46 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
47 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
71 double *
w,
double level_in,
double level_out)
73 const double a0 = bq->
a0;
74 const double a1 = bq->
a1;
75 const double a2 = bq->
a2;
76 const double b1 = bq->
b1;
77 const double b2 = bq->
b2;
81 for (
int i = 0;
i < nb_samples;
i++) {
82 double n =
src[
i] * level_in;
83 double tmp = n - w1 *
b1 - w2 *
b2;
89 dst[
i] =
out * level_out;
103 const double level_out =
s->level_out;
104 const double level_in =
s->level_in;
108 const int start = (
in->
channels * jobnr) / nb_jobs;
109 const int end = (
in->
channels * (jobnr+1)) / nb_jobs;
111 for (
int ch = start; ch < end; ch++) {
112 const double *
src = (
const double *)
in->extended_data[ch];
113 double *
w = (
double *)
s->w->extended_data[ch];
114 double *dst = (
double *)
out->extended_data[ch];
116 if (
s->rc.use_brickw) {
186 double A = sqrt(peak);
187 double w0 = freq * 2 *
M_PI / sr;
188 double alpha = sin(w0) / (2 * q);
189 double cw0 = cos(w0);
191 double b0 = 0, ib0 = 0;
193 bq->
a0 =
A*( (
A+1) + (
A-1)*cw0 +
tmp);
194 bq->
a1 = -2*
A*( (
A-1) + (
A+1)*cw0);
195 bq->
a2 =
A*( (
A+1) + (
A-1)*cw0 -
tmp);
197 bq->
b1 = 2*( (
A-1) - (
A+1)*cw0);
198 bq->
b2 = (
A+1) - (
A-1)*cw0 -
tmp;
210 double omega = 2.0 *
M_PI *
fc / sr;
211 double sn = sin(omega);
212 double cs = cos(omega);
213 double alpha = sn/(2 * q);
214 double inv = 1.0/(1.0 +
alpha);
216 bq->
a2 = bq->
a0 = gain * inv * (1.0 - cs) * 0.5;
218 bq->
b1 = (-2.0 * cs * inv);
219 bq->
b2 = ((1.0 -
alpha) * inv);
226 freq *= 2.0 *
M_PI / sr;
231 return hypot(
c->a0 +
c->a1*zr +
c->a2*(zr*zr-zi*zi),
c->a1*zi + 2*
c->a2*zr*zi) /
232 hypot(1 +
c->b1*zr +
c->b2*(zr*zr-zi*zi),
c->b1*zi + 2*
c->b2*zr*zi);
237 double i, j, k,
g, t,
a0,
a1,
a2,
b1,
b2, tau1, tau2, tau3;
238 double cutfreq, gain1kHz, gc, sr = inlink->
sample_rate;
269 i = 1. / (2. *
M_PI * tau1);
270 j = 1. / (2. *
M_PI * tau2);
271 k = 1. / (2. *
M_PI * tau3);
277 i = 1. / (2. *
M_PI * tau1);
278 j = 1. / (2. *
M_PI * tau2);
279 k = 1. / (2. *
M_PI * tau3);
285 i = 1. / (2. *
M_PI * tau1);
286 j = 1. / (2. *
M_PI * tau2);
287 k = 1. / (2. *
M_PI * tau3);
293 i = 1. / (2. *
M_PI * tau1);
294 j = 1. / (2. *
M_PI * tau2);
295 k = 1. / (2. *
M_PI * tau3);
306 if (
s->type == 7 ||
s->type == 8) {
307 double tau = (
s->type == 7 ? 0.000050 : 0.000075);
308 double f = 1.0 / (2 *
M_PI * tau);
309 double nyq = sr * 0.5;
310 double gain = sqrt(1.0 + nyq * nyq / (
f *
f));
311 double cfreq = sqrt((gain - 1.0) *
f *
f);
315 q = pow((sr / 3269.0) + 19.5, -0.25);
317 q = pow((sr / 4750.0) + 19.5, -0.25);
322 s->rc.use_brickw = 0;
324 s->rc.use_brickw = 1;
326 g = 1. / (4.+2.*
i*t+2.*k*t+
i*k*t*t);
329 a2 = (-2.*t+j*t*t)*
g;
330 b1 = (-8.+2.*
i*k*t*t)*
g;
331 b2 = (4.-2.*
i*t-2.*k*t+
i*k*t*t)*
g;
333 g = 1. / (2.*t+j*t*t);
334 a0 = (4.+2.*
i*t+2.*k*t+
i*k*t*t)*
g;
335 a1 = (-8.+2.*
i*k*t*t)*
g;
336 a2 = (4.-2.*
i*t-2.*k*t+
i*k*t*t)*
g;
338 b2 = (-2.*t+j*t*t)*
g;
350 gain1kHz =
freq_gain(&coeffs, 1000.0, sr);
353 s->rc.r1.a0 = coeffs.
a0 * gc;
354 s->rc.r1.a1 = coeffs.
a1 * gc;
355 s->rc.r1.a2 = coeffs.
a2 * gc;
356 s->rc.r1.b1 = coeffs.
b1;
357 s->rc.r1.b2 = coeffs.
b2;
360 cutfreq =
FFMIN(0.45 * sr, 21000.);
367 char *res,
int res_len,
int flags)
407 .priv_class = &aemphasis_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
AVFILTER_DEFINE_CLASS(aemphasis)
static int query_formats(AVFilterContext *ctx)
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
static int config_input(AVFilterLink *inlink)
static const AVFilterPad avfilter_af_aemphasis_inputs[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static const AVFilterPad avfilter_af_aemphasis_outputs[]
static av_cold void uninit(AVFilterContext *ctx)
static double freq_gain(BiquadCoeffs *c, double freq, double sr)
static void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples, double *w, double level_in, double level_out)
static void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
static void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
static const AVOption aemphasis_options[]
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.
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Main libavfilter public API header.
#define flags(name, subs,...)
#define fc(width, name, range_min, range_max)
mode
Use these values in ebur128_init (or'ed).
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
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.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_DBLP
double, planar
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.
static av_const double hypot(double x, double y)
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.
AVFormatInternal * internal
An opaque field for libavformat internal usage.
This structure describes decoded (raw) audio or video data.
Used for passing data between threads.
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)