I am using julius speech recognition for my application. I have one doubt regarding julius:
I have downloaded the latest version and was successful in using its lib and making it work. the problem I am facing is..once the app starts and I call the voice recognition function in my application...it takes the input from mic and displays whatever is said in the mic, but the function still continues to do so again and again. The control will never come out of that function. Here I am facing problem since the control is not returning back I am not able to proceed further. What I want is once the engine gets input from mic it should recognize and stop there.. which I tried to do by deleting the callback function but was unsuccessful.
Can anyone please guide me in this matter, what I need to do to get the desired output. It will be helpful for me.
As discussed in the same post on VoxForge:
You have a couple of choices: first to use the Julius -input control to get the sound data from a list of files (see the .jconf sample file), so that when the list (even if only length one) is exhausted then Julius stops. It is quite easy to record the voice input to a file and then feed the file into Julius. Second you can put a dialog manager in control. If you need more information on what a dialog manager does there are many posts on this forum on that subject accessible by a search.
The basic function of Julius is to start up and then keep on decoding input. When you get more experience you can run Julius as a server, and then tell the server to respond, not respond or shut down as required. It's more efficient than having Julius start and stop all the time.
When an avenue exists for a complex application to yield the required result by using an effective combination of options at run time, editing the application, while possible, might involve a lot of unnecessary work. The emphasis then shifts to passing the options correctly in whatever script is being used to access Julius.
Related
For a game website, I want a player to contest either agains a human or an AI.
I am using Django + Channels (Django-4.0.2 asgiref-3.5.0 channels-3.0.4)
This is a long way of learning...
Human vs Human: the game take place is the web browser turn by turn. Each time a player connects, it opens a websocket connexion, a move is sent through the socket, processed by the consumer (validated and saved in the database) and sent to the other player.
It is managed only with sync programming.
Human vs AI: I try to use the same route as previously. A test branch check if the game is against the computer and process a move instead of receiving it from the other end of the websocket. This AI move can be a blocking operation as it can take from 2 to 5sec.
I don't want the receive method of the consumer to wait for the AI to return its move, since I have other operations to do quickly (like update some informations on the client side).
Then I thought I could easily take advantage of the allegedly already existing event loop of the channels framework. I could send the AI thinking process to this loop and return the result later to the client through the send method of the consumer.
However, when I write:
loop = asyncio.get_event_loop()
loop.create_task(my_AI_thinking())
Django raises a runtime effort error (the same as described here: https://github.com/django/asgiref/issues/278) telling me there is no running event loop.
The solution seemed to be to upgrade asgiref to 3.5.0 which I did but issue not solved.
I think I am a little bit short of background, and some enlightments should help me to understand a little bit more what is the root cause of this fail.
My first questions would be:
In the combo django + channels + asgi: which is in charge to run the eventloop?
How to check if indeed one event loop is running whatever the thread?
Maybe your answers wil raise other questions.
Did you try running your event_loop example on Django 3.2? (and/or with different Python version)? I experienced various problems with Django 4.0 & Python 3.10, so I keep with Django 3.2 and Python3.7/3.8/3.9 for now, maybe your errors are one of these problems?
If you won't be able to get event_loop running, I see two possible alternative solutions:
Open two WS connections: one only for the moves, and the other for all the other stuff, such as updating information on Player's UI, etc.
You can also use multiprocessing to "manually" send calculating AI move to other thread, and then join the two threads again, after receiving the result (the move). To be honest, multiprocessing in Python is quite simple -- it's pretty handy, if you are familiar with the idea of multithreaded applications.
Unfortunately, I have not yet used event loops in channels myself, maybe someone more experienced in that matter will be able to better address your issue.
I am transmitting and receiving data using aux cable and GNU RADIO between two laptops.
I have implemented DQPSK using PSK mod block.
The problem is that while receiving I have to provide a delay, some integer value e.g 0,1,2 etc.
It is different every time.
Is there a way to dynamically check for the right delay value or any other workaround to this situation?
I have written 'start.' at the start of data being transmitted and 'end.' at the end.
I have to give a demo for this project and I dont want to manually change the delay at runtime.
I cannot find the .cc file of file sink in GNU RADIO, I can change the C++ code according my requirement but there is no such file.
Below is the screenshot of the grc file on the receive side.
Any help will be appreciated.
Since there's no way for the receiver to know when the transmitter started transmitting, it decodes stuff before there's actually anything to decode.
In essence, you need some kind of preamble or so to tell your receiver when to start – side effect of having something like that would be that you could correct some things (the two sound cards don't share the same oscillator, which leads to a symbol rate offset, and a center frequency offset).
You basically added that framing - your start. and end. strings.
I cannot find the .cc file of file sink in GNU RADIO, I can change the C++ code according my requirement but there is no such file.
It's in gr-blocks/lib; however, you shouldn't modify the file sink. Really,
I'd recommend you take the time to go through the guided tutorials, use gr_modtool to generate a general block which has a state machine that looks for the bits of your start string and drops everything before and including those, and then passes everything till it sees the stop string. That all can be done with a single state machine, and a bit of python or C++ code.
I'm trying to write a program to ask me to input some information periodically, so I've written a service in C to run in the background, I can watch it reporting its okay by refreshing its log file. Now I'm stuck on how to get it to open up cmd and ask for the information.
I'd like to save this information to a log file. (I'm planning on monitoring my sleeping habits)
I tried using system("getinput.exe") and that seems to do nothing, I know using system() is bad but it was a first step.
CreateProcess() I simply cannot get to work, the example on MSDN http://msdn.microsoft.com/en-us/library/ms682512.aspx doesn't work for me, i just get CreateProcess failed (2) whenever i try createprocess.exe dir for example.
Surely there must be a way?
EDIT: Thanks for the replies I will try to take a different approach then. Where would I start with writing a background application that can occasionally ask for user input in C?
It is possible to implement a feature that allows users to watch videos as they are uploaded to server by others. Is html 5 suitable for this task? But flash? Are there any read to go solutions, don't want to reinvent the wheel. The application will be hosted on a dedicated server.
Thanks.
Of course it is possible, the data is there isnt it?
However it will be very hard to implement.
Also I am not so into python and I am not aware of a library or service suiting your requirements, but I can cover the basics of video streaming.
I assume you are talking about video files that are uploaded and not streams. Because, for that, there are obviously thousands of solutions out there...
In the most simple case the video being uploaded is already ready to be served to your clients and has a so called "faststart atom". They are container format specific and there are sometimes a bunch of them. The most common is the moov-atom. It contains a lot of data and is very complex, however in our use case, in a nutshell, it holds the data that enables the client to begin playing the video right away using the data available from the beginning.
You need that if you have progressive download videos (youtube...), meaning where a file is served from a Webserver. You obviously have not downloaded the full file and the player already can start playing.
If the fastastart atom was not present, that would not be possible.
Sometimes it is, but the player for example cannot display a progress bar, because it doesnt know how long the file is.
Having that covered the file could be uploaded. You will need an upload solution that writes the data directly to a buffer or a file. (file will be easier...).
This is almost always the case, for example PHP creates a file in the tmp_dir. You can also specify it if you want to find the video while its being uploaded.
Well, now you can start reading that file byte by byte and print that data to a connection to another client. Just be sure not to go ahead of what has already been recieved and written. You would probaby initiate your upload with a metadata set in memory that holds the current recieved byte position and location of the file.
Anyone who requests the file after the uploaded has started can just recieve the entire file, or if the upload is not yet finished, get it from your application.
You will have to throttle the data delivery or pause it when the data becomes short. This will appear to the client almost as a "slow connection". However you will have to echo some data from time to time to prevent the connection from closing. But if your upload doesnt stall, and why shoud it?, that shouldnt be a problem.
Now if you want to have someting like on the fly transcoding of various input formats into your desired output format, things get interesting.
AFAIK ffmpeg has neat apis which lets you directly deal with datasterams.
Also handbrake is a very good tool, however you would need to take the long road using external executeables.
I am not really aware of your requirements, however if your clients are already tuned in, for example on a red 5 streaming server, feeding data into a stream should also work fine.
Yes, take a look at Qik, http://qik.com/
"Instant Video Sharing ... Videos can be viewed live (right as they are being recorded) or anytime later."
Qik provides developer APIs, including ones like these:
qik.stream.subscribe_public_recent -- Subscribe to the videos (live and recorded)
qik.user.following -- Provides the list of people the user is following
qik.stream.public_info -- Get public information for a specific video
It is most certainly to do this, but it won't be trivial. And no, I don't think that you will find an "out of the box" solution that will require little effort on your behalf.
You say you want to let:
users watch videos as they are uploaded to server by others
Well, this could be interpreted two different ways:
Do you mean that you don't want a user to have to refresh the page before seeing new videos that other users have just finished uploading?
Or do you mean that you want one user to be able to watch a partially uploaded video (aka another user is still in the process of uploading it and right now the server only contains a partial upload of the video)?
Implementing #1 wouldn't be hard at all whatsoever. You would just need an AJAX script to check for newly uploaded videos, and those videos could then be served to the user in whatever way you choose. HTML5 vs. Flash isn't really a consideration here.
The second scenario, on the other hand, would require quite a bit of effort. I am guessing that HTML5 might not be mature enough to handle this type of situation. If you are not looking
to reinvent the wheel and don't have a lot of time to dedicate to this feature than I would say that you would be out of luck. You may be able to use ffmpeg to parse partial video files and feed them to a Flash player, but I would think of this as a large task.
we´ve created a custom push source / parser filter that is expected to work in a directshow
editing services timeline.
Now everything is great except that the filter does not stop to deliver samples when the current
cut has reached it´s end. The rendering stops, but the downstream filter continues to consume
samples. The filter delivers samples until it reaches EOF. This causes high cpu load, so the application
is simply unusable.
After a lot of investigation I’m not able to find a suitable mechanism that can inform my filter
that the cut is over so the filter needs to be stopped :
The Deliver function on the connected decoder pins always returns S_OK, meaning the attached decoder
is also not aware the IMediaSamples are being discarded downstream
there’s no flushing in the filter graph
the IMediaSeeking::SetPositions interface is used but only the start positions are set –
our is always instructed to play up to the end of the file.
I would expect when using IAMTimelineSrc::SetMediaTimes(Start, Stop) from the application
that this would set a stop time too, but this does not happen.
I’ve also tried to manipulate the XTL timeline adding ‘mstop’ attributes to all the clip in the
hope that this would imply a stop position being set, but to no avail
In the filters point of view, the output buffers are always available (as the IMediaSamples are being discarded downstream),
so the filter is filling samples as fast as it can until the source file is finished.
Is there any way the filter can detect when to stop or can we do anything from the application side ?
Many thanks
Tilo
You can try adding a custom interface to your filter and call a method externally from your client application. See this SO question for a bit more of details on this approach. You should be careful with thread safety while implementing this method, and it is indeed possible that there is a neater way of detecting that the capturing should be stopped.
I'm not that familiar with DES, but I have tried my demux filters in DES and the stop time was set correctly when there was a "stop=" tag for the clip.
Perhaps your demux does not implement IMediaSeeking correctly. Do you expose IMediaSeeking through the pins?
I had a chance to work with DES and custom push source filter recently.
From my experience;
DES actually does return error code to Receive() call, which is in turn returned to Deliver() of the source, when the cut reaches the end.
I hit the similar situation that source does not receive it and continues to run to the end of the stream.
The problem I found (after a huge amount of ad-hoc trials) is that the source needs to call DeliverNewSegment() method at each restart after seek. DES seems to take incoming samples only after that notification. It looks like DES receives the samples as S_OK even without that notification, but it just throws away.
I don't see DES sets end time by IMediaSeeking::SetPositions, either.
I hope this helps, although this question was very old and I suppose Tilo does not care this any more...