84 for (
int i = 0;
i < n / 2;
i++)
85 q[
i] = 1. / (-2. * cos(
M_PI * (2. * (
i + 1) + n - 1.) / (2. * n)));
93 double K = tan(
M_PI * w0);
107 double omega = 2. * tan(
M_PI * w0);
109 coeffs->
b0 = 2. / (2. + omega);
110 coeffs->
b1 = -coeffs->
b0;
112 coeffs->
a1 = -(omega - 2.) / (2. + omega);
118 const int idx =
b - (s->
order & 1);
119 double norm = 1.0 / (1.0 + K / q[idx] + K * K);
122 coeffs->
b1 = -2.0 * coeffs->
b0;
123 coeffs->
b2 = coeffs->
b0;
124 coeffs->
a1 = -2.0 * (K * K - 1.0) * norm;
125 coeffs->
a2 = -(1.0 - K / q[idx] + K * K) * norm;
127 }
else if (!strcmp(ctx->
filter->
name,
"asupercut")) {
134 double omega = 2. * tan(
M_PI * w0);
136 coeffs->
b0 = omega / (2. + omega);
137 coeffs->
b1 = coeffs->
b0;
139 coeffs->
a1 = -(omega - 2.) / (2. + omega);
145 const int idx =
b - (s->
order & 1);
146 double norm = 1.0 / (1.0 + K / q[idx] + K * K);
148 coeffs->
b0 = K * K * norm;
149 coeffs->
b1 = 2.0 * coeffs->
b0;
150 coeffs->
b2 = coeffs->
b0;
151 coeffs->
a1 = -2.0 * (K * K - 1.0) * norm;
152 coeffs->
a2 = -(1.0 - K / q[idx] + K * K) * norm;
154 }
else if (!strcmp(ctx->
filter->
name,
"asuperpass")) {
155 double alpha, beta, gamma, theta;
160 d_E = (2. * tan(theta_0 / (2. * s->
qfactor))) / sin(theta_0);
164 double A = (1. + pow((d_E / 2.), 2)) / (D * d_E / 2.);
165 double d = sqrt((d_E * D) / (A + sqrt(A * A - 1.)));
166 double B = D * (d_E / 2.) / d;
167 double W = B + sqrt(B * B - 1.);
169 for (
int j = 0; j < 2; j++) {
173 theta = 2. * atan(tan(theta_0 / 2.) / W);
175 theta = 2. * atan(W * tan(theta_0 / 2.));
177 beta = 0.5 * ((1. - (d / 2.) * sin(theta)) / (1. + (d / 2.) * sin(theta)));
178 gamma = (0.5 + beta) * cos(theta);
179 alpha = 0.5 * (0.5 - beta) * sqrt(1. + pow((W - (1. / W)) / d, 2.));
181 coeffs->
a1 = 2. * gamma;
182 coeffs->
a2 = -2. * beta;
188 }
else if (!strcmp(ctx->
filter->
name,
"asuperstop")) {
189 double alpha, beta, gamma, theta;
194 d_E = (2. * tan(theta_0 / (2. * s->
qfactor))) / sin(theta_0);
198 double A = (1. + pow((d_E / 2.), 2)) / (D * d_E / 2.);
199 double d = sqrt((d_E * D) / (A + sqrt(A * A - 1.)));
200 double B = D * (d_E / 2.) / d;
201 double W = B + sqrt(B * B - 1.);
203 for (
int j = 0; j < 2; j++) {
207 theta = 2. * atan(tan(theta_0 / 2.) / W);
209 theta = 2. * atan(W * tan(theta_0 / 2.));
211 beta = 0.5 * ((1. - (d / 2.) * sin(theta)) / (1. + (d / 2.) * sin(theta)));
212 gamma = (0.5 + beta) * cos(theta);
213 alpha = 0.5 * (0.5 + beta) * ((1. - cos(theta)) / (1. - cos(theta_0)));
215 coeffs->
a1 = 2. * gamma;
216 coeffs->
a2 = -2. * beta;
218 coeffs->
b1 = -4. * alpha * cos(theta_0);
231 #define FILTER(name, type) \ 232 static int filter_channels_## name(AVFilterContext *ctx, void *arg, \ 233 int jobnr, int nb_jobs) \ 235 ASuperCutContext *s = ctx->priv; \ 236 ThreadData *td = arg; \ 237 AVFrame *out = td->out; \ 238 AVFrame *in = td->in; \ 239 const int start = (in->channels * jobnr) / nb_jobs; \ 240 const int end = (in->channels * (jobnr+1)) / nb_jobs; \ 241 const double level = s->level; \ 243 for (int ch = start; ch < end; ch++) { \ 244 const type *src = (const type *)in->extended_data[ch]; \ 245 type *dst = (type *)out->extended_data[ch]; \ 247 for (int b = 0; b < s->filter_count; b++) { \ 248 BiquadCoeffs *coeffs = &s->coeffs[b]; \ 249 const type a1 = coeffs->a1; \ 250 const type a2 = coeffs->a2; \ 251 const type b0 = coeffs->b0; \ 252 const type b1 = coeffs->b1; \ 253 const type b2 = coeffs->b2; \ 254 type *w = ((type *)s->w->extended_data[ch]) + b * 2; \ 256 for (int n = 0; n < in->nb_samples; n++) { \ 257 type sin = b ? dst[n] : src[n] * level; \ 258 type sout = sin * b0 + w[0]; \ 260 w[0] = b1 * sin + w[1] + a1 * sout; \ 261 w[1] = b2 * sin + a2 * sout; \ 279 switch (inlink->format) {
323 char *res,
int res_len,
int flags)
341 #define OFFSET(x) offsetof(ASuperCutContext, x) 342 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM 376 .priv_class = &asupercut_class,
399 .priv_class = &asubcut_class,
416 #define asuperpass_options asuperpass_asuperstop_options 420 .
name =
"asuperpass",
424 .priv_class = &asuperpass_class,
433 #define asuperstop_options asuperpass_asuperstop_options 437 .
name =
"asuperstop",
441 .priv_class = &asuperstop_class,
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
This structure describes decoded (raw) audio or video data.
static int query_formats(AVFilterContext *ctx)
Main libavfilter public API header.
static void calc_q_factors(int n, double *q)
AVFILTER_DEFINE_CLASS(asupercut)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
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 int get_coeffs(AVFilterContext *ctx)
AVFilter ff_af_asuperstop
A filter pad used for either input or output.
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.
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. ...
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options...
void * priv
private data for use by the filter
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
AVFilter ff_af_asuperpass
#define FILTER(name, type)
static av_cold void uninit(AVFilterContext *ctx)
audio channel layout utility functions
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
int(* filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
A list of supported channel layouts.
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Used for passing data between threads.
static const int16_t alpha[]
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.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static const AVFilterPad inputs[]
static const AVOption asubcut_options[]
#define flags(name, subs,...)
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
static const AVOption asuperpass_asuperstop_options[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
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
internal math functions header
int channels
Number of channels.
avfilter_execute_func * execute
static int config_input(AVFilterLink *inlink)
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad outputs[]
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 int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVOption asupercut_options[]
int nb_samples
number of audio samples (per channel) described by this frame
const AVFilter * filter
the AVFilter of which this is an instance
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.