Go to the documentation of this file.
31 #define FILTER_ORDER 4
75 #define OFFSET(x) offsetof(AudioNEqualizerContext, x)
76 #define A AV_OPT_FLAG_AUDIO_PARAM
77 #define V AV_OPT_FLAG_VIDEO_PARAM
78 #define F AV_OPT_FLAG_FILTERING_PARAM
88 {
"colors",
"set channels curves colors",
OFFSET(colors),
AV_OPT_TYPE_STRING, {.str =
"red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0,
V|
F },
104 memset(
out->data[0], 0,
s->h *
out->linesize[0]);
107 uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
115 for (
f = 0;
f <
s->w;
f++) {
116 double zr, zi, zr2, zi2;
122 w =
M_PI * (
s->fscale ? pow(
s->w - 1,
f /
s->w) :
f) / (
s->w - 1);
128 for (n = 0; n <
s->nb_filters; n++) {
129 if (
s->filters[n].channel !=
ch ||
130 s->filters[n].ignore)
139 Hr =
S->b4*(1-8*zr2*zi2) +
S->b2*(zr2-zi2) + zr*(
S->b1+
S->b3*(zr2-3*zi2))+
S->b0;
140 Hi = zi*(
S->b3*(3*zr2-zi2) +
S->b1 + 2*zr*(2*
S->b4*(zr2-zi2) +
S->b2));
141 Hmag *=
hypot(Hr, Hi);
142 Hr =
S->a4*(1-8*zr2*zi2) +
S->a2*(zr2-zi2) + zr*(
S->a1+
S->a3*(zr2-3*zi2))+
S->a0;
143 Hi = zi*(
S->a3*(3*zr2-zi2) +
S->a1 + 2*zr*(2*
S->a4*(zr2-zi2) +
S->a2));
144 Hmag /=
hypot(Hr, Hi);
148 v = av_clip((1. + -20 * log10(Hmag) /
s->mag) *
s->h / 2, 0,
s->h - 1);
153 for (y = v; y <= prev_v; y++)
156 for (y = prev_v; y <= v; y++)
202 if (
s->draw_curves) {
220 if (
s->draw_curves) {
245 if (
s->draw_curves) {
274 for (
int i = 0;
i <
ctx->nb_outputs;
i++)
283 double si,
double g,
double g0,
286 if (c0 == 1 || c0 == -1) {
287 S->b0 = (
g*
g*beta*beta + 2*
g*g0*si*beta + g0*g0)/
D;
288 S->b1 = 2*c0*(
g*
g*beta*beta - g0*g0)/
D;
289 S->b2 = (
g*
g*beta*beta - 2*g0*
g*beta*si + g0*g0)/
D;
294 S->a1 = 2*c0*(beta*beta - 1)/
D;
295 S->a2 = (beta*beta - 2*beta*si + 1)/
D;
299 S->b0 = (
g*
g*beta*beta + 2*
g*g0*si*beta + g0*g0)/
D;
300 S->b1 = -4*c0*(g0*g0 +
g*g0*si*beta)/
D;
301 S->b2 = 2*(g0*g0*(1 + 2*c0*c0) -
g*
g*beta*beta)/
D;
302 S->b3 = -4*c0*(g0*g0 -
g*g0*si*beta)/
D;
303 S->b4 = (
g*
g*beta*beta - 2*
g*g0*si*beta + g0*g0)/
D;
306 S->a1 = -4*c0*(1 + si*beta)/
D;
307 S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/
D;
308 S->a3 = -4*c0*(1 - si*beta)/
D;
309 S->a4 = (beta*beta - 2*si*beta + 1)/
D;
314 int N,
double w0,
double wb,
315 double G,
double Gb,
double G0)
317 double g, c0, g0, beta;
323 if (
G == 0 && G0 == 0) {
324 f->section[0].a0 = 1;
325 f->section[0].b0 = 1;
326 f->section[1].a0 = 1;
327 f->section[1].b0 = 1;
335 epsilon = sqrt((
G *
G - Gb * Gb) / (Gb * Gb - G0 * G0));
337 g0 = pow(G0, 1.0 /
N);
338 beta = pow(epsilon, -1.0 /
N) * tan(wb/2);
341 for (
i = 1;
i <=
L;
i++) {
342 double ui = (2.0 *
i - 1) /
N;
343 double si = sin(
M_PI *
ui / 2.0);
344 double Di = beta * beta + 2 * si * beta + 1;
351 double c,
double tetta_b,
352 double g0,
double si,
double b,
355 if (c0 == 1 || c0 == -1) {
356 S->b0 = (tetta_b*tetta_b*(
b*
b+g0*g0*
c*
c) + 2*g0*
b*si*tetta_b*tetta_b + g0*g0)/
D;
357 S->b1 = 2*c0*(tetta_b*tetta_b*(
b*
b+g0*g0*
c*
c) - g0*g0)/
D;
358 S->b2 = (tetta_b*tetta_b*(
b*
b+g0*g0*
c*
c) - 2*g0*
b*si*tetta_b + g0*g0)/
D;
363 S->a1 = 2*c0*(tetta_b*tetta_b*(
a*
a+
c*
c) - 1)/
D;
364 S->a2 = (tetta_b*tetta_b*(
a*
a+
c*
c) - 2*
a*si*tetta_b + 1)/
D;
368 S->b0 = ((
b*
b + g0*g0*
c*
c)*tetta_b*tetta_b + 2*g0*
b*si*tetta_b + g0*g0)/
D;
369 S->b1 = -4*c0*(g0*g0 + g0*
b*si*tetta_b)/
D;
370 S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (
b*
b + g0*g0*
c*
c)*tetta_b*tetta_b)/
D;
371 S->b3 = -4*c0*(g0*g0 - g0*
b*si*tetta_b)/
D;
372 S->b4 = ((
b*
b + g0*g0*
c*
c)*tetta_b*tetta_b - 2*g0*
b*si*tetta_b + g0*g0)/
D;
375 S->a1 = -4*c0*(1 +
a*si*tetta_b)/
D;
376 S->a2 = 2*(1 + 2*c0*c0 - (
a*
a +
c*
c)*tetta_b*tetta_b)/
D;
377 S->a3 = -4*c0*(1 -
a*si*tetta_b)/
D;
378 S->a4 = ((
a*
a +
c*
c)*tetta_b*tetta_b - 2*
a*si*tetta_b + 1)/
D;
383 int N,
double w0,
double wb,
384 double G,
double Gb,
double G0)
386 double a,
b, c0, g0, alfa, beta, tetta_b;
392 if (
G == 0 && G0 == 0) {
393 f->section[0].a0 = 1;
394 f->section[0].b0 = 1;
395 f->section[1].a0 = 1;
396 f->section[1].b0 = 1;
404 epsilon = sqrt((
G*
G - Gb*Gb) / (Gb*Gb - G0*G0));
406 alfa = pow(1.0/epsilon + sqrt(1 + 1/(epsilon*epsilon)), 1.0/
N);
407 beta = pow(
G/epsilon + Gb * sqrt(1 + 1/(epsilon*epsilon)), 1.0/
N);
408 a = 0.5 * (alfa - 1.0/alfa);
409 b = 0.5 * (beta - g0*g0*(1/beta));
413 for (
i = 1;
i <=
L;
i++) {
414 double ui = (2.0*
i-1.0)/
N;
415 double ci = cos(
M_PI*
ui/2.0);
416 double si = sin(
M_PI*
ui/2.0);
417 double Di = (
a*
a + ci*ci)*tetta_b*tetta_b + 2.0*
a*si*tetta_b + 1;
424 double c,
double tetta_b,
425 double g,
double si,
double b,
428 if (c0 == 1 || c0 == -1) {
429 S->b0 = (
g*
g*tetta_b*tetta_b + 2*tetta_b*
g*
b*si +
b*
b +
g*
g*
c*
c)/
D;
430 S->b1 = 2*c0*(
g*
g*tetta_b*tetta_b -
b*
b -
g*
g*
c*
c)/
D;
431 S->b2 = (
g*
g*tetta_b*tetta_b - 2*tetta_b*
g*
b*si +
b*
b +
g*
g*
c*
c)/
D;
436 S->a1 = 2*c0*(tetta_b*tetta_b -
a*
a -
c*
c)/
D;
437 S->a2 = (tetta_b*tetta_b - 2*tetta_b*
a*si +
a*
a +
c*
c)/
D;
441 S->b0 = (
g*
g*tetta_b*tetta_b + 2*
g*
b*si*tetta_b +
b*
b +
g*
g*
c*
c)/
D;
442 S->b1 = -4*c0*(
b*
b +
g*
g*
c*
c +
g*
b*si*tetta_b)/
D;
443 S->b2 = 2*((
b*
b +
g*
g*
c*
c)*(1 + 2*c0*c0) -
g*
g*tetta_b*tetta_b)/
D;
444 S->b3 = -4*c0*(
b*
b +
g*
g*
c*
c -
g*
b*si*tetta_b)/
D;
445 S->b4 = (
g*
g*tetta_b*tetta_b - 2*
g*
b*si*tetta_b +
b*
b +
g*
g*
c*
c)/
D;
448 S->a1 = -4*c0*(
a*
a +
c*
c +
a*si*tetta_b)/
D;
449 S->a2 = 2*((
a*
a +
c*
c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/
D;
450 S->a3 = -4*c0*(
a*
a +
c*
c -
a*si*tetta_b)/
D;
451 S->a4 = (tetta_b*tetta_b - 2*
a*si*tetta_b +
a*
a +
c*
c)/
D;
456 int N,
double w0,
double wb,
457 double G,
double Gb,
double G0)
459 double a,
b, c0, tetta_b;
460 double epsilon,
g, eu, ew;
465 if (
G == 0 && G0 == 0) {
466 f->section[0].a0 = 1;
467 f->section[0].b0 = 1;
468 f->section[1].a0 = 1;
469 f->section[1].b0 = 1;
477 epsilon = sqrt((
G*
G - Gb*Gb) / (Gb*Gb - G0*G0));
479 eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/
N);
480 ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/
N);
481 a = (eu - 1.0/eu)/2.0;
482 b = (ew -
g*
g/ew)/2.0;
486 for (
i = 1;
i <=
L;
i++) {
487 double ui = (2.0 *
i - 1.0)/
N;
488 double ci = cos(
M_PI *
ui / 2.0);
489 double si = sin(
M_PI *
ui / 2.0);
490 double Di = tetta_b*tetta_b + 2*
a*si*tetta_b +
a*
a + ci*ci;
502 else if(gain > -6 && gain < 6)
503 bw_gain = gain * 0.5;
516 else if(gain > -6 && gain < 6)
517 bw_gain = gain * 0.9;
530 else if(gain > -6 && gain < 6)
531 bw_gain = gain * 0.3;
569 if (
s->nb_filters >=
s->nb_allocated - 1) {
575 memcpy(
filters,
s->filters,
sizeof(*
s->filters) *
s->nb_allocated);
578 s->nb_allocated *= 2;
590 char *saveptr =
NULL;
596 s->nb_allocated = 32 *
inlink->channels;
610 s->filters[
s->nb_filters].type = 0;
611 if (sscanf(
arg,
"c%d f=%lf w=%lf g=%lf t=%d", &
s->filters[
s->nb_filters].channel,
612 &
s->filters[
s->nb_filters].freq,
613 &
s->filters[
s->nb_filters].width,
614 &
s->filters[
s->nb_filters].gain,
615 &
s->filters[
s->nb_filters].type) != 5 &&
616 sscanf(
arg,
"c%d f=%lf w=%lf g=%lf", &
s->filters[
s->nb_filters].channel,
617 &
s->filters[
s->nb_filters].freq,
618 &
s->filters[
s->nb_filters].width,
619 &
s->filters[
s->nb_filters].gain) != 4 ) {
624 if (
s->filters[
s->nb_filters].freq < 0 ||
625 s->filters[
s->nb_filters].freq >
inlink->sample_rate / 2.0)
626 s->filters[
s->nb_filters].ignore = 1;
628 if (
s->filters[
s->nb_filters].channel < 0 ||
629 s->filters[
s->nb_filters].channel >=
inlink->channels)
630 s->filters[
s->nb_filters].ignore = 1;
632 s->filters[
s->nb_filters].type = av_clip(
s->filters[
s->nb_filters].type, 0,
NB_TYPES - 1);
644 char *res,
int res_len,
int flags)
650 if (!strcmp(cmd,
"change")) {
651 double freq,
width, gain;
654 if (sscanf(args,
"%d|f=%lf|w=%lf|g=%lf", &
filter, &freq, &
width, &gain) != 4)
657 if (filter < 0 || filter >=
s->nb_filters)
660 if (freq < 0 || freq >
inlink->sample_rate / 2.0)
663 s->filters[
filter].freq = freq;
665 s->filters[
filter].gain = gain;
681 out+=
S->b1 *
S->num[0] -
S->denum[0] *
S->a1;
682 out+=
S->b2 *
S->num[1] -
S->denum[1] *
S->a2;
683 out+=
S->b3 *
S->num[2] -
S->denum[2] *
S->a3;
684 out+=
S->b4 *
S->num[3] -
S->denum[3] *
S->a4;
686 S->num[3] =
S->num[2];
687 S->num[2] =
S->num[1];
688 S->num[1] =
S->num[0];
691 S->denum[3] =
S->denum[2];
692 S->denum[2] =
S->denum[1];
693 S->denum[1] =
S->denum[0];
720 for (
i = 0;
i <
s->nb_filters;
i++) {
723 if (
f->gain == 0. ||
f->ignore)
735 if (
s->draw_curves) {
738 const int64_t
pts = buf->
pts +
767 .
name =
"anequalizer",
770 .priv_class = &anequalizer_class,
static int config_video(AVFilterLink *outlink)
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
A list of supported channel layouts.
AVPixelFormat
Pixel format.
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
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
struct AVFilterChannelLayouts * in_channel_layouts
static void butterworth_fo_section(FoSection *S, double beta, double si, double g, double g0, double D, double c0)
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
int av_parse_color(uint8_t *rgba_color, const char *color_string, int slen, void *log_ctx)
Put the RGBA values that correspond to color_string in rgba_color.
static const struct PPFilter filters[]
AVFilterFormats * in_formats
Lists of formats and channel layouts supported by the input and output filters respectively.
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
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 void chebyshev1_bp_filter(EqualizatorFilter *f, int N, double w0, double wb, double G, double Gb, double G0)
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
const char * name
Filter name.
A link between two filters.
AVFilterFormats * in_samplerates
Lists of channel layouts and sample rates used for automatic negotiation.
static void chebyshev2_bp_filter(EqualizatorFilter *f, int N, double w0, double wb, double G, double Gb, double G0)
static const AVOption anequalizer_options[]
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
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 int config_input(AVFilterLink *inlink)
AVFILTER_DEFINE_CLASS(anequalizer)
A filter pad used for either input or output.
static void equalizer(EqualizatorFilter *f, double sample_rate)
static void chebyshev2_fo_section(FoSection *S, double a, double c, double tetta_b, double g, double si, double b, double D, double c0)
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
static void chebyshev1_fo_section(FoSection *S, double a, double c, double tetta_b, double g0, double si, double b, double D, double c0)
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad outputs[]
static enum AVPixelFormat pix_fmts[]
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
static double hz_2_rad(double x, double fs)
@ AV_PIX_FMT_RGBA
packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
static av_cold void uninit(AVFilterContext *ctx)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Describe the class of an AVClass context structure.
#define fs(width, name, subs,...)
static const AVFilterPad inputs[]
Rational number (pair of numerator and denominator).
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static av_const double hypot(double x, double y)
static const uint32_t color[16+AV_CLASS_CATEGORY_NB]
AVFilter ff_af_anequalizer
AVFilterContext * src
source filter
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
static int add_filter(AudioNEqualizerContext *s, AVFilterLink *inlink)
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
int nb_samples
number of audio samples (per channel) described by this frame
#define i(width, name, range_min, range_max)
int w
agreed upon image width
uint8_t ** extended_data
pointers to the data planes/channels.
AVSampleFormat
Audio sample formats.
static double section_process(FoSection *S, double in)
static double chebyshev1_compute_bw_gain_db(double gain)
const char * name
Pad name.
static double chebyshev2_compute_bw_gain_db(double gain)
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
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
int h
agreed upon image height
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
static void butterworth_bp_filter(EqualizatorFilter *f, int N, double w0, double wb, double G, double Gb, double G0)
@ AV_SAMPLE_FMT_DBLP
double, planar
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
char * av_strdup(const char *s)
Duplicate a string.
static double process_sample(FoSection *s1, double in)
EqualizatorFilter * filters
#define flags(name, subs,...)
static av_cold int init(AVFilterContext *ctx)
static double butterworth_compute_bw_gain_db(double gain)