I am making a web server with a robotdyn arduino mega + esp8266 to control a few bo motors I flashed the firmware for the esp8266 I have 2 questions, 1: when I give AT commands through the serial monitor It doesn't respond also when I open the serial monitor its saying, "error opening COM4, port busy" how can I solve this? the second one is how can I embed the html code into the arduino code? by the way I didn't use any code for the AT commands. Can you also tell how I should orient the switches (they're only on the robotdyn arduino mega/uno + esp8266 model not on normal esp8266 chips)on the mega while talking the the mega?
Related
I am working on webrtc chat app.
system1 : peer1, peer3 | system2 : peer2, peer4
What I did ?
For video call, I am creating one peer1, peer2, and that peer1 very successfully communicate with peer2 (consider video call only).
What I want to do?
I want, peer3 can communicate with new peer4(meanwhile I don't want to disconnect peer1-->peer2).
What happening ?
When I am creating peer3, on that time " navigator.mediaDevices.getUserMedia " getting errors !!!! (I am not using SSL) and new peer3 can't create !.
So, please tell me your suggestions regarding above problem.
It is possible to only peer1 can communicate with other any peer(bidirectional communication)??
This could be the browser failing to capture video from the webcam if it's already in use by another browser or another tab in the same browser. For example, Firefox 57 on my Ubuntu 16.04 machine exhibits this behavior (Chrome does not).
I'm curious if it's possible for me to write programs that can control an Apple TV, specifically an Apple TV 4th gen running tvOS 9.1.1, like Apple's Remote app for iOS can. I'd like to send it commands for navigating in the four cardinal directions, selecting an item on the screen, going up the navigation stack -- essentially what Apple's Remote app can do.
Has anyone done any work reverse engineering the protocol it uses? Cursory Googling only has so far yielded out of date results about earlier generation Apple TVs and the DAAP protocol which looks like something different than what I want.
I captured the traffic on my iPhone using tcpdump and analyzed it with WireShark. The Remote app asks the Apple TV with normal HTTP requests on port 3689.
The workflow of the app consists in four HTTP requests:
/server-info for getting infos about the Apple TV. It responds with a Apple proprietary DAAP response (Digital Audio Access Protocol) providing some tags about the device, like the display name.
/login is performed during connection, when the app displays the "Connecting to Apple TV..." message. It responds with a DAAP about the login status.
Here's the bottleneck. /home-share-verify validates the connection between the app and the Apple TV. This call needs a Client-DAAP-Validation header with a long unknown string value. According to Wikipedia, this seems to be like an hash generated by a certificate exchange between verified sources that was introduced in iTunes 7.0+ and never reverse engineered.
/ctrl-int/1/{controlpromptupdate|controlpromptentry|playstatusupdate} seems to be the calls made for the input buttons.
Some other minor calls are fired in between (like a Bonjour service update or a /databases call).
Here and here you can find more infos. Hope this helps for getting an overview of how this simple (but protected) app works.
i wanted to tell alexa to trigger appletv and that would wake my appletv up and via HDMI & CEC turn my tv on,
in order to do that:
from your mac\linux\windows simply run:
curl -XPOST -d 'cmcc\x00\x00\x00\x01\x30cmbe\x00\x00\x00\x04menu' 'http://10.1.1.56:3689/ctrl-int/1/controlpromptentry?prompt-id=144&session-id=1'
the abstract command is:
curl -XPOST -d 'cmcc\x00\x00\x00\x01\x30cmbe\x00\x00\x00\x04menu' 'http://{APPLETV_IP}:3689/ctrl-int/1/controlpromptentry?prompt-id={CONTROL_PAIR_ID}&session-id={CONTROL_SESSION_ID}'
i extracted the CONTROL_PAIR_ID and CONTROL_SESSION_ID by setting my iphone wifi http proxy settings to my mac with fiddler on it and activated the old appletv remote app and that displayed the requests the app is executing
if you don't know how to set iphone to work with fiddler you can find it here:
http://docs.telerik.com/fiddler/Configure-Fiddler/Tasks/ConfigureForiOS
I did manage to control my Apple TV (currently running tvOS 9.2) from a python script. It turns out that you don't need to use Home Sharing to have a remote app control the Apple TV. I don't know if the following method will work if Home Sharing is enabled, but with it disabled on the Apple TV, the iOS Remote app has the option to manually add a device. (This may require removing all of the devices it is already paired with, since that was unfortunately necessary for me to get it to display the 'Add a device' button.) Once I had paired my iPhone to the Apple TV, I recorded some of its requests, copied the pairing GUID, and then constructed some of my own requests.
The only three requests necessary to make are:
/login?pairing-guid=< your pairing guid here >&hasFP=1
Logs into the Apple TV. The last four bytes of the response's is a session id, encoded as a big-endian four byte integer.
/logout?session-id=< your session id here >
Logs out. Not strictly necessary, as I found that logging in simply gets you a new session id, but probably not a bad idea to do things the way it expects.
/ctrl-int/1/controlpromptentry?prompt-id=114&session-id=< your session id here >
Send user input to the Apple TV. The data is one of several buffers that input a command, or possible a moving touch. For movement in the cardinal directions, sending several of these requests to simulate a moving touch is necessary.
I have a python script demonstrating how to do this here:
http://pastebin.com/mDHc353A
Utilizes the requests library: http://docs.python-requests.org/en/master/
Also special thanks to Adam Miskiewicz / github user skevy, since I made use of this file in his atlas-backend repo that conveniently had the right buffers to send for movement: https://github.com/skevy/atlas-backend/blob/master/atlas/services/appletv.coffee
For any people still checking out this question, I recommend checking out pyatv if they want to control their Apple TV through a python or command line interface.
I got a problem with RED5 in combination with Flash. For a personal project i am trying to make a Skype-like application. I already got a application that records the users webcam and saves it with a custom filename on the RED5 server.
But i am stuck trying to connect one more user to that window for a video chat. I made a new video container in Flash but i don't know how to connect a second client to the same stream in AS3 with Red5?
I searched on the net but i only get really old threads about RED5 in combination with Flex.
Maybe this is helping understanding my problem?
Could someone help me out? Or get me in the right direction?
Video chat? You will need 2 streams, for every client. Inbound and outbound. Outbound is a stream from the client to the media server, inbound is consumed stream of another user. So it will look like:
_streamOut = new NetStream(connection, NetStream.CONNECT_TO_FMS);
_streamIn = new NetStream(connection, NetStream.CONNECT_TO_FMS);
_streamOut.addEventListener(NetStatusEvent.NET_STATUS, onStreamOutNetStatus);
_streamIn.addEventListener(NetStatusEvent.NET_STATUS, onStreamInNetStatus);
_streamOut.attachAudio(microphone);
_streamOut.attachCamera(camera);
_streamOut.publish(hostPeerID);
_streamIn.play(companionPeerID);
Also there are some helpful examples, did you check them?
I'm working on a VoIP application for Windows Phone 8, and I want to cancel the echo produced when using speakerphone. Speex offers an AEC module, which I have tried to integrate into my application, but to no avail. My application works fine, but the echo persists. My code is based off the MS Chatterbox VoIP application, using WASAPI for capture and render. This is the form of the relevant sections (I tried to indicate what already existed and worked, and what was new):
Init:
// I've tried tail lengths between 100-500ms (800-4000 samples # 8KHz)
echoState = speex_echo_state_init (80, 800)
speex_echo_ctl(echoState, SPEEX_ECHO_SET_SAMPLING_RATE, 8000);
Render (runs every 10ms):
Read 10ms (80 samples) data from network (8KHz, 16 bit, mono)
NEW - speex_echo_playback(echoState, networkData)
Upsample data to 48KHz
Render data
Capture (runs every 10ms):
Capture 10ms of data (48KHz, 16 bit, mono)
Downsample to 8KHz
NEW - speex_echo_capture(echoState, downsampledData, echoCancelledData)
Send echoCancelledData to network
After reading the Speex documentation and looking at some posts on this site (not a lot of speex for Wp8 questions, but a few for android), I'm under the impression that this is, or is close to, the proper implementatinon of their API. So why isn't it working?
Thanks in advance
I'm trying to capture audio using WASAPI. My code is largely based on the ChatterBox VoIP sample app. I'm getting audio buffers, but they are all silent (flagged AUDCLNT_BUFFERFLAGS_SILENT).
I'm using Visual Studio Express 2012 for Windows Phone. Running on the emulator.
I had the exact same problem and managed to reproduce it in the ChatterBox sample app if I set Visual Studio to native debugging and at any point stepped through the code.
Also, closing the App without going through the "Stop" procedure and stopping the AudioClient will require you to restart the emulator/device before being able to capture audio data again.
It nearly drove me nuts before I figured out the before mentioned problems but I finally got it working.
So..
1. Be sure to NOT do native debugging
2. Always call IAudioClient->Stop(); before terminating the App.
3. Make sure you pass the correct parameters to IAudioClient->Initialize();
I've included a piece of code that works 100% of the time for me. I've left out error checking for clarity..
LPCWSTR pwstrDefaultCaptureDeviceId =
GetDefaultAudioCaptureId(AudioDeviceRole::Communications);
HRESULT hr = ActivateAudioInterface(pwstrDefaultCaptureDeviceId,
__uuidof(IAudioClient2), (void**)&m_pAudioClient);
hr = m_pAudioClient->GetMixFormat(&m_pwfx);
m_frameSizeInBytes = (m_pwfx->wBitsPerSample / 8) * m_pwfx->nChannels;
hr = m_pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_NOPERSIST | AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
latency * 10000, 0, m_pwfx, NULL);
hr = m_pAudioClient->SetEventHandle(m_hCaptureEvent);
hr = m_pAudioClient->GetService(__uuidof(IAudioCaptureClient),
(void**)&m_pCaptureClient);
And that's it.. Before calling this code I've started a worker thread that will listen to m_hCaptureEvent and call IAudioCaptureClient->GetBuffer(); whenever the capture event is triggered.
Of course using Microsoft.XNA.Audio.Microphone works fine to, but it's not always an option to reference the XNA framework.. :)
It was a really annoying problem which waste about 2 complete days of mine.My problem was solved by setting AudioClientProperties.eCatagory to AudioCategory_Communications instead of AudioCategory_Other.
After this long try and error period I am not sure that the problem won't repeat in the future because the API doesn't act very stable and every run may return a different result.
Edit:Yeah my guess was true.Restarting the wp emulator makes the buffer silent again.But changing the AudioClientProperties.eCatagory back to AudioCategory_Other again solve it.I still don't know what is wrong with it and what is the final solution.
Again I encounter the same problem and this time commenting (removing) the
properties.eCategory = AudioCategory_Communications;
solve the problem.
I can add my piece of advice for Windows Phone 8.1.
I made the following experiment.
Open capture device. Buffers are not silent.
Open render device with AudioDeviceRole::Communications. Buffers immediately go silent.
Close render device. Buffers are not silent.
Then I opened capture device with AudioDeviceRole::Communications and capture device works fine all the time.
For Windows 10 capture device works all the time, no matter if you open it with AudioDeviceRole::Communications or not.
I've had the same problem. It seems like you can either use only AudioCategory_Other or create an instance of VoipPhoneCall and use only AudioCategory_Communications.
So the solution in my case was to use AudioCategory_Communications and create an outgoing VoipPhoneCall. You should implement the background agents as in Chatterbox VoIP sample app for the VoipCallCoordinator to work .