Go to the documentation of this file.
61 #define OFFSET(x) offsetof(AudioHistogramContext, x)
62 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
125 s->shistogram =
av_calloc(
s->w,
s->dchannels *
sizeof(*
s->shistogram));
129 s->achistogram =
av_calloc(
s->w,
s->dchannels *
sizeof(*
s->achistogram));
145 s->histogram_h =
s->h *
s->phisto;
146 s->ypos =
s->h *
s->phisto;
150 if (!
s->combine_buffer)
162 const int H =
s->histogram_h;
168 if (!
s->out ||
s->out->width != outlink->
w ||
169 s->out->height != outlink->
h) {
176 for (n =
H; n <
s->h; n++) {
177 memset(
s->out->data[0] + n *
s->out->linesize[0], 0,
w);
178 memset(
s->out->data[1] + n *
s->out->linesize[0], 127,
w);
179 memset(
s->out->data[2] + n *
s->out->linesize[0], 127,
w);
180 memset(
s->out->data[3] + n *
s->out->linesize[0], 0,
w);
185 for (y = 0; y <
w; y++) {
186 s->combine_buffer[3 * y ] = 0;
187 s->combine_buffer[3 * y + 1] = 127.5;
188 s->combine_buffer[3 * y + 2] = 127.5;
192 for (n = 0; n <
H; n++) {
193 memset(
s->out->data[0] + n *
s->out->linesize[0], 0,
w);
194 memset(
s->out->data[1] + n *
s->out->linesize[0], 127,
w);
195 memset(
s->out->data[2] + n *
s->out->linesize[0], 127,
w);
196 memset(
s->out->data[3] + n *
s->out->linesize[0], 0,
w);
198 s->out->pts =
in->pts;
200 s->first =
s->frame_count;
205 const float *
src = (
const float *)
in->extended_data[
c];
206 uint64_t *achistogram = &
s->achistogram[(
s->dmode ==
SINGLE ? 0:
c) *
w];
208 for (n = 0; n <
in->nb_samples; n++) {
209 bin =
lrint(av_clipf(fabsf(
src[n]), 0, 1) * (
w - 1));
214 if (
s->in[
s->first] &&
s->count >= 0) {
215 uint64_t *shistogram = &
s->shistogram[(
s->dmode ==
SINGLE ? 0:
c) *
w];
216 const float *src2 = (
const float *)
s->in[
s->first]->extended_data[
c];
218 for (n = 0; n <
in->nb_samples; n++) {
219 bin =
lrint(av_clipf(fabsf(src2[n]), 0, 1) * (
w - 1));
228 const float *
src = (
const float *)
in->extended_data[
c];
229 uint64_t *achistogram = &
s->achistogram[(
s->dmode ==
SINGLE ? 0:
c) *
w];
231 for (n = 0; n <
in->nb_samples; n++) {
232 bin =
lrint(av_clipf(1 + log10(fabsf(
src[n])) / 6, 0, 1) * (
w - 1));
237 if (
s->in[
s->first] &&
s->count >= 0) {
238 uint64_t *shistogram = &
s->shistogram[(
s->dmode ==
SINGLE ? 0:
c) *
w];
239 const float *src2 = (
const float *)
s->in[
s->first]->extended_data[
c];
241 for (n = 0; n <
in->nb_samples; n++) {
242 bin =
lrint(av_clipf(1 + log10(fabsf(src2[n])) / 6, 0, 1) * (
w - 1));
252 s->in[
s->frame_count] =
in;
254 if (
s->frame_count >
s->count)
257 for (n = 0; n <
w *
s->dchannels; n++) {
258 acmax =
FFMAX(
s->achistogram[n] -
s->shistogram[n], acmax);
261 for (
c = 0;
c <
s->dchannels;
c++) {
262 uint64_t *shistogram = &
s->shistogram[
c *
w];
263 uint64_t *achistogram = &
s->achistogram[
c *
w];
267 yf = 256.0f /
s->dchannels;
270 uf *= 0.5 * sin((2 *
M_PI *
c) /
s->dchannels);
271 vf *= 0.5 * cos((2 *
M_PI *
c) /
s->dchannels);
274 for (n = 0; n <
w; n++) {
278 a = achistogram[n] - shistogram[n];
282 aa =
a / (double)acmax;
285 aa = sqrt(
a) / sqrt(acmax);
306 for (y =
H -
h; y <
H; y++) {
307 s->out->data[0][y *
s->out->linesize[0] + n] = 255;
308 s->out->data[3][y *
s->out->linesize[0] + n] = 255;
314 s->out->data[0][
s->ypos *
s->out->linesize[0] + n] =
h;
315 s->out->data[1][
s->ypos *
s->out->linesize[1] + n] = 127;
316 s->out->data[2][
s->ypos *
s->out->linesize[2] + n] = 127;
317 s->out->data[3][
s->ypos *
s->out->linesize[3] + n] = 255;
320 float *
out = &
s->combine_buffer[3 * n];
323 old =
s->out->data[0][(
H -
h) *
s->out->linesize[0] + n];
324 for (y =
H -
h; y <
H; y++) {
325 if (
s->out->data[0][y *
s->out->linesize[0] + n] != old)
327 old =
s->out->data[0][y *
s->out->linesize[0] + n];
328 s->out->data[0][y *
s->out->linesize[0] + n] = yf;
329 s->out->data[1][y *
s->out->linesize[1] + n] = 128+uf;
330 s->out->data[2][y *
s->out->linesize[2] + n] = 128+vf;
331 s->out->data[3][y *
s->out->linesize[3] + n] = 255;
343 for (n = 0; n <
w; n++) {
344 float *
cb = &
s->combine_buffer[3 * n];
346 s->out->data[0][
s->ypos *
s->out->linesize[0] + n] =
cb[0];
347 s->out->data[1][
s->ypos *
s->out->linesize[1] + n] =
cb[1];
348 s->out->data[2][
s->ypos *
s->out->linesize[2] + n] =
cb[2];
349 s->out->data[3][
s->ypos *
s->out->linesize[3] + n] = 255;
354 for (p = 0; p < 4; p++) {
355 for (y =
s->h; y >=
H + 1; y--) {
356 memmove(
s->out->data[p] + (y ) *
s->out->linesize[p],
357 s->out->data[p] + (y-1) *
s->out->linesize[p],
w);
363 if (
s->slide ==
SCROLL ||
s->ypos >=
s->h)
405 for (
i = 0;
i < 101;
i++)
428 .
name =
"ahistogram",
436 .priv_class = &ahistogram_class,
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
@ AV_SAMPLE_FMT_FLTP
float, planar
A list of supported channel layouts.
AVFilter ff_avf_ahistogram
AVPixelFormat
Pixel format.
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
static double cb(void *priv, double x, double y)
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
@ AV_OPT_TYPE_VIDEO_RATE
offset must point to AVRational
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
static int config_output(AVFilterLink *outlink)
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.
const char * name
Filter name.
A link between two filters.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
void * priv
private data for use by the filter
static av_cold void uninit(AVFilterContext *ctx)
static const AVFilterPad ahistogram_outputs[]
A filter pad used for either input or output.
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static const AVFilterPad outputs[]
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0,...
static enum AVPixelFormat pix_fmts[]
static int config_input(AVFilterLink *inlink)
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
Describe the class of an AVClass context structure.
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
Rational number (pair of numerator and denominator).
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
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
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 NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
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
@ AV_PIX_FMT_YUVA444P
planar YUV 4:4:4 32bpp, (1 Cr & Cb sample per 1x1 Y & A samples)
FF_FILTER_FORWARD_WANTED(outlink, inlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFILTER_DEFINE_CLASS(ahistogram)
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
#define i(width, name, range_min, range_max)
int w
agreed upon image width
#define av_malloc_array(a, b)
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
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
static const AVFilterPad ahistogram_inputs[]
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
int h
agreed upon image height
static int query_formats(AVFilterContext *ctx)
FF_FILTER_FORWARD_STATUS(inlink, outlink)
static const AVOption ahistogram_options[]
static int activate(AVFilterContext *ctx)