27 #define MAX_HAAS_DELAY 40
56 #define OFFSET(x) offsetof(HaasContext, x)
57 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
63 {
"middle_source",
"set middle source",
OFFSET(par_m_source),
AV_OPT_TYPE_INT, {.i64=2}, 0, 3,
A,
"source" },
103 size_t new_buf_size = 1;
105 while (new_buf_size < min_buf_size)
109 s->buffer =
av_calloc(new_buf_size,
sizeof(*
s->buffer));
113 s->buffer_size = new_buf_size;
116 s->delay[0] = (uint32_t)(
s->par_delay0 * 0.001 * inlink->
sample_rate);
117 s->delay[1] = (uint32_t)(
s->par_delay1 * 0.001 * inlink->
sample_rate);
119 s->phase0 =
s->par_phase0 ? 1.0 : -1.0;
120 s->phase1 =
s->par_phase1 ? 1.0 : -1.0;
122 s->balance_l[0] = (
s->par_balance0 + 1) / 2 *
s->par_gain0 *
s->phase0;
123 s->balance_r[0] = (1.0 - (
s->par_balance0 + 1) / 2) * (
s->par_gain0) *
s->phase0;
124 s->balance_l[1] = (
s->par_balance1 + 1) / 2 *
s->par_gain1 *
s->phase1;
125 s->balance_r[1] = (1.0 - (
s->par_balance1 + 1) / 2) * (
s->par_gain1) *
s->phase1;
135 const double *
src = (
const double *)
in->data[0];
136 const double level_in =
s->level_in;
137 const double level_out =
s->level_out;
138 const uint32_t
mask =
s->buffer_size - 1;
154 dst = (
double *)
out->data[0];
156 for (n = 0; n <
in->nb_samples; n++,
src += 2, dst += 2) {
157 double mid, side[2], side_l, side_r;
158 uint32_t s0_ptr, s1_ptr;
160 switch (
s->par_m_source) {
161 case 0: mid =
src[0];
break;
162 case 1: mid =
src[1];
break;
163 case 2: mid = (
src[0] +
src[1]) * 0.5;
break;
164 case 3: mid = (
src[0] -
src[1]) * 0.5;
break;
171 s0_ptr = (
s->write_ptr +
s->buffer_size -
s->delay[0]) &
mask;
172 s1_ptr = (
s->write_ptr +
s->buffer_size -
s->delay[1]) &
mask;
174 if (
s->par_middle_phase)
177 side[0] =
buffer[s0_ptr] *
s->par_side_gain;
178 side[1] =
buffer[s1_ptr] *
s->par_side_gain;
179 side_l = side[0] *
s->balance_l[0] - side[1] *
s->balance_l[1];
180 side_r = side[1] *
s->balance_r[1] - side[0] *
s->balance_r[0];
182 dst[0] = (mid + side_l) * level_out;
183 dst[1] = (mid + side_r) * level_out;
185 s->write_ptr = (
s->write_ptr + 1) &
mask;
224 .priv_class = &haas_class,
static const AVOption haas_options[]
AVFILTER_DEFINE_CLASS(haas)
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 av_cold void uninit(AVFilterContext *ctx)
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.
audio channel layout utility functions
#define AV_CH_LAYOUT_STEREO
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.
static const uint16_t mask[17]
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.