Go to the documentation of this file.
59 #define OFFSET(x) offsetof(AudioFIRSourceContext, x)
60 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
73 {
"nb_samples",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX,
FLAGS },
74 {
"n",
"set the number of samples per requested frame",
OFFSET(nb_samples),
AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX,
FLAGS },
106 if (!(
s->nb_taps & 1)) {
176 (*items)[(*nb_items)++] =
av_strtod(tail, &tail);
177 new_items =
av_fast_realloc(*items, items_size, (*nb_items + 1) *
sizeof(
float));
183 }
while (tail && *tail);
190 const float *magnitude,
194 for (
int i = 0;
i < minterp;
i++) {
195 for (
int j = 1; j < m; j++) {
196 const float x =
i / (float)minterp;
199 const float mg = (
x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
200 const float ph = (
x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
214 float overlap, scale = 1.f, compensation;
215 int fft_size, middle,
ret;
217 s->nb_freq =
s->nb_magnitude =
s->nb_phase = 0;
231 if (
s->nb_freq !=
s->nb_magnitude &&
s->nb_freq !=
s->nb_phase &&
s->nb_freq >= 2) {
236 for (
int i = 0;
i <
s->nb_freq;
i++) {
237 if (
i == 0 &&
s->freq[
i] != 0.f) {
242 if (
i ==
s->nb_freq - 1 &&
s->freq[
i] != 1.f) {
247 if (
i &&
s->freq[
i] <
s->freq[
i-1]) {
253 fft_size = 1 << (
av_log2(
s->nb_taps) + 1);
254 s->complexf =
av_calloc(fft_size * 2,
sizeof(*
s->complexf));
272 lininterp(
s->complexf,
s->freq,
s->magnitude,
s->phase,
s->nb_freq, fft_size / 2);
274 s->tx_fn(
s->tx_ctx,
s->complexf + fft_size,
s->complexf,
sizeof(
float));
276 compensation = 2.f / fft_size;
277 middle =
s->nb_taps / 2;
279 for (
int i = 0;
i <= middle;
i++) {
280 s->taps[
i] =
s->complexf[fft_size + middle -
i].re * compensation *
s->win[
i];
281 s->taps[middle +
i] =
s->complexf[fft_size +
i].re * compensation *
s->win[middle +
i];
296 nb_samples =
FFMIN(
s->nb_samples,
s->nb_taps -
s->pts);
303 memcpy(
frame->data[0],
s->taps +
s->pts, nb_samples *
sizeof(
float));
306 s->pts += nb_samples;
329 .priv_class = &afirsrc_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.
#define AV_LOG_WARNING
Something somehow does not look correct.
static void lininterp(AVComplexFloat *complexf, const float *freq, const float *magnitude, const float *phase, int m, int minterp)
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 const AVOption afirsrc_options[]
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
This structure describes decoded (raw) audio or video data.
static av_cold void uninit(AVFilterContext *ctx)
const char * name
Filter name.
A link between two filters.
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 Currently power of two lengths from 2 to ...
AVComplexFloat * complexf
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
A filter pad used for either input or output.
static int request_frame(AVFilterLink *outlink)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
void * av_fast_realloc(void *ptr, unsigned int *size, size_t min_size)
Reallocate the given buffer if it is not large enough, otherwise do nothing.
static int parse_string(char *str, float **items, int *nb_items, int *items_size)
static const AVFilterPad outputs[]
@ AV_TX_FLOAT_FFT
Standard complex to complex FFT with sample data type AVComplexFloat.
Describe the class of an AVClass context structure.
static void generate_window_func(float *lut, int N, int win_func, float *overlap)
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
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
AVFILTER_DEFINE_CLASS(afirsrc)
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.
#define i(width, name, range_min, range_max)
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
static av_cold int query_formats(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 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
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static av_cold int init(AVFilterContext *ctx)
static const AVFilterPad afirsrc_outputs[]
static av_cold int config_output(AVFilterLink *outlink)