65 bytestream_put_byte (&bs, 0x1);
66 bytestream_put_byte (&bs, avctx->
channels);
69 bytestream_put_le16 (&bs, 0x0);
70 bytestream_put_byte (&bs, 0x0);
75 int i,
tmp = 0x0, extended_toc = 0;
79 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 17, 0, 21, 25, 29 } },
80 { { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 }, { 18, 0, 22, 26, 30 } },
81 { { 1, 5, 9, 0, 0 }, { 0, 0, 0, 13, 15 }, { 19, 0, 23, 27, 31 } },
82 { { 2, 6, 10, 0, 0 }, { 0, 0, 0, 14, 16 }, { 20, 0, 24, 28, 32 } },
83 { { 3, 7, 11, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
84 { { 4, 8, 12, 0, 0 }, { 0, 0, 0, 0, 0 }, { 0, 0, 0, 0, 0 } },
102 tmp |= (cfg - 1) << 3;
107 tmp = (*fsize_needed) << 7;
112 *size = 1 + extended_toc;
125 for (ch = 0; ch < f->
channels; ch++) {
134 for (sf = 0; sf < subframes; sf++) {
135 if (sf != (subframes - 1))
140 for (ch = 0; ch < f->
channels; ch++) {
144 const size_t left = (subframesize - cur->
nb_samples)*bps;
146 memcpy(&b->
samples[sf*subframesize], input, len);
151 if (sf != (subframes - 1))
164 for (ch = 0; ch < f->
channels; ch++) {
176 for (sf = 0; sf < subframes; sf++) {
177 for (ch = 0; ch < f->
channels; ch++) {
180 for (i = 0; i < subframesize; i++) {
182 b->
samples[sf*subframesize + i] = sample - m;
185 if (sf != (subframes - 1))
198 for (ch = 0; ch < f->
channels; ch++) {
201 for (t = 0; t < f->
blocks; t++) {
213 memset(win, 0, wlen*
sizeof(
float));
214 for (ch = 0; ch < f->
channels; ch++) {
219 memcpy(win + lap_dst,
temp, CELT_OVERLAP*
sizeof(
float));
222 memcpy(&win[lap_dst + CELT_OVERLAP], b->
samples, rwin*
sizeof(
float));
227 memcpy(win + lap_dst + blk_len,
temp, CELT_OVERLAP*
sizeof(
float));
233 for (ch = 0; ch < f->
channels; ch++) {
241 for (j = 0; j < band_size; j++)
242 ener += coeffs[j]*coeffs[j];
244 block->
lin_energy[i] = sqrtf(ener) + FLT_EPSILON;
247 for (j = 0; j < band_size; j++)
260 int i, tf_select = 0,
diff = 0, tf_changed = 0, tf_select_needed;
265 for (i = f->
start_band; i < f->end_band; i++) {
287 int i, j, low, high, total, done, bandbits, remaining, tbits_8ths;
290 int intensitystereo_bit = 0;
291 int dualstereo_bit = 0;
316 for (i = f->
start_band; i < f->end_band; i++) {
317 int quanta, b_dynalloc, boost_amount = f->
alloc_boost[i];
322 quanta =
FFMIN(quanta << 3,
FFMAX(6 << 3, quanta));
323 b_dynalloc = dynalloc;
325 while (
opus_rc_tell_frac(rc) + (b_dynalloc << 3) < tbits_8ths && boost[i] < cap[i]) {
326 int is_boost = boost_amount--;
333 tbits_8ths -= quanta;
339 dynalloc =
FFMAX(2, dynalloc - 1);
354 if (tbits_8ths >= 1 << 3)
356 tbits_8ths -= skip_bit;
361 if (intensitystereo_bit <= tbits_8ths) {
362 tbits_8ths -= intensitystereo_bit;
363 if (tbits_8ths >= 1 << 3) {
364 dualstereo_bit = 1 << 3;
365 tbits_8ths -= 1 << 3;
368 intensitystereo_bit = 0;
373 for (i = f->
start_band; i < f->end_band; i++) {
377 int scale = duration + f->
channels - 1;
384 trim_offset[i] = trim * (band << scale) >> 6;
393 while (low <= high) {
394 int center = (low + high) >> 1;
402 bandbits =
FFMAX(0, bandbits + trim_offset[i]);
403 bandbits += boost[i];
405 if (bandbits >= threshold[i] || done) {
407 total +=
FFMIN(bandbits, cap[i]);
408 }
else if (bandbits >= f->
channels << 3)
412 if (total > tbits_8ths)
420 for (i = f->
start_band; i < f->end_band; i++) {
428 bits1[i] =
FFMAX(0, bits1[i] + trim_offset[i]);
430 bits2[i] =
FFMAX(0, bits2[i] + trim_offset[i]);
432 bits1[i] += boost[i];
433 bits2[i] += boost[i];
437 bits2[i] =
FFMAX(0, bits2[i] - bits1[i]);
444 int center = (low + high) >> 1;
450 if (bandbits >= threshold[j] || done) {
452 total +=
FFMIN(bandbits, cap[j]);
453 }
else if (bandbits >= f->
channels << 3)
456 if (total > tbits_8ths)
467 if (bandbits >= threshold[i] || done)
470 bandbits = (bandbits >= f->
channels << 3) ?
473 bandbits =
FFMIN(bandbits, cap[i]);
483 if (j == skip_startband) {
485 tbits_8ths += skip_bit;
490 remaining = tbits_8ths - total;
498 if (allocation >=
FFMAX(threshold[j], (f->
channels + 1) << 3)) {
505 allocation -= 1 << 3;
510 if (intensitystereo_bit) {
511 total -= intensitystereo_bit;
513 total += intensitystereo_bit;
520 if (intensitystereo_bit) {
525 tbits_8ths += dualstereo_bit;
526 else if (dualstereo_bit)
530 remaining = tbits_8ths - total;
533 for (i = f->
start_band; i < f->coded_bands; i++) {
541 for (i = f->
start_band; i < f->coded_bands; i++) {
543 int prev_extra = extrabits;
544 f->
pulses[i] += extrabits;
552 int fine_bits, max_bits;
555 f->
pulses[i] -= extrabits;
565 if (f->
pulses[i] + offset < 2 * (dof << 3))
567 else if (f->
pulses[i] + offset < 3 * (dof << 3))
570 fine_bits = (f->
pulses[i] + offset + (dof << 2)) / (dof << 3);
573 max_bits =
FFMAX(max_bits, 0);
575 f->
fine_bits[i] = av_clip(fine_bits, 0, max_bits);
586 f->
pulses[i] -= extrabits;
599 extrabits -= fineextra;
622 txval =
FFMIN(octave, 6);
626 txval = av_clip(period - (16 << octave) + 1, 0, (1 << (4 + octave)) - 1);
628 period = txval + (16 << octave) - 1;
630 txval =
FFMIN(((
int)(gain / 0.09375f)) - 1, 7);
632 gain = 0.09375f * (txval + 1);
639 for (i = 0; i < 2; i++) {
644 block->
pf_gains_new[1] = gain * ff_celt_postfilter_taps[tapset][1];
645 block->
pf_gains_new[2] = gain * ff_celt_postfilter_taps[tapset][2];
653 float alpha, beta, prev[2] = { 0, 0 };
664 beta = 1.0f - (4915.0f/32768.0f);
670 for (i = f->
start_band; i < f->end_band; i++) {
671 for (ch = 0; ch < f->
channels; ch++) {
674 const float last =
FFMAX(-9.0f, last_energy[ch][i]);
679 }
else if (left >= 2) {
680 q_en = av_clip(q_en, -1, 1);
682 }
else if (left >= 1) {
683 q_en = av_clip(q_en, -1, 0);
688 prev[
ch] += beta * q_en;
696 uint32_t inter, intra;
716 for (i = f->
start_band; i < f->end_band; i++) {
719 for (ch = 0; ch < f->
channels; ch++) {
723 quant = av_clip(floor(diff*lim), 0, lim - 1);
725 offset = 0.5f - ((quant + 0.5f) * (1 << (14 - f->
fine_bits[i])) / 16384.0f);
734 for (priority = 0; priority < 2; priority++) {
738 for (ch = 0; ch < f->
channels; ch++) {
741 const float offset = 0.5f * (1 << (14 - f->
fine_bits[i] - 1)) / 16384.0f;
742 const int sign =
FFABS(err + offset) <
FFABS(err - offset);
752 float lowband_scratch[8 * 22];
753 float norm[2 * 8 * 100];
757 int update_lowband = 1;
758 int lowband_offset = 0;
770 float *norm2 = norm + 8 * 100;
771 int effective_lowband = -1;
778 if (i <= f->coded_bands - 1) {
784 (update_lowband || lowband_offset == 0))
791 int foldstart, foldend;
796 foldstart = lowband_offset;
798 foldend = lowband_offset - 1;
802 for (j = foldstart; j < foldend; j++) {
812 norm[j] = (norm[j] + norm2[j]) / 2;
817 effective_lowband != -1 ? norm + (effective_lowband << f->
size) :
NULL, f->
size,
818 norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]);
821 effective_lowband != -1 ? norm2 + (effective_lowband << f->
size) :
NULL, f->
size,
822 norm2 + band_offset, 0, 1.0f, lowband_scratch, cm[1]);
825 effective_lowband != -1 ? norm + (effective_lowband << f->
size) :
NULL, f->
size,
826 norm + band_offset, 0, 1.0f, lowband_scratch, cm[0] | cm[1]);
835 update_lowband = (b > band_size << 3);
853 for (ch = 0; ch < s->
channels; ch++)
899 for (ch = 0; ch < f->
channels; ch++) {
909 dst[1] = v - dst[0] >> 2;
910 return 1 + (v >= 252);
915 int i,
offset, fsize_needed;
988 for (i = 0; i < pad_empty; i++) {
1022 *got_packet_ptr = 1;
1049 int i,
ch, ret, max_frames;
1068 int64_t clipped_rate = av_clip(avctx->
bit_rate, 6000, 255000 * s->
channels);
1069 av_log(avctx,
AV_LOG_ERROR,
"Unsupported bitrate %"PRId64
" kbps, clipping to %"PRId64
" kbps\n",
1070 avctx->
bit_rate/1000, clipped_rate/1000);
1095 for (ch = 0; ch < s->
channels; ch++)
1115 for (i = 0; i < max_frames; i++) {
1126 #define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1128 {
"opus_delay",
"Maximum delay in milliseconds", offsetof(
OpusEncContext,
options.max_delay_ms),
AV_OPT_TYPE_FLOAT, { .dbl =
OPUS_MAX_LOOKAHEAD }, 2.5f,
OPUS_MAX_LOOKAHEAD,
OPUSENC_FLAGS,
"max_delay_ms" },
1141 {
"compression_level",
"10" },
1158 .supported_samplerates = (
const int []){ 48000, 0 },
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
static AVFrame * ff_bufqueue_get(struct FFBufQueue *queue)
Get the first buffer from the queue and remove it.
void ff_opus_psy_celt_frame_init(OpusPsyContext *s, CeltFrame *f, int index)
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
static av_cold int opus_encode_init(AVCodecContext *avctx)
enum OpusBandwidth bandwidth
#define OPUS_RC_CHECKPOINT_SPAWN(rc)
static void celt_quant_bands(OpusRangeCoder *rc, CeltFrame *f)
static float alpha(float a)
This structure describes decoded (raw) audio or video data.
static const AVCodecDefault opusenc_defaults[]
static const AVOption opusenc_options[]
const uint8_t ff_celt_coarse_energy_dist[4][2][42]
static AVFrame * spawn_empty_frame(OpusEncContext *s)
const uint8_t ff_celt_log_freq_range[]
void ff_opus_rc_enc_init(OpusRangeCoder *rc)
float coeffs[CELT_MAX_FRAME_SIZE]
int64_t bit_rate
the average bitrate
#define LIBAVUTIL_VERSION_INT
static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
static float win(SuperEqualizerContext *s, float n, int N)
static av_cold int init(AVCodecContext *avctx)
const uint8_t ff_celt_freq_bands[]
static void celt_quant_coarse(OpusRangeCoder *rc, CeltFrame *f, float last_energy[][CELT_MAX_BANDS])
static av_cold int opus_encode_end(AVCodecContext *avctx)
#define AV_CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
void ff_opus_rc_enc_log(OpusRangeCoder *rc, int val, uint32_t bits)
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
const float ff_celt_postfilter_taps[3][3]
#define AV_CH_LAYOUT_STEREO
void ff_celt_enc_bitalloc(OpusRangeCoder *rc, CeltFrame *f)
void ff_opus_rc_enc_uint(OpusRangeCoder *rc, uint32_t val, uint32_t size)
CELT: write a uniformly distributed integer.
Structure holding the queue.
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
int fine_priority[CELT_MAX_BANDS]
static void exp_quant_coarse(OpusRangeCoder *rc, CeltFrame *f, float last_energy[][CELT_MAX_BANDS], int intra)
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
enum AVSampleFormat sample_fmt
audio sample format
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
const float * ff_celt_window
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_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),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){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) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;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)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8: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);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;unsignedm=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){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
#define OPUS_MAX_LOOKAHEAD
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
int av_cold ff_celt_pvq_init(CeltPVQ **pvq)
int ff_opus_psy_process(OpusPsyContext *s, OpusPacketInfo *p)
av_cold int ff_mdct15_init(MDCT15Context **ps, int inverse, int N, double scale)
#define CELT_POSTFILTER_MINPERIOD
static void celt_enc_tf(OpusRangeCoder *rc, CeltFrame *f)
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
static const uint8_t bits2[81]
int ff_opus_psy_celt_frame_process(OpusPsyContext *s, CeltFrame *f, int index)
float lin_energy[CELT_MAX_BANDS]
const OptionDef options[]
static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
float samples[FFALIGN(CELT_MAX_FRAME_SIZE, 16)]
const uint8_t ff_celt_log2_frac[]
int alloc_boost[CELT_MAX_BANDS]
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
#define OPUS_BLOCK_SIZE(x)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
int initial_padding
Audio only.
#define OPUS_RC_CHECKPOINT_ROLLBACK(rc)
int flags
AV_CODEC_FLAG_*.
av_cold int ff_opus_psy_init(OpusPsyContext *s, AVCodecContext *avctx, struct FFBufQueue *bufqueue, OpusEncOptions *options)
int tf_change[CELT_MAX_BANDS]
const char * name
Name of the codec implementation.
int pulses[CELT_MAX_BANDS]
static int opus_gen_toc(OpusEncContext *s, uint8_t *toc, int *size, int *fsize_needed)
static const uint8_t offset[127][2]
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
int fine_bits[CELT_MAX_BANDS]
uint64_t channel_layout
Audio channel layout.
uint64_t channel_layout
Channel layout of the audio data.
const int8_t ff_celt_tf_select[4][2][2][2]
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
#define OPUS_RC_CHECKPOINT_BITS(rc)
MDCT15Context * mdct[CELT_BLOCK_NB]
static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
float error_energy[CELT_MAX_BANDS]
static void ff_bufqueue_discard_all(struct FFBufQueue *queue)
Unref and remove all buffers from the queue.
const uint8_t ff_celt_freq_range[]
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
const uint8_t ff_opus_default_coupled_streams[]
void ff_opus_rc_enc_cdf(OpusRangeCoder *rc, int val, const uint16_t *cdf)
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
int frame_size
Number of samples per channel in an audio frame.
const uint8_t ff_celt_static_caps[4][2][21]
struct FFBufQueue bufqueue
static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f, int index)
AVSampleFormat
Audio sample formats.
unsigned short available
number of available buffers
int sample_rate
samples per second
const uint16_t ff_celt_model_spread[]
main external API structure.
static av_always_inline uint32_t opus_rc_tell(const OpusRangeCoder *rc)
CELT: estimate bits of entropy that have thus far been consumed for the current CELT frame...
#define CELT_MAX_FINE_BITS
void ff_opus_rc_enc_laplace(OpusRangeCoder *rc, int *value, uint32_t symbol, int decay)
static int write_opuslacing(uint8_t *dst, int v)
Describe the class of an AVClass context structure.
void ff_opus_psy_signal_eof(OpusPsyContext *s)
const uint8_t ff_celt_static_alloc[11][21]
#define CELT_ENERGY_SILENCE
static void opus_write_extradata(AVCodecContext *avctx)
Recommmends skipping the specified number of samples.
av_cold int ff_opus_psy_end(OpusPsyContext *s)
void ff_opus_psy_postencode_update(OpusPsyContext *s, CeltFrame *f, OpusRangeCoder *rc)
const uint16_t ff_celt_model_energy_small[]
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
#define OPUS_MAX_CHANNELS
const uint16_t ff_celt_model_alloc_trim[]
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
float energy[CELT_MAX_BANDS]
static const AVClass opusenc_class
static void celt_quant_final(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
const float ff_celt_beta_coef[]
void ff_opus_rc_put_raw(OpusRangeCoder *rc, uint32_t val, uint32_t count)
CELT: write 0 - 31 bits to the rawbits buffer.
common internal api header.
static void celt_quant_fine(OpusRangeCoder *rc, CeltFrame *f)
void ff_opus_rc_enc_end(OpusRangeCoder *rc, uint8_t *dst, int size)
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
av_cold void ff_mdct15_uninit(MDCT15Context **ps)
float last_quantized_energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS]
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static const int16_t coeffs[]
int channels
number of audio channels
const float ff_celt_alpha_coef[]
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
const float ff_celt_mean_energy[]
static void celt_enc_quant_pfilter(OpusRangeCoder *rc, CeltFrame *f)
static void ff_bufqueue_add(void *log, struct FFBufQueue *queue, AVFrame *buf)
Add a buffer to the queue.
uint8_t * av_packet_new_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int size)
Allocate new information of a packet.
uint8_t ** extended_data
pointers to the data planes/channels.
#define AV_CH_LAYOUT_MONO
void(* mdct)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t stride)
uint8_t collapse_masks[CELT_MAX_BANDS]
This structure stores compressed data.
const uint16_t ff_celt_model_tapset[]
static void celt_apply_preemph_filter(OpusEncContext *s, CeltFrame *f)
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
float overlap[FFALIGN(CELT_OVERLAP, 16)]
void av_cold ff_celt_pvq_uninit(CeltPVQ **pvq)
static const uint8_t bits1[81]
static av_always_inline uint32_t opus_rc_tell_frac(const OpusRangeCoder *rc)
static AVFrame * ff_bufqueue_peek(struct FFBufQueue *queue, unsigned index)
Get a buffer from the queue without altering it.