormatContext.
static void input_data(MLPEncodeContext *ctx, void *samples)
Wrapper function for inputting data in two different bit-depths.
#define AVERROR_EOF
End of file.
static int init_input_frame(AVFrame **frame)
Initialize one audio frame for reading from the input file.
static void output_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost)
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.
int flags
AV_CODEC_FLAG_*.
simple assert() macros that are a bit more flexible than ISO C assert().
static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context)
Load one audio frame from the FIFO buffer, encode and write it to the output file.
size_t av_strlcpy(char *dst, const char *src, size_t size)
Copy the string src to dst, but no more than size - 1 bytes, and null-terminate dst.
reference-counted frame API
uint64_t channel_layout
Audio channel layout.
Context for an Audio FIFO Buffer.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples)
Reallocate an AVAudioFifo.
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.
static int open_output_file(const char *filename, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context)
Open an output file and the required encoder.
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished)
Decode one audio frame from the input file.
#define av_err2str(errnum)
Convenience macro, the return value should be used only directly in function arguments but never stan...
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.
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.
#define AVERROR_EXIT
Immediate exit was requested; the called function should not be restarted.
static void error(const char *err)
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.
Libavcodec external API header.
void avcodec_free_context(AVCodecContext **avctx)
Free the codec context and everything associated with it and write NULL to the provided pointer...
#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.
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
static int output_frame(H264Context *h, AVFrame *dst, H264Picture *srcp)
int avcodec_parameters_from_context(AVCodecParameters *par, const AVCodecContext *codec)
Fill the parameters struct based on the values from the supplied codec context.
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.
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 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
int avformat_open_input(AVFormatContext **ps, const char *url, AVInputFormat *fmt, AVDictionary **options)
Open an input stream and read the header.
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.
AVCodecParameters * codecpar
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
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
static av_cold void cleanup(FlashSV2Context *s)