65 #define OFFSET(x) offsetof(AudioLimiterContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM
67 #define F AV_OPT_FLAG_FILTERING_PARAM
97 double peak,
double limit,
double patt,
int asc)
99 double rdelta = (1.0 -
patt) / (sample_rate * release);
105 double delta =
FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
120 const double *
src = (
const double *)in->
data[0];
121 const int channels = inlink->
channels;
124 const double release = s->
release;
125 const double limit = s->
limit;
129 const double level_in = s->
level_in;
145 dst = (
double *)
out->data[0];
150 for (
c = 0;
c < channels;
c++) {
151 double sample = src[
c] * level_in;
154 peak =
FFMAX(peak, fabs(sample));
165 peak, limit, patt, 0);
166 double delta = (limit / peak - s->
att) / buffer_size * channels;
169 if (delta < s->delta) {
173 nextdelta[0] = rdelta;
178 int j = i % buffer_size;
179 double ppeak, pdelta;
181 ppeak = fabs(
buffer[nextpos[j]]) > fabs(
buffer[nextpos[j] + 1]) ?
182 fabs(
buffer[nextpos[j]]) : fabs(
buffer[nextpos[j] + 1]);
183 pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->
pos) % buffer_size) / channels);
184 if (pdelta < nextdelta[j]) {
185 nextdelta[j] = pdelta;
202 for (
c = 0;
c < channels;
c++) {
205 peak =
FFMAX(peak, fabs(sample));
218 for (
c = 0;
c < channels;
c++)
221 if ((s->
pos + channels) % buffer_size == nextpos[s->
nextiter]) {
224 peak, limit, s->
att, 1);
226 int pnextpos = nextpos[(s->
nextiter + 1) % buffer_size];
227 double ppeak = fabs(
buffer[pnextpos]) > fabs(
buffer[pnextpos + 1]) ?
229 fabs(
buffer[pnextpos + 1]);
230 double pdelta = (limit / ppeak - s->
att) /
231 (((buffer_size + pnextpos -
232 ((s->
pos + channels) % buffer_size)) %
233 buffer_size) / channels);
234 if (pdelta < s->
delta)
239 s->
att = limit / peak;
256 s->
att = 0.0000000000001;
260 if (s->
att != 1. && (1. - s->
att) < 0.0000000000001)
263 if (s->
delta != 0. && fabs(s->
delta) < 0.00000000000001)
266 for (
c = 0;
c < channels;
c++)
267 dst[
c] = av_clipd(dst[
c], -limit, limit) *
level * level_out;
269 s->
pos = (s->
pos + channels) % buffer_size;
317 if (obuffer_size < inlink->channels)
364 .priv_class = &alimiter_class,
368 .
inputs = alimiter_inputs,
This structure describes decoded (raw) audio or video data.
Main libavfilter public API header.
static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate, double peak, double limit, double patt, int asc)
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
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(alimiter)
A filter pad used for either input or output.
A link between two filters.
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
simple assert() macros that are a bit more flexible than ISO C assert().
static av_cold int init(AVFilterContext *ctx)
audio channel layout utility functions
static int config_input(AVFilterLink *inlink)
static const AVFilterPad outputs[]
A list of supported channel layouts.
static const AVFilterPad inputs[]
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
static const int8_t patt[4]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
const char * name
Filter name.
static av_cold void uninit(AVFilterContext *ctx)
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVOption alimiter_options[]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
common internal and external API header
int channels
Number of channels.
static const AVFilterPad alimiter_inputs[]
static const AVFilterPad alimiter_outputs[]
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
#define av_malloc_array(a, b)
int nb_samples
number of audio samples (per channel) described by this frame
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
static int query_formats(AVFilterContext *ctx)