42 #define OFFSET(x) offsetof(CompensationDelayContext, x)
43 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58 #define COMP_DELAY_MAX_DISTANCE (100.0 * 100.0 + 100.0 * 1.0 + 1.0)
60 #define COMP_DELAY_SOUND_SPEED_KM_H(temp) 1.85325 * (643.95 * sqrt(((temp + 273.15) / 273.15)))
61 #define COMP_DELAY_SOUND_SPEED_CM_S(temp) (COMP_DELAY_SOUND_SPEED_KM_H(temp) * (1000.0 * 100.0) / (60.0 * 60.0) )
62 #define COMP_DELAY_SOUND_FRONT_DELAY(temp) (1.0 / COMP_DELAY_SOUND_SPEED_CM_S(temp))
64 #define COMP_DELAY_MAX_DELAY (COMP_DELAY_MAX_DISTANCE * COMP_DELAY_SOUND_FRONT_DELAY(50))
100 unsigned min_size, new_size = 1;
102 s->delay = (
s->distance_m * 100. +
s->distance_cm * 1. +
s->distance_mm * .1) *
106 while (new_size < min_size)
113 s->buf_size = new_size;
114 s->delay_frame->format = inlink->
format;
115 s->delay_frame->nb_samples = new_size;
125 const unsigned b_mask =
s->buf_size - 1;
126 const unsigned buf_size =
s->buf_size;
127 const unsigned delay =
s->delay;
128 const double dry =
s->dry;
129 const double wet =
s->wet;
130 unsigned r_ptr, w_ptr = 0;
141 for (ch = 0; ch < inlink->
channels; ch++) {
142 const double *
src = (
const double *)
in->extended_data[ch];
143 double *dst = (
double *)
out->extended_data[ch];
144 double *
buffer = (
double *)
s->delay_frame->extended_data[ch];
147 r_ptr = (w_ptr + buf_size - delay) & b_mask;
149 for (n = 0; n <
in->nb_samples; n++) {
154 w_ptr = (w_ptr + 1) & b_mask;
155 r_ptr = (r_ptr + 1) & b_mask;
190 .
name =
"compensationdelay",
194 .priv_class = &compensationdelay_class,
static enum AVSampleFormat sample_fmts[]
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
#define COMP_DELAY_SOUND_FRONT_DELAY(temp)
static const AVOption compensationdelay_options[]
AVFILTER_DEFINE_CLASS(compensationdelay)
static const AVFilterPad compensationdelay_inputs[]
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
#define COMP_DELAY_MAX_DELAY
static const AVFilterPad compensationdelay_outputs[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static av_cold void uninit(AVFilterContext *ctx)
AVFilter ff_af_compensationdelay
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
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 ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Main libavfilter public API header.
int av_frame_get_buffer(AVFrame *frame, int align)
Allocate new buffer(s) for audio or video data.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_DBLP
double, planar
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int channels
Number of channels.
int sample_rate
samples per second
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
AVFilterContext * dst
dest filter
int format
agreed upon media format
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.