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53 const float *coeffs =
s->coeffs;
60 nb_samples =
FFMIN(
s->nb_samples,
s->n -
s->pts);
61 if (nb_samples <= 0) {
69 memcpy(
frame->
data[0], coeffs +
s->pts, nb_samples *
sizeof(
float));
95 static float *
make_lpf(
int num_taps,
float Fc,
float beta,
float rho,
96 float scale,
int dc_norm)
98 int i, m = num_taps - 1;
99 float *
h =
av_calloc(num_taps,
sizeof(*
h)), sum = 0;
107 for (
i = 0;
i <= m / 2;
i++) {
108 float z =
i - .5f * m, x = z *
M_PI, y = z * mult1;
109 h[
i] = x ?
sinf(Fc * x) / x : Fc;
117 for (
i = 0; dc_norm &&
i < num_taps;
i++)
126 static const float coefs[][4] = {
127 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
128 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
129 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
130 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
131 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
132 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
133 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
134 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
135 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
136 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
138 float realm = logf(tr_bw / .0005
f) / logf(2.
f);
141 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
142 float b1 = ((
c1[0] * att +
c1[1]) * att +
c1[2]) * att +
c1[3];
144 return b0 + (
b1 -
b0) * (realm - (
int)realm);
147 return .1102f * (att - 8.7f);
149 return .58417f *
powf(att - 20.96
f, .4
f) + .07886f * (att - 20.96f);
153 static void kaiser_params(
float att,
float Fc,
float tr_bw,
float *beta,
int *num_taps)
155 *beta = *beta < 0.f ?
kaiser_beta(att, tr_bw * .5
f / Fc): *beta;
156 att = att < 60.f ? (att - 7.95f) / (2.285
f *
M_PI * 2.
f) :
157 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022
f) * *beta + .06186902f;
158 *num_taps = !*num_taps ?
ceilf(att/tr_bw + 1) : *num_taps;
161 static float *
lpf(
float Fn,
float Fc,
float tbw,
int *num_taps,
float att,
float *beta,
int round)
165 if ((Fc /= Fn) <= 0.
f || Fc >= 1.
f) {
170 att = att ? att : 120.f;
176 *num_taps =
av_clip(n, 11, 32767);
178 *num_taps = 1 + 2 * (
int)((
int)((*num_taps / 2) * Fc + .5
f) / Fc + .5f);
181 return make_lpf(*num_taps |= 1, Fc, *beta, 0.
f, 1.
f, 0);
186 for (
int i = 0;
i < n;
i++)
192 #define SQR(a) ((a) * (a))
204 float *pi_wraps, *
work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.
f;
205 int i, work_len, begin, end, imp_peak = 0, peak = 0,
ret;
206 float imp_sum = 0, peak_imp_sum = 0,
scale = 1.f;
207 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
209 for (
i = *
len, work_len = 2 * 2 * 8;
i > 1; work_len <<= 1, i >>= 1);
212 work =
av_calloc((work_len + 2) + (work_len / 2 + 1),
sizeof(
float));
215 pi_wraps = &
work[work_len + 2];
230 for (
i = 0;
i <= work_len;
i += 2) {
232 float detect = 2 *
M_PI;
233 float delta = angle - prev_angle2;
240 delta = angle - prev_angle1;
244 pi_wraps[
i >> 1] = cum_1pi;
252 for (
i = 0;
i < work_len;
i++)
253 work[
i] *= 2.
f / work_len;
255 for (
i = 1;
i < work_len / 2;
i++) {
257 work[
i + work_len / 2] = 0;
261 for (
i = 2;
i < work_len;
i += 2)
262 work[
i + 1] = phase1 *
i / work_len * pi_wraps[work_len >> 1] + (1 - phase1) * (
work[
i + 1] + pi_wraps[
i >> 1]) - pi_wraps[
i >> 1];
266 for (
i = 2;
i < work_len;
i += 2) {
274 for (
i = 0;
i < work_len;
i++)
275 work[
i] *= 2.
f / work_len;
278 for (
i = 0;
i <= (
int) (pi_wraps[work_len >> 1] /
M_PI + .5
f);
i++) {
280 if (
fabs(imp_sum) >
fabs(peak_imp_sum)) {
281 peak_imp_sum = imp_sum;
294 }
else if (phase1 == 1) {
295 begin = peak - *
len / 2;
297 begin = (.997f - (2 - phase1) * .22
f) * *
len + .5f;
298 end = (.997f + (0 - phase1) * .22
f) * *
len + .5f;
299 begin = peak - (begin & ~3);
300 end = peak + 1 + ((end + 3) & ~3);
309 for (
i = 0;
i < *
len;
i++) {
310 (*h)[
i] =
work[(begin + (phase > 50.f ? *
len - 1 -
i :
i) + work_len) & (work_len - 1)];
312 *post_len = phase > 50 ? peak - begin : begin + *
len - (peak + 1);
315 work_len, pi_wraps[work_len >> 1] /
M_PI, peak, peak_imp_sum, imp_peak,
316 work[imp_peak], *
len, *post_len, 100.
f - 100.
f * *post_len / (*
len - 1));
328 float Fn =
s->sample_rate * .5f;
330 int i, n, post_peak, longer;
335 if (
s->Fc0 >= Fn ||
s->Fc1 >= Fn) {
337 "filter frequency must be less than %d/2.\n",
s->sample_rate);
341 h[0] =
lpf(Fn,
s->Fc0,
s->tbw0, &
s->num_taps[0],
s->att, &
s->beta,
s->round);
342 h[1] =
lpf(Fn,
s->Fc1,
s->tbw1, &
s->num_taps[1],
s->att, &
s->beta,
s->round);
347 longer =
s->num_taps[1] >
s->num_taps[0];
348 n =
s->num_taps[longer];
351 for (
i = 0;
i <
s->num_taps[!longer];
i++)
352 h[longer][
i + (n -
s->num_taps[!longer]) / 2] +=
h[!longer][
i];
360 if (
s->phase != 50.f) {
374 for (
i = 0;
i < n;
i++)
375 s->coeffs[
i] =
h[longer][
i];
401 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
402 #define OFFSET(x) offsetof(SincContext, x)
407 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
408 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
415 {
"hptaps",
"set number of taps for high-pass filter",
OFFSET(num_taps[0]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
416 {
"lptaps",
"set number of taps for low-pass filter",
OFFSET(num_taps[1]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
424 .description =
NULL_IF_CONFIG_SMALL(
"Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
426 .priv_class = &sinc_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
#define AVERROR_EOF
End of file.
double av_bessel_i0(double x)
0th order modified bessel function of the first kind.
This structure describes decoded (raw) audio or video data.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static float kaiser_beta(float att, float tr_bw)
#define FILTER_QUERY_FUNC(func)
const char * name
Filter name.
A link between two filters.
static __device__ float ceilf(float a)
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static float * lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
static double b1(void *priv, double x, double y)
static av_always_inline float scale(float x, float s)
static float * make_lpf(int num_taps, float Fc, float beta, float rho, float scale, int dc_norm)
static __device__ float fabsf(float a)
A filter pad used for either input or output.
static int16_t mult(Float11 *f1, Float11 *f2)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define FF_ARRAY_ELEMS(a)
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
static int adjust(int x, int size)
#define av_assert0(cond)
assert() equivalent, that is always enabled.
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
const AVFilter ff_asrc_sinc
#define av_realloc_f(p, o, n)
static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
Describe the class of an AVClass context structure.
static __device__ float fabs(float a)
static const AVFilterPad sinc_outputs[]
@ AV_TX_INPLACE
Allows for in-place transformations, where input == output.
must be printed separately If there s no standard function for printing the type you the WRITE_1D_FUNC_ARGV macro is a very quick way to create one See libavcodec dv_tablegen c for an example The h file This file should the initialization functions should not do and instead of the variable declarations the generated *_tables h file should be included Since that will be generated in the build the path must be i e not Makefile changes To make the automatic table creation work
static int activate(AVFilterContext *ctx)
static __device__ float sqrtf(float a)
static av_cold void uninit(AVFilterContext *ctx)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
An AVChannelLayout holds information about the channel layout of audio data.
AVFilterContext * src
source filter
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
int sample_rate
samples per second
#define i(width, name, range_min, range_max)
static av_always_inline av_const double round(double x)
static void invert(float *h, int n)
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
void * av_calloc(size_t nmemb, size_t size)
static int config_output(AVFilterLink *outlink)
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
static const AVOption sinc_options[]
#define AV_CHANNEL_LAYOUT_MONO
#define FILTER_OUTPUTS(array)
static float safe_log(float x)
static int query_formats(AVFilterContext *ctx)
static double b0(void *priv, double x, double y)
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function. If this function returns true
AVFILTER_DEFINE_CLASS(sinc)