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48 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
49 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
89 const double *
src = (
const double *)
in->data[0];
90 const double level_out =
s->level_out;
91 const double level_in =
s->level_in;
106 dst = (
double *)
out->data[0];
108 for (n = 0; n <
in->nb_samples; n++) {
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);
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);
272 if (
s->type == 7 ||
s->type == 8) {
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);
288 s->rc[0].use_brickw = 0;
290 s->rc[0].use_brickw = 1;
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);
319 s->rc[0].r1.a0 = coeffs.
a0 * gc;
320 s->rc[0].r1.a1 = coeffs.
a1 * gc;
321 s->rc[0].r1.a2 = coeffs.
a2 * gc;
322 s->rc[0].r1.b1 = coeffs.
b1;
323 s->rc[0].r1.b2 = coeffs.
b2;
326 cutfreq =
FFMIN(0.45 * sr, 21000.);
327 set_lp_rbj(&
s->rc[0].brickw, cutfreq, 0.707, sr, 1.);
329 for (ch = 1; ch <
inlink->channels; ch++) {
364 .priv_class = &aemphasis_class,
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
A list of supported channel layouts.
static double biquad(BiquadD2 *bq, double in)
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 double freq_gain(BiquadCoeffs *c, double freq, double sr)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
enum MovChannelLayoutTag * layouts
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
This structure describes decoded (raw) audio or video data.
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFILTER_DEFINE_CLASS(aemphasis)
#define fc(width, name, range_min, range_max)
const char * name
Filter name.
A link between two filters.
static double b1(void *priv, double x, double y)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
A filter pad used for either input or output.
static const AVOption aemphasis_options[]
static const AVFilterPad outputs[]
static const AVFilterPad avfilter_af_aemphasis_inputs[]
Describe the class of an AVClass context structure.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
static const AVFilterPad avfilter_af_aemphasis_outputs[]
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
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
static int query_formats(AVFilterContext *ctx)
#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)
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
static double b2(void *priv, double x, double y)
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
#define i(width, name, range_min, range_max)
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
static const int16_t alpha[]
static int config_input(AVFilterLink *inlink)
static av_cold void uninit(AVFilterContext *ctx)
static void set_highshelf_rbj(BiquadD2 *bq, double freq, double q, double peak, double sr)
static void set_lp_rbj(BiquadD2 *bq, double fc, double q, double sr, double gain)
static double b0(void *priv, double x, double y)
@ AV_SAMPLE_FMT_DBL
double