36 #define FREQUENCY_DOMAIN 1 99 char *
arg, *tokenizer, *p;
100 uint64_t used_channels = 0;
103 while ((arg =
av_strtok(p,
"|", &tokenizer))) {
104 uint64_t out_channel;
111 if (used_channels & out_channel) {
115 used_channels |= out_channel;
144 const float *
const ir = td->
ir[jobnr];
150 const float *
src = (
const float *)in->
data[0];
151 float *dst = (
float *)
out->data[0];
152 const int in_channels = in->
channels;
154 const uint32_t modulo = (uint32_t)buffer_length - 1;
161 for (l = 0; l < in_channels; l++) {
166 const float *cur_ir = ir;
169 for (l = 0; l < in_channels; l++) {
170 *(buffer[l] + wr) = src[l];
173 for (l = 0; l < in_channels; cur_ir +=
air_len, l++) {
174 const float *
const bptr = buffer[l];
181 read = (wr - (ir_len - 1)) & modulo;
183 if (read + ir_len < buffer_length) {
184 memcpy(temp_src, bptr + read, ir_len *
sizeof(*temp_src));
186 int len =
FFMIN(air_len - (read % ir_len), buffer_length - read);
188 memcpy(temp_src, bptr + read, len *
sizeof(*temp_src));
189 memcpy(temp_src + len, bptr, (air_len - len) *
sizeof(*temp_src));
195 if (
fabsf(dst[0]) > 1)
200 wr = (wr + 1) & modulo;
219 const float *
src = (
const float *)in->
data[0];
220 float *dst = (
float *)
out->data[0];
221 const int in_channels = in->
channels;
223 const uint32_t modulo = (uint32_t)buffer_length - 1;
229 const float fft_scale = 1.0f / s->
n_fft;
238 for (j = 0; j < n_read; j++) {
239 dst[2 * j] = ringbuffer[wr];
240 ringbuffer[wr] = 0.0;
241 wr = (wr + 1) & modulo;
248 memset(fft_acc, 0,
sizeof(
FFTComplex) * n_fft);
250 for (i = 0; i < in_channels; i++) {
253 dst[2 * j] += src[i + j * in_channels] * s->
gain_lfe;
259 hrtf_offset = hrtf +
offset;
261 memset(fft_in, 0,
sizeof(
FFTComplex) * n_fft);
264 fft_in[j].
re = src[j * in_channels +
i];
269 for (j = 0; j <
n_fft; j++) {
271 const float re = fft_in[j].
re;
272 const float im = fft_in[j].
im;
274 fft_acc[j].
re += re * hcomplex->
re - im * hcomplex->
im;
275 fft_acc[j].
im += re * hcomplex->
im + im * hcomplex->
re;
283 dst[2 * j] += fft_acc[j].
re * fft_scale;
284 if (
fabsf(dst[2 * j]) > 1)
288 for (j = 0; j < ir_len - 1; j++) {
289 int write_pos = (wr + j) & modulo;
291 *(ringbuffer + write_pos) += fft_acc[in->
nb_samples + j].
re * fft_scale;
307 if (ir_len > max_ir_len) {
320 int n_clippings[2] = { 0 };
344 if (n_clippings[0] + n_clippings[1] > 0) {
346 n_clippings[0] + n_clippings[1], out->
nb_samples * 2);
358 float gain_lin =
expf((s->
gain - 3 * nb_input_channels) / 20 *
M_LN10);
443 for (j = 0; j <
len; j++) {
444 data_ir_l[j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
445 data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
451 for (j = 0; j <
len; j++) {
452 fft_in_l[j].
re = ptr[j * 2 ] * gain_lin;
453 fft_in_r[j].
re = ptr[j * 2 + 1] * gain_lin;
464 for (k = 0; k < N / 2; k++) {
475 for (j = 0; j <
len; j++) {
476 data_ir_l[j] = ptr[len * N - j * N - N + I ] * gain_lin;
477 data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
483 for (j = 0; j <
len; j++) {
484 fft_in_l[j].
re = ptr[j * N + I ] * gain_lin;
485 fft_in_r[j].
re = ptr[j * N + I + 1] * gain_lin;
526 "HRIR stream %d.\n", i);
719 #define OFFSET(x) offsetof(HeadphoneContext, x) 720 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM 749 .description =
NULL_IF_CONFIG_SMALL(
"Apply headphone binaural spatialization with HRTFs in additional streams."),
751 .priv_class = &headphone_class,
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
static int config_output(AVFilterLink *outlink)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
This structure describes decoded (raw) audio or video data.
av_cold void av_fft_end(FFTContext *s)
static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
#define AV_LOG_WARNING
Something somehow does not look correct.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Main libavfilter public API header.
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
FF_FILTER_FORWARD_STATUS(inlink, outlink)
static int parse_channel_name(const char *arg, uint64_t *rchannel)
void av_fft_permute(FFTContext *s, FFTComplex *z)
Do the permutation needed BEFORE calling ff_fft_calc().
AVFilterFormatsConfig outcfg
Lists of supported formats / etc.
#define AV_CH_LAYOUT_STEREO
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
static int activate(AVFilterContext *ctx)
const char * name
Pad name.
uint64_t av_get_channel_layout(const char *name)
Return a channel layout id that matches name, or 0 if no match is found.
AVFilterLink ** inputs
array of pointers to input links
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
#define AV_CH_LOW_FREQUENCY
#define AVERROR_EOF
End of file.
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
static __device__ float fabsf(float a)
static int config_input(AVFilterLink *inlink)
A filter pad used for either input or output.
A link between two filters.
AVFilterPad * input_pads
array of input pads
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
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
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
static void parse_map(AVFilterContext *ctx)
static const AVFilterPad outputs[]
FFTContext * av_fft_init(int nbits, int inverse)
Set up a complex FFT.
static av_cold void uninit(AVFilterContext *ctx)
AVFILTER_DEFINE_CLASS(headphone)
char * av_asprintf(const char *fmt,...)
static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
FFTComplex * data_hrtf[2]
int channels
number of audio channels, only used for audio.
audio channel layout utility functions
unsigned nb_inputs
number of input pads
int ff_inlink_queued_samples(AVFilterLink *link)
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
static int query_formats(AVFilterContext *ctx)
AVFilterContext * src
source filter
static const AVOption headphone_options[]
A list of supported channel layouts.
static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
Used for passing data between threads.
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
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
Describe the class of an AVClass context structure.
int ff_outlink_get_status(AVFilterLink *link)
Get the status on an output link.
const char * name
Filter name.
FFTComplex * temp_afft[2]
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
int av_get_channel_layout_channel_index(uint64_t channel_layout, uint64_t channel)
Get the index of a channel in channel_layout.
static av_cold int init(AVFilterContext *ctx)
#define flags(name, subs,...)
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
struct HeadphoneContext::hrir_inputs hrir_in[64]
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
int channels
Number of channels.
AVFilterFormatsConfig incfg
Lists of supported formats / etc.
avfilter_execute_func * execute
AVFilterContext * dst
dest filter
static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
static int check_ir(AVFilterLink *inlink, int input_number)
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
uint8_t ** extended_data
pointers to the data planes/channels.
void av_fft_calc(FFTContext *s, FFTComplex *z)
Do a complex FFT with the parameters defined in av_fft_init().
int nb_samples
number of audio samples (per channel) described by this frame
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.