76 #define OFFSET(x) offsetof(AudioVectorScopeContext, x) 77 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM 123 if (y >= s->
h || x >= s->
w)
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)
170 for (i = 0; i < s->
h; i++) {
171 for (j = 0; j < s->
w*4; j+=4) {
239 const int hw = s->
hw;
240 const int hh = s->
hh;
256 for (i = 0; i < outlink->
h; i++)
266 switch (insamples->
format) {
268 int16_t *samples = (int16_t *)insamples->
data[0];
271 float sample = samples[
i] / (float)INT16_MAX;
278 float *samples = (
float *)insamples->
data[0];
293 int16_t *samples = (int16_t *)insamples->
data[0] + i * 2;
294 float *samplesf = (
float *)insamples->
data[0] + i * 2;
297 switch (insamples->
format) {
299 src[0] = samples[0] / (float)INT16_MAX;
300 src[1] = samples[1] / (float)INT16_MAX;
303 src[0] = samplesf[0];
304 src[1] = samplesf[1];
320 src[0] =
FFSIGN(src[0]) * logf(1 +
FFABS(src[0])) / logf(2);
321 src[1] =
FFSIGN(src[1]) * logf(1 +
FFABS(src[1])) / logf(2);
332 FFSWAP(
float, src[0], src[1]);
335 x = ((src[1] - src[0]) * zoom / 2 + 1) *
hw;
336 y = (1.0 - (src[0] + src[1]) * zoom / 2) *
hh;
338 x = (src[1] * zoom + 1) * hw;
339 y = (src[0] * zoom + 1) * hh;
341 float sx, sy, cx, cy;
345 cx = sx * sqrtf(1 - 0.5 * sy * sy);
346 cy = sy * sqrtf(1 - 0.5 * sx * sx);
347 x = hw + hw *
FFSIGN(cx + cy) * (cx - cy) * .7;
348 y = s->
h - s->
h * fabsf(cx + cy) * .7;
414 .
name =
"avectorscope",
420 .
inputs = audiovectorscope_inputs,
421 .
outputs = audiovectorscope_outputs,
422 .priv_class = &avectorscope_class,
This structure describes decoded (raw) audio or video data.
Main libavfilter public API header.
static void draw_line(AudioVectorScopeContext *s, int x0, int y0, int x1, int y1)
static av_cold void uninit(AVFilterContext *ctx)
int h
agreed upon image height
#define FFERROR_NOT_READY
Filters implementation helper functions.
static int activate(AVFilterContext *ctx)
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
#define AV_CH_LAYOUT_STEREO
AVFilter ff_avf_avectorscope
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.
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static const AVFilterPad audiovectorscope_outputs[]
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
A filter pad used for either input or output.
A link between two filters.
#define i(width, name, range_min, range_max)
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0, will be automatically copied from the first input of the source filter if it exists.
int sample_rate
samples per second
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
static int config_input(AVFilterLink *inlink)
#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
simple assert() macros that are a bit more flexible than ISO C assert().
struct AVFilterChannelLayouts * out_channel_layouts
AVFilterFormats * in_formats
Lists of formats and channel layouts supported by the input and output filters respectively.
static const AVOption avectorscope_options[]
packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
int w
agreed upon image width
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
audio channel layout utility functions
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
static const AVFilterPad audiovectorscope_inputs[]
AVFilterContext * src
source filter
static void draw_dot(AudioVectorScopeContext *s, unsigned x, unsigned y)
static const AVFilterPad inputs[]
AVFilterFormats * out_samplerates
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
static const AVFilterPad outputs[]
AVFILTER_DEFINE_CLASS(avectorscope)
static int config_output(AVFilterLink *outlink)
A list of supported channel layouts.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
AVSampleFormat
Audio sample formats.
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
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.
AVRational sample_aspect_ratio
Sample aspect ratio for the video frame, 0/1 if unknown/unspecified.
static av_always_inline float cbrtf(float x)
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)\", 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.
Rational number (pair of numerator and denominator).
offset must point to AVRational
const char * name
Filter name.
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
offset must point to two consecutive integers
AVFilterLink ** outputs
array of pointers to output links
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
static enum AVPixelFormat pix_fmts[]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static int query_formats(AVFilterContext *ctx)
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
#define FFSWAP(type, a, b)
AVPixelFormat
Pixel format.
mode
Use these values in ebur128_init (or'ed).
int nb_samples
number of audio samples (per channel) described by this frame
AVFilterFormats * out_formats