70 #define OFFSET(x) offsetof(SidechainCompressContext, x) 71 #define A AV_OPT_FLAG_AUDIO_PARAM 72 #define F AV_OPT_FLAG_FILTERING_PARAM 73 #define R AV_OPT_FLAG_RUNTIME_PARAM 97 #define sidechaincompress_options options 101 #define FAKE_INFINITY (65536.0 * 65536.0) 104 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0) 112 double slope = log(lin_slope);
123 gain = (slope -
thres) / ratio + thres;
128 if (knee > 1.0 && slope > knee_start)
130 knee_stop, compressed_knee_start,
133 if (knee > 1.0 && slope < knee_stop)
135 knee_start, compressed_knee_stop,
139 return exp(gain - slope);
164 const double *
src,
double *dst,
const double *scsrc,
int nb_samples,
169 const double mix = s->
mix;
172 for (i = 0; i < nb_samples; i++) {
173 double abs_sample, gain = 1.0;
177 abs_sample =
fabs(scsrc[0] * level_sc);
180 for (c = 1; c < sclink->
channels; c++)
181 abs_sample =
FFMAX(
fabs(scsrc[c] * level_sc), abs_sample);
183 for (c = 1; c < sclink->
channels; c++)
184 abs_sample +=
fabs(scsrc[c] * level_sc);
190 abs_sample *= abs_sample;
209 for (c = 0; c < inlink->
channels; c++)
210 dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix));
219 char *res,
int res_len,
int flags)
232 #if CONFIG_SIDECHAINCOMPRESS_FILTER 237 int ret,
i, nb_samples;
261 for (i = 0; i < 2; i++) {
272 dst = (
double *)out->
data[0];
277 (
double *)in[1]->
data[0], nb_samples,
312 "No channel layout for input 1\n");
320 for (i = 0; i < 2; i++) {
341 "Inputs must have the same sample rate " 342 "%d for in0 vs %d for in1\n",
370 static const AVFilterPad sidechaincompress_inputs[] = {
381 static const AVFilterPad sidechaincompress_outputs[] = {
391 .
name =
"sidechaincompress",
394 .priv_class = &sidechaincompress_class,
398 .
inputs = sidechaincompress_inputs,
399 .
outputs = sidechaincompress_outputs,
404 #if CONFIG_ACOMPRESSOR_FILTER 407 const double *
src = (
const double *)in->
data[0];
424 dst = (
double *)
out->data[0];
427 s->level_in,
s->level_in,
465 #define acompressor_options options 472 .filter_frame = acompressor_filter_frame,
487 .
name =
"acompressor",
490 .priv_class = &acompressor_class,
492 .
inputs = acompressor_inputs,
493 .
outputs = acompressor_outputs,
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
This structure describes decoded (raw) audio or video data.
ptrdiff_t const GLvoid * data
#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.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
FF_FILTER_FORWARD_STATUS(inlink, outlink)
AVFilter ff_af_sidechaincompress
AVFilterFormatsConfig outcfg
Lists of supported formats / etc.
static int compressor_config_output(AVFilterLink *outlink)
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
const char * name
Pad name.
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.
static av_cold int uninit(AVCodecContext *avctx)
filter_frame For filters that do not use the activate() callback
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
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
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
A filter pad used for either input or output.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
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.
static __device__ float fabs(float a)
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. ...
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...
void * priv
private data for use by the filter
uint64_t * channel_layouts
list of channel layouts
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link...
simple assert() macros that are a bit more flexible than ISO C assert().
AVFILTER_DEFINE_CLASS(sidechaincompress)
Context for an Audio FIFO Buffer.
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
audio channel layout utility functions
static double hermite_interpolation(double x, double x0, double x1, double p0, double p1, double m0, double m1)
AVFilterContext * src
source filter
double compressed_knee_start
static const AVFilterPad outputs[]
int format
agreed upon media format
A list of supported channel layouts.
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)
int nb_samples
number of samples currently in the FIFO
#define IS_FAKE_INFINITY(value)
AVSampleFormat
Audio sample formats.
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
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)
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
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
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.
Rational number (pair of numerator and denominator).
const char * name
Filter name.
AVFilterLink ** outputs
array of pointers to output links
enum MovChannelLayoutTag * layouts
#define flags(name, subs,...)
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
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
static const AVOption options[]
static int query_formats(AVFilterContext *ctx)
common internal and external API header
int nb_channel_layouts
number of channel layouts
double compressed_knee_stop
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 ff_af_acompressor
AVFilterContext * dst
dest filter
static enum AVSampleFormat sample_fmts[]
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
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.
static int config_output(AVFilterLink *outlink)