59 #define OFFSET(x) offsetof(AudioHistogramContext, x)
60 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
178 for (n = H; n < s->
h; n++) {
187 for (y = 0; y < w; y++) {
194 for (n = 0; n <
H; n++) {
206 for (c = 0; c < inlink->
channels; c++) {
211 bin =
lrint(av_clipf(fabsf(src[n]), 0, 1) * (w - 1));
221 bin =
lrint(av_clipf(fabsf(src2[n]), 0, 1) * (w - 1));
229 for (c = 0; c < inlink->
channels; c++) {
234 bin =
lrint(av_clipf(1 + log10(fabsf(src[n])) / 6, 0, 1) * (w - 1));
244 bin =
lrint(av_clipf(1 + log10(fabsf(src2[n])) / 6, 0, 1) * (w - 1));
272 uf *= 0.5 * sin((2 * M_PI * c) / s->
dchannels);
273 vf *= 0.5 * cos((2 * M_PI * c) / s->
dchannels);
276 for (n = 0; n < w; n++) {
280 a = achistogram[
n] - shistogram[
n];
284 aa = a / (double)acmax;
287 aa = sqrt(a) / sqrt(acmax);
296 aa = 1. -
log2(a + 1) /
log2(acmax + 1);
308 for (y = H - h; y <
H; y++) {
326 for (y = H - h; y <
H; y++) {
345 for (n = 0; n < w; n++) {
356 for (p = 0; p < 4; p++) {
357 for (y = s->
h; y >= H + 1; y--) {
381 for (i = 0; i < 101; i++)
405 .
name =
"ahistogram",
410 .
inputs = audiovectorscope_inputs,
411 .
outputs = audiovectorscope_outputs,
412 .priv_class = &ahistogram_class,
static const AVFilterPad audiovectorscope_outputs[]
This structure describes decoded (raw) audio or video data.
Main libavfilter public API header.
int max_samples
Maximum number of samples to filter at once.
int h
agreed upon image height
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
const char * name
Pad name.
AVFilterLink ** inputs
array of pointers to input links
#define av_assert0(cond)
assert() equivalent, that is always enabled.
static double cb(void *priv, double x, double y)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static int query_formats(AVFilterContext *ctx)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static const AVOption ahistogram_options[]
static double av_q2d(AVRational a)
Convert an AVRational to a double.
static int config_output(AVFilterLink *outlink)
A filter pad used for either input or output.
A link between two filters.
int width
width and height of the video frame
int min_samples
Minimum number of samples to filter at once.
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.
#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
static int config_input(AVFilterLink *inlink)
AVFilterFormats * in_formats
Lists of formats and channel layouts supported by the input and output filters respectively.
int w
agreed upon image width
AVFILTER_DEFINE_CLASS(ahistogram)
AVFilterContext * src
source filter
int partial_buf_size
Size of the partial buffer to allocate.
static const AVFilterPad outputs[]
AVFilterFormats * out_samplerates
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
A list of supported channel layouts.
static const AVFilterPad inputs[]
AVSampleFormat
Audio sample formats.
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
planar YUV 4:4:4 32bpp, (1 Cr & Cb sample per 1x1 Y & A samples)
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.
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
enum MovChannelLayoutTag * layouts
static enum AVPixelFormat pix_fmts[]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
static const AVFilterPad audiovectorscope_inputs[]
static av_cold void uninit(AVFilterContext *ctx)
int channels
Number of channels.
AVFilterContext * dst
dest filter
AVFilter ff_avf_ahistogram
static enum AVSampleFormat sample_fmts[]
#define av_malloc_array(a, b)
uint8_t ** extended_data
pointers to the data planes/channels.
AVPixelFormat
Pixel format.
int nb_samples
number of audio samples (per channel) described by this frame
AVFilterFormats * out_formats