65 alpha = sin(w0) / 2 * sqrt((
A + 1 /
A) * (1 /
s->slope - 1) + 2);
67 s->a0 = (
A + 1) + (
A - 1) * cos(w0) + 2 * sqrt(
A) *
alpha;
68 s->a1 = -2 * ((
A - 1) + (
A + 1) * cos(w0));
69 s->a2 = (
A + 1) + (
A - 1) * cos(w0) - 2 * sqrt(
A) *
alpha;
70 s->b0 =
A * ((
A + 1) - (
A - 1) * cos(w0) + 2 * sqrt(
A) *
alpha);
71 s->b1 = 2 *
A * ((
A - 1) - (
A + 1) * cos(w0));
72 s->b2 =
A * ((
A + 1) - (
A - 1) * cos(w0) - 2 * sqrt(
A) *
alpha);
88 const double *
src = (
const double *)
in->data[0];
89 const double level_in =
s->level_in;
90 const double level_out =
s->level_out;
91 const double b0 =
s->b0;
92 const double b1 =
s->b1;
93 const double b2 =
s->b2;
94 const double a1 = -
s->a1;
95 const double a2 = -
s->a2;
110 dst = (
double *)
out->data[0];
112 for (n = 0; n <
out->nb_samples; n++,
src += 2, dst += 2) {
113 double mid = (
src[0] +
src[1]) * level_in * .5;
114 double side = (
src[0] -
src[1]) * level_in * .5;
115 double oside = side *
b0 +
s->w1;
117 s->w1 =
b1 * side +
s->w2 +
a1 * oside;
118 s->w2 =
b2 * side +
a2 * oside;
120 if (
ctx->is_disabled) {
124 dst[0] = (mid + oside) * level_out;
125 dst[1] = (mid - oside) * level_out;
135 char *res,
int res_len,
int flags)
146 #define OFFSET(x) offsetof(CrossfeedContext, x)
147 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
183 .priv_class = &crossfeed_class,
static const AVOption crossfeed_options[]
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
AVFILTER_DEFINE_CLASS(crossfeed)
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.
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Main libavfilter public API header.
#define flags(name, subs,...)
audio channel layout utility functions
internal math functions header
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_CH_LAYOUT_STEREO
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
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.
@ AV_SAMPLE_FMT_DBL
double
static const int16_t alpha[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Describe the class of an AVClass context structure.
A list of supported channel layouts.
A link between two filters.
int sample_rate
samples per second
AVFilterContext * dst
dest filter
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.
static double b1(void *priv, double x, double y)
static double b2(void *priv, double x, double y)
static double b0(void *priv, double x, double y)