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)
#define FFERROR_NOT_READY
Filters implementation helper functions.
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.
#define i(width, name, range_min, range_max)
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)
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
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 inputs[]
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[]
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)\", 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
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
AVFILTER_DEFINE_CLASS(superequalizer)
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
AVFilterContext * dst
dest filter
static av_cold void uninit(AVFilterContext *ctx)
static enum AVSampleFormat sample_fmts[]
static int config_input(AVFilterLink *inlink)
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 av_cold int init(AVFilterContext *ctx)