audio-streaming

问题 I have developed video streaming application with RED5 media server(RTMP). Instead of RTMP need to stream live video through HTTP. Any open source HTTP media server?? Is any open source server which supports both RTMP and HTTP ? Thanks in advance. 回答1: Primarily, HTTP and RTMP are different protocols. You won't serve RTMP inside the HTTP. (Although you can do this for a tunneling solution). Exist several ways to do HTTP Streaming. Such as HLS, DASH, Smooth and Progresive Download. If you need

问题 I have developed video streaming application with RED5 media server(RTMP). Instead of RTMP need to stream live video through HTTP. Any open source HTTP media server?? Is any open source server which supports both RTMP and HTTP ? Thanks in advance. 回答1: Primarily, HTTP and RTMP are different protocols. You won't serve RTMP inside the HTTP. (Although you can do this for a tunneling solution). Exist several ways to do HTTP Streaming. Such as HLS, DASH, Smooth and Progresive Download. If you need

问题 I'm trying to open 2 microphone streams with google's Oboe library like this, for each microphone: oboe::AudioStreamBuilder builder; builder.setChannelCount(channelCount) ->setDirection(isInput ? oboe::Direction::Input : oboe::Direction::Output) ->setSharingMode((oboe::SharingMode) sharingMode) ->setPerformanceMode((oboe::PerformanceMode) performanceMode) ->setInputPreset((oboe::InputPreset)inputPreset) ->setDeviceId(deviceId) ->setSessionId((oboe::SessionId) sessionId) ->setSampleRate

问题 I'm facing issues with m3u8 playlists on chrome and firefox (edge works like a charm) I'm trying to send data converted with ffmpeg the line I use for conversion is the following making the m3u8 file at same time: ffmpeg -i .\european-anthem-2012.mp3 -c:a aac -b:a 64k -f segment -segment_time 15 -segment_time_delta 3 -segment_list_type m3u8 -segment_list playlist_aac.m3u8 outfile_aac-%03d.aac As destination formats I've tried also mp3: -c:a mp3 -b:a 64k and vorbis: -c:a libvorbis -b:a 64k the

问题 I'm trying to implement simple HTTP server for real-time audio (in Java). Suppose there is a website where you can see a list of songs which are playing one after another. When client connects to server - lets say in the middle of the song - I'm thinking to use "Range" HTTP header and send the data range starting from that part of song. But if during download connection is temporary lost (and song finished) - should the server send previous song part and finish it - or should the server send

问题 I am currently working in using the HTML5 audio player to provide a audio stream (24/7 radio stream) via the (mobile) browser. Loading in the stream and playing it works fine. The major problem is that the HTML5 <audio> tag will keep downloading (buffering) content even when its not active. This could be a major issue for mobile users since most of them pay for data use. So far I have not been able to find a decent solutions that works cross browser to prevent this. I tried so far: Unload the

问题 I am currently working in using the HTML5 audio player to provide a audio stream (24/7 radio stream) via the (mobile) browser. Loading in the stream and playing it works fine. The major problem is that the HTML5 <audio> tag will keep downloading (buffering) content even when its not active. This could be a major issue for mobile users since most of them pay for data use. So far I have not been able to find a decent solutions that works cross browser to prevent this. I tried so far: Unload the

问题 My goal is to enable screen sharing in the middle of a video or audio call using webrtc web application . Well I found that I can use MediaStreamTrack.applyConstraints() to change video property but is it possible to change the video source ? further more how can I add video to an existing audio stream . I need this to work on chrome only for now . 回答1: As of today, I feel a better approach for your case would be to use the RTCRtpSender.replaceTrack method. Assuming your camera stream is

问题 My goal is to enable screen sharing in the middle of a video or audio call using webrtc web application . Well I found that I can use MediaStreamTrack.applyConstraints() to change video property but is it possible to change the video source ? further more how can I add video to an existing audio stream . I need this to work on chrome only for now . 回答1: As of today, I feel a better approach for your case would be to use the RTCRtpSender.replaceTrack method. Assuming your camera stream is

问题 i have a java application that records audio from a mixer and store it on a byte array, or save it to a file. What I need is to get audio from two mixers simultaneously, and save it to an audio file (i am trying with .wav). The thing is that I can get the two byte arrays, but don't know how to merge them (by "merge" i don't mean concatenate). To be specific, it is an application that handles conversations over an USB modem and I need to record them (the streams are the voices for each talking