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64 #define MAX_DURATION (24*60*60*1000000LL)
65 #define OFFSET(x) offsetof(AudioPhaseMeterContext, x)
66 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
67 #define get_duration(index) (index[1] - index[0])
79 {
"phasing",
"set mono and out-of-phase detection output",
OFFSET(do_phasing_detection),
AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1,
FLAGS },
135 inlink->partial_buf_size =
137 inlink->max_samples = nb_samples;
153 if (!strcmp(
s->mpc_str,
"none"))
154 s->draw_median_phase = 0;
156 s->draw_median_phase = 1;
163 static inline int get_x(
float phase,
int w)
165 return (phase + 1.) / 2. * (
w - 1);
172 snprintf(buf,
sizeof(buf),
"lavfi.aphasemeter.%s",
key);
178 int64_t mono_duration;
179 if (!
s->is_mono && mono_measurement) {
181 s->start_mono_presence = 1;
182 s->mono_idx[0] = insamples->
pts;
184 if (
s->is_mono && mono_measurement &&
s->start_mono_presence) {
185 s->mono_idx[1] =
s->frame_end;
187 if (mono_duration >=
s->duration) {
190 s->start_mono_presence = 0;
193 if (
s->is_mono && !mono_measurement) {
194 s->mono_idx[1] = insamples ? insamples->
pts :
s->frame_end;
196 if (mono_duration >=
s->duration) {
209 int64_t out_phase_duration;
210 if (!
s->is_out_phase && out_phase_measurement) {
212 s->start_out_phase_presence = 1;
213 s->out_phase_idx[0] = insamples->
pts;
215 if (
s->is_out_phase && out_phase_measurement &&
s->start_out_phase_presence) {
216 s->out_phase_idx[1] =
s->frame_end;
218 if (out_phase_duration >=
s->duration) {
221 s->start_out_phase_presence = 0;
224 if (
s->is_out_phase && !out_phase_measurement) {
225 s->out_phase_idx[1] = insamples ? insamples->
pts :
s->frame_end;
227 if (out_phase_duration >=
s->duration) {
245 const int rc =
s->contrast[0];
246 const int gc =
s->contrast[1];
247 const int bc =
s->contrast[2];
252 int mono_measurement;
253 int out_phase_measurement;
254 float tolerance = 1.0f -
s->tolerance;
255 float angle =
cosf(
s->angle/180.0f*
M_PI);
257 if (
s->do_video && (!
s->out ||
s->out->width != outlink->
w ||
258 s->out->height != outlink->
h)) {
267 for (
i = 0;
i < outlink->
h;
i++)
268 memset(
out->data[0] +
i *
out->linesize[0], 0, outlink->
w * 4);
269 }
else if (
s->do_video) {
271 for (
i = outlink->
h - 1;
i >= 10;
i--)
272 memmove(
out->data[0] + (
i ) *
out->linesize[0],
273 out->data[0] + (
i-1) *
out->linesize[0],
275 for (
i = 0;
i < outlink->
w;
i++)
279 for (
i = 0;
i <
in->nb_samples;
i++) {
280 const float *
src = (
float *)
in->data[0] +
i * 2;
282 const float phase =
isnan(
f) ? 1 :
f;
283 const int x =
get_x(phase,
s->w);
286 dst =
out->data[0] + x * 4;
287 dst[0] =
FFMIN(255, dst[0] + rc);
288 dst[1] =
FFMIN(255, dst[1] + gc);
289 dst[2] =
FFMIN(255, dst[2] + bc);
294 fphase /=
in->nb_samples;
298 if (
s->draw_median_phase) {
299 dst =
out->data[0] +
get_x(fphase,
s->w) * 4;
303 for (
i = 1;
i < 10 &&
i < outlink->
h;
i++)
304 memcpy(
out->data[0] +
i *
out->linesize[0],
out->data[0], outlink->
w * 4);
307 metadata = &
in->metadata;
315 if (
s->do_phasing_detection) {
316 s->time_base =
inlink->time_base;
320 mono_measurement = (tolerance - fphase) < FLT_EPSILON;
321 out_phase_measurement = (angle - fphase) > FLT_EPSILON;
330 s->out->pts =
in->pts;
343 if (
s->do_phasing_detection) {
389 .
name =
"aphasemeter",
397 .priv_class = &aphasemeter_class,
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
static const AVFilterPad inputs[]
static int config_video_output(AVFilterLink *outlink)
A list of supported channel layouts.
AVPixelFormat
Pixel format.
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
static av_cold int init(AVFilterContext *ctx)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
int av_parse_color(uint8_t *rgba_color, const char *color_string, int slen, void *log_ctx)
Put the RGBA values that correspond to color_string in rgba_color.
@ AV_OPT_TYPE_VIDEO_RATE
offset must point to AVRational
AVFilter ff_avf_aphasemeter
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
This structure describes decoded (raw) audio or video data.
static int config_input(AVFilterLink *inlink)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
const char * name
Filter name.
A link between two filters.
int start_out_phase_presence
#define AV_CH_LAYOUT_STEREO
A filter pad used for either input or output.
static void update_out_phase_detection(AudioPhaseMeterContext *s, AVFrame *insamples, int out_phase_measurement)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVRational sample_aspect_ratio
agreed upon sample aspect ratio
static const AVFilterPad outputs[]
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0,...
static enum AVPixelFormat pix_fmts[]
static int query_formats(AVFilterContext *ctx)
AVFrame * av_frame_clone(const AVFrame *src)
Create a new frame that references the same data as src.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
@ AV_PIX_FMT_RGBA
packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
Describe the class of an AVClass context structure.
Rational number (pair of numerator and denominator).
static const AVOption aphasemeter_options[]
@ AV_OPT_TYPE_IMAGE_SIZE
offset must point to two consecutive integers
static void update_mono_detection(AudioPhaseMeterContext *s, AVFrame *insamples, int mono_measurement)
AVFILTER_DEFINE_CLASS(aphasemeter)
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
static int get_x(float phase, int w)
#define av_ts2timestr(ts, tb)
Convenience macro, the return value should be used only directly in function arguments but never stan...
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
AVFilterContext * src
source filter
AVFilterFormatsConfig incfg
Lists of supported formats / etc.
#define AV_LOG_INFO
Standard information.
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 layout
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
int w
agreed upon image width
#define AV_TIME_BASE
Internal time base represented as integer.
AVSampleFormat
Audio sample formats.
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
const char * name
Pad name.
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
static av_cold void uninit(AVFilterContext *ctx)
int h
agreed upon image height
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
AVDictionary * metadata
metadata.
static void add_metadata(AVFrame *insamples, const char *key, char *value)
#define get_duration(index)
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
#define flags(name, subs,...)