FFmpeg  4.3
Typedefs | Functions | Variables
doc/multithreading.txt File Reference

Typedefs

using clients = =============================================Slice threading - *The client 's draw_horiz_band() must be thread-safe according to the comment in avcodec.h. Frame threading - *Restrictions with slice threading also apply. *For best performance, the client should set thread_safe_callbacks if it provides a thread-safe get_buffer() callback. *There is one frame of delay added for every thread beyond the first one. Clients must be able to handle this
 

Functions

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before ff_thread_await_progress () has been called on them. reget_buffer() and buffer age optimizations no longer work. *The contents of buffers must not be written to after ff_thread_report_progress() has been called on them. This includes draw_edges(). Porting codecs to frame threading
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling get_buffer ()
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call ff_thread_finish_setup () afterwards. If some code can 't be moved
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have update_thread_context () run it in the next thread. Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little speed gain at this point but it should work. If there are inter-frame dependencies
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report await_progress ()
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use ff_thread_get_buffer () to allocate frames. The frames must then be freed with ff_thread_release_buffer(). Otherwise decode directly into the user-supplied frames. Call ff_thread_report_progress() after some part of the current picture has decoded. A good place to put this is where draw_horiz_band() is called - add this if it isn 't called anywhere
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use as it s useful too and the implementation is trivial when you re doing this Note that draw_edges () needs to be called before reporting progress. Before accessing a reference frame or its MVs
 

Variables

FFmpeg multithreading methods
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec implementations
 
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across frames
 

Typedef Documentation

◆ clients

using clients = ============================================= Slice threading - * The client's draw_horiz_band() must be thread-safe according to the comment in avcodec.h. Frame threading - * Restrictions with slice threading also apply. * For best performance, the client should set thread_safe_callbacks if it provides a thread-safe get_buffer() callback. * There is one frame of delay added for every thread beyond the first one. Clients must be able to handle this

Definition at line 26 of file multithreading.txt.

Function Documentation

◆ ff_thread_await_progress()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before ff_thread_await_progress ( )

◆ get_buffer()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling get_buffer ( )

◆ ff_thread_finish_setup()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling up to before the decode process starts Call ff_thread_finish_setup ( )

◆ update_thread_context()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have update_thread_context ( )

◆ await_progress()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report await_progress ( )

◆ ff_thread_get_buffer()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use ff_thread_get_buffer ( )

◆ draw_edges()

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel* The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use as it s useful too and the implementation is trivial when you re doing this Note that draw_edges ( )

Variable Documentation

◆ methods

FFmpeg multithreading methods
Initial value:
==============================================
FFmpeg provides two methods for multithreading codecs.
Slice threading decodes multiple parts of a frame at the same time

Definition at line 2 of file multithreading.txt.

◆ implementations

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec implementations
Initial value:
==============================================
Slice threading -
None except that there must be something worth executing in parallel.
Frame threading -
* Codecs can only accept entire pictures per packet.
* Codecs similar to ffv1

Definition at line 30 of file multithreading.txt.

◆ frames

the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across frames

Definition at line 37 of file multithreading.txt.

be
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it be(in the first position) for now. Options ------- Then comes the options array. This is what will define the user accessible options. For example
codecs
static struct codec_string codecs[]
to
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 X fate suite ffmpeg Duo ug o o X fate suite fate suite ffmpeg Duo ug o o X fate suite fate suite ffmpeg can be set to
Definition: fate.txt:178
Slice
Definition: magicyuv.c:36
methods
FFmpeg multithreading methods
Definition: multithreading.txt:2
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
that
if it could not because there are no more it should return AVERROR_EOF The typical implementation of request_frame for a filter with several inputs will look like that
Definition: filter_design.txt:273
in
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
Definition: audio_convert.c:326
frame
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
Definition: filter_design.txt:264