56 int k, previous, present;
59 base =
powf((
float)stop / start, 1.0f / num_bands);
63 for (k = 0; k < num_bands-1; k++) {
66 bands[k] = present - previous;
69 bands[num_bands-1] = stop - previous;
77 static const double exp2_tab[2] = {1,
M_SQRT2};
82 float temp1, temp2, fac;
97 fac = temp1 / (1.0f + temp2);
103 for (k = 0; k < sbr->
n_q; k++) {
108 fac = temp1 / (1.0f + temp2);
114 for (ch = 0; ch < (id_aac ==
TYPE_CPE) + 1; ch++) {
129 for (k = 0; k < sbr->
n_q; k++)
141 float (*alpha0)[2],
float (*alpha1)[2],
142 const float X_low[32][40][2],
int k0)
145 for (k = 0; k < k0; k++) {
151 dk = phi[2][1][0] * phi[1][0][0] -
152 (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
158 float temp_real, temp_im;
159 temp_real = phi[0][0][0] * phi[1][1][0] -
160 phi[0][0][1] * phi[1][1][1] -
161 phi[0][1][0] * phi[1][0][0];
162 temp_im = phi[0][0][0] * phi[1][1][1] +
163 phi[0][0][1] * phi[1][1][0] -
164 phi[0][1][1] * phi[1][0][0];
166 alpha1[k][0] = temp_real / dk;
167 alpha1[k][1] = temp_im / dk;
174 float temp_real, temp_im;
175 temp_real = phi[0][0][0] + alpha1[k][0] * phi[1][1][0] +
176 alpha1[k][1] * phi[1][1][1];
177 temp_im = phi[0][0][1] + alpha1[k][1] * phi[1][1][0] -
178 alpha1[k][0] * phi[1][1][1];
180 alpha0[k][0] = -temp_real / phi[1][0][0];
181 alpha0[k][1] = -temp_im / phi[1][0][0];
184 if (alpha1[k][0] * alpha1[k][0] + alpha1[k][1] * alpha1[k][1] >= 16.0f ||
185 alpha0[k][0] * alpha0[k][0] + alpha0[k][1] * alpha0[k][1] >= 16.0f) {
199 static const float bw_tab[] = { 0.0f, 0.75f, 0.9f, 0.98f };
201 for (i = 0; i < sbr->
n_q; i++) {
207 if (new_bw < ch_data->bw_array[i]) {
208 new_bw = 0.75f * new_bw + 0.25f * ch_data->
bw_array[i];
210 new_bw = 0.90625f * new_bw + 0.09375f * ch_data->
bw_array[i];
211 ch_data->
bw_array[i] = new_bw < 0.015625f ? 0.0f : new_bw;
220 SBRData *ch_data,
const int e_a[2])
224 static const float limgain[4] = { 0.70795, 1.0, 1.41254, 10000000000 };
227 int delta = !((e == e_a[1]) || (e == e_a[0]));
228 for (k = 0; k < sbr->
n_lim; k++) {
229 float gain_boost, gain_max;
230 float sum[2] = { 0.0f, 0.0f };
231 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
237 ((1.0f + sbr->
e_curr[e][m]) *
238 (1.0f + sbr->
q_mapped[e][m] * delta)));
241 ((1.0f + sbr->
e_curr[e][m]) *
245 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
249 gain_max = limgain[sbr->
bs_limiter_gains] * sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
250 gain_max =
FFMIN(100000.f, gain_max);
251 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
252 float q_m_max = sbr->
q_m[e][
m] * gain_max / sbr->
gain[e][
m];
256 sum[0] = sum[1] = 0.0f;
257 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
261 + (delta && !sbr->
s_m[e][
m]) * sbr->
q_m[e][m] * sbr->
q_m[e][m];
263 gain_boost = sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
264 gain_boost =
FFMIN(1.584893192f, gain_boost);
265 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
266 sbr->
gain[e][
m] *= gain_boost;
267 sbr->
q_m[e][
m] *= gain_boost;
268 sbr->
s_m[e][
m] *= gain_boost;
276 const float X_high[64][40][2],
282 const int kx = sbr->
kx[1];
283 const int m_max = sbr->
m[1];
284 static const float h_smooth[5] = {
291 float (*g_temp)[48] = ch_data->
g_temp, (*q_temp)[48] = ch_data->
q_temp;
296 for (i = 0; i < h_SL; i++) {
297 memcpy(g_temp[i + 2*ch_data->
t_env[0]], sbr->
gain[0], m_max *
sizeof(sbr->
gain[0][0]));
298 memcpy(q_temp[i + 2*ch_data->
t_env[0]], sbr->
q_m[0], m_max *
sizeof(sbr->
q_m[0][0]));
301 for (i = 0; i < 4; i++) {
302 memcpy(g_temp[i + 2 * ch_data->
t_env[0]],
305 memcpy(q_temp[i + 2 * ch_data->
t_env[0]],
312 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
313 memcpy(g_temp[h_SL + i], sbr->
gain[e], m_max *
sizeof(sbr->
gain[0][0]));
314 memcpy(q_temp[h_SL + i], sbr->
q_m[e], m_max *
sizeof(sbr->
q_m[0][0]));
319 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
322 float *g_filt, *q_filt;
324 if (h_SL && e != e_a[0] && e != e_a[1]) {
327 for (m = 0; m < m_max; m++) {
328 const int idx1 = i + h_SL;
331 for (j = 0; j <= h_SL; j++) {
332 g_filt[
m] += g_temp[idx1 - j][
m] * h_smooth[j];
333 q_filt[
m] += q_temp[idx1 - j][
m] * h_smooth[j];
337 g_filt = g_temp[i + h_SL];
341 sbr->
dsp.
hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
344 if (e != e_a[0] && e != e_a[1]) {
349 int idx = indexsine&1;
350 int A = (1-((indexsine+(kx & 1))&2));
351 int B = (A^(-idx)) + idx;
352 float *
out = &Y1[i][kx][idx];
353 float *
in = sbr->
s_m[e];
354 for (m = 0; m+1 < m_max; m+=2) {
355 out[2*
m ] += in[
m ] *
A;
356 out[2*m+2] += in[m+1] *
B;
359 out[2*
m ] += in[
m ] *
A;
361 indexnoise = (indexnoise + m_max) & 0x1ff;
362 indexsine = (indexsine + 1) & 3;
uint8_t s_indexmapped[8][48]
static void sbr_hf_assemble(float Y1[38][64][2], const float X_high[64][40][2], SpectralBandReplication *sbr, SBRData *ch_data, const int e_a[2])
Assembling HF Signals (14496-3 sp04 p220)
unsigned bs_smoothing_mode
INTFLOAT bw_array[5]
Chirp factors.
static void aacsbr_func_ptr_init(AACSBRContext *c)
AAC_SIGNE kx[2]
kx', and kx respectively, kx is the first QMF subband where SBR is used.
uint8_t noise_facs_q[3][5]
Noise scalefactors.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
AAC_FLOAT noise_facs[3][5]
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.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void sbr_hf_inverse_filter(SBRDSPContext *dsp, float(*alpha0)[2], float(*alpha1)[2], const float X_low[32][40][2], int k0)
High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering (14496-3 sp04 p214) Warning: Thi...
AAC_SIGNE m[2]
M' and M respectively, M is the number of QMF subbands that use SBR.
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
Dequantization and stereo decoding (14496-3 sp04 p203)
Spectral Band Replication definitions and structures.
simple assert() macros that are a bit more flexible than ISO C assert().
static av_always_inline float ff_exp2fi(int x)
2^(x) for integer x
Reference: libavcodec/aacsbr.c.
uint8_t env_facs_q[6][48]
Envelope scalefactors.
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
AAC_FLOAT e_origmapped[7][48]
Dequantized envelope scalefactors, remapped.
uint8_t s_mapped[7][48]
Sinusoidal presence, remapped.
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 void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
Chirp Factors (14496-3 sp04 p214)
AAC_FLOAT q_m[7][48]
Amplitude adjusted noise scalefactors.
AAC_FLOAT env_facs[6][48]
#define NOISE_FLOOR_OFFSET
AAC_FLOAT e_curr[7][48]
Estimated envelope.
uint8_t bs_invf_mode[2][5]
common internal api header.
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 void make_bands(int16_t *bands, int start, int stop, int num_bands)
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])
#define LOCAL_ALIGNED_16(t, v,...)
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.