65 #define OFFSET(x) offsetof(AudioLimiterContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM
67 #define F AV_OPT_FLAG_FILTERING_PARAM
91 s->asc_coeff = pow(0.5,
s->asc_coeff - 0.5) * 2 * -1;
97 double peak,
double limit,
double patt,
int asc)
101 if (asc &&
s->auto_release &&
s->asc_c > 0) {
102 double a_att = limit / (
s->asc_coeff *
s->asc) * (
double)
s->asc_c;
120 const double *
src = (
const double *)
in->data[0];
122 const int buffer_size =
s->buffer_size;
123 double *dst, *
buffer =
s->buffer;
124 const double release =
s->release;
125 const double limit =
s->limit;
126 double *nextdelta =
s->nextdelta;
127 double level =
s->auto_level ? 1 / limit : 1;
128 const double level_out =
s->level_out;
129 const double level_in =
s->level_in;
130 int *nextpos =
s->nextpos;
145 dst = (
double *)
out->data[0];
147 for (n = 0; n <
in->nb_samples; n++) {
157 if (
s->auto_release && peak > limit) {
165 peak, limit,
patt, 0);
169 if (delta < s->
delta) {
173 nextdelta[0] = rdelta;
177 for (
i =
s->nextiter; i < s->nextiter +
s->nextlen;
i++) {
178 int j =
i % buffer_size;
179 double ppeak = 0, pdelta;
184 pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] +
s->pos) % buffer_size) /
channels);
185 if (pdelta < nextdelta[j]) {
186 nextdelta[j] = pdelta;
192 s->nextlen =
i -
s->nextiter + 1;
193 nextpos[(
s->nextiter +
s->nextlen) % buffer_size] =
s->pos;
194 nextdelta[(
s->nextiter +
s->nextlen) % buffer_size] = rdelta;
195 nextpos[(
s->nextiter +
s->nextlen + 1) % buffer_size] = -1;
201 buf = &
s->buffer[(
s->pos +
channels) % buffer_size];
209 if (
s->pos ==
s->asc_pos && !
s->asc_changed)
212 if (
s->auto_release &&
s->asc_pos == -1 && peak > limit) {
220 dst[
c] = buf[
c] *
s->att;
222 if ((
s->pos +
channels) % buffer_size == nextpos[
s->nextiter]) {
223 if (
s->auto_release) {
225 peak, limit,
s->att, 1);
226 if (
s->nextlen > 1) {
227 int pnextpos = nextpos[(
s->nextiter + 1) % buffer_size];
231 double pdelta = (limit / ppeak -
s->att) /
232 (((buffer_size + pnextpos -
235 if (pdelta < s->
delta)
239 s->delta = nextdelta[
s->nextiter];
240 s->att = limit / peak;
244 nextpos[
s->nextiter] = -1;
245 s->nextiter = (
s->nextiter + 1) % buffer_size;
257 s->att = 0.0000000000001;
261 if (
s->att != 1. && (1. -
s->att) < 0.0000000000001)
264 if (
s->delta != 0. &&
fabs(
s->delta) < 0.00000000000001)
318 if (obuffer_size < inlink->
channels)
321 s->buffer =
av_calloc(obuffer_size,
sizeof(*
s->buffer));
322 s->nextdelta =
av_calloc(obuffer_size,
sizeof(*
s->nextdelta));
324 if (!
s->buffer || !
s->nextdelta || !
s->nextpos)
327 memset(
s->nextpos, -1, obuffer_size *
sizeof(*
s->nextpos));
329 s->buffer_size -=
s->buffer_size % inlink->
channels;
331 if (
s->buffer_size <= 0) {
370 .priv_class = &alimiter_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static const AVFilterPad alimiter_inputs[]
static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate, double peak, double limit, double patt, int asc)
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
AVFILTER_DEFINE_CLASS(alimiter)
static const AVFilterPad alimiter_outputs[]
static const AVOption alimiter_options[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static av_cold int init(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
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
simple assert() macros that are a bit more flexible than ISO C assert().
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Main libavfilter public API header.
audio channel layout utility functions
common internal and external API header
static __device__ float fabs(float a)
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.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_DBL
double
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
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.
This structure describes decoded (raw) audio or video data.
#define av_malloc_array(a, b)
static const int8_t patt[4]