Hi, I'm an intern of a well-known telecommunication french operator. The subject of my intership is evaluating the WebRTC technology's performances through google Chrome and Mozilla Firefox. First I studied how work this technology with Firefox Hello, to understand what is WebRTC, how it works, how media-stream are exchanged etc.
Then I started to work with the website https://apprtc.appspot.com/ On chrome and Firefox, to compare the behavior of every brower through the same webRTC app. Till now everything was good except the fact that I didn't master content injection (video resolution), the A/V bitrate and A/V encoders. 'caus as you know it, the video bitrate is blocked around 2Mb/sec. And we would like to see if it's possible to unleash the video bitrate to have a better quality and by using another less greedy encoder (VP8 ? H.264?) A formation was given to my colleagues to develop their own WebRTC app using javascript language. It allows us to manage all parameters of a WebRTC communication (media constraints etc.). So now I can inject 1080p content into the communication, unleash the bitrate, manage the displayed resolution. Just to specify, I used Chrome first. Anytime I modify the content of the SDP packet to manage de max bitrate : localSDP.sdp = localSDP.sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:4000\r\n'); So I could therefore make my own tests without any problems on Chrome. But I have to compare my results with another A/V encoder for subjectives tests. On Mozilla Firefox, the H.264 encoder works, even if it's not selected by default into a WebRTC communication (using Firefox Hello or apprtc). So I modify the content of the SDP packet to force the choice of the A/V encoder (replacing VP8 with H.264): localSDP.sdp = localSDP.sdp.replace( / 120 126 /g,' 126 '); I can do the same thing to replace the opus encoder with the G.722 encoder. If forcing A/V encoders works without problems, the media-stream parts are giving me headaches. In fact, if I inject 1080p content (with Firefox obviously) during a WebRTC app, the resolution in return has been downscaled to 640x480. And it doesn't depends on the resolution injected, even if the source resolution is smaller than 640x480, what I see in output is 640x480. But if I modify the displayed resolution, it's good in output but I lose the audio stream. As a matter of fact, till i try to modify the resolution, when the communication begin, the browser doesn't ask me to share the microphone. To explain a bit more how work the app I use, i give you some sourcecode (not all because i didn't develop it, so I can't share all the code) : var mediaConstraints = {video:true,audio:true}; [...] mediaConstraints = generateMediaConstraints(); [...] function generateMediaConstraints(){ if (replaceResolution == true && replaceFPS == true && replaceRatio == true){ return { video: { mandatory: { maxWidth: userMaxWidht.value, maxHeight: userMaxHeight.value, minWidth: userMinWidht.value, minHeight: userMinHeight.value, maxAspectRatio: userRatio.value, minAspectRatio: userRatio.value, minFrameRate: userFPS.value, maxFrameRate: userFPS.value } } }; } else if(replaceResolution == true && replaceFPS == true && replaceRatio == false){ return { video: { mandatory: { maxWidth: userMaxWidht.value, maxHeight: userMaxHeight.value, minWidth: userMinWidht.value, minHeight: userMinHeight.value, minFrameRate: userFPS.value, maxFrameRate: userFPS.value } } }; } else if(replaceResolution == true && replaceFPS == false && replaceRatio == true){ return { video: { mandatory: { maxWidth: userMaxWidht.value, maxHeight: userMaxHeight.value, minWidth: userMinWidht.value, minHeight: userMinHeight.value, maxAspectRatio: userRatio.value, minAspectRatio: userRatio.value } } }; } else if(replaceResolution == true && replaceFPS == false && replaceRatio == false){ return { video: { mandatory: { maxWidth: userMaxWidht.value, maxHeight: userMaxHeight.value, minWidth: userMinWidht.value, minHeight: userMinHeight.value } } }; } else if(replaceResolution == false && replaceFPS == true && replaceRatio == true){ return { video: { mandatory: { maxAspectRatio: userRatio.value, minAspectRatio: userRatio.value, minFrameRate: userFPS.value, maxFrameRate: userFPS.value } } }; } else if(replaceResolution == false && replaceFPS == true && replaceRatio == false){ return { video: { mandatory: { minFrameRate: userFPS.value, maxFrameRate: userFPS.value } } }; } else if(replaceResolution == false && replaceFPS == false && replaceRatio == true){ return { video: { mandatory: { maxAspectRatio: userRatio.value, minAspectRatio: userRatio.value } } }; } else{ return {video:true,audio:true} } } I tried to put the audio variable into one of the subconditions' function, ie : else if(replaceResolution == true && replaceFPS == false && replaceRatio == false){...} It works because the audio stream work, but I can't manage the resolution anymore. I don't underderstand why and how. Colleagues who développed this app left the compagny. I'm a really ban developper (well, I'm not a dev). I'm blocked on this f****** since one week. I don't see what to do. Could you please help me to fix that ? maybe give me the solution I dunno. Thanks in advance. _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
