Re: [discuss-webrtc] Missing "Enable Mediastream" flag in chrome

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Punyabrata Ray

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Jun 22, 2012, 12:22:05 PM6/22/12
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Hi Ram,

Enable Mediastream has come out of the flag and is available by default. As soon as you try to place a call now, you should see a bar at the top asking the user that this application is trying to access the camera and microphone.

/Ray

On Fri, Jun 22, 2012 at 7:25 AM, Ramasundar Kandasamy <krama...@yahoo.com> wrote:
Hi,

I am running Chrome 22.0.1183.0 canary.

This is my shortcut "C:\Users\****\AppData\Local\Google\Chrome SxS\Application\chrome.exe" --enable-media-stream --enable-peer-connection.

But when i navigate to chrome:\\flags page, "Enable Mediastream" flag is missing.

PS: "Enable PeerConnection" is available.

Do you have any suggestion to fix this?

Thanks,
Ram

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Serge Lachapelle

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Jun 22, 2012, 2:22:50 PM6/22/12
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good point. I have updated this one: 


Thanks!!!

/S

On Fri, Jun 22, 2012 at 7:06 PM, qzhua <qzh...@gmail.com> wrote:
The WebRTC document should be updated.


On Friday, June 22, 2012 10:25:13 PM UTC+8, Ramasundar Kandasamy wrote:
Hi,

I am running Chrome 22.0.1183.0 canary.

This is my shortcut "C:\Users\****\AppData\Local\Google\Chrome SxS\Application\chrome.exe" --enable-media-stream --enable-peer-connection.

But when i navigate to chrome:\\flags page, "Enable Mediastream" flag is missing.

PS: "Enable PeerConnection" is available.

Do you have any suggestion to fix this?

Thanks,
Ram

On Friday, June 22, 2012 10:25:13 PM UTC+8, Ramasundar Kandasamy wrote:
Hi,

I am running Chrome 22.0.1183.0 canary.

This is my shortcut "C:\Users\****\AppData\Local\Google\Chrome SxS\Application\chrome.exe" --enable-media-stream --enable-peer-connection.

But when i navigate to chrome:\\flags page, "Enable Mediastream" flag is missing.

PS: "Enable PeerConnection" is available.

Do you have any suggestion to fix this?

Thanks,
Ram

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Ramasundar Kandasamy

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Jun 22, 2012, 4:32:40 PM6/22/12
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Thanks a lot guys...


On Friday, June 22, 2012 8:22:50 PM UTC+2, Serge Lachapelle wrote:
good point. I have updated this one: 


Thanks!!!

/S

Michael Adeyeye

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Jun 23, 2012, 5:33:32 AM6/23/12
to discuss...@googlegroups.com

Hi,
Chrome canary does not work on any Linux OS. What's my best bet browser for webrtc works? I run Fedora 17. Please advice.

Regards,
Michael.

On 06/22/2012 07:28 PM, discuss...@googlegroups.com wrote:

Group: http://groups.google.com/group/discuss-webrtc/topics

    Raju <mmr...@gmail.com> Jun 22 09:39AM -0700  

    Thanks for the response.
    >You can't do that. Why would you want to?
    We wanted to interwork browser and native-IMS clients. Though, we can
    ignore the parts of SDP offer and exclude themi n SDP answer.
    But, in our scenario that approach may not work always.
    So, we resorted to updating the SDP generated by createOffer() and/or
    ignore parts of SDP in SDP offer to suit our needs.
     
    Thanks again!
    -Raju
     
    On Wednesday, June 20, 2012 11:57:33 PM UTC-5, Harald Alvestrand wrote:

     



    Hi,
     
    I am running Chrome 22.0.1183.0 canary.
     
    This is my shortcut "C:\Users\****\AppData\Local\Google\Chrome
    SxS\Application\chrome.exe" --enable-media-stream --enable-peer-connection.
     
    But when i navigate to chrome:\\flags page, "Enable Mediastream" flag is
    missing.
     
    PS: "Enable PeerConnection" is available.
     
    Do you have any suggestion to fix this?
     
    Thanks,
    Ram

     



      Hi Ram,
       
      Enable Mediastream has come out of the flag and is available by default. As
      soon as you try to place a call now, you should see a bar at the top asking
      the user that this application is trying to access the camera and
      microphone.
       
      /Ray
       
      On Fri, Jun 22, 2012 at 7:25 AM, Ramasundar Kandasamy <krama...@yahoo.com

       

        qzhua <qzh...@gmail.com> Jun 22 07:49AM -0700  

        Does WebRTC provide such functions in its JS-APIs. I can find any
        information anywhere? can any one help me?

         

        qzhua <qzh...@gmail.com> Jun 22 07:41AM -0700  

        When I start to look into the WebRTC APIs in W3C, I find it lack of details
        of how to control the media stream, for example how to specify the
        video/audio compression format, and how to enable/disable jitter buffering
        or error concealment. With out these details how can I establish talk with
        other? It seems that the WebRTC source code has provide such kind of
        functions, but how to invoke them with JavaScritp? The current WebRTC
        documents lack of basic information to JS programmers. Does any one can
        kindly provide some help?

         

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        manohar <manoh...@gmail.com> Jun 22 04:41AM -0700  

        Hi Emanuele,
        I also had a similar requirement of using ExternalPlayoutGetData
        (external audio sink). I implemented a thread in my application which calls ExternalPlayoutGetData
        for every ~10ms.
        I also do observe loss of audio sometimes. Is this the issue you are also
        facing ?
        I suspect this could be to deal something with ~10ms accuracy. Are there
        any other ways like callbacks or other to get the audio packets to my
        application asynchronously instead of application probing it for every
        ~10ms ??
         
        On Monday, October 3, 2011 6:19:40 PM UTC+5:30, emanuele bizzarri wrote:

         

        manohar <manoh...@gmail.com> Jun 22 04:43AM -0700  

        Hi Emanuele,
        I also had a similar requirement of using ExternalPlayoutGetData
        (external audio sink). I implemented
        a thread in my application which calls ExternalPlayoutGetData for every
        ~10ms.
        I also do observe loss of audio sometimes. Is this the issue you are also
        facing ?
        I suspect this could be to deal something with ~10ms accuracy. Are there
        any other ways like
        callbacks or other to get the audio packets to my application
        asynchronously instead of application
        probing it for every ~10ms ??
         
         
        On Monday, October 3, 2011 6:19:40 PM UTC+5:30, emanuele bizzarri wrote:

         

        Emanuele Bizzarri <emab...@gmail.com> Jun 22 02:59PM +0200  

        Hi manohar,
        my app plays webrtc recordings and converts them in other formats,
        using ffmpeg.
        I use webrtc voe ExternalPlayoutGetData with TimeSetEvent on windows, in
        order to have max precision.
        The result is good.
        During realtime playback I have no problem.
        My problem is in conversion. I'd like to convert the recording at the
        max speed possible, but I failed to achieve this result.
        My application converts the recording in a time that is at least equal
        to the duration of the recording, because the only way I've found, to
        play audio data without loss, is to use ExternalPlayoutGetData in
        "playback mode"
         
        Bye
        Emanuele
         
         
         
         
        Il 22/06/2012 13:43, manohar ha scritto:

         

        crmoratelli <crmor...@gmail.com> Jun 22 05:08AM -0700  

        Hi Leo,
         
        I tested the PixelView 8000GT2 with chrome canary and WebRTC but it doesn't
        work.
         
        I could see it works well against a typical webcam USB.
         
        Suggestions?
         
        Thanks.
         
        Moratelli
         
         
         
        On Wednesday, June 20, 2012 5:02:50 PM UTC-3, Leo Wang wrote:

         

        Dmitriy Berezovskiy <dberez...@gmail.com> Jun 21 11:20PM -0700  

        Hi all! Where can I find a detailed description of the classes and methods
        for these applications?

         

        kadam <ad...@kornafeld.com> Jun 21 04:35PM -0700  

        The warning disappeared after AudioSessionInitialize().
         
        ==Adam
         
        On Tuesday, June 19, 2012 5:17:35 PM UTC-4, kadam wrote:

         

        Luke Weber <luke....@gmail.com> Jun 21 11:55AM -0700  

        Just ignore it with an ifdef.
         
        #ifdef ANDROID is where this offending code lives, but you could substitute
        these undefined code blocks, at least for now, with #ifdef
        ANDROID_MEDIA_ENGINE
         
        As well if you want audio only like me, then you'll have to define VIDEO_ENG_NAME
        to be NullVideoEngine, less you'd need to compile in video for no reason.
         
        Example of the changes I made to make this work:
        https://github.com/lukeweber/webrtc-jingle/commit/95677857268dcb61016aed72a2435655a7390f67
         
        Luke
         
        On Thursday, June 21, 2012 4:01:23 PM UTC+2, kaiduan wrote:

         

        wesbos <wes...@gmail.com> Jun 21 10:51AM -0700  

        Pretty sure opera 12 just implemented the getUserMedia to access the device
        webcam/microphone. There are no current implementations for recording.
         
         
        On Thursday, 21 June 2012 00:52:39 UTC-4, Eric Thomas wrote:

         

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      PhistucK

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      Jun 25, 2012, 5:10:35 AM6/25/12
      to discuss...@googlegroups.com
      You can use the dev channel on Linux. It updates weekly (this is not set in stone, though).
      Either that, or download a continuous build, or a snapshot build of Chromium, manually. Note that you have to re-download them every time, because Chromium does not update automatically (there are unofficial update scripts, Google them).

      ☆PhistucK
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