FFmpeg
sbcenc.c
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1 /*
2  * Bluetooth low-complexity, subband codec (SBC)
3  *
4  * Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
5  * Copyright (C) 2012-2013 Intel Corporation
6  * Copyright (C) 2008-2010 Nokia Corporation
7  * Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
8  * Copyright (C) 2004-2005 Henryk Ploetz <henryk@ploetzli.ch>
9  * Copyright (C) 2005-2008 Brad Midgley <bmidgley@xmission.com>
10  *
11  * This file is part of FFmpeg.
12  *
13  * FFmpeg is free software; you can redistribute it and/or
14  * modify it under the terms of the GNU Lesser General Public
15  * License as published by the Free Software Foundation; either
16  * version 2.1 of the License, or (at your option) any later version.
17  *
18  * FFmpeg is distributed in the hope that it will be useful,
19  * but WITHOUT ANY WARRANTY; without even the implied warranty of
20  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
21  * Lesser General Public License for more details.
22  *
23  * You should have received a copy of the GNU Lesser General Public
24  * License along with FFmpeg; if not, write to the Free Software
25  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26  */
27 
28 /**
29  * @file
30  * SBC encoder implementation
31  */
32 
33 #include <stdbool.h>
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "internal.h"
37 #include "profiles.h"
38 #include "put_bits.h"
39 #include "sbc.h"
40 #include "sbcdsp.h"
41 
42 typedef struct SBCEncContext {
43  AVClass *class;
44  int64_t max_delay;
45  int msbc;
47  DECLARE_ALIGNED(SBC_ALIGN, SBCDSPContext, dsp);
49 
50 static int sbc_analyze_audio(SBCDSPContext *s, struct sbc_frame *frame)
51 {
52  int ch, blk;
53  int16_t *x;
54 
55  switch (frame->subbands) {
56  case 4:
57  for (ch = 0; ch < frame->channels; ch++) {
58  x = &s->X[ch][s->position - 4 *
59  s->increment + frame->blocks * 4];
60  for (blk = 0; blk < frame->blocks;
61  blk += s->increment) {
62  s->sbc_analyze_4s(
63  s, x,
64  frame->sb_sample_f[blk][ch],
65  frame->sb_sample_f[blk + 1][ch] -
66  frame->sb_sample_f[blk][ch]);
67  x -= 4 * s->increment;
68  }
69  }
70  return frame->blocks * 4;
71 
72  case 8:
73  for (ch = 0; ch < frame->channels; ch++) {
74  x = &s->X[ch][s->position - 8 *
75  s->increment + frame->blocks * 8];
76  for (blk = 0; blk < frame->blocks;
77  blk += s->increment) {
78  s->sbc_analyze_8s(
79  s, x,
80  frame->sb_sample_f[blk][ch],
81  frame->sb_sample_f[blk + 1][ch] -
82  frame->sb_sample_f[blk][ch]);
83  x -= 8 * s->increment;
84  }
85  }
86  return frame->blocks * 8;
87 
88  default:
89  return AVERROR(EIO);
90  }
91 }
92 
93 /*
94  * Packs the SBC frame from frame into the memory in avpkt.
95  * Returns the length of the packed frame.
96  */
97 static size_t sbc_pack_frame(AVPacket *avpkt, struct sbc_frame *frame,
98  int joint, bool msbc)
99 {
100  PutBitContext pb;
101 
102  /* Will copy the header parts for CRC-8 calculation here */
103  uint8_t crc_header[11] = { 0 };
104  int crc_pos;
105 
106  uint32_t audio_sample;
107 
108  int ch, sb, blk; /* channel, subband, block and bit counters */
109  int bits[2][8]; /* bits distribution */
110  uint32_t levels[2][8]; /* levels are derived from that */
111  uint32_t sb_sample_delta[2][8];
112 
113  if (msbc) {
114  avpkt->data[0] = MSBC_SYNCWORD;
115  avpkt->data[1] = 0;
116  avpkt->data[2] = 0;
117  } else {
118  avpkt->data[0] = SBC_SYNCWORD;
119 
120  avpkt->data[1] = (frame->frequency & 0x03) << 6;
121  avpkt->data[1] |= (((frame->blocks >> 2) - 1) & 0x03) << 4;
122  avpkt->data[1] |= (frame->mode & 0x03) << 2;
123  avpkt->data[1] |= (frame->allocation & 0x01) << 1;
124  avpkt->data[1] |= ((frame->subbands == 8) & 0x01) << 0;
125 
126  avpkt->data[2] = frame->bitpool;
127 
128  if (frame->bitpool > frame->subbands << (4 + (frame->mode == STEREO
129  || frame->mode == JOINT_STEREO)))
130  return -5;
131  }
132 
133  /* Can't fill in crc yet */
134  crc_header[0] = avpkt->data[1];
135  crc_header[1] = avpkt->data[2];
136  crc_pos = 16;
137 
138  init_put_bits(&pb, avpkt->data + 4, avpkt->size);
139 
140  if (frame->mode == JOINT_STEREO) {
141  put_bits(&pb, frame->subbands, joint);
142  crc_header[crc_pos >> 3] = joint;
143  crc_pos += frame->subbands;
144  }
145 
146  for (ch = 0; ch < frame->channels; ch++) {
147  for (sb = 0; sb < frame->subbands; sb++) {
148  put_bits(&pb, 4, frame->scale_factor[ch][sb] & 0x0F);
149  crc_header[crc_pos >> 3] <<= 4;
150  crc_header[crc_pos >> 3] |= frame->scale_factor[ch][sb] & 0x0F;
151  crc_pos += 4;
152  }
153  }
154 
155  /* align the last crc byte */
156  if (crc_pos % 8)
157  crc_header[crc_pos >> 3] <<= 8 - (crc_pos % 8);
158 
159  avpkt->data[3] = ff_sbc_crc8(frame->crc_ctx, crc_header, crc_pos);
160 
162 
163  for (ch = 0; ch < frame->channels; ch++) {
164  for (sb = 0; sb < frame->subbands; sb++) {
165  levels[ch][sb] = ((1 << bits[ch][sb]) - 1) <<
166  (32 - (frame->scale_factor[ch][sb] +
167  SCALE_OUT_BITS + 2));
168  sb_sample_delta[ch][sb] = (uint32_t) 1 <<
169  (frame->scale_factor[ch][sb] +
170  SCALE_OUT_BITS + 1);
171  }
172  }
173 
174  for (blk = 0; blk < frame->blocks; blk++) {
175  for (ch = 0; ch < frame->channels; ch++) {
176  for (sb = 0; sb < frame->subbands; sb++) {
177 
178  if (bits[ch][sb] == 0)
179  continue;
180 
181  audio_sample = ((uint64_t) levels[ch][sb] *
182  (sb_sample_delta[ch][sb] +
183  frame->sb_sample_f[blk][ch][sb])) >> 32;
184 
185  put_bits(&pb, bits[ch][sb], audio_sample);
186  }
187  }
188  }
189 
190  flush_put_bits(&pb);
191 
192  return (put_bits_count(&pb) + 7) / 8;
193 }
194 
195 static int sbc_encode_init(AVCodecContext *avctx)
196 {
197  SBCEncContext *sbc = avctx->priv_data;
198  struct sbc_frame *frame = &sbc->frame;
199 
200  if (avctx->profile == FF_PROFILE_SBC_MSBC)
201  sbc->msbc = 1;
202 
203  if (sbc->msbc) {
204  if (avctx->channels != 1) {
205  av_log(avctx, AV_LOG_ERROR, "mSBC require mono channel.\n");
206  return AVERROR(EINVAL);
207  }
208 
209  if (avctx->sample_rate != 16000) {
210  av_log(avctx, AV_LOG_ERROR, "mSBC require 16 kHz samplerate.\n");
211  return AVERROR(EINVAL);
212  }
213 
214  frame->mode = SBC_MODE_MONO;
215  frame->subbands = 8;
216  frame->blocks = MSBC_BLOCKS;
217  frame->allocation = SBC_AM_LOUDNESS;
218  frame->bitpool = 26;
219 
220  avctx->frame_size = 8 * MSBC_BLOCKS;
221  } else {
222  int d;
223 
224  if (avctx->global_quality > 255*FF_QP2LAMBDA) {
225  av_log(avctx, AV_LOG_ERROR, "bitpool > 255 is not allowed.\n");
226  return AVERROR(EINVAL);
227  }
228 
229  if (avctx->channels == 1) {
230  frame->mode = SBC_MODE_MONO;
231  if (sbc->max_delay <= 3000 || avctx->bit_rate > 270000)
232  frame->subbands = 4;
233  else
234  frame->subbands = 8;
235  } else {
236  if (avctx->bit_rate < 180000 || avctx->bit_rate > 420000)
238  else
239  frame->mode = SBC_MODE_STEREO;
240  if (sbc->max_delay <= 4000 || avctx->bit_rate > 420000)
241  frame->subbands = 4;
242  else
243  frame->subbands = 8;
244  }
245  /* sbc algorithmic delay is ((blocks + 10) * subbands - 2) / sample_rate */
246  frame->blocks = av_clip(((sbc->max_delay * avctx->sample_rate + 2)
247  / (1000000 * frame->subbands)) - 10, 4, 16) & ~3;
248 
249  frame->allocation = SBC_AM_LOUDNESS;
250 
251  d = frame->blocks * ((frame->mode == SBC_MODE_DUAL_CHANNEL) + 1);
252  frame->bitpool = (((avctx->bit_rate * frame->subbands * frame->blocks) / avctx->sample_rate)
253  - 4 * frame->subbands * avctx->channels
254  - (frame->mode == SBC_MODE_JOINT_STEREO)*frame->subbands - 32 + d/2) / d;
255  if (avctx->global_quality > 0)
256  frame->bitpool = avctx->global_quality / FF_QP2LAMBDA;
257 
258  avctx->frame_size = 4*((frame->subbands >> 3) + 1) * 4*(frame->blocks >> 2);
259  }
260 
261  for (int i = 0; avctx->codec->supported_samplerates[i]; i++)
262  if (avctx->sample_rate == avctx->codec->supported_samplerates[i])
263  frame->frequency = i;
264 
265  frame->channels = avctx->channels;
266  frame->codesize = frame->subbands * frame->blocks * avctx->channels * 2;
267  frame->crc_ctx = av_crc_get_table(AV_CRC_8_EBU);
268 
269  memset(&sbc->dsp.X, 0, sizeof(sbc->dsp.X));
270  sbc->dsp.position = (SBC_X_BUFFER_SIZE - frame->subbands * 9) & ~7;
271  sbc->dsp.increment = sbc->msbc ? 1 : 4;
272  ff_sbcdsp_init(&sbc->dsp);
273 
274  return 0;
275 }
276 
277 static int sbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
278  const AVFrame *av_frame, int *got_packet_ptr)
279 {
280  SBCEncContext *sbc = avctx->priv_data;
281  struct sbc_frame *frame = &sbc->frame;
283  uint8_t dual = frame->mode == SBC_MODE_DUAL_CHANNEL;
284  int ret, j = 0;
285 
286  int frame_length = 4 + (4 * frame->subbands * frame->channels) / 8
287  + ((frame->blocks * frame->bitpool * (1 + dual)
288  + joint * frame->subbands) + 7) / 8;
289 
290  /* input must be large enough to encode a complete frame */
291  if (av_frame->nb_samples * frame->channels * 2 < frame->codesize)
292  return 0;
293 
294  if ((ret = ff_alloc_packet2(avctx, avpkt, frame_length, 0)) < 0)
295  return ret;
296 
297  /* Select the needed input data processing function and call it */
298  if (frame->subbands == 8)
299  sbc->dsp.position = sbc->dsp.sbc_enc_process_input_8s(
300  sbc->dsp.position, av_frame->data[0], sbc->dsp.X,
301  frame->subbands * frame->blocks, frame->channels);
302  else
303  sbc->dsp.position = sbc->dsp.sbc_enc_process_input_4s(
304  sbc->dsp.position, av_frame->data[0], sbc->dsp.X,
305  frame->subbands * frame->blocks, frame->channels);
306 
307  sbc_analyze_audio(&sbc->dsp, &sbc->frame);
308 
309  if (frame->mode == JOINT_STEREO)
310  j = sbc->dsp.sbc_calc_scalefactors_j(frame->sb_sample_f,
311  frame->scale_factor,
312  frame->blocks,
313  frame->subbands);
314  else
315  sbc->dsp.sbc_calc_scalefactors(frame->sb_sample_f,
316  frame->scale_factor,
317  frame->blocks,
318  frame->channels,
319  frame->subbands);
320  emms_c();
321  sbc_pack_frame(avpkt, frame, j, sbc->msbc);
322 
323  *got_packet_ptr = 1;
324  return 0;
325 }
326 
327 #define OFFSET(x) offsetof(SBCEncContext, x)
328 #define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
329 static const AVOption options[] = {
330  { "sbc_delay", "set maximum algorithmic latency",
331  OFFSET(max_delay), AV_OPT_TYPE_DURATION, {.i64 = 13000}, 1000,13000, AE },
332  { "msbc", "use mSBC mode (wideband speech mono SBC)",
333  OFFSET(msbc), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AE },
334  { NULL },
335 };
336 
337 static const AVClass sbc_class = {
338  .class_name = "sbc encoder",
339  .item_name = av_default_item_name,
340  .option = options,
341  .version = LIBAVUTIL_VERSION_INT,
342 };
343 
345  .name = "sbc",
346  .long_name = NULL_IF_CONFIG_SMALL("SBC (low-complexity subband codec)"),
347  .type = AVMEDIA_TYPE_AUDIO,
348  .id = AV_CODEC_ID_SBC,
349  .priv_data_size = sizeof(SBCEncContext),
351  .encode2 = sbc_encode_frame,
352  .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME,
353  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
354  .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
356  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
358  .supported_samplerates = (const int[]) { 16000, 32000, 44100, 48000, 0 },
359  .priv_class = &sbc_class,
361 };
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2245
AVCodec
AVCodec.
Definition: avcodec.h:3481
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:40
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
AV_CRC_8_EBU
@ AV_CRC_8_EBU
Definition: crc.h:57
MSBC_SYNCWORD
#define MSBC_SYNCWORD
Definition: sbc.h:69
JOINT_STEREO
#define JOINT_STEREO
Definition: atrac3.c:55
OFFSET
#define OFFSET(x)
Definition: sbcenc.c:327
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2225
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
ch
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 &&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
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
put_bits
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:208
internal.h
AVPacket::data
uint8_t * data
Definition: avcodec.h:1477
AVOption
AVOption.
Definition: opt.h:246
AV_OPT_TYPE_DURATION
@ AV_OPT_TYPE_DURATION
Definition: opt.h:237
STEREO
#define STEREO
Definition: cook.c:61
sbc_analyze_audio
static int sbc_analyze_audio(SBCDSPContext *s, struct sbc_frame *frame)
Definition: sbcenc.c:50
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
MSBC_BLOCKS
#define MSBC_BLOCKS
Definition: sbc.h:39
SBCEncContext::dsp
SBCDSPContext dsp
Definition: sbcenc.c:47
options
static const AVOption options[]
Definition: sbcenc.c:329
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:1574
sbcdsp.h
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
SBCEncContext::msbc
int msbc
Definition: sbcenc.c:45
AV_CODEC_ID_SBC
@ AV_CODEC_ID_SBC
Definition: avcodec.h:652
AVCodec::supported_samplerates
const int * supported_samplerates
array of supported audio samplerates, or NULL if unknown, array is terminated by 0
Definition: avcodec.h:3503
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1631
sbc_pack_frame
static size_t sbc_pack_frame(AVPacket *avpkt, struct sbc_frame *frame, int joint, bool msbc)
Definition: sbcenc.c:97
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AE
#define AE
Definition: sbcenc.c:328
bits
uint8_t bits
Definition: vp3data.h:202
ff_sbcdsp_init
av_cold void ff_sbcdsp_init(SBCDSPContext *s)
Definition: sbcdsp.c:364
SBCEncContext::max_delay
int64_t max_delay
Definition: sbcenc.c:44
SBC_SYNCWORD
#define SBC_SYNCWORD
Definition: sbc.h:68
blk
#define blk(i)
Definition: sha.c:185
PutBitContext
Definition: put_bits.h:35
SBCEncContext
Definition: sbcenc.c:42
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
ff_sbc_encoder
AVCodec ff_sbc_encoder
Definition: sbcenc.c:344
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1615
SBC_MODE_STEREO
#define SBC_MODE_STEREO
Definition: sbc.h:56
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
profiles.h
SBC_AM_LOUDNESS
#define SBC_AM_LOUDNESS
Definition: sbc.h:60
SBC_ALIGN
#define SBC_ALIGN
Definition: sbc.h:78
AVPacket::size
int size
Definition: avcodec.h:1478
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
sbc_encode_frame
static int sbc_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *av_frame, int *got_packet_ptr)
Definition: sbcenc.c:277
SBC_MODE_MONO
#define SBC_MODE_MONO
Definition: sbc.h:54
av_crc_get_table
const AVCRC * av_crc_get_table(AVCRCId crc_id)
Get an initialized standard CRC table.
Definition: crc.c:374
sbc_encode_init
static int sbc_encode_init(AVCodecContext *avctx)
Definition: sbcenc.c:195
sbc.h
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2226
DECLARE_ALIGNED
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:112
SBC_X_BUFFER_SIZE
#define SBC_X_BUFFER_SIZE
Definition: sbcdsp.h:39
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
sbc_frame
Definition: sbc.h:82
put_bits_count
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:85
sbc_frame::joint
uint8_t joint
Definition: sbc.h:101
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t
uint8_t
Definition: audio_convert.c:194
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
ff_sbc_profiles
const AVProfile ff_sbc_profiles[]
Definition: profiles.c:149
ff_sbc_crc8
uint8_t ff_sbc_crc8(const AVCRC *ctx, const uint8_t *data, size_t len)
Definition: sbc.c:55
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVClass::class_name
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
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
SBCEncContext::frame
struct sbc_frame frame
Definition: sbcenc.c:46
sbc_class
static const AVClass sbc_class
Definition: sbcenc.c:337
AVCodecContext
main external API structure.
Definition: avcodec.h:1565
AVCodecContext::profile
int profile
profile
Definition: avcodec.h:2898
SCALE_OUT_BITS
#define SCALE_OUT_BITS
Definition: sbcdsp.h:38
FF_PROFILE_SBC_MSBC
#define FF_PROFILE_SBC_MSBC
Definition: avcodec.h:3001
ff_sbc_calculate_bits
void ff_sbc_calculate_bits(const struct sbc_frame *frame, int(*bits)[8])
Definition: sbc.c:79
SBC_MODE_DUAL_CHANNEL
#define SBC_MODE_DUAL_CHANNEL
Definition: sbc.h:55
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1592
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1454
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:240
channel_layouts
static const uint16_t channel_layouts[7]
Definition: dca_lbr.c:113
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
FF_QP2LAMBDA
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
AV_CODEC_CAP_SMALL_LAST_FRAME
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:1011
SBC_MODE_JOINT_STEREO
#define SBC_MODE_JOINT_STEREO
Definition: sbc.h:57
ff_alloc_packet2
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:32
put_bits.h