Go to the documentation of this file.
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;
121 for (
i = 0;
i <= m / 2;
i++) {
122 float z =
i - .5f * m, x = z *
M_PI, y = z * mult1;
123 h[
i] = x ?
sinf(Fc * x) / x : Fc;
131 for (
i = 0; dc_norm &&
i < num_taps;
i++)
140 static const float coefs[][4] = {
141 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
142 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
143 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
144 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
145 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
146 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
147 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
148 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
149 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
150 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
152 float realm = logf(tr_bw / .0005
f) / logf(2.
f);
155 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
156 float b1 = ((
c1[0] * att +
c1[1]) * att +
c1[2]) * att +
c1[3];
158 return b0 + (
b1 -
b0) * (realm - (
int)realm);
161 return .1102f * (att - 8.7f);
163 return .58417f *
powf(att - 20.96
f, .4
f) + .07886f * (att - 20.96f);
167 static void kaiser_params(
float att,
float Fc,
float tr_bw,
float *beta,
int *num_taps)
169 *beta = *beta < 0.f ?
kaiser_beta(att, tr_bw * .5
f / Fc): *beta;
170 att = att < 60.f ? (att - 7.95f) / (2.285
f *
M_PI * 2.
f) :
171 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022
f) * *beta + .06186902f;
172 *num_taps = !*num_taps ?
ceilf(att/tr_bw + 1) : *num_taps;
175 static float *
lpf(
float Fn,
float Fc,
float tbw,
int *num_taps,
float att,
float *beta,
int round)
179 if ((Fc /= Fn) <= 0.
f || Fc >= 1.
f) {
184 att = att ? att : 120.f;
190 *num_taps =
av_clip(n, 11, 32767);
192 *num_taps = 1 + 2 * (
int)((
int)((*num_taps / 2) * Fc + .5
f) / Fc + .5f);
195 return make_lpf(*num_taps |= 1, Fc, *beta, 0.
f, 1.
f, 0);
200 for (
int i = 0;
i < n;
i++)
206 #define SQR(a) ((a) * (a))
218 float *pi_wraps, *
work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.
f;
219 int i, work_len, begin, end, imp_peak = 0, peak = 0;
220 float imp_sum = 0, peak_imp_sum = 0,
scale = 1.f;
221 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
223 for (
i = *
len, work_len = 2 * 2 * 8;
i > 1; work_len <<= 1, i >>= 1);
226 work =
av_calloc((work_len + 2) + (work_len / 2 + 1),
sizeof(
float));
229 pi_wraps = &
work[work_len + 2];
237 if (!
s->tx || !
s->itx) {
244 for (
i = 0;
i <= work_len;
i += 2) {
246 float detect = 2 *
M_PI;
247 float delta = angle - prev_angle2;
254 delta = angle - prev_angle1;
258 pi_wraps[
i >> 1] = cum_1pi;
266 for (
i = 0;
i < work_len;
i++)
267 work[
i] *= 2.
f / work_len;
269 for (
i = 1;
i < work_len / 2;
i++) {
271 work[
i + work_len / 2] = 0;
275 for (
i = 2;
i < work_len;
i += 2)
276 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];
280 for (
i = 2;
i < work_len;
i += 2) {
288 for (
i = 0;
i < work_len;
i++)
289 work[
i] *= 2.
f / work_len;
292 for (
i = 0;
i <= (
int) (pi_wraps[work_len >> 1] /
M_PI + .5
f);
i++) {
294 if (
fabs(imp_sum) >
fabs(peak_imp_sum)) {
295 peak_imp_sum = imp_sum;
308 }
else if (phase1 == 1) {
309 begin = peak - *
len / 2;
311 begin = (.997f - (2 - phase1) * .22
f) * *
len + .5f;
312 end = (.997f + (0 - phase1) * .22
f) * *
len + .5f;
313 begin = peak - (begin & ~3);
314 end = peak + 1 + ((end + 3) & ~3);
323 for (
i = 0;
i < *
len;
i++) {
324 (*h)[
i] =
work[(begin + (phase > 50.f ? *
len - 1 -
i :
i) + work_len) & (work_len - 1)];
326 *post_len = phase > 50 ? peak - begin : begin + *
len - (peak + 1);
329 work_len, pi_wraps[work_len >> 1] /
M_PI, peak, peak_imp_sum, imp_peak,
330 work[imp_peak], *
len, *post_len, 100.
f - 100.
f * *post_len / (*
len - 1));
341 float Fn =
s->sample_rate * .5f;
343 int i, n, post_peak, longer;
348 if (
s->Fc0 >= Fn ||
s->Fc1 >= Fn) {
350 "filter frequency must be less than %d/2.\n",
s->sample_rate);
354 h[0] =
lpf(Fn,
s->Fc0,
s->tbw0, &
s->num_taps[0],
s->att, &
s->beta,
s->round);
355 h[1] =
lpf(Fn,
s->Fc1,
s->tbw1, &
s->num_taps[1],
s->att, &
s->beta,
s->round);
360 longer =
s->num_taps[1] >
s->num_taps[0];
361 n =
s->num_taps[longer];
364 for (
i = 0;
i <
s->num_taps[!longer];
i++)
365 h[longer][
i + (n -
s->num_taps[!longer]) / 2] +=
h[!longer][
i];
373 if (
s->phase != 50.f) {
387 for (
i = 0;
i < n;
i++)
388 s->coeffs[
i] =
h[longer][
i];
414 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
415 #define OFFSET(x) offsetof(SincContext, x)
420 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
421 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
428 {
"hptaps",
"set number of taps for high-pass filter",
OFFSET(num_taps[0]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
429 {
"lptaps",
"set number of taps for low-pass filter",
OFFSET(num_taps[1]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
437 .description =
NULL_IF_CONFIG_SMALL(
"Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
439 .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.
This structure describes decoded (raw) audio or video data.
static float kaiser_beta(float att, float tr_bw)
#define FILTER_QUERY_FUNC(func)
#define AV_CHANNEL_LAYOUT_MONO
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...
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
Performs an in-place transformation on the input.
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)
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)
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
static const AVOption sinc_options[]
#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)
static float bessel_I_0(float x)