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_q[e][k] + 2;
194 temp1.
mant = 0x20000000;
197 temp2.
mant = 0x20000000;
204 for (ch = 0; ch < (id_aac ==
TYPE_CPE) + 1; ch++) {
212 temp1.
mant = 759250125;
214 temp1.
mant = 0x20000000;
215 temp1.
exp = (temp1.
exp >> 1) + 1;
216 if (temp1.
exp > 66) {
223 for (k = 0; k < sbr->
n_q; k++){
225 sbr->data[
ch].noise_facs_q[e][k] + 1;
237 int (*alpha0)[2],
int (*alpha1)[2],
238 const int X_low[32][40][2],
int k0)
243 for (k = 0; k < k0; k++) {
270 if (!phi[1][0][0].mant) {
284 a00 =
av_div_sf(temp_real, phi[1][0][0]);
290 alpha0[k][0] = 0x7fffffff;
291 else if (shift <= -30)
298 round = 1 << (shift-1);
299 alpha0[k][0] = (a00.
mant +
round) >> shift;
305 alpha0[k][1] = 0x7fffffff;
306 else if (shift <= -30)
313 round = 1 << (shift-1);
314 alpha0[k][1] = (a01.
mant +
round) >> shift;
319 alpha1[k][0] = 0x7fffffff;
320 else if (shift <= -30)
327 round = 1 << (shift-1);
328 alpha1[k][0] = (a10.
mant +
round) >> shift;
334 alpha1[k][1] = 0x7fffffff;
335 else if (shift <= -30)
342 round = 1 << (shift-1);
343 alpha1[k][1] = (a11.
mant +
round) >> shift;
347 shift = (
int)(((int64_t)(alpha1[k][0]>>1) * (alpha1[k][0]>>1) + \
348 (int64_t)(alpha1[k][1]>>1) * (alpha1[k][1]>>1) + \
350 if (shift >= 0x20000000){
357 shift = (
int)(((int64_t)(alpha0[k][0]>>1) * (alpha0[k][0]>>1) + \
358 (int64_t)(alpha0[k][1]>>1) * (alpha0[k][1]>>1) + \
360 if (shift >= 0x20000000){
374 static const int bw_tab[] = { 0, 1610612736, 1932735283, 2104533975 };
377 for (i = 0; i < sbr->
n_q; i++) {
383 if (new_bw < ch_data->bw_array[i]){
384 accu = (int64_t)new_bw * 1610612736;
385 accu += (int64_t)ch_data->
bw_array[i] * 0x20000000;
386 new_bw = (
int)((accu + 0x40000000) >> 31);
388 accu = (int64_t)new_bw * 1946157056;
389 accu += (int64_t)ch_data->
bw_array[i] * 201326592;
390 new_bw = (
int)((accu + 0x40000000) >> 31);
392 ch_data->
bw_array[i] = new_bw < 0x2000000 ? 0 : new_bw;
401 SBRData *ch_data,
const int e_a[2])
405 static const SoftFloat limgain[4] = { { 760155524, 0 }, { 0x20000000, 1 },
406 { 758351638, 1 }, { 625000000, 34 } };
409 int delta = !((e == e_a[1]) || (e == e_a[0]));
410 for (k = 0; k < sbr->
n_lim; k++) {
414 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
438 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
449 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
454 sbr->
q_m[e][m] = q_m_max;
456 sbr->
gain[e][m] = gain_max;
459 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
468 if (delta && !sbr->
s_m[e][m].mant)
479 for (m = sbr->
f_tablelim[k] - sbr->
kx[1]; m < sbr->f_tablelim[k + 1] - sbr->
kx[1]; m++) {
490 const int X_high[64][40][2],
496 const int kx = sbr->
kx[1];
497 const int m_max = sbr->
m[1];
510 for (i = 0; i < h_SL; i++) {
511 memcpy(g_temp[i + 2*ch_data->
t_env[0]], sbr->
gain[0], m_max *
sizeof(sbr->
gain[0][0]));
512 memcpy(q_temp[i + 2*ch_data->
t_env[0]], sbr->
q_m[0], m_max *
sizeof(sbr->
q_m[0][0]));
515 for (i = 0; i < 4; i++) {
516 memcpy(g_temp[i + 2 * ch_data->
t_env[0]],
519 memcpy(q_temp[i + 2 * ch_data->
t_env[0]],
526 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
527 memcpy(g_temp[h_SL + i], sbr->
gain[e], m_max *
sizeof(sbr->
gain[0][0]));
528 memcpy(q_temp[h_SL + i], sbr->
q_m[e], m_max *
sizeof(sbr->
q_m[0][0]));
533 for (i = 2 * ch_data->
t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
538 if (h_SL && e != e_a[0] && e != e_a[1]) {
541 for (m = 0; m < m_max; m++) {
542 const int idx1 = i + h_SL;
543 g_filt[m].
mant = g_filt[m].
exp = 0;
544 q_filt[m].
mant = q_filt[m].
exp = 0;
545 for (j = 0; j <= h_SL; j++) {
555 g_filt = g_temp[i + h_SL];
559 sbr->
dsp.
hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
562 if (e != e_a[0] && e != e_a[1]) {
567 int idx = indexsine&1;
568 int A = (1-((indexsine+(kx & 1))&2));
569 int B = (A^(-idx)) + idx;
570 unsigned *
out = &Y1[i][kx][idx];
575 for (m = 0; m+1 < m_max; m+=2) {
577 shift = 22 - in[m ].
exp;
578 shift2= 22 - in[m+1].
exp;
579 if (shift < 1 || shift2 < 1) {
584 round = 1 << (shift-1);
585 out[2*m ] += (
int)(in[m ].mant * A + round) >>
shift;
589 round = 1 << (shift2-1);
590 out[2*m+2] += (
int)(in[m+1].mant * B + round) >>
shift2;
595 shift = 22 - in[m ].
exp;
599 }
else if (shift < 32) {
600 round = 1 << (shift-1);
601 out[2*m ] += (
int)(in[m ].mant * A + round) >>
shift;
605 indexnoise = (indexnoise + m_max) & 0x1ff;
606 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)
static float alpha(float a)
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.
uint8_t noise_facs_q[3][5]
Noise scalefactors.
static const SoftFloat FLOAT_0
0.0
#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.
static int fixed_log(int x)
static const SoftFloat FLOAT_100000
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...
#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
1.0
static const SoftFloat FLOAT_0999999
0.999999
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)
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
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)
Compares two SoftFloats.
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.
static const float bands[]
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]
#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
1.584893192 (10^.2)
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
A small value.
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)
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.
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