51 int curve0,
int curve1);
54 enum CurveType {
TRI,
QSIN,
ESIN,
HSIN,
LOG,
IPAR,
QUA,
CUB,
SQU,
CBR,
PAR,
EXP,
IQSIN,
IHSIN,
DESE,
DESI,
LOSI,
NONE,
NB_CURVES };
56 #define OFFSET(x) offsetof(AudioFadeContext, x) 57 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 94 #define CUBE(a) ((a)*(a)*(a)) 97 gain = av_clipd(1.0 * index / range, 0, 1.0);
101 gain = sin(gain *
M_PI / 2.0);
105 gain = 0.6366197723675814 * asin(gain);
108 gain = 1.0 - cos(
M_PI / 4.0 * (
CUBE(2.0*gain - 1) + 1));
111 gain = (1.0 - cos(gain *
M_PI)) / 2.0;
115 gain = 0.3183098861837907 * acos(1 - 2 * gain);
119 gain =
exp(-11.512925464970227 * (1 - gain));
122 gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
125 gain = 1 - sqrt(1 - gain);
128 gain = (1 - (1 - gain) * (1 - gain));
143 gain = gain <= 0.5 ?
cbrt(2 * gain) / 2: 1 -
cbrt(2 * (1 - gain)) / 2;
146 gain = gain <= 0.5 ?
CUBE(2 * gain) / 2: 1 -
CUBE(2 * (1 - gain)) / 2;
149 const double a = 1. / (1. - 0.787) - 1;
150 double A = 1. / (1.0 +
exp(0 -((gain-0.5) * a * 2.0)));
151 double B = 1. / (1.0 +
exp(a));
152 double C = 1. / (1.0 +
exp(0-a));
153 gain = (A -
B) / (C - B);
164 #define FADE_PLANAR(name, type) \ 165 static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \ 166 int nb_samples, int channels, int dir, \ 167 int64_t start, int64_t range, int curve) \ 171 for (i = 0; i < nb_samples; i++) { \ 172 double gain = fade_gain(curve, start + i * dir, range); \ 173 for (c = 0; c < channels; c++) { \ 174 type *d = (type *)dst[c]; \ 175 const type *s = (type *)src[c]; \ 177 d[i] = s[i] * gain; \ 182 #define FADE(name, type) \ 183 static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \ 184 int nb_samples, int channels, int dir, \ 185 int64_t start, int64_t range, int curve) \ 187 type *d = (type *)dst[0]; \ 188 const type *s = (type *)src[0]; \ 191 for (i = 0; i < nb_samples; i++) { \ 192 double gain = fade_gain(curve, start + i * dir, range); \ 193 for (c = 0; c < channels; c++, k++) \ 194 d[k] = s[k] * gain; \ 213 switch (outlink->format) {
232 #if CONFIG_AFADE_FILTER 234 static const AVOption afade_options[] = {
303 if ((!s->
type && (cur_sample + nb_samples < s->start_sample)) ||
317 s->
type ? -1 : 1, start,
327 static const AVFilterPad avfilter_af_afade_inputs[] = {
336 static const AVFilterPad avfilter_af_afade_outputs[] = {
351 .
inputs = avfilter_af_afade_inputs,
352 .
outputs = avfilter_af_afade_outputs,
353 .priv_class = &afade_class,
359 #if CONFIG_ACROSSFADE_FILTER 361 static const AVOption acrossfade_options[] = {
395 #define CROSSFADE_PLANAR(name, type) \ 396 static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \ 397 uint8_t * const *cf1, \ 398 int nb_samples, int channels, \ 399 int curve0, int curve1) \ 403 for (i = 0; i < nb_samples; i++) { \ 404 double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \ 405 double gain1 = fade_gain(curve1, i, nb_samples); \ 406 for (c = 0; c < channels; c++) { \ 407 type *d = (type *)dst[c]; \ 408 const type *s0 = (type *)cf0[c]; \ 409 const type *s1 = (type *)cf1[c]; \ 411 d[i] = s0[i] * gain0 + s1[i] * gain1; \ 416 #define CROSSFADE(name, type) \ 417 static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \ 418 uint8_t * const *cf1, \ 419 int nb_samples, int channels, \ 420 int curve0, int curve1) \ 422 type *d = (type *)dst[0]; \ 423 const type *s0 = (type *)cf0[0]; \ 424 const type *s1 = (type *)cf1[0]; \ 427 for (i = 0; i < nb_samples; i++) { \ 428 double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \ 429 double gain1 = fade_gain(curve1, i, nb_samples); \ 430 for (c = 0; c < channels; c++, k++) \ 431 d[k] = s0[k] * gain0 + s1[k] * gain1; \ 435 CROSSFADE_PLANAR(dbl,
double)
436 CROSSFADE_PLANAR(flt,
float)
437 CROSSFADE_PLANAR(s16, int16_t)
440 CROSSFADE(dbl,
double)
441 CROSSFADE(flt,
float)
442 CROSSFADE(s16, int16_t)
462 }
else if (ret < 0) {
507 cf[1]->extended_data,
575 static int acrossfade_config_output(
AVFilterLink *outlink)
582 "Inputs must have the same sample rate " 583 "%d for in0 vs %d for in1\n",
593 switch (outlink->
format) {
609 static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
611 .
name =
"crossfade0",
615 .name =
"crossfade1",
621 static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
625 .config_props = acrossfade_config_output,
631 .
name =
"acrossfade",
636 .priv_class = &acrossfade_class,
637 .
inputs = avfilter_af_acrossfade_inputs,
638 .
outputs = avfilter_af_acrossfade_outputs,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
This structure describes decoded (raw) audio or video data.
#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.
static av_cold int init(AVCodecContext *avctx)
static int config_output(AVFilterLink *outlink)
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the 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...
const char * name
Pad name.
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.
filter_frame For filters that do not use the activate() callback
static int query_formats(AVFilterContext *ctx)
#define FADE_PLANAR(name, type)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
void(* fade_samples)(uint8_t **dst, uint8_t *const *src, int nb_samples, int channels, int direction, int64_t start, int64_t range, int curve)
#define AVERROR_EOF
End of file.
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
A link between two filters.
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
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
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
void(* crossfade_samples)(uint8_t **dst, uint8_t *const *cf0, uint8_t *const *cf1, int nb_samples, int channels, int curve0, int curve1)
int channels
number of audio channels, only used for audio.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
#define AV_TIME_BASE
Internal time base represented as integer.
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int ff_inlink_queued_samples(AVFilterLink *link)
AVFilterContext * src
source filter
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
s EdgeDetect Foobar g libavfilter vf_edgedetect c libavfilter vf_foobar c edit libavfilter and add an entry for foobar following the pattern of the other filters edit libavfilter allfilters and add an entry for foobar following the pattern of the other filters configure make j< whatever > ffmpeg ffmpeg i you should get a foobar png with Lena edge detected That s your new playground is ready Some little details about what s going which in turn will define variables for the build system and the C
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
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
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.
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
Rational number (pair of numerator and denominator).
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
#define flags(name, subs,...)
static int filter_frame(DBEContext *s, AVFrame *frame)
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
static double fade_gain(int curve, int64_t index, int64_t range)
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[]
AVFilter ff_af_acrossfade
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
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
#define AV_NOPTS_VALUE
Undefined timestamp value.