51 #define OFFSET(x) offsetof(AExciterContext, x)
52 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
68 static inline double M(
double x)
70 return (
fabs(x) > 0.00000001) ? x : 0.0;
73 static inline double D(
double x)
77 return (x > 0.00000001) ? sqrt(x) : 0.0;
81 double blend,
double drive,
82 double srate,
double freq,
88 p->
rbdr = p->
rdrive / (10.5 - blend) * 780.0 / 33.0;
90 p->
kpb = (2.0 - p->
kpa) / 2.0;
94 p->
srct = (0.1 * srate) / (0.1 * srate + 1.0);
99 p->
imr = 2.0 * p->
knb +
D(2.0 * p->
kna + 4.0 * p->
an - 1.0);
100 p->
pwrq = 2.0 / (p->
imr + 1.0);
102 w0 = 2 *
M_PI * freq / srate;
103 alpha = sin(w0) / (2. * 0.707);
107 b0 = (1 + cos(w0)) / 2;
109 b2 = (1 + cos(w0)) / 2;
118 alpha = sin(w0) / (2. * 0.707);
122 b0 = (1 - cos(w0)) / 2;
124 b2 = (1 - cos(w0)) / 2;
134 double *w1,
double *w2)
136 double out =
c[2] *
in + *w1;
138 *w1 =
c[3] *
in + *w2 +
c[0] *
out;
146 double proc =
in, med;
152 med = (
D(p->
ap + proc * (p->
kpa - proc)) + p->
kpb) * p->
pwrq;
154 med = (
D(p->
an - proc * (p->
kna + proc)) + p->
knb) * p->
pwrq * -1.0;
164 if (
s->ceil >= 10000.) {
178 const double *
src = (
const double *)
in->data[0];
179 const double level_in =
s->level_in;
180 const double level_out =
s->level_out;
181 const double amount =
s->amount;
182 const double listen = 1.0 -
s->listen;
196 dst = (
double *)
out->data[0];
197 for (
int n = 0; n <
in->nb_samples; n++) {
205 if (
ctx->is_disabled)
276 char *res,
int res_len,
int flags)
310 .priv_class = &aexciter_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static double distortion_process(AExciterContext *s, ChannelParams *p, double in)
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
static const AVFilterPad avfilter_af_aexciter_inputs[]
AVFILTER_DEFINE_CLASS(aexciter)
static double D(double x)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static void set_params(ChannelParams *p, double blend, double drive, double srate, double freq, double ceil)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static av_cold void uninit(AVFilterContext *ctx)
static const AVOption aexciter_options[]
static double M(double x)
static double bprocess(double in, const double *const c, double *w1, double *w2)
static const AVFilterPad avfilter_af_aexciter_outputs[]
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
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.
Main libavfilter public API header.
#define flags(name, subs,...)
static __device__ float ceil(float a)
static __device__ float fabs(float a)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_DBL
double
static const int16_t alpha[]
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int channels
Number of channels.
int sample_rate
samples per second
AVFilterContext * dst
dest filter
A filter pad used for either input or output.
const char * name
Pad name.
const char * name
Filter name.
This structure describes decoded (raw) audio or video data.
sample data coding information
static double b1(void *priv, double x, double y)
static double b2(void *priv, double x, double y)
static double b0(void *priv, double x, double y)