FFmpeg
wmaenc.c
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1 /*
2  * WMA compatible encoder
3  * Copyright (c) 2007 Michael Niedermayer
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 "libavutil/attributes.h"
23 #include "libavutil/ffmath.h"
24 
25 #include "avcodec.h"
26 #include "internal.h"
27 #include "wma.h"
28 #include "libavutil/avassert.h"
29 
30 
32 {
33  WMACodecContext *s = avctx->priv_data;
34  int i, flags1, flags2, block_align;
35  uint8_t *extradata;
36  int ret;
37 
38  s->avctx = avctx;
39 
40  if (avctx->channels > MAX_CHANNELS) {
41  av_log(avctx, AV_LOG_ERROR,
42  "too many channels: got %i, need %i or fewer\n",
43  avctx->channels, MAX_CHANNELS);
44  return AVERROR(EINVAL);
45  }
46 
47  if (avctx->sample_rate > 48000) {
48  av_log(avctx, AV_LOG_ERROR, "sample rate is too high: %d > 48kHz\n",
49  avctx->sample_rate);
50  return AVERROR(EINVAL);
51  }
52 
53  if (avctx->bit_rate < 24 * 1000) {
54  av_log(avctx, AV_LOG_ERROR,
55  "bitrate too low: got %"PRId64", need 24000 or higher\n",
56  avctx->bit_rate);
57  return AVERROR(EINVAL);
58  }
59 
60  /* extract flag info */
61  flags1 = 0;
62  flags2 = 1;
63  if (avctx->codec->id == AV_CODEC_ID_WMAV1) {
64  extradata = av_malloc(4);
65  if (!extradata)
66  return AVERROR(ENOMEM);
67  avctx->extradata_size = 4;
68  AV_WL16(extradata, flags1);
69  AV_WL16(extradata + 2, flags2);
70  } else if (avctx->codec->id == AV_CODEC_ID_WMAV2) {
71  extradata = av_mallocz(10);
72  if (!extradata)
73  return AVERROR(ENOMEM);
74  avctx->extradata_size = 10;
75  AV_WL32(extradata, flags1);
76  AV_WL16(extradata + 4, flags2);
77  } else {
78  av_assert0(0);
79  }
80  avctx->extradata = extradata;
81  s->use_exp_vlc = flags2 & 0x0001;
82  s->use_bit_reservoir = flags2 & 0x0002;
83  s->use_variable_block_len = flags2 & 0x0004;
84  if (avctx->channels == 2)
85  s->ms_stereo = 1;
86 
87  if ((ret = ff_wma_init(avctx, flags2)) < 0)
88  return ret;
89 
90  /* init MDCT */
91  for (i = 0; i < s->nb_block_sizes; i++)
92  ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 0, 1.0);
93 
94  block_align = avctx->bit_rate * (int64_t) s->frame_len /
95  (avctx->sample_rate * 8);
96  block_align = FFMIN(block_align, MAX_CODED_SUPERFRAME_SIZE);
97  avctx->block_align = block_align;
98  avctx->frame_size = avctx->initial_padding = s->frame_len;
99 
100  return 0;
101 }
102 
104 {
105  WMACodecContext *s = avctx->priv_data;
106  float **audio = (float **) frame->extended_data;
107  int len = frame->nb_samples;
108  int window_index = s->frame_len_bits - s->block_len_bits;
109  FFTContext *mdct = &s->mdct_ctx[window_index];
110  int ch;
111  const float *win = s->windows[window_index];
112  int window_len = 1 << s->block_len_bits;
113  float n = 2.0 * 32768.0 / window_len;
114 
115  for (ch = 0; ch < avctx->channels; ch++) {
116  memcpy(s->output, s->frame_out[ch], window_len * sizeof(*s->output));
117  s->fdsp->vector_fmul_scalar(s->frame_out[ch], audio[ch], n, len);
118  s->fdsp->vector_fmul_reverse(&s->output[window_len], s->frame_out[ch],
119  win, len);
120  s->fdsp->vector_fmul(s->frame_out[ch], s->frame_out[ch], win, len);
121  mdct->mdct_calc(mdct, s->coefs[ch], s->output);
122  if (!isfinite(s->coefs[ch][0])) {
123  av_log(avctx, AV_LOG_ERROR, "Input contains NaN/+-Inf\n");
124  return AVERROR(EINVAL);
125  }
126  }
127 
128  return 0;
129 }
130 
131 // FIXME use for decoding too
132 static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
133 {
134  int n;
135  const uint16_t *ptr;
136  float v, *q, max_scale, *q_end;
137 
138  ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
139  q = s->exponents[ch];
140  q_end = q + s->block_len;
141  max_scale = 0;
142  while (q < q_end) {
143  /* XXX: use a table */
144  v = ff_exp10(*exp_param++ *(1.0 / 16.0));
145  max_scale = FFMAX(max_scale, v);
146  n = *ptr++;
147  do {
148  *q++ = v;
149  } while (--n);
150  }
151  s->max_exponent[ch] = max_scale;
152 }
153 
154 static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
155 {
156  int last_exp;
157  const uint16_t *ptr;
158  float *q, *q_end;
159 
160  ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
161  q = s->exponents[ch];
162  q_end = q + s->block_len;
163  if (s->version == 1) {
164  last_exp = *exp_param++;
165  av_assert0(last_exp - 10 >= 0 && last_exp - 10 < 32);
166  put_bits(&s->pb, 5, last_exp - 10);
167  q += *ptr++;
168  } else
169  last_exp = 36;
170  while (q < q_end) {
171  int exp = *exp_param++;
172  int code = exp - last_exp + 60;
173  av_assert1(code >= 0 && code < 120);
176  /* XXX: use a table */
177  q += *ptr++;
178  last_exp = exp;
179  }
180 }
181 
182 static int encode_block(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
183  int total_gain)
184 {
185  int v, bsize, ch, coef_nb_bits, parse_exponents;
186  float mdct_norm;
187  int nb_coefs[MAX_CHANNELS];
188  static const int fixed_exp[25] = {
189  20, 20, 20, 20, 20,
190  20, 20, 20, 20, 20,
191  20, 20, 20, 20, 20,
192  20, 20, 20, 20, 20,
193  20, 20, 20, 20, 20
194  };
195 
196  // FIXME remove duplication relative to decoder
197  if (s->use_variable_block_len) {
198  av_assert0(0); // FIXME not implemented
199  } else {
200  /* fixed block len */
201  s->next_block_len_bits = s->frame_len_bits;
202  s->prev_block_len_bits = s->frame_len_bits;
203  s->block_len_bits = s->frame_len_bits;
204  }
205 
206  s->block_len = 1 << s->block_len_bits;
207 // av_assert0((s->block_pos + s->block_len) <= s->frame_len);
208  bsize = s->frame_len_bits - s->block_len_bits;
209 
210  // FIXME factor
211  v = s->coefs_end[bsize] - s->coefs_start;
212  for (ch = 0; ch < s->avctx->channels; ch++)
213  nb_coefs[ch] = v;
214  {
215  int n4 = s->block_len / 2;
216  mdct_norm = 1.0 / (float) n4;
217  if (s->version == 1)
218  mdct_norm *= sqrt(n4);
219  }
220 
221  if (s->avctx->channels == 2)
222  put_bits(&s->pb, 1, !!s->ms_stereo);
223 
224  for (ch = 0; ch < s->avctx->channels; ch++) {
225  // FIXME only set channel_coded when needed, instead of always
226  s->channel_coded[ch] = 1;
227  if (s->channel_coded[ch])
228  init_exp(s, ch, fixed_exp);
229  }
230 
231  for (ch = 0; ch < s->avctx->channels; ch++) {
232  if (s->channel_coded[ch]) {
233  WMACoef *coefs1;
234  float *coefs, *exponents, mult;
235  int i, n;
236 
237  coefs1 = s->coefs1[ch];
238  exponents = s->exponents[ch];
239  mult = ff_exp10(total_gain * 0.05) / s->max_exponent[ch];
240  mult *= mdct_norm;
241  coefs = src_coefs[ch];
242  if (s->use_noise_coding && 0) {
243  av_assert0(0); // FIXME not implemented
244  } else {
245  coefs += s->coefs_start;
246  n = nb_coefs[ch];
247  for (i = 0; i < n; i++) {
248  double t = *coefs++ / (exponents[i] * mult);
249  if (t < -32768 || t > 32767)
250  return -1;
251 
252  coefs1[i] = lrint(t);
253  }
254  }
255  }
256  }
257 
258  v = 0;
259  for (ch = 0; ch < s->avctx->channels; ch++) {
260  int a = s->channel_coded[ch];
261  put_bits(&s->pb, 1, a);
262  v |= a;
263  }
264 
265  if (!v)
266  return 1;
267 
268  for (v = total_gain - 1; v >= 127; v -= 127)
269  put_bits(&s->pb, 7, 127);
270  put_bits(&s->pb, 7, v);
271 
272  coef_nb_bits = ff_wma_total_gain_to_bits(total_gain);
273 
274  if (s->use_noise_coding) {
275  for (ch = 0; ch < s->avctx->channels; ch++) {
276  if (s->channel_coded[ch]) {
277  int i, n;
278  n = s->exponent_high_sizes[bsize];
279  for (i = 0; i < n; i++) {
280  put_bits(&s->pb, 1, s->high_band_coded[ch][i] = 0);
281  if (0)
282  nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
283  }
284  }
285  }
286  }
287 
288  parse_exponents = 1;
289  if (s->block_len_bits != s->frame_len_bits)
290  put_bits(&s->pb, 1, parse_exponents);
291 
292  if (parse_exponents) {
293  for (ch = 0; ch < s->avctx->channels; ch++) {
294  if (s->channel_coded[ch]) {
295  if (s->use_exp_vlc) {
297  } else {
298  av_assert0(0); // FIXME not implemented
299 // encode_exp_lsp(s, ch);
300  }
301  }
302  }
303  } else
304  av_assert0(0); // FIXME not implemented
305 
306  for (ch = 0; ch < s->avctx->channels; ch++) {
307  if (s->channel_coded[ch]) {
308  int run, tindex;
309  WMACoef *ptr, *eptr;
310  tindex = (ch == 1 && s->ms_stereo);
311  ptr = &s->coefs1[ch][0];
312  eptr = ptr + nb_coefs[ch];
313 
314  run = 0;
315  for (; ptr < eptr; ptr++) {
316  if (*ptr) {
317  int level = *ptr;
318  int abs_level = FFABS(level);
319  int code = 0;
320  if (abs_level <= s->coef_vlcs[tindex]->max_level)
321  if (run < s->coef_vlcs[tindex]->levels[abs_level - 1])
322  code = run + s->int_table[tindex][abs_level - 1];
323 
324  av_assert2(code < s->coef_vlcs[tindex]->n);
325  put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[code],
326  s->coef_vlcs[tindex]->huffcodes[code]);
327 
328  if (code == 0) {
329  if (1 << coef_nb_bits <= abs_level)
330  return -1;
331 
332  put_bits(&s->pb, coef_nb_bits, abs_level);
333  put_bits(&s->pb, s->frame_len_bits, run);
334  }
335  // FIXME the sign is flipped somewhere
336  put_bits(&s->pb, 1, level < 0);
337  run = 0;
338  } else
339  run++;
340  }
341  if (run)
342  put_bits(&s->pb, s->coef_vlcs[tindex]->huffbits[1],
343  s->coef_vlcs[tindex]->huffcodes[1]);
344  }
345  if (s->version == 1 && s->avctx->channels >= 2)
346  avpriv_align_put_bits(&s->pb);
347  }
348  return 0;
349 }
350 
351 static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
352  uint8_t *buf, int buf_size, int total_gain)
353 {
354  init_put_bits(&s->pb, buf, buf_size);
355 
356  if (s->use_bit_reservoir)
357  av_assert0(0); // FIXME not implemented
358  else if (encode_block(s, src_coefs, total_gain) < 0)
359  return INT_MAX;
360 
361  avpriv_align_put_bits(&s->pb);
362 
363  return put_bits_count(&s->pb) / 8 - s->avctx->block_align;
364 }
365 
366 static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
367  const AVFrame *frame, int *got_packet_ptr)
368 {
369  WMACodecContext *s = avctx->priv_data;
370  int i, total_gain, ret, error;
371 
372  s->block_len_bits = s->frame_len_bits; // required by non variable block len
373  s->block_len = 1 << s->block_len_bits;
374 
375  ret = apply_window_and_mdct(avctx, frame);
376 
377  if (ret < 0)
378  return ret;
379 
380  if (s->ms_stereo) {
381  float a, b;
382  int i;
383 
384  for (i = 0; i < s->block_len; i++) {
385  a = s->coefs[0][i] * 0.5;
386  b = s->coefs[1][i] * 0.5;
387  s->coefs[0][i] = a + b;
388  s->coefs[1][i] = a - b;
389  }
390  }
391 
392  if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE, 0)) < 0)
393  return ret;
394 
395  total_gain = 128;
396  for (i = 64; i; i >>= 1) {
397  error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
398  total_gain - i);
399  if (error <= 0)
400  total_gain -= i;
401  }
402 
403  while(total_gain <= 128 && error > 0)
404  error = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain++);
405  if (error > 0) {
406  av_log(avctx, AV_LOG_ERROR, "Invalid input data or requested bitrate too low, cannot encode\n");
407  avpkt->size = 0;
408  return AVERROR(EINVAL);
409  }
410  av_assert0((put_bits_count(&s->pb) & 7) == 0);
411  i= avctx->block_align - (put_bits_count(&s->pb)+7)/8;
412  av_assert0(i>=0);
413  while(i--)
414  put_bits(&s->pb, 8, 'N');
415 
416  flush_put_bits(&s->pb);
417  av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == avctx->block_align);
418 
419  if (frame->pts != AV_NOPTS_VALUE)
420  avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
421 
422  avpkt->size = avctx->block_align;
423  *got_packet_ptr = 1;
424  return 0;
425 }
426 
427 #if CONFIG_WMAV1_ENCODER
429  .name = "wmav1",
430  .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
431  .type = AVMEDIA_TYPE_AUDIO,
432  .id = AV_CODEC_ID_WMAV1,
433  .priv_data_size = sizeof(WMACodecContext),
434  .init = encode_init,
435  .encode2 = encode_superframe,
436  .close = ff_wma_end,
437  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
439 };
440 #endif
441 #if CONFIG_WMAV2_ENCODER
443  .name = "wmav2",
444  .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
445  .type = AVMEDIA_TYPE_AUDIO,
446  .id = AV_CODEC_ID_WMAV2,
447  .priv_data_size = sizeof(WMACodecContext),
448  .init = encode_init,
449  .encode2 = encode_superframe,
450  .close = ff_wma_end,
451  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
453 };
454 #endif
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
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
ff_exp10
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
level
uint8_t level
Definition: svq3.c:207
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
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
coef_vlcs
static const CoefVLCTable coef_vlcs[6]
Definition: wmadata.h:1375
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2225
n
int n
Definition: avisynth_c.h:760
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:686
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
apply_window_and_mdct
static int apply_window_and_mdct(AVCodecContext *avctx, const AVFrame *frame)
Definition: wmaenc.c:103
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
b
#define b
Definition: input.c:41
init_exp
static void init_exp(WMACodecContext *s, int ch, const int *exp_param)
Definition: wmaenc.c:132
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:169
AV_CODEC_ID_WMAV2
@ AV_CODEC_ID_WMAV2
Definition: avcodec.h:572
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
win
static float win(SuperEqualizerContext *s, float n, int N)
Definition: af_superequalizer.c:119
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:1574
WMACoef
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
Definition: wma.h:57
encode_block
static int encode_block(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], int total_gain)
Definition: wmaenc.c:182
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:3096
WMACodecContext
Definition: wma.h:67
encode_init
static av_cold int encode_init(AVCodecContext *avctx)
Definition: wmaenc.c:31
mult
static int16_t mult(Float11 *f1, Float11 *f2)
Definition: g726.c:55
AV_CODEC_ID_WMAV1
@ AV_CODEC_ID_WMAV1
Definition: avcodec.h:571
avassert.h
lrint
#define lrint
Definition: tablegen.h:53
ff_samples_to_time_base
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx, int64_t samples)
Rescale from sample rate to AVCodecContext.time_base.
Definition: internal.h:288
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
buf
void * buf
Definition: avisynth_c.h:766
av_cold
#define av_cold
Definition: attributes.h:84
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1667
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
BLOCK_MAX_SIZE
#define BLOCK_MAX_SIZE
Definition: wma.h:35
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
isfinite
#define isfinite(x)
Definition: libm.h:359
wma.h
ff_wma_total_gain_to_bits
int ff_wma_total_gain_to_bits(int total_gain)
Definition: wma.c:354
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
run
uint8_t run
Definition: svq3.c:206
parse_exponents
static int parse_exponents(DBEContext *s, DBEChannel *c)
Definition: dolby_e.c:174
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1615
encode_superframe
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: wmaenc.c:366
exp
int8_t exp
Definition: eval.c:72
MAX_CODED_SUPERFRAME_SIZE
#define MAX_CODED_SUPERFRAME_SIZE
Definition: wma.h:45
error
static void error(const char *err)
Definition: target_dec_fuzzer.c:61
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
ff_wma_end
int ff_wma_end(AVCodecContext *avctx)
Definition: wma.c:368
avpriv_align_put_bits
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: bitstream.c:48
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:92
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
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
ff_wma_init
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
Definition: wma.c:81
AV_WL16
#define AV_WL16(p, v)
Definition: intreadwrite.h:412
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
fixed_exp
static int fixed_exp(int x)
Definition: aacsbr_fixed.c:112
attributes.h
MAX_CHANNELS
#define MAX_CHANNELS
Definition: aac.h:47
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2226
AVCodec::id
enum AVCodecID id
Definition: avcodec.h:3495
av_assert2
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
FFTContext
Definition: fft.h:88
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1470
code
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
Definition: filter_design.txt:178
put_bits_count
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:85
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t
uint8_t
Definition: audio_convert.c:194
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
AVCodec::name
const char * name
Name of the codec implementation.
Definition: avcodec.h:3488
len
int len
Definition: vorbis_enc_data.h:452
avcodec.h
ret
ret
Definition: filter_design.txt:187
encode_frame
static int encode_frame(WMACodecContext *s, float(*src_coefs)[BLOCK_MAX_SIZE], uint8_t *buf, int buf_size, int total_gain)
Definition: wmaenc.c:351
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:2262
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
AVCodecContext
main external API structure.
Definition: avcodec.h:1565
put_bits_ptr
static uint8_t * put_bits_ptr(PutBitContext *s)
Return the pointer to the byte where the bitstream writer will put the next bit.
Definition: put_bits.h:324
encode_exp_vlc
static void encode_exp_vlc(WMACodecContext *s, int ch, const int *exp_param)
Definition: wmaenc.c:154
ffmath.h
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1454
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1592
ff_wmav2_encoder
AVCodec ff_wmav2_encoder
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ff_wmav1_encoder
AVCodec ff_wmav1_encoder
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
ff_aac_scalefactor_code
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:73