stereo | Perform stereo matching algorithm using Direct 3D | GPU library
kandi X-RAY | stereo Summary
kandi X-RAY | stereo Summary
Performs stereo matching algorithm using Direct 3D (level 9.3) so it can run on a mobile device without CUDA support.
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QUESTION
I have run some Python code in Windows 10 and gotten the Traceback error below. Does it mean the string that is supposed to be an int occurs in line 347
of /DeOldify/deoldify/filters.py
or line 1943
of /lib/site-packages/PIL/Image.py
?
For context, below the traceback error, I've also included the steps that lead to this error (I attempted to colorize a black and white film clip based on the DeOldify Colab https://colab.research.google.com/github/jantic/DeOldify/blob/master/VideoColorizerColab.ipynb but only succeeded at colorizing the first frame as a .jpeg) and the full terminal output.
EDIT: Thanks to @Daweo's response, I discovered the problem was with the arguments I entered into the terminal as indicated by "", line 1
. The correct command should have been:
video_path = colorizer.colorize_from_file_name(file_name='my_video.mp4', render_factor=render_factor)
Also, the code for colorizing a photograph is from https://colab.research.google.com/github/jantic/DeOldify/blob/master/ImageColorizerColab.ipynb
The error:
...ANSWER
Answered 2021-May-31 at 07:59Does it mean the string that is supposed to be an int occurs in line 347 of /DeOldify/deoldify/filters.py or line 1943 of /lib/site-packages/PIL/Image.py
This imply something in filters.py
is responsbile for such usage of something from Image.py
that raised TypeError
.
Consider simple example let zerodiv.py
content be:
QUESTION
We're trying to use xdmp:document-filter
function to extract metadata about PDF files, in particularly we want to know the number of pages in a PDF. It seems that currently MarkLogic is not able to retrieve this information for PDFs (nor for Word documents) but is able to get the number of slides for a PowerPoint. Maybe there is there a hidden option?
https://docs.marklogic.com/guide/search-dev/binary-document-metadata#id_98155 https://docs.marklogic.com/xdmp:document-filter
At some stage we may want to also extract metadata from audio files (MP3), such as duration in seconds and stereo/mono. Is this something that may be possible?
...ANSWER
Answered 2021-Jun-04 at 15:20Please note that there are 2 approaches within Marklogic when it comes to extraction from files:
xdmp:document-filter() as you have already oncovered.
The eternally bundled document conversion libraries offering the xdmp:xxx-convert() functions
For the second option, there is a completely different engine for whiich one option is to generate an xhtml document per page. I would suggest that you explore options using xdmp:pdf-convert()
This can have a unexpected effect of creating multiple documents in the system, but it may still serve your purpose once you work through and try the various options. The first node returned is the manifest, so that may have enough info to count pages if you extract per page. The trick will be to get the information you need without overhead of extracting items that are not needed.. If this helps, then you can also explore the other convert functions in the same family such as the one for word.
QUESTION
I am currently trying to make a .wav file that will play sos in morse.
The way I went about this is: I have a byte array that contains one wave of a beep. I then repeated that until I had the desired length. After that I inserted those bytes into a new array and put bytes containing 00 (in hexadecimal) to separate the beeps.
If I add 1 beep to a WAVE file, it creates the file correctly (i.e. I get a beep of the desired length). Here is a picture of the waves zoomed in (I opened the file in Audacity): And here is a picture of the entire wave part:
The problem now is that when I add a second beep, the second one becomes completely distorted: So this is what the entire file looks like now:
If I add another beep, it will be the correct beep again, If I add yet another beep it's going to be distorted again, etc. So basically, every other wave is distorted.
Does anyone know why this happens?
Here is a link to a .txt file I generated containing the the audio data of the wave file I created: byteTest19.txt
And here is a lint to a .txt file that I generated using file format.info that is a hexadecimal representation of the bytes in the .wav file I generated containing 5 beeps (with two of them, the even beeps being distorted): test3.txt
You can tell when a new beep starts because it is preceded by a lot of 00's.
As far as I can see, the bytes of the second beep does not differ from the first one, which is why I am asking this question.
If anyone knows why this happens, please help me. If you need more information, don't hesitate to ask. I hope I explained well what I'm doing, if not, that's my bad.
EDIT Here is my code:
...ANSWER
Answered 2021-Jun-04 at 09:07The problem
Your .wav file is Signed 16 bit Little Endian, Rate 44100 Hz, Mono
- which means that each sample in the file is 2 bytes long, and describes a signed amplitude. So you can copy-and-paste chunks of samples without any problems, as long as their lengths are divisible by 2 (your block size). Your silences are likely of odd length, so that the 1st sample after a silence is interpreted as
QUESTION
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 03:01With those changes, I'm able to acheive 3 to 4 stable delay ;)
LINE 79 of
I REPLACED
QUESTION
I have ripped files from an audio CD I just bought. I ripped using the Music
app on my Macbook Pro, Catalina 10.15.6 - output format was .wav
as there was no option for FLAC
. My plan was to change format using ffmpeg
:
ANSWER
Answered 2021-Jun-01 at 18:34Add -disposition:v attached_pic
:
QUESTION
I am streaming audio using FFMPeg and need to mix two audio sources using FFMpeg and set the volume level dynamically. I.e. once the stream starts, I need to be able update the ratio of the two volumes.
Currently, I have the volume mixing and streaming working using the CLI version of FFMPeg but the volume mix ratio is static.
Is there a way to dynamically set the volume ratio using the CLI tool? Perhaps something with an FFMpeg expression?
Or is using the API the only option? If so, can anyone point me towards an example of dynamically mixing audio? I haven't been able to find one.
Edit: here are the params I currently pass to mix audio. Again, this works fine, so not including the log as there is no error to fix. The question is how can I adjust the ratio of the amix
mix after the process has launched. Willing to use the API if need be.
ANSWER
Answered 2021-Jun-01 at 15:46- Some filters support commands which allow temporal control of some filter options. Not all filters have commands. Sometimes commands get added so always use a recent
ffmpeg
to take advantage of new features. - Refer to
ffmpeg -filters
and look forC
next to the filter name, or viewman ffmpeg-filters
or FFmpeg Filters documentation and search for commands under each filter. - Not all options have a command equivalent.
- asendcmd / sendcmd - does scheduled commands or interactive commands via interactive mode.
- azmq / zmq - does interactive, on demand, live commands. To enable (a)zmq you need to install the libzmq library and headers and configure ffmpeg with
--enable-libzmq
.
Halve the volume at timestamp 10:
QUESTION
I am trying to add xfade
filter and the command is working but audio of second video is missing in output video.
command is -
...ANSWER
Answered 2021-May-27 at 21:54You didn't tell ffmpeg what to do with the audio so it just picked the audio from the first input (see stream selection).
Because you are using xfade you probably want to use acrossfade as shown in Merging multiple video files with ffmpeg and xfade filter:
QUESTION
I'm trying to convert a .ts file with this output to mkv:
...ANSWER
Answered 2021-May-02 at 08:14Try to run with the -ss
flag.
QUESTION
I am able to stream with ffmpeg an mp4 file with h264 encoded video and aac encoded audio to flash flv output. The command I used is:
ffmpeg -re -i bigbuckbunny_HD_60fps.mp4 -c copy -f flv rtmp://192.168.3.64:1935/main/stream0
When I am trying to use the tee option to duplicate the output I am failing with an error. This is the command I use:
ffmpeg -re -i bigbuckbunny_HD_60fps.mp4 -map 0 -c copy -f tee "[f=flv]rtmp://192.168.3.64:1935/main/stream0|[f=flv]rtmp://192.168.3.64:1935/main/stream1"
However, it fails. What I am missing? The output I have from ffmpeg is:
...ANSWER
Answered 2021-May-19 at 19:28According to ffmpeg -f tee
refuses to do anything useful add -tag:v 7
(or -vtag 7
, same thing):
QUESTION
I am able to stream with ffmpeg an mp4 file with h264 encoded video and aac encoded audio to flash flv output. The command I used is:
...ANSWER
Answered 2021-May-19 at 18:09You have to add the -map
option when using the tee muxer. From the tee muxer documentation:
Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select output streams. So all streams intended for output must be specified using
-map
.
Example to include all streams from the input (-map 0
):
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