65 double coefs[3 *(
N+1)];
94 #define OFFSET(x) offsetof(MCompandContext, x) 95 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 98 {
"args",
"set parameters for each band",
OFFSET(args),
AV_OPT_TYPE_STRING, { .str =
"0.005,0.1 6 -47/-40,-34/-34,-17/-33 100 | 0.003,0.05 6 -47/-40,-34/-34,-17/-33 400 | 0.000625,0.0125 6 -47/-40,-34/-34,-15/-33 1600 | 0.0001,0.025 6 -47/-40,-34/-34,-31/-31,-0/-30 6400 | 0,0.025 6 -38/-31,-28/-28,-0/-25 22000" }, 0, 0,
A },
156 static void count_items(
char *item_str,
int *nb_items,
char delimiter)
161 for (p = item_str; *p; p++) {
180 double in_log, out_log;
183 if (in_lin <= s->in_min_lin)
186 in_log = log(in_lin);
189 if (in_log <= s->segments[i].
x)
193 out_log = cs->
y + in_log * (cs->
a * in_log + cs->
b);
201 int new_nb_items, num;
202 char *saveptr =
NULL;
206 #define S(x) s->segments[2 * ((x) + 1)] 207 for (i = 0, new_nb_items = 0; i < nb_points; i++) {
208 char *tstr =
av_strtok(p,
",", &saveptr);
210 if (!tstr || sscanf(tstr,
"%lf/%lf", &
S(i).
x, &
S(i).
y) != 2) {
212 "Invalid and/or missing input/output value.\n");
215 if (i &&
S(i - 1).x >
S(i).x) {
217 "Transfer function input values must be increasing.\n");
227 if (num == 0 ||
S(num - 1).
x)
231 #define S(x) s->segments[2 * (x)] 238 for (i = 2; i < num; i++) {
239 double g1 = (
S(i - 1).y -
S(i - 2).y) * (
S(i - 0).x -
S(i - 1).x);
240 double g2 = (
S(i - 0).y -
S(i - 1).y) * (
S(i - 1).x -
S(i - 2).x);
246 for (j = --i; j < num; j++)
256 #define L(x) s->segments[i - (x)] 258 double x,
y, cx, cy, in1, in2, out1, out2, theta,
len,
r;
261 L(4).b = (
L(2).y -
L(4).y) / (
L(2).x -
L(4).x);
264 L(2).b = (
L(0).y -
L(2).y) / (
L(0).x -
L(2).x);
266 theta = atan2(
L(2).y -
L(4).y,
L(2).x -
L(4).x);
267 len =
hypot(
L(2).x -
L(4).x,
L(2).y -
L(4).y);
268 r =
FFMIN(radius, len);
269 L(3).x =
L(2).x - r * cos(theta);
270 L(3).y =
L(2).y - r * sin(theta);
272 theta = atan2(
L(0).y -
L(2).y,
L(0).x -
L(2).x);
273 len =
hypot(
L(0).x -
L(2).x,
L(0).y -
L(2).y);
274 r =
FFMIN(radius, len / 2);
275 x =
L(2).x + r * cos(theta);
276 y =
L(2).y + r * sin(theta);
278 cx = (
L(3).x +
L(2).x +
x) / 3;
279 cy = (
L(3).y +
L(2).y +
y) / 3;
286 in2 =
L(2).x -
L(3).x;
287 out2 =
L(2).y -
L(3).y;
288 L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
289 L(3).b = out1 / in1 -
L(3).a * in1;
303 y[1] = 2 * x[0] * x[1];
304 y[2] = 2 * x[0] * x[2] + x[1] * x[1];
305 y[3] = 2 * x[1] * x[2];
312 double Q = sqrt(.5),
alpha = sin(w0) / (2*Q);
319 x[0] = (1 - cos(w0))/2;
321 x[2] = (1 - cos(w0))/2;
322 x[3] = (1 + cos(w0))/2;
323 x[4] = -(1 + cos(w0));
324 x[5] = (1 + cos(w0))/2;
329 for (norm = x[6], i = 0; i < 9; ++
i)
347 int ret, ch,
i, k, new_nb_items, nb_bands;
349 int max_delay_size = 0;
358 for (i = 0, new_nb_items = 0; i < nb_bands; i++) {
359 int nb_points, nb_attacks, nb_items = 0;
360 char *tstr2, *tstr =
av_strtok(p,
"|", &saveptr);
361 char *p2, *p3, *saveptr2 =
NULL, *saveptr3 =
NULL;
379 if (!nb_attacks || nb_attacks & 1) {
387 for (k = 0; k <
FFMIN(nb_attacks / 2, outlink->
channels); k++) {
388 char *tstr3 =
av_strtok(p3,
",", &saveptr3);
408 for (ch = k; ch < outlink->
channels; ch++) {
447 new_nb_items += sscanf(tstr2,
"%lf", &s->
bands[i].
topfreq) == 1;
466 double initial_volume;
468 sscanf(tstr2,
"%lf", &initial_volume);
469 initial_volume = pow(10.0, initial_volume / 20);
471 for (k = 0; k < outlink->
channels; k++) {
484 for (i = 0; max_delay_size > 0 && i < s->
nb_bands; i++) {
494 #define CONVOLVE _ _ _ _ 497 double *ibuf,
double *obuf_low,
498 double *obuf_high,
size_t len)
500 double out_low, out_high;
504 #define _ out_low += p->coefs[j] * p->previous[ch][p->pos + j].in \ 505 - p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_low, j++; 508 out_low = p->
coefs[0] * *ibuf;
510 *obuf_low++ = out_low;
513 #define _ out_high += p->coefs[j+N+1] * p->previous[ch][p->pos + j].in \ 514 - p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_high, j++; 517 out_high = p->
coefs[
N+1] * *ibuf;
519 *obuf_high++ = out_high;
531 for (i = 0; i <
len; i++) {
532 double level_in_lin, level_out_lin, checkbuf;
537 level_in_lin = l->
volume[ch];
541 checkbuf = ibuf[
i] * level_out_lin;
605 for (ch = 0; ch < outlink->
channels; ch++) {
669 "Multiband Compress or expand audio dynamic range."),
672 .priv_class = &mcompand_class,
674 .
inputs = mcompand_inputs,
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
This structure describes decoded (raw) audio or video data.
static double get_volume(CompandT *s, double in_lin)
static void update_volume(CompBand *cb, double in, int ch)
Main libavfilter public API header.
static int query_formats(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
static double cb(void *priv, double x, double y)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static void count_items(char *item_str, int *nb_items, char delimiter)
static int request_frame(AVFilterLink *outlink)
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).
static void square_quadratic(double const *x, double *y)
A filter pad used for either input or output.
A link between two filters.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
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.
static __device__ float fabs(float a)
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
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
simple assert() macros that are a bit more flexible than ISO C assert().
CompandSegment * segments
static av_const double hypot(double x, double y)
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
static int crossover_setup(AVFilterLink *outlink, Crossover *p, double frequency)
AVFilterContext * src
source filter
static const AVFilterPad outputs[]
A list of supported channel layouts.
AVSampleFormat
Audio sample formats.
static const int16_t alpha[]
static const AVFilterPad mcompand_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)\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.
const char * name
Filter name.
AVFILTER_DEFINE_CLASS(mcompand)
AVFilterLink ** outputs
array of pointers to output links
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
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()...
internal math functions header
static void crossover(int ch, Crossover *p, double *ibuf, double *obuf_low, double *obuf_high, size_t len)
static av_cold void uninit(AVFilterContext *ctx)
int channels
Number of channels.
AVFilterContext * dst
dest filter
static int parse_points(char *points, int nb_points, double radius, CompandT *s, AVFilterContext *ctx)
static const AVOption mcompand_options[]
static enum AVSampleFormat sample_fmts[]
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
#define FFSWAP(type, a, b)
static int mcompand_channel(MCompandContext *c, CompBand *l, double *ibuf, double *obuf, int len, int ch)
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
uint8_t ** extended_data
pointers to the data planes/channels.
int nb_samples
number of audio samples (per channel) described by this frame
static const AVFilterPad mcompand_inputs[]