streaming

问题 I would like create a server to create subtitles for live videos on Facebook. I use Google Speech to convert sound to text. However, in order to do that, I need to read the facebook live streams. Using Facebook Live API, with me/live_videos , I get the following response: { "status": "LIVE", "stream_url": "rtmp://rtmp-api.facebook.com:80/rtmp/{id}", "secure_stream_url": "rtmps://rtmp-api.facebook.com:443/rtmp/{id}, "embed_html": "<iframe src=\"https://www.facebook.com/video/embed?video_id=

问题 I'm using a fairly limited embedded system so I can't use any of the libraries and I'm on my own building HTTP requests. I'm able to handle the stats pretty well with polling but I'm trying to turn on Rest Streaming The Nest site directs you to the Firebase site which directs you to the W3C site and all I get through all of that is to 'include the header: Accept: text/event-stream in your request'. If I send the request (after the redirect): GET /?auth=[auth code] HTTP/1.1 Host: firebase

问题 Is it possible to use progressive download for near real-time playback of a live video stream recorded with a webcam? What I need is that a video stream is recorded on one end, uploaded in real-time to a server and downloaded with a short delay, but in real-time, using progressive download (i.e., HTTP streaming) on another end for playback. Is it possible or does it require the use of RTMP? If it's possible, then does it require that Flash Media Server runs on a server? Thanks! 回答1: What you

问题 In the end what I want to do is to use a streaming interface with single precision floating point arrays in Vivado Design Suite to build hardware accelerators. HLS User Guide UG902 shows that is possible to create HW accelerators (starting from C, C++, SystemC, OpenCL code) using different interfaces. If you want to use an AXI4 streaming interface, HLS synthesizes the signals TREADY and TVALID but it doesn't synthesize the signal TLAST necessary to connect the RTL interface generated to Zynq

问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 12 months ago . I want to write an app to stream the current playing music to another device. The connection between the two devices does work and I also can stream some strings over wifi but I got problems with getting the current track. I used the code of this blog to get some info of the

问题 This is a problem I ran into using the Google Speech to Text Engine. I am currently streaming 16 bit / 16 kHz audio real time in 32kB chunks. But there is an average 25 second latency between sending audio and receiving transcripts, defeating the purpose of real time transcription. Why is there such high latency? 回答1: The Google Speech to Text documentation recommends using a 100 ms frame size to minimize latency. 32kB * (8 bits / 1 byte) * ( 1 sample / 16 bits ) * (1 sec / 16000 samples ) =

问题 My problem: When a user clicks an image button on an aspx page the codebehind creates a zip file and then I attempt to stream that zip file to the user. To stream the file I'm using the following code: FileInfo toDownload = new FileInfo(fullFileName); if (toDownload.Exists) { Response.Clear(); Response.ContentType = "application/zip"; Response.AppendHeader("Content-Disposition", "attachment;filename=" + toDownload.Name); Response.AppendHeader("Content-Length", toDownload.Length.ToString());

问题 I'm trying to upload big file by streaming, recently I got this error log: Error Domain=kCFErrorDomainCFNetwork Code=303 "The operation couldn’t be completed. (kCFErrorDomainCFNetwork error 303.)" UserInfo=0x103c0610 {NSErrorFailingURLKey=/adv,/cgi-bin/file_upload-cgic, NSErrorFailingURLStringKey/adv,/cgi-bin/file_upload-cgic}<br> this is where I set bodystream: -(void)finishedRequestBody{ // set bodyinput stream [self appendBodyString:[NSString stringWithFormat:@"\r\n--%@--\r\n",[self

问题 I am trying to stream large content (say 200 MB) of formatted data to PDF with minimal memory footprint (say 20 MB per Client/Thread). The PDF structure is written in Adobe postscript and it is complex to directly write in PDF postscript format. I have been using the following APIs to stream content to PDF. Jasper Reports iText The problem I am facing with Jasper reports is that it needs all the input data to be taken into in-memory and only supports OutputStream. There is a function

问题 I use AVPlayer that plays stream content. I want to know the time when buffer is empty and the time when it is ready to play. But the observers "playbackBufferEmpty" and "playbackLikelyToKeepUp" do not work every time as it needed. They sometimes work but often do not work. I use only iPad simulator iOS 6.1 under OSX 10.7.5. Here is how i set and listen observers: - (void)playAudioStream:(NSURL *)audioStreamURL { if(_audioPlayer && _audioPlayer.currentItem) { [_audioPlayer removeObserver:self