63 #define OFFSET(x) offsetof(AudioGateContext, x)
64 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
90 double lin_knee_sqrt = sqrt(s->
knee);
91 double lin_knee_start;
94 lin_threshold *= lin_threshold;
99 lin_knee_start = lin_threshold / lin_knee_sqrt;
100 s->
thres = log(lin_threshold);
108 #define FAKE_INFINITY (65536.0 * 65536.0)
111 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
113 static double output_gain(
double lin_slope,
double ratio,
double thres,
114 double knee,
double knee_start,
double knee_stop,
115 double lin_knee_stop,
double range)
117 if (lin_slope < lin_knee_stop) {
118 double slope = log(lin_slope);
119 double tratio = ratio;
125 gain = (slope - thres) * tratio + thres;
128 if (knee > 1. && slope > knee_start) {
129 gain =
hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
131 return FFMAX(range,
exp(gain - slope));
138 const double *
src,
double *dst,
const double *scsrc,
139 int nb_samples,
double level_in,
double level_sc,
142 const double makeup = s->
makeup;
148 double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
151 for (c = 1; c < sclink->
channels; c++)
152 abs_sample =
FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
154 for (c = 1; c < sclink->
channels; c++)
155 abs_sample += fabs(scsrc[c] * level_sc);
161 abs_sample *= abs_sample;
169 for (c = 0; c < inlink->
channels; c++)
170 dst[c] = src[c] * level_in * gain * makeup;
174 #if CONFIG_AGATE_FILTER
176 #define agate_options options
207 const double *
src = (
const double *)in->
data[0];
224 dst = (
double *)
out->data[0];
227 s->level_in,
s->level_in, inlink, inlink);
257 .priv_class = &agate_class,
264 #if CONFIG_SIDECHAINGATE_FILTER
266 #define sidechaingate_options options
273 int ret, i, nb_samples;
297 for (i = 0; i < 2; i++) {
308 dst = (
double *)out->
data[0];
310 s->
pts += nb_samples;
312 gate(s, (
double *)in[0]->data[0], dst,
313 (
double *)in[1]->
data[0], nb_samples,
348 "No channel layout for input 1\n");
356 for (i = 0; i < 2; i++) {
377 "Inputs must have the same sample rate "
378 "%d for in0 vs %d for in1\n",
407 static const AVFilterPad sidechaingate_inputs[] = {
418 static const AVFilterPad sidechaingate_outputs[] = {
422 .config_props = scconfig_output,
428 .
name =
"sidechaingate",
431 .priv_class = &sidechaingate_class,
435 .inputs = sidechaingate_inputs,
436 .outputs = sidechaingate_outputs,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
#define AV_LOG_WARNING
Something somehow does not look correct.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
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)
struct AVFilterChannelLayouts * in_channel_layouts
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
const char * name
Pad name.
#define IS_FAKE_INFINITY(value)
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static int activate(AVFilterContext *ctx)
static av_cold int uninit(AVCodecContext *avctx)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
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.
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
uint64_t * channel_layouts
list of channel layouts
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
simple assert() macros that are a bit more flexible than ISO C assert().
struct AVFilterChannelLayouts * out_channel_layouts
Context for an Audio FIFO Buffer.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
audio channel layout utility functions
static double hermite_interpolation(double x, double x0, double x1, double p0, double p1, double m0, double m1)
AVFilter ff_af_sidechaingate
AVFilterContext * src
source filter
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
int nb_samples
number of samples currently in the FIFO
static const AVOption options[]
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
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.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
static int filter_frame(DBEContext *s, AVFrame *frame)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
static int query_formats(AVFilterContext *ctx)
int nb_channel_layouts
number of channel layouts
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
AVFilterContext * dst
dest filter
#define AVFILTER_DEFINE_CLASS(fname)
static enum AVSampleFormat sample_fmts[]
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double lin_knee_stop, double range)
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.