Go to the documentation of this file.
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)
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){
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;
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)
305 alpha0[k][1] = 0x7fffffff;
306 else if (
shift <= -30)
319 alpha1[k][0] = 0x7fffffff;
320 else if (
shift <= -30)
334 alpha1[k][1] = 0x7fffffff;
335 else if (
shift <= -30)
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++) {
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];
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) {
599 }
else if (
shift < 32) {
605 indexnoise = (indexnoise + m_max) & 0x1ff;
606 indexsine = (indexsine + 1) & 3;
unsigned bs_limiter_gains
AAC_FLOAT e_origmapped[7][48]
Dequantized envelope scalefactors, remapped.
static const SoftFloat FLOAT_EPSILON
A small value.
AAC_FLOAT env_facs[6][48]
static void aacsbr_func_ptr_init(AACSBRContext *c)
AAC_SIGNE m[2]
M' and M respectively, M is the number of QMF subbands that use SBR.
static av_const SoftFloat av_sub_sf(SoftFloat a, SoftFloat b)
AAC_FLOAT q_m[7][48]
Amplitude adjusted noise scalefactors.
uint8_t t_env_num_env_old
Envelope time border of the last envelope of the previous frame.
uint8_t t_env[8]
Envelope time borders.
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
Chirp Factors (14496-3 sp04 p214)
static const SoftFloat FLOAT_1
1.0
static av_const int av_gt_sf(SoftFloat a, SoftFloat b)
Compares two SoftFloats.
static av_always_inline SoftFloat av_sqrt_sf(SoftFloat val)
Rounding-to-nearest used.
static av_const SoftFloat av_div_sf(SoftFloat a, SoftFloat b)
b has to be normalized and not zero.
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)
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo x
AAC_FLOAT noise_facs[3][5]
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.
static const SoftFloat FLOAT_MIN
#define av_assert0(cond)
assert() equivalent, that is always enabled.
#define NOISE_FLOOR_OFFSET
void(* autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2])
static const float bands[]
AAC_FLOAT s_m[7][48]
Sinusoidal levels.
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) 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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
AAC_SIGNE n_lim
Number of limiter bands.
uint8_t env_facs_q[6][48]
Envelope scalefactors.
static const SoftFloat FLOAT_0
0.0
uint16_t f_tablelim[30]
Frequency borders for the limiter.
aacsbr functions pointers
static int fixed_log(int x)
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_SIGNE n[2]
N_Low and N_High respectively, the number of frequency bands for low and high resolution.
uint8_t s_indexmapped[8][48]
unsigned bs_smoothing_mode
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
static const SoftFloat FLOAT_0999999
0.999999
static const SoftFloat FLOAT_1584893192
1.584893192 (10^.2)
static const SoftFloat FLOAT_100000
100000
Spectral Band Replication.
uint8_t bs_invf_mode[2][5]
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)
AAC_SIGNE n_q
Number of noise floor bands.
static int fixed_exp(int x)
static const int CONST_076923
static const int shift2[6]
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
AAC_FLOAT e_curr[7][48]
Estimated envelope.
static const int CONST_RECIP_LN2
#define i(width, name, range_min, range_max)
static av_always_inline av_const double round(double x)
Spectral Band Replication per channel data.
INTFLOAT bw_array[5]
Chirp factors.
static av_const SoftFloat av_int2sf(int v, int frac_bits)
Converts a mantisse and exponent to a SoftFloat.
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 av_const SoftFloat av_add_sf(SoftFloat a, SoftFloat b)
static const int fixed_exp_table[7]
static void make_bands(int16_t *bands, int start, int stop, int num_bands)
uint8_t noise_facs_q[3][5]
Noise scalefactors.
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
Dequantization and stereo decoding (14496-3 sp04 p203)
static const int CONST_LN2
static int shift(int a, int b)
#define ENVELOPE_ADJUSTMENT_OFFSET
static const int16_t alpha[]
static const int fixed_log_table[10]
AAC_FLOAT q_mapped[7][48]
Dequantized noise scalefactors, remapped.
AAC_SIGNE kx[2]
kx', and kx respectively, kx is the first QMF subband where SBR is used.
static av_const SoftFloat av_mul_sf(SoftFloat a, SoftFloat b)
uint8_t s_mapped[7][48]
Sinusoidal presence, remapped.