Go to the documentation of this file.
50 const float *coeffs =
s->coeffs;
54 nb_samples =
FFMIN(
s->nb_samples,
s->n -
s->pts);
61 memcpy(
frame->data[0], coeffs +
s->pts, nb_samples *
sizeof(
float));
102 float term = 1, sum = 1, last_sum, x2 =
x / 2;
109 sum += term *= y * y;
110 }
while (sum != last_sum);
115 static float *
make_lpf(
int num_taps,
float Fc,
float beta,
float rho,
116 float scale,
int dc_norm)
118 int i, m = num_taps - 1;
119 float *
h =
av_calloc(num_taps,
sizeof(*
h)), sum = 0;
120 float mult = scale /
bessel_I_0(beta), mult1 = 1.f / (.5f * m + rho);
124 for (
i = 0;
i <= m / 2;
i++) {
125 float z =
i - .5f * m,
x = z *
M_PI, y = z * mult1;
134 for (
i = 0; dc_norm &&
i < num_taps;
i++)
143 static const float coefs[][4] = {
144 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
145 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
146 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
147 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
148 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
149 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
150 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
151 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
152 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
153 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
155 float realm = logf(tr_bw / .0005
f) / logf(2.
f);
156 float const *c0 = coefs[av_clip((
int)realm, 0,
FF_ARRAY_ELEMS(coefs) - 1)];
157 float const *
c1 = coefs[av_clip(1 + (
int)realm, 0,
FF_ARRAY_ELEMS(coefs) - 1)];
158 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
159 float b1 = ((
c1[0] * att +
c1[1]) * att +
c1[2]) * att +
c1[3];
161 return b0 + (
b1 -
b0) * (realm - (
int)realm);
164 return .1102f * (att - 8.7f);
166 return .58417f *
powf(att - 20.96
f, .4
f) + .07886f * (att - 20.96f);
170 static void kaiser_params(
float att,
float Fc,
float tr_bw,
float *beta,
int *num_taps)
172 *beta = *beta < 0.f ?
kaiser_beta(att, tr_bw * .5
f / Fc): *beta;
173 att = att < 60.f ? (att - 7.95f) / (2.285
f *
M_PI * 2.
f) :
174 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022
f) * *beta + .06186902f;
175 *num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
178 static float *
lpf(
float Fn,
float Fc,
float tbw,
int *num_taps,
float att,
float *beta,
int round)
182 if ((Fc /= Fn) <= 0.
f || Fc >= 1.
f) {
187 att = att ? att : 120.f;
193 *num_taps = av_clip(n, 11, 32767);
195 *num_taps = 1 + 2 * (
int)((
int)((*num_taps / 2) * Fc + .5
f) / Fc + .5f);
198 return make_lpf(*num_taps |= 1, Fc, *beta, 0.
f, 1.
f, 0);
203 for (
int i = 0;
i < n;
i++)
209 #define PACK(h, n) h[1] = h[n]
210 #define UNPACK(h, n) h[n] = h[1], h[n + 1] = h[1] = 0;
211 #define SQR(a) ((a) * (a))
223 float *pi_wraps, *
work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.
f;
224 int i, work_len, begin,
end, imp_peak = 0, peak = 0;
225 float imp_sum = 0, peak_imp_sum = 0;
226 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
228 for (
i = *
len, work_len = 2 * 2 * 8;
i > 1; work_len <<= 1, i >>= 1);
231 work =
av_calloc((work_len + 2) + (work_len / 2 + 1),
sizeof(
float));
234 pi_wraps = &
work[work_len + 2];
240 s->rdft =
s->irdft =
NULL;
243 if (!
s->rdft || !
s->irdft) {
251 for (
i = 0;
i <= work_len;
i += 2) {
253 float detect = 2 *
M_PI;
254 float delta = angle - prev_angle2;
261 delta = angle - prev_angle1;
265 pi_wraps[
i >> 1] = cum_1pi;
274 for (
i = 0;
i < work_len;
i++)
275 work[
i] *= 2.
f / work_len;
277 for (
i = 1;
i < work_len / 2;
i++) {
279 work[
i + work_len / 2] = 0;
283 for (
i = 2;
i < work_len;
i += 2)
284 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];
288 for (
i = 2;
i < work_len;
i += 2) {
296 for (
i = 0;
i < work_len;
i++)
297 work[
i] *= 2.
f / work_len;
300 for (
i = 0;
i <= (
int) (pi_wraps[work_len >> 1] /
M_PI + .5
f);
i++) {
302 if (fabs(imp_sum) > fabs(peak_imp_sum)) {
303 peak_imp_sum = imp_sum;
310 while (peak && fabsf(
work[peak - 1]) > fabsf(
work[peak]) && (
work[peak - 1] *
work[peak] > 0)) {
316 }
else if (phase1 == 1) {
317 begin = peak - *
len / 2;
319 begin = (.997f - (2 - phase1) * .22
f) * *
len + .5f;
320 end = (.997f + (0 - phase1) * .22
f) * *
len + .5f;
321 begin = peak - (begin & ~3);
322 end = peak + 1 + ((
end + 3) & ~3);
331 for (
i = 0;
i < *
len;
i++) {
332 (*h)[
i] =
work[(begin + (phase > 50.f ? *
len - 1 -
i :
i) + work_len) & (work_len - 1)];
334 *post_len = phase > 50 ? peak - begin : begin + *
len - (peak + 1);
337 work_len, pi_wraps[work_len >> 1] /
M_PI, peak, peak_imp_sum, imp_peak,
338 work[imp_peak], *
len, *post_len, 100.
f - 100.
f * *post_len / (*
len - 1));
349 float Fn =
s->sample_rate * .5f;
351 int i, n, post_peak, longer;
356 if (
s->Fc0 >= Fn ||
s->Fc1 >= Fn) {
358 "filter frequency must be less than %d/2.\n",
s->sample_rate);
362 h[0] =
lpf(Fn,
s->Fc0,
s->tbw0, &
s->num_taps[0],
s->att, &
s->beta,
s->round);
363 h[1] =
lpf(Fn,
s->Fc1,
s->tbw1, &
s->num_taps[1],
s->att, &
s->beta,
s->round);
368 longer =
s->num_taps[1] >
s->num_taps[0];
369 n =
s->num_taps[longer];
372 for (
i = 0;
i <
s->num_taps[!longer];
i++)
373 h[longer][
i + (n -
s->num_taps[!longer]) / 2] +=
h[!longer][
i];
381 if (
s->phase != 50.f) {
395 for (
i = 0;
i < n;
i++)
396 s->coeffs[
i] =
h[longer][
i];
401 s->rdft =
s->irdft =
NULL;
413 s->rdft =
s->irdft =
NULL;
426 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
427 #define OFFSET(x) offsetof(SincContext, x)
432 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
433 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX,
AF },
440 {
"hptaps",
"set number of taps for high-pass filter",
OFFSET(num_taps[0]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
441 {
"lptaps",
"set number of taps for low-pass filter",
OFFSET(num_taps[1]),
AV_OPT_TYPE_INT, {.i64=0}, 0, 32768,
AF },
449 .description =
NULL_IF_CONFIG_SMALL(
"Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
451 .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.
A list of supported channel layouts.
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[]
enum MovChannelLayoutTag * layouts
#define AVERROR_EOF
End of file.
#define AV_CH_LAYOUT_MONO
static av_cold int end(AVCodecContext *avctx)
This structure describes decoded (raw) audio or video data.
static float kaiser_beta(float att, float tr_bw)
const char * name
Filter name.
A link between two filters.
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)
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
static float * make_lpf(int num_taps, float Fc, float beta, float rho, float scale, int dc_norm)
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 av_assert0(cond)
assert() equivalent, that is always enabled.
static const AVFilterPad outputs[]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
void av_rdft_calc(RDFTContext *s, FFTSample *data)
#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 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 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)
static int request_frame(AVFilterLink *outlink)
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.
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)
#define FF_ARRAY_ELEMS(a)
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static const AVOption sinc_options[]
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)
AVFILTER_DEFINE_CLASS(sinc)
static float bessel_I_0(float x)