57 65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
58 1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
66 for (m = 1; m <=
M; m++) {
69 t = pow(x / 2, m) / s->
fact[m];
79 float omega = 2 *
M_PI *
f;
81 if (n * omega * t == 0)
83 return 2 * f * t *
sinf(n * omega * t) / (n * omega * t);
88 return n == 0 ? 1.f : 0.f;
97 ret = param[0].
gain*lhn;
99 for (i = 1; i <
NBANDS + 1 && param[
i].
upper < fs / 2; i++) {
100 float lhn2 =
hn_lpf(n, param[i].upper, fs);
101 ret += param[
i].
gain * (lhn2 - lhn);
115 return .5842f * pow(a - 21, 0.4
f) + 0.07886f * (a - 21);
116 return .1102f * (a - 8.7f);
121 return izero(s,
alpha(s->
aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->
iza;
128 for (i = 0; i <=
NBANDS; i++) {
145 s->
winlen = (1 << (wb-1))-1;
152 for (i = 0; i <=
M; i++) {
154 for (j = 1; j <=
i; j++)
165 const int winlen = s->
winlen;
166 const int tabsize = s->
tabsize;
174 for (i = 0; i < winlen; i++)
175 s->
irest[i] =
hn(i - winlen / 2, param, fs) *
win(s, i - winlen / 2, winlen);
176 for (; i < tabsize; i++)
181 for (i = 0; i < tabsize; i++)
182 nires[i] = s->
irest[i];
190 const float *ires = s->
ires;
195 float *
src, *dst, *ptr;
202 for (ch = 0; ch < in->
channels; ch++) {
208 fsamples[i] = src[i];
214 fsamples[0] = ires[0] * fsamples[0];
215 fsamples[1] = ires[1] * fsamples[1];
216 for (i = 1; i < s->
tabsize / 2; i++) {
219 re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
220 im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
223 fsamples[i*2+1] =
im;
228 for (i = 0; i < s->
winlen; i++)
229 dst[i] += fsamples[i] / s->
tabsize * 2;
232 for (i = 0; i < s->
winlen; i++)
234 for (i = 0; i < s->
winlen; i++)
235 dst[i] = dst[i+s->
winlen];
350 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 351 #define OFFSET(x) offsetof(SuperEqualizerContext, x) 378 .
name =
"superequalizer",
381 .priv_class = &superequalizer_class,
386 .
inputs = superequalizer_inputs,
387 .
outputs = superequalizer_outputs,
static float alpha(float a)
This structure describes decoded (raw) audio or video data.
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad superequalizer_outputs[]
static const AVOption superequalizer_options[]
Main libavfilter public API header.
static float win(SuperEqualizerContext *s, float n, int N)
FF_FILTER_FORWARD_STATUS(inlink, outlink)
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
EqParameter params[NBANDS+1]
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static float hn_imp(int n)
#define fs(width, name, subs,...)
static int equ_init(SuperEqualizerContext *s, int wb)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static float hn(int n, EqParameter *param, float fs)
static int config_output(AVFilterLink *outlink)
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
A filter pad used for either input or output.
static float hn_lpf(int n, float f, float fs)
A link between two filters.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
static const AVFilterPad superequalizer_inputs[]
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
void * priv
private data for use by the filter
static float izero(SuperEqualizerContext *s, float x)
void av_rdft_calc(RDFTContext *s, FFTSample *data)
int channels
number of audio channels, only used for audio.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
void av_rdft_end(RDFTContext *s)
AVFilterContext * src
source filter
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
static const AVFilterPad outputs[]
A list of supported channel layouts.
AVFilter ff_af_superequalizer
AVSampleFormat
Audio sample formats.
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.
static const float bands[]
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
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
Describe the class of an AVClass context structure.
const char * name
Filter name.
static void process_param(float *bc, EqParameter *param, float fs)
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
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
AVFILTER_DEFINE_CLASS(superequalizer)
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
FF_FILTER_FORWARD_WANTED(outlink, inlink)
AVFilterContext * dst
dest filter
static av_cold void uninit(AVFilterContext *ctx)
static enum AVSampleFormat sample_fmts[]
static int config_input(AVFilterLink *inlink)
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
uint8_t ** extended_data
pointers to the data planes/channels.
static int activate(AVFilterContext *ctx)
int nb_samples
number of audio samples (per channel) described by this frame
static double fact(double i)
static av_cold int init(AVFilterContext *ctx)