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28 #ifndef AVCODEC_AACENC_UTILS_H
29 #define AVCODEC_AACENC_UTILS_H
36 #define ROUND_STANDARD 0.4054f
37 #define ROUND_TO_ZERO 0.1054f
38 #define C_QUANT 0.4054f
44 float a = fabsf(
in[
i]);
45 out[
i] = sqrtf(
a * sqrtf(
a));
51 return sqrtf(
a * sqrtf(
a));
59 static inline int quant(
float coef,
const float Q,
const float rounding)
62 return sqrtf(
a * sqrtf(
a)) + rounding;
66 int size,
int is_signed,
int maxval,
const float Q34,
71 float qc = scaled[
i] * Q34;
73 if (is_signed &&
in[
i] < 0.0
f) {
80 static inline float find_max_val(
int group_len,
int swb_size,
const float *scaled)
84 for (w2 = 0; w2 < group_len; w2++) {
85 for (
i = 0;
i < swb_size;
i++) {
86 maxval =
FFMAX(maxval, scaled[w2*128+
i]);
96 qmaxval = maxval * Q34 +
C_QUANT;
105 const float *scaled,
float nzslope) {
106 const float iswb_size = 1.0f / swb_size;
107 const float iswb_sizem1 = 1.0f / (swb_size - 1);
108 const float ethresh = thresh;
111 for (w2 = 0; w2 < group_len; w2++) {
112 float e = 0.0f, e2 = 0.0f, var = 0.0f, maxval = 0.0f;
114 for (
i = 0;
i < swb_size;
i++) {
115 float s = fabsf(scaled[w2*128+
i]);
116 maxval =
FFMAX(maxval,
s);
127 nzl += (
s / ethresh) * (
s / ethresh);
137 for (
i = 0;
i < swb_size;
i++) {
138 float d = fabsf(scaled[w2*128+
i]) - e;
141 var = sqrtf(var * iswb_sizem1);
144 frm = e /
FFMIN(e+4*var,maxval);
175 for (
i = 0;
i < num;
i++) {
177 if (
error < quant_min_err) {
178 quant_min_err =
error;
190 return 0.001f + 0.0035f * (
b*
b*
b) / (15.5
f*15.5
f*15.5
f);
201 unsigned char prevband = 0;
204 for (
g = 0;
g < 128;
g++)
211 prevband = nextband[prevband] =
w*16+
g;
214 nextband[prevband] = prevband;
223 nextband[prevband] = nextband[band];
233 const uint8_t *nextband,
int prev_sf,
int band)
247 const uint8_t *nextband,
int prev_sf,
int new_sf,
int band)
264 union {
unsigned u;
int s; } v = { previous_val * 1664525
u + 1013904223 };
268 #define ERROR_IF(cond, ...) \
270 av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
271 return AVERROR(EINVAL); \
274 #define WARN_IF(cond, ...) \
276 av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
static void error(const char *err)
static double cb(void *priv, double x, double y)
#define u(width, name, range_min, range_max)
static void abs_pow34_v(float *out, const float *in, const int size)
uint8_t zeroes[128]
band is not coded (used by encoder)
static av_always_inline float bval2bmax(float b)
approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
static av_always_inline float ff_fast_powf(float x, float y)
Compute x^y for floating point x, y.
static int ff_sfdelta_can_remove_band(const SingleChannelElement *sce, const uint8_t *nextband, int prev_sf, int band)
static uint8_t coef2maxsf(float coef)
Return the maximum scalefactor where the quantized coef is not zero.
int num_swb
number of scalefactor window bands
#define SCALE_DIV_512
scalefactor difference that corresponds to scale difference in 512 times
static int ff_sfdelta_can_replace(const SingleChannelElement *sce, const uint8_t *nextband, int prev_sf, int new_sf, int band)
static float find_form_factor(int group_len, int swb_size, float thresh, const float *scaled, float nzslope)
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
static double val(void *priv, double ch)
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
IndividualChannelStream ics
This is the more generic form
static void ff_init_nextband_map(const SingleChannelElement *sce, uint8_t *nextband)
int sf_idx[128]
scalefactor indices (used by encoder)
static int weight(int i, int blen, int offset)
static uint8_t coef2minsf(float coef)
Return the minimum scalefactor where the quantized coef does not clip.
static void quantize_bands(int *out, const float *in, const float *scaled, int size, int is_signed, int maxval, const float Q34, const float rounding)
static int quant_array_idx(const float val, const float *arr, const int num)
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
#define SCALE_MAX_DIFF
maximum scalefactor difference allowed by standard
static float pos_pow34(float a)
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
Single Channel Element - used for both SCE and LFE elements.
#define i(width, name, range_min, range_max)
#define SCALE_ONE_POS
scalefactor index that corresponds to scale=1.0
static int find_min_book(float maxval, int sf)
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
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 ug o o w
@ RESERVED_BT
Band types following are encoded differently from others.
#define FF_ARRAY_ELEMS(a)
static const unsigned char aac_maxval_cb[]
float ff_aac_pow34sf_tab[428]
static float find_max_val(int group_len, int swb_size, const float *scaled)
enum BandType band_type[128]
band types
static void ff_nextband_remove(uint8_t *nextband, int prevband, int band)