59 #define OFFSET(x) offsetof(StereoToolsContext, x)
60 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
142 const double *
src = (
const double *)in->
data[0];
143 const double sb = s->
base < 0 ? s->
base * 0.5 : s->
base;
144 const double sbal = 1 + s->
sbal;
145 const double mpan = 1 + s->
mpan;
146 const double slev = s->
slev;
147 const double mlev = s->
mlev;
150 const double level_in = s->
level_in;
152 const double sc_level = s->
sc_level;
153 const double delay = s->
delay;
155 const int mute_l = s->
mute_l;
156 const int mute_r = s->
mute_r;
157 const int phase_l = s->
phase_l;
158 const int phase_r = s->
phase_r;
162 int nbuf = inlink->
sample_rate * (fabs(delay) / 1000.);
176 dst = (
double *)
out->data[0];
179 double L = src[0],
R = src[1], l,
r, m,
S, gl, gr, gd;
184 gl = 1. -
FFMAX(0., balance_in);
185 gr = 1. +
FFMIN(0., balance_in);
193 if (balance_in < 0.) {
196 }
else if (balance_in > 0.) {
214 l = m * mlev *
FFMIN(1., 2. - mpan) + S * slev *
FFMIN(1., 2. - sbal);
215 r = m * mlev *
FFMIN(1., mpan) - S * slev *
FFMIN(1., sbal);
220 l = L *
FFMIN(1., 2. - sbal);
221 r = R *
FFMIN(1., sbal);
222 L = 0.5 * (l +
r) * mlev;
223 R = 0.5 * (l -
r) * slev;
226 l = L * mlev *
FFMIN(1., 2. - mpan) + R * slev *
FFMIN(1., 2. - sbal);
227 r = L * mlev *
FFMIN(1., mpan) - R * slev *
FFMIN(1., sbal);
247 l = m * mlev *
FFMIN(1., 2. - mpan) + S * slev *
FFMIN(1., 2. - sbal);
248 r = m * mlev *
FFMIN(1., mpan) - S * slev *
FFMIN(1., sbal);
253 l = L * mlev *
FFMIN(1., 2. - mpan) + R * slev *
FFMIN(1., 2. - sbal);
258 r = L * mlev *
FFMIN(1., mpan) - R * slev *
FFMIN(1., sbal);
267 L *= (2. * (1. - phase_l)) - 1.;
268 R *= (2. * (1. - phase_r)) - 1.;
275 }
else if (delay < 0.) {
279 l = L + sb * L - sb *
R;
280 r = R + sb * R - sb *
L;
293 gl = 1. -
FFMAX(0., balance_out);
294 gr = 1. +
FFMIN(0., balance_out);
302 if (balance_out < 0.) {
305 }
else if (balance_out > 0.) {
353 .
name =
"stereotools",
357 .priv_class = &stereotools_class,
This structure describes decoded (raw) audio or video data.
Main libavfilter public API header.
#define AV_CH_LAYOUT_STEREO
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
const char * name
Pad name.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
A filter pad used for either input or output.
A link between two filters.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
int sample_rate
samples per second
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
#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
audio channel layout utility functions
A list of supported channel layouts.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
AVFilterContext * dst
dest filter
mode
Use these values in ebur128_init (or'ed).
int nb_samples
number of audio samples (per channel) described by this frame
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.