76 #define OFFSET(x) offsetof(AudioVectorScopeContext, x)
77 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
119 const int linesize =
s->outpicref->linesize[0];
123 if (y >=
s->h || x >=
s->w)
130 dst = &
s->outpicref->data[0][y * linesize + x * 4];
131 dst[0] =
FFMIN(dst[0] +
s->contrast[0], 255);
132 dst[1] =
FFMIN(dst[1] +
s->contrast[1], 255);
133 dst[2] =
FFMIN(dst[2] +
s->contrast[2], 255);
134 dst[3] =
FFMIN(dst[3] +
s->contrast[3], 255);
139 int dx =
FFABS(x1-x0), sx = x0 < x1 ? 1 : -1;
140 int dy =
FFABS(y1-y0), sy = y0 < y1 ? 1 : -1;
141 int err = (dx>dy ? dx : -dy) / 2, e2;
146 if (x0 == x1 && y0 == y1)
165 const int linesize =
s->outpicref->linesize[0];
168 if (
s->fade[0] ||
s->fade[1] ||
s->fade[2]) {
170 for (
i = 0;
i <
s->h;
i++) {
171 for (j = 0; j <
s->w*4; j+=4) {
172 d[j+0] =
FFMAX(d[j+0] -
s->fade[0], 0);
173 d[j+1] =
FFMAX(d[j+1] -
s->fade[1], 0);
174 d[j+2] =
FFMAX(d[j+2] -
s->fade[2], 0);
175 d[j+3] =
FFMAX(d[j+3] -
s->fade[3], 0);
228 s->prev_x =
s->hw =
s->w / 2;
229 s->prev_y =
s->hh =
s->mode ==
POLAR ?
s->h - 1 :
s->h / 2;
239 const int hw =
s->hw;
240 const int hh =
s->hh;
243 unsigned prev_x =
s->prev_x, prev_y =
s->prev_y;
244 double zoom =
s->zoom;
247 if (!
s->outpicref ||
s->outpicref->width != outlink->
w ||
248 s->outpicref->height != outlink->
h) {
256 s->outpicref->sample_aspect_ratio = (
AVRational){1,1};
257 for (
i = 0;
i < outlink->
h;
i++)
258 memset(
s->outpicref->data[0] +
i *
s->outpicref->linesize[0], 0, outlink->
w * 4);
260 s->outpicref->pts = insamples->
pts;
267 switch (insamples->
format) {
269 int16_t *samples = (int16_t *)insamples->
data[0];
272 float sample = samples[
i] / (float)INT16_MAX;
279 float *samples = (
float *)insamples->
data[0];
294 int16_t *samples = (int16_t *)insamples->
data[0] +
i * 2;
295 float *samplesf = (
float *)insamples->
data[0] +
i * 2;
298 switch (insamples->
format) {
300 src[0] = samples[0] / (float)INT16_MAX;
301 src[1] = samples[1] / (float)INT16_MAX;
304 src[0] = samplesf[0];
305 src[1] = samplesf[1];
336 x = ((
src[1] -
src[0]) * zoom / 2 + 1) * hw;
337 y = (1.0 - (
src[0] +
src[1]) * zoom / 2) * hh;
339 x = (
src[1] * zoom + 1) * hw;
340 y = (
src[0] * zoom + 1) * hh;
342 float sx, sy, cx, cy;
346 cx = sx * sqrtf(1 - 0.5 * sy * sy);
347 cy = sy * sqrtf(1 - 0.5 * sx * sx);
348 x = hw + hw *
FFSIGN(cx + cy) * (cx - cy) * .7;
349 y =
s->h -
s->h *
fabsf(cx + cy) * .7;
352 if (
s->draw ==
DOT) {
361 s->prev_x = x,
s->prev_y = y;
419 .
name =
"avectorscope",
427 .priv_class = &avectorscope_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
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().
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
static void draw_dot(AudioVectorScopeContext *s, unsigned x, unsigned y)
static void draw_line(AudioVectorScopeContext *s, int x0, int y0, int x1, int y1)
static int activate(AVFilterContext *ctx)
static void fade(AudioVectorScopeContext *s)
static av_cold void uninit(AVFilterContext *ctx)
static const AVFilterPad audiovectorscope_outputs[]
AVFilter ff_avf_avectorscope
AVFILTER_DEFINE_CLASS(avectorscope)
static int config_output(AVFilterLink *outlink)
static const AVOption avectorscope_options[]
static const AVFilterPad audiovectorscope_inputs[]
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
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.
Main libavfilter public API header.
audio channel layout utility functions
#define FFSWAP(type, a, b)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static __device__ float fabsf(float a)
mode
Use these values in ebur128_init (or'ed).
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
#define FFERROR_NOT_READY
Filters implementation helper functions.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
@ AV_OPT_TYPE_VIDEO_RATE
offset must point to AVRational
#define AV_CH_LAYOUT_STEREO
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_S16
signed 16 bits
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static enum AVPixelFormat pix_fmts[]
static av_always_inline float cbrtf(float x)
AVPixelFormat
Pixel format.
@ AV_PIX_FMT_RGBA
packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
Describe the class of an AVClass context structure.
A list of supported channel layouts.
void * priv
private data for use by the filter
A link between two filters.
AVFilterFormatsConfig incfg
Lists of supported formats / etc.
int w
agreed upon image width
int h
agreed upon image height
AVFilterFormatsConfig outcfg
Lists of supported formats / etc.
AVFilterContext * src
source filter
int sample_rate
samples per second
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0,...
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.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Rational number (pair of numerator and denominator).
static void mirror(const float *modifier, float *vec)
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.