59 #define OFFSET(x) offsetof(StereoToolsContext, x)
60 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
129 s->inv_atan_shape = 1.0 / atan(
s->sc_level);
130 s->phase_cos_coef = cos(
s->phase / 180 *
M_PI);
131 s->phase_sin_coef = sin(
s->phase / 180 *
M_PI);
141 const double *
src = (
const double *)
in->data[0];
142 const double sb =
s->base < 0 ?
s->base * 0.5 :
s->base;
143 const double sbal = 1 +
s->sbal;
144 const double mpan = 1 +
s->mpan;
145 const double slev =
s->slev;
146 const double mlev =
s->mlev;
147 const double balance_in =
s->balance_in;
148 const double balance_out =
s->balance_out;
149 const double level_in =
s->level_in;
150 const double level_out =
s->level_out;
151 const double sc_level =
s->sc_level;
152 const double delay =
s->delay;
153 const int length =
s->length;
154 const int mute_l =
s->mute_l;
155 const int mute_r =
s->mute_r;
156 const int phase_l =
s->phase_l;
157 const int phase_r =
s->phase_r;
175 dst = (
double *)
out->data[0];
177 for (n = 0; n <
in->nb_samples; n++,
src += 2, dst += 2) {
178 double L =
src[0],
R =
src[1], l,
r, m,
S, gl, gr, gd;
183 gl = 1. -
FFMAX(0., balance_in);
184 gr = 1. +
FFMIN(0., balance_in);
185 switch (
s->bmode_in) {
192 if (balance_in < 0.) {
195 }
else if (balance_in > 0.) {
205 R =
s->inv_atan_shape * atan(
R * sc_level);
206 L =
s->inv_atan_shape * atan(
L * sc_level);
213 l = m * mlev *
FFMIN(1., 2. - mpan) +
S * slev *
FFMIN(1., 2. - sbal);
214 r = m * mlev *
FFMIN(1., mpan) -
S * slev *
FFMIN(1., sbal);
219 l =
L *
FFMIN(1., 2. - sbal);
221 L = 0.5 * (l +
r) * mlev;
222 R = 0.5 * (l -
r) * slev;
225 l =
L * mlev *
FFMIN(1., 2. - mpan) +
R * slev *
FFMIN(1., 2. - sbal);
246 l = m * mlev *
FFMIN(1., 2. - mpan) +
S * slev *
FFMIN(1., 2. - sbal);
247 r = m * mlev *
FFMIN(1., mpan) -
S * slev *
FFMIN(1., sbal);
252 l =
L * mlev *
FFMIN(1., 2. - mpan) +
R * slev *
FFMIN(1., 2. - sbal);
262 l =
L * mlev *
FFMIN(1., 2. - mpan) +
R * slev *
FFMIN(1., 2. - sbal);
276 L *= (2. * (1. - phase_l)) - 1.;
277 R *= (2. * (1. - phase_r)) - 1.;
283 R =
buffer[(
s->pos - (
int)nbuf + 1 + length) % length];
284 }
else if (delay < 0.) {
285 L =
buffer[(
s->pos - (
int)nbuf + length) % length];
288 l =
L + sb *
L - sb *
R;
289 r =
R + sb *
R - sb *
L;
294 l =
L *
s->phase_cos_coef -
R *
s->phase_sin_coef;
295 r =
L *
s->phase_sin_coef +
R *
s->phase_cos_coef;
300 s->pos = (
s->pos + 2) %
s->length;
302 gl = 1. -
FFMAX(0., balance_out);
303 gr = 1. +
FFMIN(0., balance_out);
304 switch (
s->bmode_out) {
311 if (balance_out < 0.) {
314 }
else if (balance_out > 0.) {
327 if (
ctx->is_disabled) {
342 char *res,
int res_len,
int flags)
379 .
name =
"stereotools",
383 .priv_class = &stereotools_class,
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
static __device__ float fabs(float a)
mode
Use these values in ebur128_init (or'ed).
#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.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
@ AV_SAMPLE_FMT_DBL
double
#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.