80 #define PFILTER(name, type, sin, cos, cc) \
81 static void pfilter_channel_## name(AVFilterContext *ctx, \
83 AVFrame *in, AVFrame *out) \
85 AFreqShift *s = ctx->priv; \
86 const int nb_samples = in->nb_samples; \
87 const type *src = (const type *)in->extended_data[ch]; \
88 type *dst = (type *)out->extended_data[ch]; \
89 type *i1 = (type *)s->i1->extended_data[ch]; \
90 type *o1 = (type *)s->o1->extended_data[ch]; \
91 type *i2 = (type *)s->i2->extended_data[ch]; \
92 type *o2 = (type *)s->o2->extended_data[ch]; \
93 const type *c = s->cc; \
94 const type level = s->level; \
95 type shift = s->shift * M_PI; \
96 type cos_theta = cos(shift); \
97 type sin_theta = sin(shift); \
99 for (int n = 0; n < nb_samples; n++) { \
100 type xn1 = src[n], xn2 = src[n]; \
103 for (int j = 0; j < NB_COEFS / 2; j++) { \
104 I = c[j] * (xn1 + o2[j]) - i2[j]; \
112 for (int j = NB_COEFS / 2; j < NB_COEFS; j++) { \
113 Q = c[j] * (xn2 + o2[j]) - i2[j]; \
120 Q = o2[NB_COEFS - 1]; \
122 dst[n] = (I * cos_theta - Q * sin_theta) * level; \
126 PFILTER(flt,
float, sin, cos, cf)
127 PFILTER(dbl,
double, sin, cos, cd)
129 #define FFILTER(name, type, sin, cos, fmod, cc) \
130 static void ffilter_channel_## name(AVFilterContext *ctx, \
132 AVFrame *in, AVFrame *out) \
134 AFreqShift *s = ctx->priv; \
135 const int nb_samples = in->nb_samples; \
136 const type *src = (const type *)in->extended_data[ch]; \
137 type *dst = (type *)out->extended_data[ch]; \
138 type *i1 = (type *)s->i1->extended_data[ch]; \
139 type *o1 = (type *)s->o1->extended_data[ch]; \
140 type *i2 = (type *)s->i2->extended_data[ch]; \
141 type *o2 = (type *)s->o2->extended_data[ch]; \
142 const type *c = s->cc; \
143 const type level = s->level; \
144 type ts = 1. / in->sample_rate; \
145 type shift = s->shift; \
146 int64_t N = s->in_samples; \
148 for (int n = 0; n < nb_samples; n++) { \
149 type xn1 = src[n], xn2 = src[n]; \
152 for (int j = 0; j < NB_COEFS / 2; j++) { \
153 I = c[j] * (xn1 + o2[j]) - i2[j]; \
161 for (int j = NB_COEFS / 2; j < NB_COEFS; j++) { \
162 Q = c[j] * (xn2 + o2[j]) - i2[j]; \
169 Q = o2[NB_COEFS - 1]; \
171 theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.); \
172 dst[n] = (I * cos(theta) - Q * sin(theta)) * level; \
177 FFILTER(dbl,
double, sin, cos, fmod, cd)
181 double kksqrt, e, e2, e4, k, q;
183 k = tan((1. - transition * 2.) *
M_PI / 4.);
185 kksqrt = pow(1 - k * k, 0.25);
186 e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
189 q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));
218 q_ii1 *= sin((
i * 2 + 1) *
c *
M_PI / order) * j;
223 }
while (
fabs(q_ii1) > 1e-100);
237 q_i2 *= cos(
i * 2 *
c *
M_PI / order) * j;
242 }
while (
fabs(q_i2) > 1e-100);
252 const double ww = num / den;
253 const double wwsq = ww * ww;
255 const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
256 const double coef = (1 - x) / (1 + x);
261 static void compute_coefs(
double *coef_arrd,
float *coef_arrf,
int nbr_coefs,
double transition)
263 const int order = nbr_coefs * 2 + 1;
268 for (
int n = 0; n < nbr_coefs; n++) {
269 const int idx = (n / 2) + (n & 1) * nbr_coefs / 2;
272 coef_arrf[idx] = coef_arrd[idx];
287 if (!
s->i1 || !
s->o1 || !
s->i2 || !
s->o2)
291 if (!strcmp(
ctx->filter->name,
"afreqshift"))
292 s->filter_channel = ffilter_channel_dbl;
294 s->filter_channel = pfilter_channel_dbl;
296 if (!strcmp(
ctx->filter->name,
"afreqshift"))
297 s->filter_channel = ffilter_channel_flt;
299 s->filter_channel = pfilter_channel_flt;
315 const int start = (
in->
channels * jobnr) / nb_jobs;
316 const int end = (
in->
channels * (jobnr+1)) / nb_jobs;
318 for (
int ch = start; ch < end; ch++)
347 s->in_samples +=
in->nb_samples;
364 #define OFFSET(x) offsetof(AFreqShift, x)
365 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
394 .
name =
"afreqshift",
398 .priv_class = &afreqshift_class,
416 .
name =
"aphaseshift",
420 .priv_class = &aphaseshift_class,
static enum AVSampleFormat sample_fmts[]
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
#define FFILTER(name, type, sin, cos, fmod, cc)
static double compute_coef(int index, double k, double q, int order)
static void compute_coefs(double *coef_arrd, float *coef_arrf, int nbr_coefs, double transition)
static int query_formats(AVFilterContext *ctx)
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
AVFilter ff_af_afreqshift
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
static const AVOption aphaseshift_options[]
static const AVFilterPad outputs[]
#define PFILTER(name, type, sin, cos, cc)
AVFILTER_DEFINE_CLASS(afreqshift)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static const AVOption afreqshift_options[]
AVFilter ff_af_aphaseshift
static void compute_transition_param(double *K, double *Q, double transition)
static av_cold void uninit(AVFilterContext *ctx)
static double compute_acc_den(double q, int order, int c)
static double compute_acc_num(double q, int order, int c)
static double ipowp(double x, int64_t n)
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.
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Main libavfilter public API header.
#define flags(name, subs,...)
audio channel layout utility functions
static __device__ float fabs(float a)
channel
Use these values when setting the channel map with ebur128_set_channel().
internal math functions header
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
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.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
@ AV_SAMPLE_FMT_DBLP
double, planar
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
enum MovChannelLayoutTag * layouts
typedef void(RENAME(mix_any_func_type))
static int shift(int a, int b)
void(* filter_channel)(AVFilterContext *ctx, int channel, AVFrame *in, AVFrame *out)
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
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.
AVFormatInternal * internal
An opaque field for libavformat internal usage.
This structure describes decoded (raw) audio or video data.
Used for passing data between threads.