Go to the documentation of this file.
55 (
desc->log2_chroma_w !=
desc->log2_chroma_h &&
56 desc->comp[0].plane ==
desc->comp[1].plane)) &&
68 for (j = 0; j <
w; j++)
74 const uint16_t *
src = (
const uint16_t *)ssrc;
75 uint16_t *dst = (uint16_t *)ddst;
78 for (j = 0; j <
w; j++)
84 const uint32_t *
src = (
const uint32_t *)ssrc;
85 uint32_t *dst = (uint32_t *)ddst;
88 for (j = 0; j <
w; j++)
98 for (j = 0; j <
w; j++,
out += 3,
in -= 3) {
111 for (j = 0; j <
w; j++,
out += 6,
in -= 6) {
120 const uint64_t *
src = (
const uint64_t *)ssrc;
121 uint64_t *dst = (uint64_t *)ddst;
124 for (j = 0; j <
w; j++)
137 s->planewidth[0] =
s->planewidth[3] =
inlink->w;
139 s->planeheight[0] =
s->planeheight[3] =
inlink->h;
151 for (
i = 0;
i < nb_planes;
i++) {
182 for (plane = 0; plane < 4 &&
in->data[plane] &&
in->linesize[plane]; plane++) {
183 const int width =
s->planewidth[plane];
184 const int height =
s->planeheight[plane];
185 const int start = (
height * job ) / nb_jobs;
186 const int end = (
height * (job+1)) / nb_jobs;
188 step =
s->max_step[plane];
190 outrow =
out->data[plane] + start *
out->linesize[plane];
191 inrow =
in ->data[plane] + start *
in->linesize[plane] + (
width - 1) *
step;
192 for (
i = start;
i <
end;
i++) {
193 s->flip_line[plane](inrow, outrow,
width);
195 inrow +=
in ->linesize[plane];
196 outrow +=
out->linesize[plane];
250 .priv_class = &hflip_class,
AVFrame * ff_get_video_buffer(AVFilterLink *link, int w, int h)
Request a picture buffer with a specific set of permissions.
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.
const AVPixFmtDescriptor * av_pix_fmt_desc_get(enum AVPixelFormat pix_fmt)
static const AVOption hflip_options[]
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.
static av_cold int end(AVCodecContext *avctx)
This structure describes decoded (raw) audio or video data.
trying all byte sequences megabyte in length and selecting the best looking sequence will yield cases to try But a word about which is also called distortion Distortion can be quantified by almost any quality measurement one chooses the sum of squared differences is used but more complex methods that consider psychovisual effects can be used as well It makes no difference in this discussion First step
static void hflip_qword_c(const uint8_t *ssrc, uint8_t *ddst, int w)
const char * name
Filter name.
AVFormatInternal * internal
An opaque field for libavformat internal usage.
A link between two filters.
int av_pix_fmt_count_planes(enum AVPixelFormat pix_fmt)
#define AV_PIX_FMT_FLAG_HWACCEL
Pixel format is an HW accelerated format.
A filter pad used for either input or output.
void ff_hflip_init_x86(FlipContext *s, int step[4], int nb_planes)
#define AV_CEIL_RSHIFT(a, b)
static const AVFilterPad outputs[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
static enum AVPixelFormat pix_fmts[]
uint8_t log2_chroma_w
Amount to shift the luma width right to find the chroma width.
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
AVFILTER_DEFINE_CLASS(hflip)
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
uint64_t flags
Combination of AV_PIX_FMT_FLAG_...
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
static void hflip_dword_c(const uint8_t *ssrc, uint8_t *ddst, int w)
#define AV_PIX_FMT_FLAG_BITSTREAM
All values of a component are bit-wise packed end to end.
int ff_hflip_init(FlipContext *s, int step[4], int nb_planes)
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
static void hflip_b24_c(const uint8_t *src, uint8_t *dst, int w)
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
#define i(width, name, range_min, range_max)
int w
agreed upon image width
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Used for passing data between threads.
const char * name
Pad name.
static const AVFilterPad avfilter_vf_hflip_inputs[]
static int config_props(AVFilterLink *inlink)
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo ug o o w
int h
agreed upon image height
void av_image_fill_max_pixsteps(int max_pixsteps[4], int max_pixstep_comps[4], const AVPixFmtDescriptor *pixdesc)
Compute the max pixel step for each plane of an image with a format described by pixdesc.
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Descriptor that unambiguously describes how the bits of a pixel are stored in the up to 4 data planes...
static void hflip_byte_c(const uint8_t *src, uint8_t *dst, int w)
static void hflip_b48_c(const uint8_t *src, uint8_t *dst, int w)
static const AVFilterPad avfilter_vf_hflip_outputs[]
static int filter_slice(AVFilterContext *ctx, void *arg, int job, int nb_jobs)
#define flags(name, subs,...)
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
#define AV_PIX_FMT_FLAG_PAL
Pixel format has a palette in data[1], values are indexes in this palette.
uint8_t log2_chroma_h
Amount to shift the luma height right to find the chroma height.
static int query_formats(AVFilterContext *ctx)
static void hflip_short_c(const uint8_t *ssrc, uint8_t *ddst, int w)