44 "NB_CONSUMED_SAMPLES",
176 #define d2istr(v) double2int64str((char[BUF_SIZE]){0}, v)
188 "N:%"PRId64
" PTS:%s T:%f POS:%s",
193 switch (inlink->
type) {
264 #define OFFSET(x) offsetof(SetPTSContext, x)
265 #define V AV_OPT_FLAG_VIDEO_PARAM
266 #define A AV_OPT_FLAG_AUDIO_PARAM
267 #define F AV_OPT_FLAG_FILTERING_PARAM
269 #if CONFIG_SETPTS_FILTER
270 static const AVOption setpts_options[] = {
271 {
"expr",
"Expression determining the frame timestamp",
OFFSET(expr_str),
AV_OPT_TYPE_STRING, { .str =
"PTS" }, .flags =
V|
F },
276 static const AVFilterPad avfilter_vf_setpts_inputs[] = {
285 static const AVFilterPad avfilter_vf_setpts_outputs[] = {
301 .priv_class = &setpts_class,
303 .
inputs = avfilter_vf_setpts_inputs,
304 .
outputs = avfilter_vf_setpts_outputs,
308 #if CONFIG_ASETPTS_FILTER
310 static const AVOption asetpts_options[] = {
311 {
"expr",
"Expression determining the frame timestamp",
OFFSET(expr_str),
AV_OPT_TYPE_STRING, { .str =
"PTS" }, .flags =
A|
F },
340 .priv_class = &asetpts_class,
static const AVFilterPad inputs[]
static const AVFilterPad outputs[]
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_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
Main libavfilter public API header.
void av_expr_free(AVExpr *e)
Free a parsed expression previously created with av_expr_parse().
double av_expr_eval(AVExpr *e, const double *const_values, void *opaque)
Evaluate a previously parsed expression.
int av_expr_parse(AVExpr **expr, const char *s, const char *const *const_names, const char *const *func1_names, double(*const *funcs1)(void *, double), const char *const *func2_names, double(*const *funcs2)(void *, double, double), int log_offset, void *log_ctx)
Parse an expression.
simple arithmetic expression evaluator
#define FF_FILTER_FORWARD_WANTED(outlink, inlink)
Forward the frame_wanted_out flag from an output link to an input link.
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
#define FFERROR_NOT_READY
Filters implementation helper functions.
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
#define AV_LOG_TRACE
Extremely verbose debugging, useful for libav* development.
#define AV_LOG_VERBOSE
Detailed information.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static double av_q2d(AVRational a)
Convert an AVRational to a double.
#define AVFILTER_DEFINE_CLASS(fname)
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
@ VAR_NB_CONSUMED_SAMPLES
static char * double2int64str(char *buf, double v)
static int config_input(AVFilterLink *inlink)
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
static double eval_pts(SetPTSContext *setpts, AVFilterLink *inlink, AVFrame *frame, int64_t pts)
static const char *const var_names[]
static int activate(AVFilterContext *ctx)
static av_cold int init(AVFilterContext *ctx)
static av_cold void uninit(AVFilterContext *ctx)
Describe the class of an AVClass context structure.
void * priv
private data for use by the filter
AVFilterLink ** outputs
array of pointers to output links
A link between two filters.
enum AVMediaType type
filter media type
AVFilterContext * src
source filter
AVRational time_base
Define the time base used by the PTS of the frames/samples which will pass through this link.
int sample_rate
samples per second
AVRational frame_rate
Frame rate of the stream on the link, or 1/0 if unknown or variable; if left to 0/0,...
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.
int nb_samples
number of audio samples (per channel) described by this frame
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
int64_t pkt_pos
reordered pos from the last AVPacket that has been input into the decoder
int interlaced_frame
The content of the picture is interlaced.
double var_values[VAR_VARS_NB]
int64_t av_gettime(void)
Get the current time in microseconds.