40 #define MAX_BANDS MAX_SPLITS + 1 84 #define OFFSET(x) offsetof(AudioCrossoverContext, x) 85 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM 110 char *p, *
arg, *saveptr =
NULL;
119 if (!(arg =
av_strtok(p,
" |", &saveptr)))
124 if (
av_sscanf(arg,
"%f%2s", &gain, c) < 1) {
130 if (c[0] ==
'd' && c[1] ==
'B')
145 char *p, *
arg, *saveptr =
NULL;
156 if (!(arg =
av_strtok(p,
" |", &saveptr)))
170 if (i > 0 && freq <= s->splits[i-1]) {
205 double omega = 2. *
M_PI * fc / sr;
206 double cosine = cos(omega);
207 double alpha = sin(omega) / (2. * q);
209 double b0 = (1. - cosine) / 2.;
210 double b1 = 1. - cosine;
211 double b2 = (1. - cosine) / 2.;
213 double a1 = -2. * cosine;
231 double omega = 2. *
M_PI * fc / sr;
232 double cosine = cos(omega);
233 double alpha = sin(omega) / (2. * q);
235 double b0 = (1. + cosine) / 2.;
236 double b1 = -1. - cosine;
237 double b2 = (1. + cosine) / 2.;
239 double a1 = -2. * cosine;
257 double omega = 2. *
M_PI * fc / sr;
258 double cosine = cos(omega);
259 double alpha = sin(omega) / (2. * q);
262 double a1 = -2. * cosine;
283 double omega = 2. *
M_PI * fc / sr;
300 double n = order / 2.;
302 for (
int i = 0;
i < n / 2;
i++)
303 q[
i] = 1. / (-2. * cos(
M_PI * (2. * (
i + 1) + n - 1.) / (2. * n)));
336 #define BIQUAD_PROCESS(name, type) \ 337 static void biquad_process_## name(const type *const c, \ 339 type *dst, const type *src, \ 342 const type b0 = c[B0]; \ 343 const type b1 = c[B1]; \ 344 const type b2 = c[B2]; \ 345 const type a1 = c[A1]; \ 346 const type a2 = c[A2]; \ 350 for (int n = 0; n + 1 < nb_samples; n++) { \ 354 out = in * b0 + z1; \ 355 z1 = b1 * in + z2 + a1 * out; \ 356 z2 = b2 * in + a2 * out; \ 361 out = in * b0 + z1; \ 362 z1 = b1 * in + z2 + a1 * out; \ 363 z2 = b2 * in + a2 * out; \ 367 if (nb_samples & 1) { \ 368 const int n = nb_samples - 1; \ 369 const type in = src[n]; \ 372 out = in * b0 + z1; \ 373 z1 = b1 * in + z2 + a1 * out; \ 374 z2 = b2 * in + a2 * out; \ 385 #define XOVER_PROCESS(name, type, one, ff) \ 386 static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \ 388 AudioCrossoverContext *s = ctx->priv; \ 389 AVFrame *in = s->input_frame; \ 390 AVFrame **frames = s->frames; \ 391 const int start = (in->channels * jobnr) / nb_jobs; \ 392 const int end = (in->channels * (jobnr+1)) / nb_jobs; \ 393 const int nb_samples = in->nb_samples; \ 394 const int nb_outs = ctx->nb_outputs; \ 395 const int first_order = s->first_order; \ 397 for (int ch = start; ch < end; ch++) { \ 398 const type *src = (const type *)in->extended_data[ch]; \ 399 type *xover = (type *)s->xover->extended_data[ch]; \ 401 s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \ 402 s->level_in, FFALIGN(nb_samples, sizeof(type))); \ 404 for (int band = 0; band < nb_outs; band++) { \ 405 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \ 406 const type *prv = (const type *)frames[band]->extended_data[ch]; \ 407 type *dst = (type *)frames[band + 1]->extended_data[ch]; \ 408 const type *hsrc = f == 0 ? prv : dst; \ 409 type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \ 410 const type *const hpc = (type *)&s->hp[band][f].c ## ff; \ 412 biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \ 415 for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \ 416 type *dst = (type *)frames[band]->extended_data[ch]; \ 417 const type *lsrc = dst; \ 418 type *lp = xover + band * 20 + f * 2; \ 419 const type *const lpc = (type *)&s->lp[band][f].c ## ff; \ 421 biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \ 424 for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \ 426 const type *asrc = (const type *)frames[band]->extended_data[ch]; \ 427 type *dst = (type *)frames[band]->extended_data[ch]; \ 428 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \ 429 const type *const apc = (type *)&s->ap[aband][0].c ## ff; \ 431 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \ 434 for (int f = first_order; f < s->ap_filter_count; f++) { \ 435 const type *asrc = (const type *)frames[band]->extended_data[ch]; \ 436 type *dst = (type *)frames[band]->extended_data[ch]; \ 437 type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\ 438 const type *const apc = (type *)&s->ap[aband][f].c ## ff; \ 440 biquad_process_## name(apc, ap, dst, asrc, nb_samples); \ 445 for (int band = 0; band < nb_outs; band++) { \ 446 const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \ 447 type *dst = (type *)frames[band]->extended_data[ch]; \ 449 s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \ 450 FFALIGN(nb_samples, sizeof(type))); \ 473 for (
int band = 0; band <= s->
nb_splits; band++) {
479 for (
int n = s->
first_order; n < s->filter_count; n++) {
489 for (
int n = s->
first_order; n < s->ap_filter_count; n++) {
496 switch (inlink->format) {
573 .
name =
"acrossover",
576 .priv_class = &acrossover_class,
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
This structure describes decoded (raw) audio or video data.
static av_cold int init(AVFilterContext *ctx)
static void set_ap1(BiquadCoeffs *b, double fc, double sr)
AVFrame * frames[MAX_BANDS]
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
Main libavfilter public API header.
BiquadCoeffs hp[MAX_BANDS][20]
enum AVMediaType type
AVFilterPad type.
if it could not because there are no more frames
Macro definitions for various function/variable attributes.
const char * name
Pad name.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
AVFilterPad * output_pads
array of output pads
int(* filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
A filter pad used for either input or output.
A link between two filters.
#define fc(width, name, range_min, range_max)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
BiquadCoeffs ap[MAX_BANDS][20]
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
unsigned nb_outputs
number of output pads
static int parse_gains(AVFilterContext *ctx)
#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
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
#define BIQUAD_PROCESS(name, type)
static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
char * av_asprintf(const char *fmt,...)
common internal API header
audio channel layout utility functions
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad outputs[]
A list of supported channel layouts.
AVSampleFormat
Audio sample formats.
static const int16_t alpha[]
AVFILTER_DEFINE_CLASS(acrossover)
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.
BiquadCoeffs lp[MAX_BANDS][20]
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
#define flags(name, subs,...)
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
AVFilter ff_af_acrossover
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
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
static av_cold void uninit(AVFilterContext *ctx)
int channels
Number of channels.
avfilter_execute_func * execute
AVFilterContext * dst
dest filter
static void calc_q_factors(int order, double *q)
static enum AVSampleFormat sample_fmts[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
#define XOVER_PROCESS(name, type, one, ff)
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 acrossover_options[]
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
int nb_samples
number of audio samples (per channel) described by this frame
simple arithmetic expression evaluator