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52 const float *coeffs =
s->coeffs;
59 nb_samples =
FFMIN(
s->nb_samples,
s->n -
s->pts);
60 if (nb_samples <= 0) {
68 memcpy(
frame->data[0], coeffs +
s->pts, nb_samples *
sizeof(
float));
96 float term = 1, sum = 1, last_sum, x2 = x / 2;
103 sum += term *= y * y;
104 }
while (sum != last_sum);
109 static float *
make_lpf(
int num_taps,
float Fc,
float beta,
float rho,
110 float scale,
int dc_norm)
112 int i, m = num_taps - 1;
113 float *
h =
av_calloc(num_taps,
sizeof(*
h)), sum = 0;
118 for (
i = 0;
i <= m / 2;
i++) {
119 float z =
i - .5f * m, x = z *
M_PI, y = z * mult1;
120 h[
i] = x ?
sinf(Fc * x) / x : Fc;
128 for (
i = 0; dc_norm &&
i < num_taps;
i++)
137 static const float coefs[][4] = {
138 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
139 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
140 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
141 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
142 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
143 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
144 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
145 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
146 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
147 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
149 float realm = logf(tr_bw / .0005
f) / logf(2.
f);
152 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
153 float b1 = ((
c1[0] * att +
c1[1]) * att +
c1[2]) * att +
c1[3];
155 return b0 + (
b1 -
b0) * (realm - (
int)realm);
158 return .1102f * (att - 8.7f);
160 return .58417f *
powf(att - 20.96
f, .4
f) + .07886f * (att - 20.96f);
164 static void kaiser_params(
float att,
float Fc,
float tr_bw,
float *beta,
int *num_taps)
166 *beta = *beta < 0.f ?
kaiser_beta(att, tr_bw * .5
f / Fc): *beta;
167 att = att < 60.f ? (att - 7.95f) / (2.285
f *
M_PI * 2.
f) :
168 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022
f) * *beta + .06186902f;
169 *num_taps = !*num_taps ?
ceilf(att/tr_bw + 1) : *num_taps;
172 static float *
lpf(
float Fn,
float Fc,
float tbw,
int *num_taps,
float att,
float *beta,
int round)
176 if ((Fc /= Fn) <= 0.
f || Fc >= 1.
f) {
181 att = att ? att : 120.f;
187 *num_taps =
av_clip(n, 11, 32767);
189 *num_taps = 1 + 2 * (
int)((
int)((*num_taps / 2) * Fc + .5
f) / Fc + .5f);
192 return make_lpf(*num_taps |= 1, Fc, *beta, 0.
f, 1.
f, 0);
197 for (
int i = 0;
i < n;
i++)
203 #define PACK(h, n) h[1] = h[n]
204 #define UNPACK(h, n) h[n] = h[1], h[n + 1] = h[1] = 0;
205 #define SQR(a) ((a) * (a))
217 float *pi_wraps, *
work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.
f;
218 int i, work_len, begin, end, imp_peak = 0, peak = 0;
219 float imp_sum = 0, peak_imp_sum = 0;
220 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
222 for (
i = *
len, work_len = 2 * 2 * 8;
i > 1; work_len <<= 1, i >>= 1);
225 work =
av_calloc((work_len + 2) + (work_len / 2 + 1),
sizeof(
float));
228 pi_wraps = &
work[work_len + 2];
234 s->rdft =
s->irdft =
NULL;
237 if (!
s->rdft || !
s->irdft) {
245 for (
i = 0;
i <= work_len;
i += 2) {
247 float detect = 2 *
M_PI;
248 float delta = angle - prev_angle2;
255 delta = angle - prev_angle1;
259 pi_wraps[
i >> 1] = cum_1pi;
268 for (
i = 0;
i < work_len;
i++)
269 work[
i] *= 2.
f / work_len;
271 for (
i = 1;
i < work_len / 2;
i++) {
273 work[
i + work_len / 2] = 0;
277 for (
i = 2;
i < work_len;
i += 2)
278 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];
282 for (
i = 2;
i < work_len;
i += 2) {
290 for (
i = 0;
i < work_len;
i++)
291 work[
i] *= 2.
f / work_len;
294 for (
i = 0;
i <= (
int) (pi_wraps[work_len >> 1] /
M_PI + .5
f);
i++) {
296 if (
fabs(imp_sum) >
fabs(peak_imp_sum)) {
297 peak_imp_sum = imp_sum;
310 }
else if (phase1 == 1) {
311 begin = peak - *
len / 2;
313 begin = (.997f - (2 - phase1) * .22
f) * *
len + .5f;
314 end = (.997f + (0 - phase1) * .22
f) * *
len + .5f;
315 begin = peak - (begin & ~3);
316 end = peak + 1 + ((end + 3) & ~3);
325 for (
i = 0;
i < *
len;
i++) {
326 (*h)[
i] =
work[(begin + (phase > 50.f ? *
len - 1 -
i :
i) + work_len) & (work_len - 1)];
328 *post_len = phase > 50 ? peak - begin : begin + *
len - (peak + 1);
331 work_len, pi_wraps[work_len >> 1] /
M_PI, peak, peak_imp_sum, imp_peak,
332 work[imp_peak], *
len, *post_len, 100.
f - 100.
f * *post_len / (*
len - 1));
343 float Fn =
s->sample_rate * .5f;
345 int i, n, post_peak, longer;
350 if (
s->Fc0 >= Fn ||
s->Fc1 >= Fn) {
352 "filter frequency must be less than %d/2.\n",
s->sample_rate);
356 h[0] =
lpf(Fn,
s->Fc0,
s->tbw0, &
s->num_taps[0],
s->att, &
s->beta,
s->round);
357 h[1] =
lpf(Fn,
s->Fc1,
s->tbw1, &
s->num_taps[1],
s->att, &
s->beta,
s->round);
362 longer =
s->num_taps[1] >
s->num_taps[0];
363 n =
s->num_taps[longer];
366 for (
i = 0;
i <
s->num_taps[!longer];
i++)
367 h[longer][
i + (n -
s->num_taps[!longer]) / 2] +=
h[!longer][
i];
375 if (
s->phase != 50.f) {
389 for (
i = 0;
i < n;
i++)
390 s->coeffs[
i] =
h[longer][
i];
395 s->rdft =
s->irdft =
NULL;
407 s->rdft =
s->irdft =
NULL;
418 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
419 #define OFFSET(x) offsetof(SincContext, x)
424 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
425 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
432 {
"hptaps",
"set number of taps for high-pass filter",
OFFSET(num_taps[0]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
433 {
"lptaps",
"set number of taps for low-pass filter",
OFFSET(num_taps[1]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
441 .description =
NULL_IF_CONFIG_SMALL(
"Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
443 .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.
#define AV_CH_LAYOUT_MONO
This structure describes decoded (raw) audio or video data.
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)
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)
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.
void av_rdft_calc(RDFTContext *s, FFTSample *data)
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[]
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 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)
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
AVFilterContext * src
source filter
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)
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 the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static int config_output(AVFilterLink *outlink)
static const AVOption sinc_options[]
#define FILTER_OUTPUTS(array)
static float safe_log(float x)
static int query_formats(AVFilterContext *ctx)
void av_rdft_end(RDFTContext *s)
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)
static float bessel_I_0(float x)