88 int i, ret, xpow,
tmp;
92 for (
i=0;
i<10;
i+=2){
93 xpow = (
int)(((int64_t)xpow * x + 0x40000000) >> 31);
97 xpow = (
int)(((int64_t)xpow * x + 0x40000000) >> 31);
108 Q31(1.0/720),
Q31(1.0/5040),
Q31(1.0/40320)
113 int i, ret, xpow,
tmp;
118 xpow = (
int)(((int64_t)xpow * x + 0x400000) >> 23);
128 int k, previous, present;
129 int base, prod, nz = 0;
131 base = (stop << 23) / start;
132 while (
base < 0x40000000){
143 for (k = 0; k < num_bands-1; k++) {
144 prod = (
int)(((int64_t)prod *
base + 0x400000) >> 23);
145 present = (prod + 0x400000) >> 23;
146 bands[k] = present - previous;
149 bands[num_bands-1] = stop - previous;
167 temp1.
mant = 759250125;
169 temp1.
mant = 0x20000000;
170 temp1.
exp = (temp1.
exp >> 1) + 1;
171 if (temp1.
exp > 66) {
178 temp2.
mant = 759250125;
180 temp2.
mant = 0x20000000;
181 temp2.
exp = (temp2.
exp >> 1) + 1;
188 for (k = 0; k < sbr->
n_q; k++) {
192 sbr->data[0].noise_facs_q[e][k] + 2;
193 temp1.
mant = 0x20000000;
196 temp2.
mant = 0x20000000;
203 for (ch = 0; ch < (id_aac ==
TYPE_CPE) + 1; ch++) {
211 temp1.
mant = 759250125;
213 temp1.
mant = 0x20000000;
214 temp1.
exp = (temp1.
exp >> 1) + 1;
215 if (temp1.
exp > 66) {
222 for (k = 0; k < sbr->
n_q; k++){
224 sbr->data[ch].noise_facs_q[e][k] + 1;
236 int (*alpha0)[2],
int (*alpha1)[2],
237 const int X_low[32][40][2],
int k0)
242 for (k = 0; k < k0; k++) {
269 if (!phi[1][0][0].mant) {
283 a00 =
av_div_sf(temp_real, phi[1][0][0]);
289 alpha0[k][0] = 0x7fffffff;
290 else if (
shift <= -30)
304 alpha0[k][1] = 0x7fffffff;
305 else if (
shift <= -30)
318 alpha1[k][0] = 0x7fffffff;
319 else if (
shift <= -30)
333 alpha1[k][1] = 0x7fffffff;
334 else if (
shift <= -30)
346 shift = (
int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
347 (int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
349 if (
shift >= 0x20000000){
356 shift = (
int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
357 (int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
359 if (
shift >= 0x20000000){
373 static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
376 for (
i = 0;
i < sbr->
n_q;
i++) {
382 if (new_bw < ch_data->bw_array[
i]){
383 accu = (int64_t)new_bw * 1610612736;
384 accu += (int64_t)ch_data->
bw_array[
i] * 0x20000000;
385 new_bw = (
int)((accu + 0x40000000) >> 31);
387 accu = (int64_t)new_bw * 1946157056;
388 accu += (int64_t)ch_data->
bw_array[
i] * 201326592;
389 new_bw = (
int)((accu + 0x40000000) >> 31);
391 ch_data->
bw_array[
i] = new_bw < 0x2000000 ? 0 : new_bw;
400 SBRData *ch_data,
const int e_a[2])
404 static const SoftFloat limgain[4] = { { 760155524, 0 }, { 0x20000000, 1 },
405 { 758351638, 1 }, { 625000000, 34 } };
408 int delta = !((e == e_a[1]) || (e == e_a[0]));
409 for (k = 0; k < sbr->
n_lim; k++) {
413 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
437 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
448 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
453 sbr->
q_m[e][m] = q_m_max;
455 sbr->
gain[e][m] = gain_max;
458 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
478 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
489 const int X_high[64][40][2],
495 const int kx = sbr->
kx[1];
496 const int m_max = sbr->
m[1];
509 for (
i = 0;
i < h_SL;
i++) {
510 memcpy(g_temp[
i + 2*ch_data->
t_env[0]], sbr->
gain[0], m_max *
sizeof(sbr->
gain[0][0]));
511 memcpy(q_temp[
i + 2*ch_data->
t_env[0]], sbr->
q_m[0], m_max *
sizeof(sbr->
q_m[0][0]));
514 for (
i = 0;
i < 4;
i++) {
515 memcpy(g_temp[
i + 2 * ch_data->
t_env[0]],
518 memcpy(q_temp[
i + 2 * ch_data->
t_env[0]],
525 for (
i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1];
i++) {
526 memcpy(g_temp[h_SL +
i], sbr->
gain[e], m_max *
sizeof(sbr->
gain[0][0]));
527 memcpy(q_temp[h_SL +
i], sbr->
q_m[e], m_max *
sizeof(sbr->
q_m[0][0]));
532 for (
i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1];
i++) {
537 if (h_SL && e != e_a[0] && e != e_a[1]) {
540 for (m = 0; m < m_max; m++) {
541 const int idx1 =
i + h_SL;
542 g_filt[m].
mant = g_filt[m].
exp = 0;
543 q_filt[m].
mant = q_filt[m].
exp = 0;
544 for (j = 0; j <= h_SL; j++) {
554 g_filt = g_temp[
i + h_SL];
561 if (e != e_a[0] && e != e_a[1]) {
566 int idx = indexsine&1;
567 int A = (1-((indexsine+(kx & 1))&2));
568 int B = (
A^(-idx)) + idx;
569 unsigned *
out = &Y1[
i][kx][idx];
574 for (m = 0; m+1 < m_max; m+=2) {
598 }
else if (
shift < 32) {
604 indexnoise = (indexnoise + m_max) & 0x1ff;
605 indexsine = (indexsine + 1) & 3;
AAC Spectral Band Replication function declarations.
#define ENVELOPE_ADJUSTMENT_OFFSET
#define NOISE_FLOOR_OFFSET
static int fixed_log(int x)
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 ...
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 int fixed_log_table[10]
static const int CONST_076923
static void aacsbr_func_ptr_init(AACSBRContext *c)
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 const int CONST_RECIP_LN2
static const int fixed_exp_table[7]
static void make_bands(int16_t *bands, int start, int stop, int num_bands)
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
Dequantization and stereo decoding (14496-3 sp04 p203)
static const int CONST_LN2
static int fixed_exp(int x)
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
Chirp Factors (14496-3 sp04 p214)
AAC Spectral Band Replication decoding functions.
AAC Spectral Band Replication decoding data.
static const float bands[]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static const int16_t alpha[]
static const int shift2[6]
common internal API header
Replacements for frequently missing libm functions.
static av_always_inline av_const double round(double x)
Spectral Band Replication definitions and structures.
static av_const SoftFloat av_add_sf(SoftFloat a, SoftFloat b)
static av_const SoftFloat av_sub_sf(SoftFloat a, SoftFloat b)
static const SoftFloat FLOAT_1
1.0
static av_const SoftFloat av_div_sf(SoftFloat a, SoftFloat b)
b has to be normalized and not zero.
static const SoftFloat FLOAT_MIN
static av_const SoftFloat av_mul_sf(SoftFloat a, SoftFloat b)
static const SoftFloat FLOAT_0
0.0
static av_const SoftFloat av_int2sf(int v, int frac_bits)
Converts a mantisse and exponent to a SoftFloat.
static const SoftFloat FLOAT_1584893192
1.584893192 (10^.2)
static const SoftFloat FLOAT_EPSILON
A small value.
static av_always_inline SoftFloat av_sqrt_sf(SoftFloat val)
Rounding-to-nearest used.
static const SoftFloat FLOAT_100000
100000
static const SoftFloat FLOAT_0999999
0.999999
static av_const int av_gt_sf(SoftFloat a, SoftFloat b)
Compares two SoftFloats.
static int shift(int a, int b)
aacsbr functions pointers
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)
void(* autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2])
void(* hf_g_filt)(INTFLOAT(*Y)[2], const INTFLOAT(*X_high)[40][2], const AAC_FLOAT *g_filt, int m_max, intptr_t ixh)
Spectral Band Replication per channel data.
INTFLOAT bw_array[5]
Chirp factors.
AAC_FLOAT env_facs[6][48]
uint8_t s_indexmapped[8][48]
uint8_t noise_facs_q[3][5]
Noise scalefactors.
uint8_t t_env[8]
Envelope time borders.
uint8_t t_env_num_env_old
Envelope time border of the last envelope of the previous frame.
uint8_t bs_invf_mode[2][5]
AAC_FLOAT noise_facs[3][5]
uint8_t env_facs_q[6][48]
Envelope scalefactors.
Spectral Band Replication.
AAC_SIGNE m[2]
M' and M respectively, M is the number of QMF subbands that use SBR.
uint8_t s_mapped[7][48]
Sinusoidal presence, remapped.
AAC_FLOAT q_mapped[7][48]
Dequantized noise scalefactors, remapped.
AAC_FLOAT e_origmapped[7][48]
Dequantized envelope scalefactors, remapped.
unsigned bs_smoothing_mode
unsigned bs_limiter_gains
AAC_SIGNE kx[2]
kx', and kx respectively, kx is the first QMF subband where SBR is used.
AAC_SIGNE n_q
Number of noise floor bands.
AAC_SIGNE n_lim
Number of limiter bands.
uint16_t f_tablelim[30]
Frequency borders for the limiter.
AAC_FLOAT s_m[7][48]
Sinusoidal levels.
AAC_FLOAT e_curr[7][48]
Estimated envelope.
AAC_SIGNE n[2]
N_Low and N_High respectively, the number of frequency bands for low and high resolution.
AAC_FLOAT q_m[7][48]
Amplitude adjusted noise scalefactors.