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61 #define OFFSET(x) offsetof(SetTBContext, x)
62 #define DEFINE_OPTIONS(filt_name, filt_type) \
63 static const AVOption filt_name##_options[] = { \
64 { "expr", "set expression determining the output timebase", OFFSET(tb_expr), AV_OPT_TYPE_STRING, {.str="intb"}, \
65 .flags=AV_OPT_FLAG_##filt_type##_PARAM|AV_OPT_FLAG_FILTERING_PARAM }, \
66 { "tb", "set expression determining the output timebase", OFFSET(tb_expr), AV_OPT_TYPE_STRING, {.str="intb"}, \
67 .flags=AV_OPT_FLAG_##filt_type##_PARAM|AV_OPT_FLAG_FILTERING_PARAM }, \
92 time_base =
av_d2q(res, INT_MAX);
93 if (time_base.
num <= 0 || time_base.
den <= 0) {
95 "Invalid non-positive values for the timebase num:%d or den:%d.\n",
96 time_base.
num, time_base.
den);
111 int64_t new_pts = orig_pts;
116 inlink ->time_base.num,
inlink ->time_base.den, orig_pts,
160 #if CONFIG_SETTB_FILTER
165 static const AVFilterPad avfilter_vf_settb_inputs[] = {
173 static const AVFilterPad avfilter_vf_settb_outputs[] = {
186 .priv_class = &settb_class,
187 .
inputs = avfilter_vf_settb_inputs,
188 .
outputs = avfilter_vf_settb_outputs,
193 #if CONFIG_ASETTB_FILTER
198 static const AVFilterPad avfilter_af_asettb_inputs[] = {
206 static const AVFilterPad avfilter_af_asettb_outputs[] = {
219 .
inputs = avfilter_af_asettb_inputs,
220 .
outputs = avfilter_af_asettb_outputs,
221 .priv_class = &asettb_class,
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
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
#define DEFINE_OPTIONS(filt_name, filt_type)
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
#define AV_TIME_BASE_Q
Internal time base represented as fractional value.
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
This structure describes decoded (raw) audio or video data.
#define AV_LOG_VERBOSE
Detailed information.
const char * name
Filter name.
static const char *const var_names[]
A link between two filters.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
static int config_output_props(AVFilterLink *outlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
A filter pad used for either input or output.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
static double av_q2d(AVRational a)
Convert an AVRational to a double.
static const AVFilterPad outputs[]
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
static int activate(AVFilterContext *ctx)
Describe the class of an AVClass context structure.
Rational number (pair of numerator and denominator).
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
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
int av_expr_parse_and_eval(double *d, const char *s, const char *const *const_names, const double *const_values, const char *const *func1_names, double(*const *funcs1)(void *, double), const char *const *func2_names, double(*const *funcs2)(void *, double, double), void *opaque, int log_offset, void *log_ctx)
Parse and evaluate an expression.
double var_values[VAR_VARS_NB]
AVFilterContext * src
source filter
FF_FILTER_FORWARD_WANTED(outlink, inlink)
#define AVFILTER_DEFINE_CLASS(fname)
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
static int64_t rescale_pts(AVFilterLink *inlink, AVFilterLink *outlink, int64_t orig_pts)
int w
agreed upon image width
AVRational av_d2q(double d, int max)
Convert a double precision floating point number to a rational.
const char * name
Pad name.
static int av_cmp_q(AVRational a, AVRational b)
Compare two rationals.
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
int h
agreed upon image height
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.