FFmpeg  4.2.1
libopusdec.c
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1 /*
2  * Opus decoder using libopus
3  * Copyright (c) 2012 Nicolas George
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include <opus.h>
23 #include <opus_multistream.h>
24 
25 #include "libavutil/internal.h"
26 #include "libavutil/intreadwrite.h"
27 #include "libavutil/ffmath.h"
28 #include "libavutil/opt.h"
29 
30 #include "avcodec.h"
31 #include "internal.h"
32 #include "vorbis.h"
33 #include "mathops.h"
34 #include "libopus.h"
35 
37  AVClass *class;
38  OpusMSDecoder *dec;
39  int pre_skip;
40 #ifndef OPUS_SET_GAIN
41  union { int i; double d; } gain;
42 #endif
43 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
44  int apply_phase_inv;
45 #endif
46 };
47 
48 #define OPUS_HEAD_SIZE 19
49 
51 {
52  struct libopus_context *opus = avc->priv_data;
53  int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
54  uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
55 
56  avc->channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->channels == 1) ? 1 : 2;
57  if (avc->channels <= 0) {
59  "Invalid number of channels %d, defaulting to stereo\n", avc->channels);
60  avc->channels = 2;
61  }
62 
63  avc->sample_rate = 48000;
66  avc->channel_layout = avc->channels > 8 ? 0 :
68 
69  if (avc->extradata_size >= OPUS_HEAD_SIZE) {
70  opus->pre_skip = AV_RL16(avc->extradata + 10);
71  gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16);
72  channel_map = AV_RL8 (avc->extradata + 18);
73  }
74  if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
76  nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
77  if (nb_streams + nb_coupled != avc->channels)
78  av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
79  mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
80  } else {
81  if (avc->channels > 2 || channel_map) {
82  av_log(avc, AV_LOG_ERROR,
83  "No channel mapping for %d channels.\n", avc->channels);
84  return AVERROR(EINVAL);
85  }
86  nb_streams = 1;
87  nb_coupled = avc->channels > 1;
88  mapping = mapping_arr;
89  }
90 
91  if (avc->channels > 2 && avc->channels <= 8) {
92  const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
93  int ch;
94 
95  /* Remap channels from Vorbis order to ffmpeg order */
96  for (ch = 0; ch < avc->channels; ch++)
97  mapping_arr[ch] = mapping[vorbis_offset[ch]];
98  mapping = mapping_arr;
99  }
100 
101  opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
102  nb_streams, nb_coupled,
103  mapping, &ret);
104  if (!opus->dec) {
105  av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
106  opus_strerror(ret));
107  return ff_opus_error_to_averror(ret);
108  }
109 
110 #ifdef OPUS_SET_GAIN
111  ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
112  if (ret != OPUS_OK)
113  av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
114  opus_strerror(ret));
115 #else
116  {
117  double gain_lin = ff_exp10(gain_db / (20.0 * 256));
118  if (avc->sample_fmt == AV_SAMPLE_FMT_FLT)
119  opus->gain.d = gain_lin;
120  else
121  opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX);
122  }
123 #endif
124 
125 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
126  ret = opus_multistream_decoder_ctl(opus->dec,
127  OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv));
128  if (ret != OPUS_OK)
129  av_log(avc, AV_LOG_WARNING,
130  "Unable to set phase inversion: %s\n",
131  opus_strerror(ret));
132 #endif
133 
134  /* Decoder delay (in samples) at 48kHz */
135  avc->delay = avc->internal->skip_samples = opus->pre_skip;
136 
137  return 0;
138 }
139 
141 {
142  struct libopus_context *opus = avc->priv_data;
143 
144  if (opus->dec) {
145  opus_multistream_decoder_destroy(opus->dec);
146  opus->dec = NULL;
147  }
148  return 0;
149 }
150 
151 #define MAX_FRAME_SIZE (960 * 6)
152 
153 static int libopus_decode(AVCodecContext *avc, void *data,
154  int *got_frame_ptr, AVPacket *pkt)
155 {
156  struct libopus_context *opus = avc->priv_data;
157  AVFrame *frame = data;
158  int ret, nb_samples;
159 
160  frame->nb_samples = MAX_FRAME_SIZE;
161  if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
162  return ret;
163 
164  if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
165  nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
166  (opus_int16 *)frame->data[0],
167  frame->nb_samples, 0);
168  else
169  nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
170  (float *)frame->data[0],
171  frame->nb_samples, 0);
172 
173  if (nb_samples < 0) {
174  av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
175  opus_strerror(nb_samples));
176  return ff_opus_error_to_averror(nb_samples);
177  }
178 
179 #ifndef OPUS_SET_GAIN
180  {
181  int i = avc->channels * nb_samples;
182  if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
183  float *pcm = (float *)frame->data[0];
184  for (; i > 0; i--, pcm++)
185  *pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
186  } else {
187  int16_t *pcm = (int16_t *)frame->data[0];
188  for (; i > 0; i--, pcm++)
189  *pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
190  }
191  }
192 #endif
193 
194  frame->nb_samples = nb_samples;
195  *got_frame_ptr = 1;
196 
197  return pkt->size;
198 }
199 
200 static void libopus_flush(AVCodecContext *avc)
201 {
202  struct libopus_context *opus = avc->priv_data;
203 
204  opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE);
205  /* The stream can have been extracted by a tool that is not Opus-aware.
206  Therefore, any packet can become the first of the stream. */
207  avc->internal->skip_samples = opus->pre_skip;
208 }
209 
210 
211 #define OFFSET(x) offsetof(struct libopus_context, x)
212 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
213 static const AVOption libopusdec_options[] = {
214 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
215  { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
216 #endif
217  { NULL },
218 };
219 
220 static const AVClass libopusdec_class = {
221  .class_name = "libopusdec",
222  .item_name = av_default_item_name,
223  .option = libopusdec_options,
224  .version = LIBAVUTIL_VERSION_INT,
225 };
226 
227 
229  .name = "libopus",
230  .long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
231  .type = AVMEDIA_TYPE_AUDIO,
232  .id = AV_CODEC_ID_OPUS,
233  .priv_data_size = sizeof(struct libopus_context),
234  .init = libopus_decode_init,
235  .close = libopus_decode_close,
236  .decode = libopus_decode,
237  .flush = libopus_flush,
238  .capabilities = AV_CODEC_CAP_DR1,
239  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
240  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
243  .priv_class = &libopusdec_class,
244  .wrapper_name = "libopus",
245 };
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:48
static int libopus_decode(AVCodecContext *avc, void *data, int *got_frame_ptr, AVPacket *pkt)
Definition: libopusdec.c:153
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define OFFSET(x)
Definition: libopusdec.c:211
int size
Definition: avcodec.h:1478
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/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_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 *(INT64_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 *(INT64_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 &&HAVE_MMX) 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
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
#define AV_RL16
Definition: intreadwrite.h:42
static AVPacket pkt
AVCodec.
Definition: avcodec.h:3481
AVCodec ff_libopus_decoder
Definition: libopusdec.c:228
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
int ff_opus_error_to_averror(int err)
Definition: libopus.c:28
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
uint8_t
static int nb_streams
Definition: ffprobe.c:280
#define av_cold
Definition: attributes.h:82
AVOptions.
#define MAX_FRAME_SIZE
Definition: libopusdec.c:151
union libopus_context::@109 gain
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
static AVFrame * frame
const char data[16]
Definition: mxf.c:91
uint8_t * data
Definition: avcodec.h:1477
#define av_log(a,...)
#define AV_RL8(x)
Definition: intreadwrite.h:398
OpusMSDecoder * dec
Definition: libopusdec.c:38
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define AVERROR(e)
Definition: error.h:43
enum AVSampleFormat request_sample_fmt
desired sample format
Definition: avcodec.h:2298
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2276
common internal API header
#define FFMIN(a, b)
Definition: common.h:96
static av_cold int libopus_decode_init(AVCodecContext *avc)
Definition: libopusdec.c:50
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int sample_rate
samples per second
Definition: avcodec.h:2225
main external API structure.
Definition: avcodec.h:1565
const uint64_t ff_vorbis_channel_layouts[9]
Definition: vorbis_data.c:47
static void libopus_flush(AVCodecContext *avc)
Definition: libopusdec.c:200
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1964
int extradata_size
Definition: avcodec.h:1667
Describe the class of an AVClass context structure.
Definition: log.h:67
static const AVClass libopusdec_class
Definition: libopusdec.c:220
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:185
#define OPUS_HEAD_SIZE
Definition: libopusdec.c:48
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
internal math functions header
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
static const AVOption libopusdec_options[]
Definition: libopusdec.c:213
void * priv_data
Definition: avcodec.h:1592
int channels
number of audio channels
Definition: avcodec.h:2226
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1600
static av_cold int libopus_decode_close(AVCodecContext *avc)
Definition: libopusdec.c:140
const uint8_t ff_vorbis_channel_layout_offsets[8][8]
Definition: vorbis_data.c:25
#define FLAGS
Definition: libopusdec.c:212
This structure stores compressed data.
Definition: avcodec.h:1454
int delay
Codec delay.
Definition: avcodec.h:1721
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:981
for(j=16;j >0;--j)