65 #define OFFSET(x) offsetof(AudioGateContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
94 double lin_threshold =
s->threshold;
95 double lin_knee_sqrt = sqrt(
s->knee);
98 lin_threshold *= lin_threshold;
102 s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
103 s->lin_knee_start = lin_threshold / lin_knee_sqrt;
104 s->thres = log(lin_threshold);
105 s->knee_start = log(
s->lin_knee_start);
106 s->knee_stop = log(
s->lin_knee_stop);
112 #define FAKE_INFINITY (65536.0 * 65536.0)
115 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
117 static double output_gain(
double lin_slope,
double ratio,
double thres,
118 double knee,
double knee_start,
double knee_stop,
119 double range,
int mode)
121 double slope = log(lin_slope);
122 double tratio = ratio;
128 gain = (slope - thres) * tratio + thres;
132 if (knee > 1. && slope < knee_stop)
135 if (knee > 1. && slope > knee_start)
138 return FFMAX(range,
exp(gain - slope));
142 const double *
src,
double *dst,
const double *scsrc,
143 int nb_samples,
double level_in,
double level_sc,
146 const double makeup =
s->makeup;
147 const double attack_coeff =
s->attack_coeff;
148 const double release_coeff =
s->release_coeff;
152 double abs_sample =
fabs(scsrc[0] * level_sc), gain = 1.0;
157 abs_sample =
FFMAX(
fabs(scsrc[
c] * level_sc), abs_sample);
160 abs_sample +=
fabs(scsrc[
c] * level_sc);
166 abs_sample *= abs_sample;
168 s->lin_slope += (abs_sample -
s->lin_slope) * (abs_sample >
s->lin_slope ? attack_coeff : release_coeff);
171 detected =
s->lin_slope >
s->lin_knee_start;
173 detected =
s->lin_slope <
s->lin_knee_stop;
175 if (
s->lin_slope > 0.0 && detected)
177 s->knee,
s->knee_start,
s->knee_stop,
181 dst[
c] =
src[
c] * level_in * gain * makeup;
185 #if CONFIG_AGATE_FILTER
187 #define agate_options options
218 const double *
src = (
const double *)
in->data[0];
235 dst = (
double *)
out->data[0];
238 s->level_in,
s->level_in, inlink, inlink);
268 .priv_class = &agate_class,
277 #if CONFIG_SIDECHAINGATE_FILTER
279 #define sidechaingate_options options
286 int ret,
i, nb_samples;
310 for (
i = 0;
i < 2;
i++) {
321 dst = (
double *)
out->data[0];
326 (
double *)
in[1]->
data[0], nb_samples,
327 s->level_in,
s->level_sc,
328 ctx->inputs[0],
ctx->inputs[1]);
358 if (!
ctx->inputs[0]->incfg.channel_layouts ||
359 !
ctx->inputs[0]->incfg.channel_layouts->nb_channel_layouts) {
361 "No channel layout for input 1\n");
369 for (
i = 0;
i < 2;
i++) {
388 if (
ctx->inputs[0]->sample_rate !=
ctx->inputs[1]->sample_rate) {
390 "Inputs must have the same sample rate "
391 "%d for in0 vs %d for in1\n",
392 ctx->inputs[0]->sample_rate,
ctx->inputs[1]->sample_rate);
403 if (!
s->fifo[0] || !
s->fifo[1])
420 static const AVFilterPad sidechaingate_inputs[] = {
431 static const AVFilterPad sidechaingate_outputs[] = {
435 .config_props = scconfig_output,
441 .
name =
"sidechaingate",
444 .priv_class = &sidechaingate_class,
448 .
inputs = sidechaingate_inputs,
449 .
outputs = sidechaingate_outputs,
static enum AVSampleFormat sample_fmts[]
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static int activate(AVFilterContext *ctx)
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double range, int mode)
static const AVOption options[]
static int agate_config_input(AVFilterLink *inlink)
static void gate(AudioGateContext *s, const double *src, double *dst, const double *scsrc, int nb_samples, double level_in, double level_sc, AVFilterLink *inlink, AVFilterLink *sclink)
#define IS_FAKE_INFINITY(value)
AVFilter ff_af_sidechaingate
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.
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Main libavfilter public API header.
#define flags(name, subs,...)
audio channel layout utility functions
static av_cold int uninit(AVCodecContext *avctx)
static __device__ float fabs(float a)
static int filter_frame(DBEDecodeContext *s, AVFrame *frame)
mode
Use these values in ebur128_init (or'ed).
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
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_WARNING
Something somehow does not look correct.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_DBL
double
static double hermite_interpolation(double x, double x0, double x1, double p0, double p1, double m0, double m1)
#define AVFILTER_DEFINE_CLASS(fname)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
Context for an Audio FIFO Buffer.
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int channels
Number of channels.
AVFilterContext * src
source filter
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
int sample_rate
samples per second
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
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.
Rational number (pair of numerator and denominator).