49 #define OFFSET(x) offsetof(AudioPhaseMeterContext, x) 50 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM 128 if (!strcmp(s->
mpc_str,
"none"))
138 static inline int get_x(
float phase,
int w)
140 return (phase + 1.) / 2. * (w - 1);
168 for (i = 0; i < outlink->
h; i++)
169 memset(out->
data[0] + i * out->
linesize[0], 0, outlink->
w * 4);
172 for (i = outlink->
h - 1; i >= 10; i--)
176 for (i = 0; i < outlink->
w; i++)
181 const float *
src = (
float *)in->
data[0] + i * 2;
182 const float f = src[0] * src[1] / (src[0]*src[0] + src[1] * src[1]) * 2;
183 const float phase =
isnan(f) ? 1 : f;
184 const int x =
get_x(phase, s->
w);
187 dst = out->
data[0] + x * 4;
188 dst[0] =
FFMIN(255, dst[0] + rc);
189 dst[1] =
FFMIN(255, dst[1] + gc);
190 dst[2] =
FFMIN(255, dst[2] + bc);
203 for (i = 1; i < 10 && i < outlink->
h; i++)
211 snprintf(value,
sizeof(value),
"%f", fphase);
212 av_dict_set(metadata,
"lavfi.aphasemeter.phase", value, 0);
270 .
name =
"aphasemeter",
278 .priv_class = &aphasemeter_class,
This structure describes decoded (raw) audio or video data.
AVFILTER_DEFINE_CLASS(aphasemeter)
Main libavfilter public API header.
int max_samples
Maximum number of samples to filter at once.
static int get_x(float phase, int w)
int h
agreed upon image height
static av_cold int init(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
static const AVOption aphasemeter_options[]
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
struct AVFilterChannelLayouts * in_channel_layouts
static int query_formats(AVFilterContext *ctx)
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.
AVFilterPad * output_pads
array of output pads
static int config_video_output(AVFilterLink *outlink)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
static double av_q2d(AVRational a)
Convert an AVRational to a double.
AVDictionary * metadata
metadata.
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
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.
A filter pad used for either input or output.
A link between two filters.
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.
unsigned nb_outputs
number of output pads
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
#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.
packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
int w
agreed upon image width
audio channel layout utility functions
AVFilterContext * src
source filter
int partial_buf_size
Size of the partial buffer to allocate.
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[]
A list of supported channel layouts.
AVFilterFormats * in_samplerates
Lists of channel layouts and sample rates used for automatic negotiation.
char * av_strdup(const char *s)
Duplicate a string.
AVSampleFormat
Audio sample formats.
int linesize[AV_NUM_DATA_POINTERS]
For video, size in bytes of each picture line.
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
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
static enum AVPixelFormat pix_fmts[]
static const AVFilterPad inputs[]
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
static av_cold void uninit(AVFilterContext *ctx)
AVFilter ff_avf_aphasemeter
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
AVPixelFormat
Pixel format.
int nb_samples
number of audio samples (per channel) described by this frame
AVFilterFormats * out_formats