48 #define OFFSET(x) offsetof(AudioEmphasisContext, x) 49 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 76 double out = tmp * bq->
a0 + bq->
w1 * bq->
a1 + bq->
w2 * bq->
a2;
89 const double *
src = (
const double *)in->
data[0];
106 dst = (
double *)
out->data[0];
152 double A = sqrt(peak);
153 double w0 = freq * 2 *
M_PI / sr;
154 double alpha = sin(w0) / (2 * q);
155 double cw0 = cos(w0);
157 double b0 = 0, ib0 = 0;
159 bq->
a0 = A*( (A+1) + (A-1)*cw0 +
tmp);
160 bq->
a1 = -2*A*( (A-1) + (A+1)*cw0);
161 bq->
a2 = A*( (A+1) + (A-1)*cw0 -
tmp);
162 b0 = (A+1) - (A-1)*cw0 +
tmp;
163 bq->
b1 = 2*( (A-1) - (A+1)*cw0);
164 bq->
b2 = (A+1) - (A-1)*cw0 -
tmp;
176 double omega = 2.0 *
M_PI * fc / sr;
177 double sn = sin(omega);
178 double cs = cos(omega);
179 double alpha = sn/(2 * q);
180 double inv = 1.0/(1.0 +
alpha);
182 bq->
a2 = bq->
a0 = gain * inv * (1.0 - cs) * 0.5;
184 bq->
b1 = (-2.0 * cs * inv);
185 bq->
b2 = ((1.0 -
alpha) * inv);
192 freq *= 2.0 *
M_PI / sr;
197 return hypot(c->
a0 + c->
a1*zr + c->
a2*(zr*zr-zi*zi), c->
a1*zi + 2*c->
a2*zr*zi) /
198 hypot(1 + c->
b1*zr + c->
b2*(zr*zr-zi*zi), c->
b1*zi + 2*c->
b2*zr*zi);
203 double i, j, k,
g, t,
a0,
a1,
a2,
b1,
b2, tau1, tau2, tau3;
204 double cutfreq, gain1kHz, gc, sr = inlink->
sample_rate;
235 i = 1. / (2. *
M_PI * tau1);
236 j = 1. / (2. *
M_PI * tau2);
237 k = 1. / (2. *
M_PI * tau3);
243 i = 1. / (2. *
M_PI * tau1);
244 j = 1. / (2. *
M_PI * tau2);
245 k = 1. / (2. *
M_PI * tau3);
251 i = 1. / (2. *
M_PI * tau1);
252 j = 1. / (2. *
M_PI * tau2);
253 k = 1. / (2. *
M_PI * tau3);
259 i = 1. / (2. *
M_PI * tau1);
260 j = 1. / (2. *
M_PI * tau2);
261 k = 1. / (2. *
M_PI * tau3);
273 double tau = (s->
type == 7 ? 0.000050 : 0.000075);
274 double f = 1.0 / (2 *
M_PI * tau);
275 double nyq = sr * 0.5;
276 double gain = sqrt(1.0 + nyq * nyq / (f * f));
277 double cfreq = sqrt((gain - 1.0) * f * f);
281 q = pow((sr / 3269.0) + 19.5, -0.25);
283 q = pow((sr / 4750.0) + 19.5, -0.25);
292 g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
295 a2 = (-2.*t+j*t*t)*g;
296 b1 = (-8.+2.*i*k*t*t)*g;
297 b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
299 g = 1. / (2.*t+j*t*t);
300 a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
301 a1 = (-8.+2.*i*k*t*t)*g;
302 a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
304 b2 = (-2.*t+j*t*t)*g;
316 gain1kHz =
freq_gain(&coeffs, 1000.0, sr);
326 cutfreq =
FFMIN(0.45 * sr, 21000.);
329 for (ch = 1; ch < inlink->
channels; ch++) {
364 .priv_class = &aemphasis_class,
367 .
inputs = avfilter_af_aemphasis_inputs,
368 .
outputs = avfilter_af_aemphasis_outputs,
This structure describes decoded (raw) audio or video data.
Main libavfilter public API header.
static const AVOption aemphasis_options[]
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static double freq_gain(BiquadCoeffs *c, double freq, double sr)
const char * name
Pad name.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
AVFILTER_DEFINE_CLASS(aemphasis)
A filter pad used for either input or output.
A link between two filters.
#define fc(width, name, range_min, range_max)
#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.
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 av_const double hypot(double x, double y)
static const AVFilterPad avfilter_af_aemphasis_inputs[]
static double b0(void *priv, double x, double y)
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static int query_formats(AVFilterContext *ctx)
A list of supported channel layouts.
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
static const int16_t alpha[]
static const AVFilterPad avfilter_af_aemphasis_outputs[]
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.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int channels
Number of channels.
static void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
static int config_input(AVFilterLink *inlink)
static av_cold void uninit(AVFilterContext *ctx)
static void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
mode
Use these values in ebur128_init (or'ed).
int nb_samples
number of audio samples (per channel) described by this frame
static double biquad(BiquadD2 *bq, double in)
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)