FFmpeg
decode_audio.c
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1 /*
2  * Copyright (c) 2001 Fabrice Bellard
3  *
4  * Permission is hereby granted, free of charge, to any person obtaining a copy
5  * of this software and associated documentation files (the "Software"), to deal
6  * in the Software without restriction, including without limitation the rights
7  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
8  * copies of the Software, and to permit persons to whom the Software is
9  * furnished to do so, subject to the following conditions:
10  *
11  * The above copyright notice and this permission notice shall be included in
12  * all copies or substantial portions of the Software.
13  *
14  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
15  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
16  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
17  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
18  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
19  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
20  * THE SOFTWARE.
21  */
22 
23 /**
24  * @file
25  * audio decoding with libavcodec API example
26  *
27  * @example decode_audio.c
28  */
29 
30 #include <stdio.h>
31 #include <stdlib.h>
32 #include <string.h>
33 
34 #include <libavutil/frame.h>
35 #include <libavutil/mem.h>
36 
37 #include <libavcodec/avcodec.h>
38 
39 #define AUDIO_INBUF_SIZE 20480
40 #define AUDIO_REFILL_THRESH 4096
41 
42 static int get_format_from_sample_fmt(const char **fmt,
43  enum AVSampleFormat sample_fmt)
44 {
45  int i;
46  struct sample_fmt_entry {
47  enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
48  } sample_fmt_entries[] = {
49  { AV_SAMPLE_FMT_U8, "u8", "u8" },
50  { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
51  { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
52  { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
53  { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
54  };
55  *fmt = NULL;
56 
57  for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
58  struct sample_fmt_entry *entry = &sample_fmt_entries[i];
59  if (sample_fmt == entry->sample_fmt) {
60  *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
61  return 0;
62  }
63  }
64 
65  fprintf(stderr,
66  "sample format %s is not supported as output format\n",
67  av_get_sample_fmt_name(sample_fmt));
68  return -1;
69 }
70 
72  FILE *outfile)
73 {
74  int i, ch;
75  int ret, data_size;
76 
77  /* send the packet with the compressed data to the decoder */
78  ret = avcodec_send_packet(dec_ctx, pkt);
79  if (ret < 0) {
80  fprintf(stderr, "Error submitting the packet to the decoder\n");
81  exit(1);
82  }
83 
84  /* read all the output frames (in general there may be any number of them */
85  while (ret >= 0) {
86  ret = avcodec_receive_frame(dec_ctx, frame);
87  if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
88  return;
89  else if (ret < 0) {
90  fprintf(stderr, "Error during decoding\n");
91  exit(1);
92  }
93  data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
94  if (data_size < 0) {
95  /* This should not occur, checking just for paranoia */
96  fprintf(stderr, "Failed to calculate data size\n");
97  exit(1);
98  }
99  for (i = 0; i < frame->nb_samples; i++)
100  for (ch = 0; ch < dec_ctx->channels; ch++)
101  fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
102  }
103 }
104 
105 int main(int argc, char **argv)
106 {
107  const char *outfilename, *filename;
108  const AVCodec *codec;
110  AVCodecParserContext *parser = NULL;
111  int len, ret;
112  FILE *f, *outfile;
114  uint8_t *data;
115  size_t data_size;
116  AVPacket *pkt;
117  AVFrame *decoded_frame = NULL;
118  enum AVSampleFormat sfmt;
119  int n_channels = 0;
120  const char *fmt;
121 
122  if (argc <= 2) {
123  fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
124  exit(0);
125  }
126  filename = argv[1];
127  outfilename = argv[2];
128 
129  pkt = av_packet_alloc();
130 
131  /* find the MPEG audio decoder */
133  if (!codec) {
134  fprintf(stderr, "Codec not found\n");
135  exit(1);
136  }
137 
138  parser = av_parser_init(codec->id);
139  if (!parser) {
140  fprintf(stderr, "Parser not found\n");
141  exit(1);
142  }
143 
144  c = avcodec_alloc_context3(codec);
145  if (!c) {
146  fprintf(stderr, "Could not allocate audio codec context\n");
147  exit(1);
148  }
149 
150  /* open it */
151  if (avcodec_open2(c, codec, NULL) < 0) {
152  fprintf(stderr, "Could not open codec\n");
153  exit(1);
154  }
155 
156  f = fopen(filename, "rb");
157  if (!f) {
158  fprintf(stderr, "Could not open %s\n", filename);
159  exit(1);
160  }
161  outfile = fopen(outfilename, "wb");
162  if (!outfile) {
163  av_free(c);
164  exit(1);
165  }
166 
167  /* decode until eof */
168  data = inbuf;
169  data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
170 
171  while (data_size > 0) {
172  if (!decoded_frame) {
173  if (!(decoded_frame = av_frame_alloc())) {
174  fprintf(stderr, "Could not allocate audio frame\n");
175  exit(1);
176  }
177  }
178 
179  ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
180  data, data_size,
182  if (ret < 0) {
183  fprintf(stderr, "Error while parsing\n");
184  exit(1);
185  }
186  data += ret;
187  data_size -= ret;
188 
189  if (pkt->size)
190  decode(c, pkt, decoded_frame, outfile);
191 
192  if (data_size < AUDIO_REFILL_THRESH) {
193  memmove(inbuf, data, data_size);
194  data = inbuf;
195  len = fread(data + data_size, 1,
196  AUDIO_INBUF_SIZE - data_size, f);
197  if (len > 0)
198  data_size += len;
199  }
200  }
201 
202  /* flush the decoder */
203  pkt->data = NULL;
204  pkt->size = 0;
205  decode(c, pkt, decoded_frame, outfile);
206 
207  /* print output pcm infomations, because there have no metadata of pcm */
208  sfmt = c->sample_fmt;
209 
210  if (av_sample_fmt_is_planar(sfmt)) {
211  const char *packed = av_get_sample_fmt_name(sfmt);
212  printf("Warning: the sample format the decoder produced is planar "
213  "(%s). This example will output the first channel only.\n",
214  packed ? packed : "?");
215  sfmt = av_get_packed_sample_fmt(sfmt);
216  }
217 
218  n_channels = c->channels;
219  if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
220  goto end;
221 
222  printf("Play the output audio file with the command:\n"
223  "ffplay -f %s -ac %d -ar %d %s\n",
224  fmt, n_channels, c->sample_rate,
225  outfilename);
226 end:
227  fclose(outfile);
228  fclose(f);
229 
231  av_parser_close(parser);
232  av_frame_free(&decoded_frame);
233  av_packet_free(&pkt);
234 
235  return 0;
236 }
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
const char * fmt
Definition: avisynth_c.h:861
Memory handling functions.
int size
Definition: avcodec.h:1481
static AVPacket pkt
AVCodec.
Definition: avcodec.h:3492
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
void av_packet_free(AVPacket **pkt)
Free the packet, if the packet is reference counted, it will be unreferenced first.
Definition: avpacket.c:62
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
AV_SAMPLE_FMT_U8
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:189
#define f(width, name)
Definition: cbs_vp9.c:255
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
#define AUDIO_REFILL_THRESH
Definition: decode_audio.c:40
#define AV_NE(be, le)
Definition: common.h:50
uint8_t * data
Definition: avcodec.h:1480
#define AVERROR_EOF
End of file.
Definition: error.h:55
signed 32 bits
Definition: samplefmt.h:62
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
enum AVCodecID id
Definition: avcodec.h:3506
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
int avcodec_receive_frame(AVCodecContext *avctx, AVFrame *frame)
Return decoded output data from a decoder.
Definition: decode.c:739
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
Definition: decode_audio.c:42
reference-counted frame API
AVCodecContext * avcodec_alloc_context3(const AVCodec *codec)
Allocate an AVCodecContext and set its fields to default values.
Definition: options.c:156
int av_parser_parse2(AVCodecParserContext *s, AVCodecContext *avctx, uint8_t **poutbuf, int *poutbuf_size, const uint8_t *buf, int buf_size, int64_t pts, int64_t dts, int64_t pos)
Parse a packet.
Definition: parser.c:120
void av_parser_close(AVCodecParserContext *s)
Definition: parser.c:224
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
#define FF_ARRAY_ELEMS(a)
int avcodec_send_packet(AVCodecContext *avctx, const AVPacket *avpkt)
Supply raw packet data as input to a decoder.
Definition: decode.c:676
AVCodecParserContext * av_parser_init(int codec_id)
Definition: parser.c:34
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
void avcodec_free_context(AVCodecContext **avctx)
Free the codec context and everything associated with it and write NULL to the provided pointer...
Definition: options.c:171
int sample_rate
samples per second
Definition: avcodec.h:2228
main external API structure.
Definition: avcodec.h:1568
AVCodec * avcodec_find_decoder(enum AVCodecID id)
Find a registered decoder with a matching codec ID.
Definition: allcodecs.c:894
#define AUDIO_INBUF_SIZE
Definition: decode_audio.c:39
int avcodec_open2(AVCodecContext *avctx, const AVCodec *codec, AVDictionary **options)
Initialize the AVCodecContext to use the given AVCodec.
Definition: utils.c:548
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:75
signed 16 bits
Definition: samplefmt.h:61
static AVCodecContext * dec_ctx
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:793
#define av_free(p)
int len
int channels
number of audio channels
Definition: avcodec.h:2229
printf("static const uint8_t my_array[100] = {\n")
AVPacket * av_packet_alloc(void)
Allocate an AVPacket and set its fields to default values.
Definition: avpacket.c:51
int main(int argc, char **argv)
Definition: decode_audio.c:105
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
This structure stores compressed data.
Definition: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
FILE * outfile
Definition: audiogen.c:96
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248