audio-streaming

问题 I would like to generate 2 audio tones using audiotestsrc but then duplicate those two channels across 16 channels (i.e 8channels of the one tone and 8channels of the other). I have a command that generates 2 tones for 2 channels: gst-launch-1.0 interleave name=i ! audioconvert ! wavenc ! filesink location=file.wav audiotestsrc wave=0 freq=100 volume=0.4 ! decodebin ! audioconvert ! "audio/x-raw,channels=1,channel-mask=(bitmask)0x1" ! queue ! i.sink_0 audiotestsrc wave=1 freq=150 volume=0.4 !

问题 If you load remote files with the AVPlayer it sends a http request with a range field in the request header, something like Range: bytes=0-8148096 I like to use the SevenDigital commercial API for streaming songs but they cannot handle this Range header. Is there a way to change the URL requests the AVPlayer sends? 回答1: Nope, and it is an apple standard that media providers need to support http 1.1 with the range header (check out the iTunes store guidelines for podcasts for example), so I

问题 If you load remote files with the AVPlayer it sends a http request with a range field in the request header, something like Range: bytes=0-8148096 I like to use the SevenDigital commercial API for streaming songs but they cannot handle this Range header. Is there a way to change the URL requests the AVPlayer sends? 回答1: Nope, and it is an apple standard that media providers need to support http 1.1 with the range header (check out the iTunes store guidelines for podcasts for example), so I

问题 If you load remote files with the AVPlayer it sends a http request with a range field in the request header, something like Range: bytes=0-8148096 I like to use the SevenDigital commercial API for streaming songs but they cannot handle this Range header. Is there a way to change the URL requests the AVPlayer sends? 回答1: Nope, and it is an apple standard that media providers need to support http 1.1 with the range header (check out the iTunes store guidelines for podcasts for example), so I

问题 If you load remote files with the AVPlayer it sends a http request with a range field in the request header, something like Range: bytes=0-8148096 I like to use the SevenDigital commercial API for streaming songs but they cannot handle this Range header. Is there a way to change the URL requests the AVPlayer sends? 回答1: Nope, and it is an apple standard that media providers need to support http 1.1 with the range header (check out the iTunes store guidelines for podcasts for example), so I

问题 I'd like to capture audio streaming from a live radio on internet using ffmpeg. If you have some examples or documentation it will be very appreciated. 回答1: Assuming the protocol is HTTP and audio format is MP3 it can be as simple as: ffmpeg -i http://server:port -c copy output.mp3 See: FFmpeg Protocols Documentation ffmpeg Documentation: Stream copy 来源： https://stackoverflow.com/questions/45239954/how-to-capture-audio-streaming-with-ffmpeg

问题 So, I'm building a threaded IMA ADPCM decoder streaming audio data to OpenAL (see below for short description) but I've run into some trouble. One of my issues is that sometimes my call to alBufferData: alBufferData(*bufferID, format, pcmData, sizeInBytes, bitRate); returns AL_INVALID_VALUE even though, when checking the parameters they look, e.g., like this: bufferID='109770616', format='AL_FORMAT_STEREO16', dataPtr='109754188', sizeInBytes='8164' Any clues, anyone? The actual sound being

问题 I am trying to set up a server to receive audio from a client browser using SocketIO , then process it through Google Speech-to-Text, and finally reply back to the client with the text. Originally and ideally, I wanted to set up to function somewhat like the tool on this page: https://cloud.google.com/speech-to-text/ I tried using getUserMedia and streaming it through SocketIO-Stream , but I couldn't figure out how to 'pipe' MediaStream . Instead, now I've decided to use MediaRecorder on the

问题 I am trying to set up a server to receive audio from a client browser using SocketIO , then process it through Google Speech-to-Text, and finally reply back to the client with the text. Originally and ideally, I wanted to set up to function somewhat like the tool on this page: https://cloud.google.com/speech-to-text/ I tried using getUserMedia and streaming it through SocketIO-Stream , but I couldn't figure out how to 'pipe' MediaStream . Instead, now I've decided to use MediaRecorder on the

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