ddR | Standard API for Distributed Data Structures in R | Runtime Evironment library

by vertica R Version: Current License: GPL-2.0

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kandi X-RAY | ddR Summary

ddR is a R library typically used in Server, Runtime Evironment applications. ddR has no bugs, it has no vulnerabilities, it has a Strong Copyleft License and it has low support. You can download it from GitHub.

The 'ddR' package aims to provide an unified R interface for writing parallel and distributed applications. Our goal is to ensure that R programs written using the 'ddR' API work across different distributed backends, therefore, reducing the effort required by users to understand and program on different backends. Currently 'ddR' programs can be executed on R's default 'parallel' package as well as the open source HP Distributed R. We plan to add support for SparkR. This package is an outcome of feedback and collaboration across different companies and R-core members!. Through funding provided by the R-consortium this package is under active development for the summer of 2016. Check out the mailing list to see the latest discussions. 'ddR' is an API, and includes a default execution engine, to express and execute distributed applications. Users can declare distributed objects (i.e., dlist, dframe, darray), and execute parallel operations on these data structures using R-style apply functions. It also allows different backends (that support ddR, and have ddR "drivers" written for them) to be dynamically activated in the R user's environment to execute applications. Please refer to the user guide under vignettes/ for a detailed description on how to use the package.

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Support

ddR has a low active ecosystem.

It has 118 star(s) with 17 fork(s). There are 48 watchers for this library.

It had no major release in the last 6 months.

There are 15 open issues and 1 have been closed. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of ddR is current.

Quality

ddR has no bugs reported.

Security

ddR has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

License

ddR is licensed under the GPL-2.0 License. This license is Strong Copyleft.

Strong Copyleft licenses enforce sharing, and you can use them when creating open source projects.

Reuse

ddR releases are not available. You will need to build from source code and install.

Installation instructions are not available. Examples and code snippets are available.

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Community Discussions

Trending Discussions on ddR

ffmpeg x11grab to streamable format

Is it possible to unit test C macros for embedded software?

ATmega32 analog comparator code with interrupts not working

How to include audio in an overlay ffmpeg command?

Intel MAX 10 DDR output

FFmpeg stream stops after a certain time

Prevent ffmpeg from changing the intensity of colors while downscaling the resolution of the video

How could I achieve a good ratio of compression and time while converting video files with ffmpeg?

FFMPEG keeping quality when reducing FPS and streaming over RTSP with rtsp-simple-server

Unable to play recorded video and audio using ffmpeg

QUESTION

ffmpeg x11grab to streamable format

Asked 2021-Jun-02 at 03:01

2 FFMPEG process

(1) generating a ffmpeg x11grab to a .mp4 (2) take the .mp4 and restream it simultaneously to multiple rtmp endpoints

ISSUE the generated file in (1) have this error "moov atom not found"

This is the command that generate (1) :

...

ANSWER

Answered 2021-Jun-02 at 03:01

With those changes, I'm able to acheive 3 to 4 stable delay ;)

LINE 79 of

https://github.com/OpenVidu/openvidu/blob/master/openvidu-server/docker/openvidu-recording/scripts/composed.sh

I REPLACED

Source https://stackoverflow.com/questions/66837454

QUESTION

Is it possible to unit test C macros for embedded software?

Asked 2021-May-06 at 10:34

I was how it's possible to test some specific C macros for embedded SW.

For example if I have the following macro:

...

ANSWER

Answered 2021-May-06 at 10:16

No, this is not possible to be done at compile time. You cannot use say, a #if inside of a macro substitution.

You can technically pass anything to the two arguments of _SET_INPUT_PIN. They are just blindly substituted by the pre-processor wherever they occur in the macro definition. You may pass constants to these, but they can also be variables. If it is a variable then it has no meaning at compile time since variables exist only at run-time. So there is no way of implementing this check at compile time. You can of course, implement a run-time check.

For example,

Source https://stackoverflow.com/questions/67415449

QUESTION

ATmega32 analog comparator code with interrupts not working

Asked 2021-Apr-27 at 17:47

I wrote a code for my ATmega32 to light an LED using the analog comparator's interrupts but the ISR won't execute. I'm testing on Proteus.

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ANSWER

Answered 2021-Apr-27 at 17:47

Edit:

The ATMega32 bit macros are the actual bit numbers (0, 1) and need to be leftshifted when used in bitwise expressions with the registers.

ACSR |= (ACIE)|(ACIS1);
should be
ACSR |= (1 << ACIE)|(1 << ACIS1);

There's lots of things that could go wrong here. and I don't see anything obviously wrong (other than the above).

I added a line of code to toggle another pin between the while loop to see if any interrupts at all occur but the pin inside the while loop keeps toggling

I don't understand what this means - could you add a code snippet for this test?

Here's a couple of troubleshooting steps I'd take.

Can you verify that this code can light up the LED?

This could isolate a problem with the LED code / circuit.
Try some main function like this:

Source https://stackoverflow.com/questions/67286073

QUESTION

How to include audio in an overlay ffmpeg command?

Asked 2021-Apr-26 at 23:42

Using ffmpeg I add a video overlay successfully over an origin video (origin has audio, overlay doesn't). However the audio of the origin video doesn't appear in the result video.

...

ANSWER

Answered 2021-Apr-26 at 23:42

Tell it which audio you want with -map:

Source https://stackoverflow.com/questions/67271854

QUESTION

Intel MAX 10 DDR output

Asked 2021-Apr-08 at 06:03

I want to output a clock signal via a DDR register. The target FPGA is an Intel MAX 10 (10M16DAU324I7G) FPGA. I instantiate an ALTDDIO_OUT component as shown in the code below. However, the output Pin stays permanently low. Clock is running, Pin in R15. Can anyone provide a hint what my problem could be?

...

ANSWER

Answered 2021-Apr-08 at 06:02

Direct instantiation of the ALTDDIO_OUT primitive does not seem to work reliably on the chosen FPGA and/or tool chain (MAX 10, Quartus Prime 18.1). The solution is to generate an IP core with the MegaWizard GPIO Lite Intel FPGA IP using a DDR register output.

Source https://stackoverflow.com/questions/66851707

QUESTION

FFmpeg stream stops after a certain time

Asked 2021-Apr-07 at 06:11

We have a little Nodejs app, which starts a stream process, with a child_process.spawn. On the client-side, we have an HTML5-canvas element, which records the video data new MediaRecorder(canvas.captureStream(30), config), then this client sends its data to our Nodejs server over a WebSocket connection. We using FFmpeg for video encoding and decoding, then we send the data to our 3-rd party service (MUX), which accepts the stream and broadcasts them. Sadly the process continuously loses its fps, and after in general 1 minute, stops with an interesting error code. (when we save the video result locally instead of streaming via rtmps, it works perfectly.

*The whole system is in docker.

The error:

...

ANSWER

Answered 2021-Apr-07 at 06:11

Im found another FFmpeg config that works perfectly.

Source https://stackoverflow.com/questions/66236584

QUESTION

Prevent ffmpeg from changing the intensity of colors while downscaling the resolution of the video

Asked 2021-Feb-18 at 14:29

I have a use case where I need to downscale a 716x1280 mp4 video to 358x640 (half of the original). Command that I used is

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ANSWER

Answered 2021-Feb-18 at 14:29

You need to use Bit Stream Video Filter for setting h264 metadata.

When a video player plays a video file, it looks for metadata that attached to the video stream (h264 metadata for example).
The H.264 metadata parameters that affects the colors and brightness are: video_format, colour_primaries, transfer_characteris and matrix_coefficients.

If the parameters are not set, there are defaults.
The defaults for low resolution video are "Limited Range" BT.601 (in most player - I am not sure about MAC OS).
The default gamma curve (affects the brightness) is sRGB gamma curve.
The player converts the pixels from YUV color space to RGB (for displaying the video). The conversion formula is done according to the metadata.

Your input video file input.mp4 has H.264 metadata parameters that are far from the default.

We can assume that scale video filter does not change the color characteristics (the filter applies the YUV elements without converting to RGB).

The characteristics of input.mp4 applies BT.2020, and HLG gamma curve, but converted as if they were default (BT.601 and sRGB gamma), so the colors and brightness are very different from what they should have been.

When FFmpeg encodes a video stream, it does not copy the metadata parameters from the input to the output - you need to set the parameters explicitly.

The solution is using a Bit Stream Video Filter for setting the metadata parameters.

Try using the following command:

Source https://stackoverflow.com/questions/66240097

QUESTION

How could I achieve a good ratio of compression and time while converting video files with ffmpeg?

Asked 2021-Feb-17 at 17:55

So, I made 2 scripts that convert CCTV footage in mp4 videos. One of them is just a -vcodec copy and creates a mp4 with the same size of the footage (huge, btw) and my other alternative was tweak with some parameters and figure out what was the best I could do without sacrifice too much quality and make it "fast". Then I come up with c:v libx264 -crf 30 -preset veryfast -filter:v fps=fps=20 which took something like 2 secs in my machine to run an average 6MB file and transform into a 600kB file.

Happy with the results I decided to put it on AWS lambda (to avoid bottlenecks), and then people started to complain about missing files, so I increase the timeout and the memory to 380MB. And even after that, I am still getting a few lambda errors...

Anyway, the lambda is going to cost me too much compared to just store the file without compression, there is another way to decrease size without sacrificing time?

[UPDATE] I crunch some numbers and even tho using lambda is not what I expected, I am still saving a lot of cash monthly by reducing the file size 10x times.

As asked, this is the logs for the ffmpeg.

...

ANSWER

Answered 2021-Feb-17 at 17:55

libx264

You have to choose a balance between encoding speed and encoding efficiency.

Choose the slowest -preset you have patience for.
Choose the highest -crf value that provides an acceptable quality.

See FFmpeg Wiki: H.264 for more info.

libx265

If libx264 does not make a small enough file try libx265, but it takes longer to encode.

See FFmpeg Wiki: HEVC/H.265 for more info.

Hardware accelerated encoder

If you have the proper hardware, then you can use NVENC, QuickSync, or some other implementation.

Encoding will be fast, but it will not match the quality per bit provided by libx264 or libx265.

See FFmpeg Wiki: Hardware for more info.

Source https://stackoverflow.com/questions/65891087

QUESTION

FFMPEG keeping quality when reducing FPS and streaming over RTSP with rtsp-simple-server

Asked 2021-Jan-25 at 12:11

I'm using the rtsp-simple-server (https://github.com/aler9/rtsp-simple-server) and feed the RTSP Server with a FFMPEG stream.

I use a docker compose file to start the stream:

...

ANSWER

Answered 2021-Jan-25 at 12:11

When you say, "the quality of the video becomes pretty bad," I guess you mean your transcoded output video has a lot of block artifacts in it. That's generally because you haven't allocated enough bandwidth to your output video stream. Without enough output bandwidth to play with, the coder quantizes and eliminates higher-frequency stuff so it looks nasty.

You didn't mention what sort of program material you have. But it's worth mentioning this: in material with lots of motion (think James Bond flick) it doesn't save much bandwidth to reduce the frame rate: we're coding the difference between successive frames. The longer you wait between frames, the more differences there are to code (and the harder the motion estimator has to work). If you radically reduce the frame rate (from 24 to 2 for example) it gets much worse.

Talking-heads material is generally less sensitive to framerate.

You might try setting your bandwidth -- your output bitrate -- explicitly like this.

Source https://stackoverflow.com/questions/65852931

QUESTION

Unable to play recorded video and audio using ffmpeg

Asked 2021-Jan-11 at 17:44

I’m trying to generate a video and audio for every 40ms in a separate files and sending it to the cloud for a live stream, but created videos and audio’s are unable to play using ffplay.

Command:

ffmpeg -f alsa -thread_queue_size 1024 -i hw:0 -f video4linux2 -i /dev/video0 -c:a aac -ar 48k -t 0:10 -segment_time 00:00.04 -f segment sample-%003d.aac -c:v h264 -force_key_frames "expr:gte(t,n_forced*0.04)" -pix_fmt yuv420p -s:v 640x480 -t 0:10 -r 25 -g 1 -segment_time 00:00.04 -f segment frame-%003d.h264

Error:

frame-001.h264: Invalid data found when processing input.

Console output:

configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared libavutil 55. 78.100 / 55. 78.100 libavcodec 57.107.100 / 57.107.100 libavformat 57. 83.100 / 57. 83.100 libavdevice 57. 10.100 / 57. 10.100 libavfilter 6.107.100 / 6.107.100 libavresample 3. 7. 0 / 3. 7. 0 libswscale 4. 8.100 / 4. 8.100 libswresample 2. 9.100 / 2. 9.100 libpostproc 54. 7.100 / 54. 7.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, alsa, from 'hw:0': Duration: N/A, start: 1610338632.931406, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Input #1, video4linux2,v4l2, from '/dev/video0': Duration: N/A, start: 3405.427360, bitrate: 147456 kb/s Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 147456 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Stream #1:0 -> #1:0 (rawvideo (native) -> h264 (libx264)) Press [q] to stop, [?] for help [alsa @ 0x55777d96fe00] ALSA buffer xrun. [segment @ 0x55777d983d80] Opening 'sample-000.aac' for writing Output #0, segment, to 'sample-%003d.aac': Metadata: encoder : Lavf57.83.100 Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s Metadata: encoder : Lavc57.107.100 aac [libx264 @ 0x55777d98fa20] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 [libx264 @ 0x55777d98fa20] profile High, level 3.0 [libx264 @ 0x55777d98fa20] 264

core 152 r2854 e9a5903 - H.264/MPEG-4 AVC codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=1 keyint_min=1 scenecut=40 intra_refresh=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 [segment @ 0x55777d98dda0] Opening 'frame-000.h264' for writing Output #1, segment, to 'frame-%003d.h264': Metadata: encoder : Lavf57.83.100 Stream #1:0: Video: h264 (libx264), yuv420p, 640x480, q=-1--1, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc57.107.100 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 [segment @ 0x55777d983d80] Opening 'sample-001.aac' for writing [segment @ 0x55777d98dda0] Opening 'frame-001.h264' for writing [segment @ 0x55777d98dda0] Opening 'frame-002.h264' for writing [segment @ 0x55777d98dda0] Opening 'frame-003.h264' for writing [segment @ 0x55777d98dda0] Opening 'frame-004.h264' for writing [segment @ 0x55777d98dda0] Opening 'frame-005.h264' for writing [segment @ 0x55777d98dda0] Opening 'frame-006.h264' for writingA dup=1 drop=0 speed=1.07x ... [segment @ 0x55777d98dda0] Opening 'frame-018.h264' for writingA dup=5 drop=0 speed=0.714x ...
[segment @ 0x55777d98dda0] Opening 'frame-029.h264' for writingA dup=12 drop=0 speed=0.768x ... [segment @ 0x55777d98dda0] Opening 'frame-042.h264' for writingA dup=21 drop=0 speed=0.834x ... [segment @ 0x55777d983d80] Opening 'sample-055.aac' for writingA dup=31 drop=0 speed=0.89x ... [segment @ 0x55777d98dda0] Opening 'frame-067.h264' for writingA dup=39 drop=0 speed=0.887x ... [segment @ 0x55777d98dda0] Opening 'frame-081.h264' for writingA dup=49 drop=0 speed=0.92x ... [segment @ 0x55777d98dda0] Opening 'frame-091.h264' for writingA dup=56 drop=0 speed=0.904x ... [segment @ 0x55777d98dda0] Opening 'frame-105.h264' for writingA dup=66 drop=0 speed=0.927x ... [segment @ 0x55777d98dda0] Opening 'frame-119.h264' for writingA dup=76 drop=0 speed=0.944x ... [segment @ 0x55777d98dda0] Opening 'frame-130.h264' for writingA dup=84 drop=0 speed=0.938x ... [segment @ 0x55777d98dda0] Opening 'frame-144.h264' for writingA dup=94 drop=0 speed=0.952x ... [segment @ 0x55777d983d80] Opening 'sample-154.aac' for writingA dup=103 drop=0 speed=0.958x ... [segment @ 0x55777d98dda0] Opening 'frame-168.h264' for writingA dup=111 drop=0 speed=0.952x ... [segment @ 0x55777d98dda0] Opening 'frame-182.h264' for writingA dup=121 drop=0 speed=0.962x ... [segment @ 0x55777d98dda0] Opening 'frame-193.h264' for writingA dup=129 drop=0 speed=0.956x ... [segment @ 0x55777d98dda0] Opening 'frame-207.h264' for writingA dup=139 drop=0 speed=0.965x ... [segment @ 0x55777d983d80] Opening 'sample-218.aac' for writingA dup=149 drop=0 speed=0.974x ... [segment @ 0x55777d98dda0] Opening 'frame-231.h264' for writingA dup=156 drop=0 speed=0.964x ... [segment @ 0x55777d98dda0] Opening 'frame-249.h264' for writing frame= 250 fps= 24 q=-1.0 Lsize=N/A time=00:00:10.00 bitrate=N/A dup=168 drop=0 speed=0.98x
video:2707kB audio:149kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown [aac @ 0x55777d98cf00] Qavg: 260.500 [libx264 @ 0x55777d98fa20] frame I:250 Avg QP:26.77 size: 11085 [libx264 @ 0x55777d98fa20] mb I I16..4: 13.4% 72.3% 14.3% [libx264 @ 0x55777d98fa20] 8x8 transform intra:72.3% [libx264 @ 0x55777d98fa20] coded y,uvDC,uvAC intra: 54.2% 91.6% 64.5% [libx264 @ 0x55777d98fa20] i16 v,h,dc,p: 13% 18% 6% 62% [libx264 @ 0x55777d98fa20] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 19% 18% 16% 6% 7% 6% 12% 5% 10% [libx264 @ 0x55777d98fa20] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 18% 11% 6% 8% 8% 10% 5% 6% [libx264 @ 0x55777d98fa20] i8c dc,h,v,p: 58% 20% 15% 7% [libx264 @ 0x55777d98fa20] kb/s:2216.90

...

ANSWER

Answered 2021-Jan-11 at 17:44

Use -f stream_segment (or the alias -f ssegment). From the documentation:

stream_segment is a variant of the segment muxer used to write to streaming output formats, i.e. which do not require global headers, and is recommended for outputting e.g. to MPEG transport stream segments. ssegment is a shorter alias for stream_segment.

Example command:

Source https://stackoverflow.com/questions/65670670

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