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70 #define OFFSET(x) offsetof(SidechainCompressContext, x)
71 #define A AV_OPT_FLAG_AUDIO_PARAM
72 #define F AV_OPT_FLAG_FILTERING_PARAM
96 #define sidechaincompress_options options
100 #define FAKE_INFINITY (65536.0 * 65536.0)
103 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
105 static double output_gain(
double lin_slope,
double ratio,
double thres,
106 double knee,
double knee_start,
double knee_stop,
107 double compressed_knee_start,
108 double compressed_knee_stop,
109 int detection,
int mode)
111 double slope = log(lin_slope);
122 gain = (slope - thres) / ratio + thres;
127 if (knee > 1.0 && slope > knee_start)
129 knee_stop, compressed_knee_start,
132 if (knee > 1.0 && slope < knee_stop)
134 knee_start, compressed_knee_stop,
138 return exp(gain - slope);
146 s->thres = log(
s->threshold);
147 s->lin_knee_start =
s->threshold / sqrt(
s->knee);
148 s->lin_knee_stop =
s->threshold * sqrt(
s->knee);
149 s->adj_knee_start =
s->lin_knee_start *
s->lin_knee_start;
150 s->adj_knee_stop =
s->lin_knee_stop *
s->lin_knee_stop;
151 s->knee_start = log(
s->lin_knee_start);
152 s->knee_stop = log(
s->lin_knee_stop);
153 s->compressed_knee_start = (
s->knee_start -
s->thres) /
s->ratio +
s->thres;
154 s->compressed_knee_stop = (
s->knee_stop -
s->thres) /
s->ratio +
s->thres;
163 const double *
src,
double *dst,
const double *scsrc,
int nb_samples,
164 double level_in,
double level_sc,
167 const double makeup =
s->makeup;
168 const double mix =
s->mix;
171 for (
i = 0;
i < nb_samples;
i++) {
172 double abs_sample, gain = 1.0;
176 abs_sample = fabs(scsrc[0] * level_sc);
180 abs_sample =
FFMAX(fabs(scsrc[
c] * level_sc), abs_sample);
183 abs_sample += fabs(scsrc[
c] * level_sc);
189 abs_sample *= abs_sample;
191 s->lin_slope += (abs_sample -
s->lin_slope) * (abs_sample >
s->lin_slope ?
s->attack_coeff :
s->release_coeff);
194 detector = (
s->detection ?
s->adj_knee_stop :
s->lin_knee_stop);
195 detected =
s->lin_slope < detector;
197 detector = (
s->detection ?
s->adj_knee_start :
s->lin_knee_start);
198 detected =
s->lin_slope > detector;
201 if (
s->lin_slope > 0.0 && detected)
203 s->knee_start,
s->knee_stop,
204 s->compressed_knee_start,
205 s->compressed_knee_stop,
206 s->detection,
s->mode);
209 dst[
c] =
src[
c] * level_in * (gain * makeup *
mix + (1. -
mix));
217 #if CONFIG_SIDECHAINCOMPRESS_FILTER
222 int ret,
i, nb_samples;
246 for (
i = 0;
i < 2;
i++) {
257 dst = (
double *)
out->data[0];
259 s->pts += nb_samples;
262 (
double *)
in[1]->
data[0], nb_samples,
263 s->level_in,
s->level_sc,
264 ctx->inputs[0],
ctx->inputs[1]);
294 if (!
ctx->inputs[0]->in_channel_layouts ||
295 !
ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) {
297 "No channel layout for input 1\n");
305 for (
i = 0;
i < 2;
i++) {
324 if (
ctx->inputs[0]->sample_rate !=
ctx->inputs[1]->sample_rate) {
326 "Inputs must have the same sample rate "
327 "%d for in0 vs %d for in1\n",
328 ctx->inputs[0]->sample_rate,
ctx->inputs[1]->sample_rate);
339 if (!
s->fifo[0] || !
s->fifo[1])
355 static const AVFilterPad sidechaincompress_inputs[] = {
366 static const AVFilterPad sidechaincompress_outputs[] = {
376 .
name =
"sidechaincompress",
379 .priv_class = &sidechaincompress_class,
383 .
inputs = sidechaincompress_inputs,
384 .
outputs = sidechaincompress_outputs,
388 #if CONFIG_ACOMPRESSOR_FILTER
391 const double *
src = (
const double *)
in->data[0];
408 dst = (
double *)
out->data[0];
411 s->level_in,
s->level_in,
449 #define acompressor_options options
456 .filter_frame = acompressor_filter_frame,
471 .
name =
"acompressor",
474 .priv_class = &acompressor_class,
476 .
inputs = acompressor_inputs,
477 .
outputs = acompressor_outputs,
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
A list of supported channel layouts.
#define AV_LOG_WARNING
Something somehow does not look correct.
static int mix(int c0, int c1)
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
static enum AVSampleFormat sample_fmts[]
enum MovChannelLayoutTag * layouts
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
This structure describes decoded (raw) audio or video data.
AVFILTER_DEFINE_CLASS(sidechaincompress)
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double compressed_knee_start, double compressed_knee_stop, int detection, int mode)
const char * name
Filter name.
A link between two filters.
int channels
Number of channels.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Context for an Audio FIFO Buffer.
static int config_output(AVFilterLink *outlink)
A filter pad used for either input or output.
static const AVOption options[]
static void compressor(SidechainCompressContext *s, const double *src, double *dst, const double *scsrc, int nb_samples, double level_in, double level_sc, AVFilterLink *inlink, AVFilterLink *sclink)
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
static double hermite_interpolation(double x, double x0, double x1, double p0, double p1, double m0, double m1)
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
static const AVFilterPad outputs[]
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 link
Describe the class of an AVClass context structure.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
filter_frame For filters that do not use the activate() callback
uint64_t channel_layout
channel layout of current buffer (see libavutil/channel_layout.h)
AVFilter ff_af_sidechaincompress
double compressed_knee_stop
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
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
AVFilterContext * src
source filter
double compressed_knee_start
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
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 sample_rate
samples per second
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
static int compressor_config_output(AVFilterLink *outlink)
#define i(width, name, range_min, range_max)
AVSampleFormat
Audio sample formats.
const char * name
Pad name.
AVFilter ff_af_acompressor
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
FF_FILTER_FORWARD_STATUS(inlink, outlink)
#define IS_FAKE_INFINITY(value)
static int query_formats(AVFilterContext *ctx)
static av_cold int uninit(AVCodecContext *avctx)
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
@ AV_SAMPLE_FMT_DBL
double