89 int i, ret, xpow, tmp;
93 for (i=0; i<10; i+=2){
94 xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
98 xpow = (int)(((int64_t)xpow * x + 0x40000000) >> 31);
109 Q31(1.0/720),
Q31(1.0/5040),
Q31(1.0/40320)
114 int i, ret, xpow, tmp;
119 xpow = (int)(((int64_t)xpow * x + 0x400000) >> 23);
129 int k, previous, present;
130 int base, prod, nz = 0;
132 base = (stop << 23) / start;
133 while (base < 0x40000000){
138 base = (((base + 0x80) >> 8) + (8-nz)*
CONST_LN2) / num_bands;
144 for (k = 0; k < num_bands-1; k++) {
145 prod = (int)(((int64_t)prod * base + 0x400000) >> 23);
146 present = (prod + 0x400000) >> 23;
147 bands[k] = present - previous;
150 bands[num_bands-1] = stop - previous;
168 temp1.
mant = 759250125;
170 temp1.
mant = 0x20000000;
171 temp1.
exp = (temp1.
exp >> 1) + 1;
172 if (temp1.
exp > 66) {
179 temp2.
mant = 759250125;
181 temp2.
mant = 0x20000000;
182 temp2.
exp = (temp2.
exp >> 1) + 1;
189 for (k = 0; k < sbr->
n_q; k++) {
193 sbr->data[0].noise_facs[e][k].mant + 2;
194 temp1.
mant = 0x20000000;
195 if (temp1.
exp > 66) {
200 temp2.
mant = 0x20000000;
207 for (ch = 0; ch < (id_aac ==
TYPE_CPE) + 1; ch++) {
215 temp1.
mant = 759250125;
217 temp1.
mant = 0x20000000;
218 temp1.
exp = (temp1.
exp >> 1) + 1;
219 if (temp1.
exp > 66) {
226 for (k = 0; k < sbr->
n_q; k++){
228 sbr->data[ch].noise_facs[e][k].mant + 1;
240 int (*alpha0)[2],
int (*alpha1)[2],
241 const int X_low[32][40][2],
int k0)
246 for (k = 0; k < k0; k++) {
273 if (!phi[1][0][0].mant) {
287 a00 =
av_div_sf(temp_real, phi[1][0][0]);
293 alpha0[k][0] = 0x7fffffff;
294 else if (shift <= -30)
300 alpha0[k][0] = a00.
mant;
302 round = 1 << (shift-1);
303 alpha0[k][0] = (a00.
mant +
round) >> shift;
309 alpha0[k][1] = 0x7fffffff;
310 else if (shift <= -30)
316 alpha0[k][1] = a01.
mant;
318 round = 1 << (shift-1);
319 alpha0[k][1] = (a01.
mant +
round) >> shift;
324 alpha1[k][0] = 0x7fffffff;
325 else if (shift <= -30)
331 alpha1[k][0] = a10.
mant;
333 round = 1 << (shift-1);
334 alpha1[k][0] = (a10.
mant +
round) >> shift;
340 alpha1[k][1] = 0x7fffffff;
341 else if (shift <= -30)
347 alpha1[k][1] = a11.
mant;
349 round = 1 << (shift-1);
350 alpha1[k][1] = (a11.
mant +
round) >> shift;
354 shift = (int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
355 (int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
357 if (shift >= 0x20000000){
364 shift = (int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
365 (int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
367 if (shift >= 0x20000000){
381 static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
384 for (i = 0; i < sbr->
n_q; i++) {
390 if (new_bw < ch_data->bw_array[i]){
391 accu = (int64_t)new_bw * 1610612736;
392 accu += (int64_t)ch_data->
bw_array[i] * 0x20000000;
393 new_bw = (
int)((accu + 0x40000000) >> 31);
395 accu = (int64_t)new_bw * 1946157056;
396 accu += (int64_t)ch_data->
bw_array[i] * 201326592;
397 new_bw = (
int)((accu + 0x40000000) >> 31);
399 ch_data->
bw_array[i] = new_bw < 0x2000000 ? 0 : new_bw;
408 SBRData *ch_data,
const int e_a[2])
412 static const SoftFloat limgain[4] = { { 760155524, 0 }, { 0x20000000, 1 },
413 { 758351638, 1 }, { 625000000, 34 } };
416 int delta = !((e == e_a[1]) || (e == e_a[0]));
417 for (k = 0; k < sbr->
n_lim; k++) {
421 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
445 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
456 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
461 sbr->
q_m[e][
m] = q_m_max;
463 sbr->
gain[e][
m] = gain_max;
466 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
475 if (delta && !sbr->
s_m[e][m].mant)
486 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
497 const int X_high[64][40][2],
503 const int kx = sbr->
kx[1];
504 const int m_max = sbr->
m[1];
517 for (i = 0; i < h_SL; i++) {
518 memcpy(g_temp[i + 2*ch_data->
t_env[0]], sbr->
gain[0], m_max *
sizeof(sbr->
gain[0][0]));
519 memcpy(q_temp[i + 2*ch_data->
t_env[0]], sbr->
q_m[0], m_max *
sizeof(sbr->
q_m[0][0]));
522 for (i = 0; i < 4; i++) {
523 memcpy(g_temp[i + 2 * ch_data->
t_env[0]],
526 memcpy(q_temp[i + 2 * ch_data->
t_env[0]],
533 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
534 memcpy(g_temp[h_SL + i], sbr->
gain[e], m_max *
sizeof(sbr->
gain[0][0]));
535 memcpy(q_temp[h_SL + i], sbr->
q_m[e], m_max *
sizeof(sbr->
q_m[0][0]));
540 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
545 if (h_SL && e != e_a[0] && e != e_a[1]) {
548 for (m = 0; m < m_max; m++) {
549 const int idx1 = i + h_SL;
552 for (j = 0; j <= h_SL; j++) {
562 g_filt = g_temp[i + h_SL];
566 sbr->
dsp.
hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
569 if (e != e_a[0] && e != e_a[1]) {
574 int idx = indexsine&1;
575 int A = (1-((indexsine+(kx & 1))&2));
576 int B = (A^(-idx)) + idx;
577 unsigned *
out = &Y1[i][kx][idx];
582 for (m = 0; m+1 < m_max; m+=2) {
584 shift = 22 - in[
m ].
exp;
585 shift2= 22 - in[m+1].
exp;
586 if (shift < 1 || shift2 < 1) {
591 round = 1 << (shift-1);
592 out[2*
m ] += (int)(in[m ].mant * A + round) >>
shift;
596 round = 1 << (shift2-1);
597 out[2*m+2] += (int)(in[m+1].mant * B + round) >>
shift2;
602 shift = 22 - in[
m ].
exp;
606 }
else if (shift < 32) {
607 round = 1 << (shift-1);
608 out[2*
m ] += (int)(in[m ].mant * A + round) >>
shift;
612 indexnoise = (indexnoise + m_max) & 0x1ff;
613 indexsine = (indexsine + 1) & 3;
uint8_t s_indexmapped[8][48]
static av_always_inline SoftFloat av_sqrt_sf(SoftFloat val)
Rounding-to-nearest used.
static int shift(int a, int b)
unsigned bs_smoothing_mode
INTFLOAT bw_array[5]
Chirp factors.
static av_const SoftFloat av_div_sf(SoftFloat a, SoftFloat b)
b has to be normalized and not zero.
static const int fixed_log_table[10]
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
Chirp Factors (14496-3 sp04 p214)
AAC_SIGNE kx[2]
kx', and kx respectively, kx is the first QMF subband where SBR is used.
static const SoftFloat FLOAT_0
AAC_FLOAT noise_facs[3][5]
Noise scalefactors.
AAC_SIGNE n_lim
Number of limiter bands.
#define ENVELOPE_ADJUSTMENT_OFFSET
void(* hf_g_filt)(INTFLOAT(*Y)[2], const INTFLOAT(*X_high)[40][2], const AAC_FLOAT *g_filt, int m_max, intptr_t ixh)
AAC Spectral Band Replication decoding data.
static int fixed_log(int x)
static const SoftFloat FLOAT_100000
static void sbr_hf_inverse_filter(SBRDSPContext *dsp, int(*alpha0)[2], int(*alpha1)[2], const int X_low[32][40][2], int k0)
High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering (14496-3 sp04 p214) Warning: Thi...
static double alpha(void *priv, double x, double y)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
AAC_SIGNE m[2]
M' and M respectively, M is the number of QMF subbands that use SBR.
static void sbr_hf_assemble(int Y1[38][64][2], const int X_high[64][40][2], SpectralBandReplication *sbr, SBRData *ch_data, const int e_a[2])
Assembling HF Signals (14496-3 sp04 p220)
static const SoftFloat FLOAT_1
static const SoftFloat FLOAT_0999999
Spectral Band Replication definitions and structures.
simple assert() macros that are a bit more flexible than ISO C assert().
static av_always_inline av_const double round(double x)
AAC Spectral Band Replication decoding functions.
common internal API header
uint8_t t_env_num_env_old
Envelope time border of the last envelope of the previous frame.
AAC Spectral Band Replication function declarations.
unsigned bs_limiter_gains
static const int CONST_RECIP_LN2
AAC_FLOAT e_origmapped[7][48]
Dequantized envelope scalefactors, remapped.
uint8_t s_mapped[7][48]
Sinusoidal presence, remapped.
static av_const int av_gt_sf(SoftFloat a, SoftFloat b)
AAC definitions and structures.
static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr, SBRData *ch_data, const int e_a[2])
Calculation of levels of additional HF signal components (14496-3 sp04 p219) and Calculation of gain ...
AAC_FLOAT q_mapped[7][48]
Dequantized noise scalefactors, remapped.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Replacements for frequently missing libm functions.
static int fixed_exp(int x)
static const int CONST_076923
AAC_FLOAT q_m[7][48]
Amplitude adjusted noise scalefactors.
AAC_FLOAT env_facs[6][48]
Envelope scalefactors.
#define NOISE_FLOOR_OFFSET
static av_const SoftFloat av_sub_sf(SoftFloat a, SoftFloat b)
AAC_FLOAT e_curr[7][48]
Estimated envelope.
uint8_t bs_invf_mode[2][5]
static av_const SoftFloat av_add_sf(SoftFloat a, SoftFloat b)
static const int CONST_LN2
static const SoftFloat FLOAT_1584893192
static const int shift2[6]
static av_const SoftFloat av_mul_sf(SoftFloat a, SoftFloat b)
uint8_t t_env[8]
Envelope time borders.
aacsbr functions pointers
AAC_FLOAT s_m[7][48]
Sinusoidal levels.
uint16_t f_tablelim[30]
Frequency borders for the limiter.
Spectral Band Replication per channel data.
static const SoftFloat FLOAT_EPSILON
static const int fixed_exp_table[7]
void(* hf_apply_noise[4])(INTFLOAT(*Y)[2], const AAC_FLOAT *s_m, const AAC_FLOAT *q_filt, int noise, int kx, int m_max)
static void make_bands(int16_t *bands, int start, int stop, int num_bands)
static const SoftFloat FLOAT_MIN
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
Dequantization and stereo decoding (14496-3 sp04 p203)
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
void(* autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2])
static av_const SoftFloat av_int2sf(int v, int frac_bits)
Converts a mantisse and exponent to a SoftFloat.
static void aacsbr_func_ptr_init(AACSBRContext *c)
AAC_SIGNE n_q
Number of noise floor bands.
Spectral Band Replication.
AAC_SIGNE n[2]
N_Low and N_High respectively, the number of frequency bands for low and high resolution.