I'm wondering how I can use Sauce Connect and their rest api to disable disable video recording and screen shots. Thanks!
The only way I know to disable video recording and screenshots is when you create a WebDriver instance with Selenium, you have to set the desired capabilities named record-screenshots and record-video to "false". For instance, in Python:
from selenium import webdriver
desired_capabilities = dict(
webdriver.DesiredCapabilities.CHROME)
desired_capabilities["record-screenshots"] = "false"
desired_capabilities["record-video"] = "false"
driver = webdriver.Remote(
desired_capabilities=desired_capabilities,
command_executor="http://localhost:4444/wd/hub")
The REST API is meant to be used after a test has started so it would not be able to prevent the creation of the video and screenshots in the first place. I've seen no evidence that Sauce Connect would be able to do anything about this.
Here is a link to the Sauce Labs documentation (https://docs.saucelabs.com/reference/test-configuration/#disabling-video-recording) explaining how to disable the video recoding and screen captures. It's actually a desired capability that is passed as part of the test. Can you please provide more clarity on the Sauce Connect question.
You can set a boolean value as part of DesiredCapabilities to turn video recording on or off.
Suggest that it makes sense to only record video when the test fails, which is what Saucery does. It does work. Have a look at this class
Related
Is there possibility to add Extra Voice Commands to Voice Guidance which is running from here maps?
-like Turn Right (From Here maps) something like- I want (Stop After turn Right)
NMAAudioManager is the central class that is used by SDK for iOS to modify the application AVAudioSession and play audio. It is the interface that the NMANavigationManager uses to play audio feedback such as voice instructions. You can also use NMAAudioManager to change whether hardware keys directly control HERE SDK volume, and also use it to set volume as a factor relative to the user's device volume.
The NMAAudioManager contains a queue of audio output objects. You can add to this queue by calling playOutput: with NMAAudioFileOutput, NMATTSAudioOutput, or your own NMAAudioOutput implementation. You can also use NMAAudioManager methods such as clearQueue, skipCurrentOutput, and stopOutputAndClearQueue to manage audio output in this queue.
Please refer below link for detailed implementation :
developer.here.com/documentation/ios-premium/dev_guide/topics/audio-management.html
I have a DirectShow webcam application. I make use of Sample Grabber to get the buffer callbacks and IVideoWindow to control the display co-ordinates for the Preview. I have Preview and Capture Streams which I run as below.
g_pBuild->RenderStream(&PIN_CATEGORY_CAPTURE, &MEDIATYPE_Video,cam,g_pGrabberF,pNullRenderer2); g_pBuild->RenderStream(&PIN_CATEGORY_PREVIEW, &MEDIATYPE_Video,cam,NULL,NULL);
On certain On board cameras, IMediaControl::Run followed by IMediaControl::Stop followed by IMediaCOntrol::Run doesn't switch on the camera.
Extenal USB cameras work properly here. How can I diagnose more on this? Any pointers, please help.
Maybe its specific to a certain hardware issue in the unit.
Do a quick test by adding sleep of 1 sec between calls.
If it does help than you need to find a way to know when to unit state in idle or not.
There are two important parts of the question which you did not provide:
Filter graph topologies
HRESULTs of the method calls
A problem you might be having is that one of the filters in the topology does not handle well state transitions and fails somewhere between states. Supposedly your second Run meets it still trying to complete Stop. You might get a HRESULT there which indicates the issue (better for you) or the filter fails silently.
The filter graph's is the unlikely source of the bug itself. Chances are high that it does everything flawlessly, however since internally it distributes the calls between filters, one of the filter is letting you down.
I have to configure my video camera display resolution before capturing and processing the data. Initially I did it as follows.
Created all necessary interfaces.
Added camera and renderer filters
Did RenderStream with Capture and Preview PIN Categories.
Then did the looping through AM_MEDIA_TYPE structures and setting the params.
This worked for a lot of cameras, but a few cameras failed. Then I changed the order of 3 and 4 given above. That is, I did the setting of params before the RenderStream. This time, the error cases went through, but a few On board cameras in SONY VAIO laptop etc seem to fail.
Now, my questions are
Which is the optimal and correct method of getting and setting AM_MEDIA_TYPE parameters and running the graph?
If there are different cameras, if I get an indication of which order is the best for a particular camera by going through the camera's DirectShow interfaces, that will also serve my purpose.
Please help me in this at the earliest,
Thanks and regards,
Shiju
IAMStreamConfig::SetFormat needs to be used to set capture format before the pin is connected and rendered. This way the downstream subchain of filters is built with proper media types.
FMLE = Flash live media encoder 3.0
i have posted this question on Adobe Forum, but not sure if they have people on that forum with programming experience.
I am a developer writing a video capture and audio capture device. The devices already work in other encoders. The devices are written in directshow. I am integrating with FMLE and encountered this problem.
The audio device doesnt have a usable volume bar in FMLE. The FMLE error is "The selected audio device "censored (company secret)" doesn't allow setting volume intensity. Disabling the volume slider control."
my audio device implements these interfaces along with the standard directshow filter interfaces
IBasicAudio
IAMAudioInputMixer
I put tracepoints in queryinterface and found FMLE query's for (my comments in comment string)
{IID_IUnknown}
{IID_IPersistPropertyBag}
{IID_IBaseFilter}
{IID_IAMOpenProgress}
{IID_IAMDeviceRemoval}
{IID_IMediaFilter}
{IID_IAMBufferNegotiation}
{IID_IAMStreamConfig}
{IID_IPin}
{IID_IReferenceClock}
{IID_IMediaSeeking}
{IID_IMediaPosition}
{IID_IVideoWindow} // WTF ?? query video window ?
{IID_IBasicAudio}
{2DD74950-A890-11D1-ABE8-00A0C905F375} // i think this is async stream,
What am i missing ? FMLE doesnt use IAMAudioInputMixer ?
Anyone know the exact interface which FMLE uses for Volume intensity ? . .I assumed it was IBasicAudio, but it doesnt seem to call any methods in there.
Answer provided by Ram Gupta of adobe forum.
"FMLE does not query for CLSID_AudioInputMixerProperties interface.
FMLE enumerates all the pin of audio source filter(using EnumPins) and then it extracts each pin info using QueryPinInfo Function.
FMLE searches for the audio filter Pin whose direction is PINDIR_INPUT(using QueryPinInfo) and then it queries for IAMAudioInputMixer interface to set the volume level.
Could you pls chk if the following functions are properly implemented
-->get_enable: it should set its parameter value to true.
-->put_MixLevel
-->QueryPinInfo:"
This solution did work. My problem was that because i never declared an input pin (since i dont have any directshow related input).
1-we are trying to write an application which dial a number and play a voice file instead of microphone input. Is it possible in Maemo (N900)?
we can not find any ""Answering Machine " like program in N900. is this means that there is no way to play a voice file instead of Microphone input?
There is a way. Play that voice file and make pulseaudio believe it's a proper input, and disable the microphone input. For more information see my question:
How to redirect from Audio Output to Mic Input using PulseAudio?
It is possible but you need a good pulseaudio knowledge to do it, I can already set it easily on my PC using pavucontrol. Drop me a message (or better, answer my question) if you bite the bullet and decide to learn how to use pactl/pacmd.