web-audio-api

问题 Is it possible to play the whole audio instance at the same time? I know, I could create every time an new instance and call .play() on every instance, but I think its dirty. like sound1 = new Audio('sound1.mp3'); sound2 = new Audio('sound2.mp3); sound1.play(); sound2.play; Its more or less the same as in this thread: Play multiple sound at the same time But there has to be and elegant way? 回答1: Edit: MediaController and mediagroup turned out to be vaporware. You will want to stick with the

问题 I am attempting to make a request to the SoundCloud API. Then when I get the response I set the stream_url as the source of an < audio > element. This works: http://matthiasdv.org/beta/ But not always... When you search for 'Bonobo' for example, you can play the first few tracks without any issue. But when you try to play 'London Grammar - Hey Now (Bonobo remix)' - the 7th result - it won't play. It throws no errors whatsoever. I've been tinkering around with Chrome's webdev-tools and under

问题 I'm trying to play an infinite stream coming from the fetch API using Chrome 51. (a webcam audio stream as Microsoft PCM, 16 bit, mono 11025 Hz) The code works almost OK with mp3s files, except some glitches, but it does not work at all with wav files for some reason i get "DOMException: Unable to decode audio data" The code is adapted from this answer Choppy/inaudible playback with chunked audio through Web Audio API Any idea if its possible to make it work with WAV streams ? function play

问题 I created an ionic webrtc app that runs perfectly when using ionic serve (on web browser, witch is normal). but was not working at all on the device since the getUserMedia function was not able to execute. The solution i found is to install crosswalk, update permissions in the AndroidManifest.xml and add the meta tag on the index.html for content security: <meta http-equiv="Content-Security-Policy: media-src: 'self' mediastream"> Now, i have a working ionic webrtc app but only video, the

问题 I'm trying to use the Web Audio API's bandpass filter functionality, but I believe my question is more general. I don't understand the "Q" value of the bandpass filter. I would like to be able to configure the filter to pass frequencies that are within Y hertz of a middle frequency X hertz. I'm very new to audio programming, so are there other variables I need to consider to compute Q? 回答1: Let's say you have a filter at 1000Hz, and you want it to start at 500Hz and end at 2000Hz. First off,

问题 I'm trying to use the Web Audio API's bandpass filter functionality, but I believe my question is more general. I don't understand the "Q" value of the bandpass filter. I would like to be able to configure the filter to pass frequencies that are within Y hertz of a middle frequency X hertz. I'm very new to audio programming, so are there other variables I need to consider to compute Q? 回答1: Let's say you have a filter at 1000Hz, and you want it to start at 500Hz and end at 2000Hz. First off,

问题 The Problem I'm working on a web application where users can sequence audio samples and optionally apply effects to the musical patterns they create using the Web Audio API. The patterns are stored as JSON data, and I'd like to do some analysis of the rendered audio of each pattern server-side. This leaves me with two options, as far as I can see: Run my own rendering code server-side, trying to make it as faithful as possible to the in-browser rendering. Maybe I could even pull out the Web

问题 How can I specify the codecs used with the MediaRecorder API? The only option I see is for mimeType which isn't really sufficient. Cramming in the codecs in the mimeType option doesn't seem to work. var mediaRecorder = new MediaRecorder( outputMediaStream ), { mimeType: 'video/webm; codecs="opus,vp8"' } ); This results in a WebM stream with Vorbis and VP8: FFMPEG STDERR: Input #0, matroska,webm, from 'pipe:': Metadata: encoder : QTmuxingAppLibWebM-0.0.1 Duration: N/A, start: 0.000000, bitrate

问题 I've been checking Web Audio API documentation and the tutorials but haven't quiet figured out how to approach this problem. Let's say I load few wav files via XMLHttpRequest and then create buffersources. I know I can schedule when the playback starts precisely. But what if I don't want to play them, but instead want to store and schedule them in a buffer. A real example: I want to create a simple sequencer where you schedule drums and than export the whole mix to wav (without recording it

问题 Firefox 25 says to bring Web Audio but an important functions seems to be missing - createJavaScriptNode. I'm trying to build a analyser but I get the error in console that createJavaScriptNode is not a function. Demo - http://jsbin.com/olugOri/3/edit 回答1: You can try using createScriptProcessor instead. Firefox is still not getting correct values, but at least that error is no longer present. Demo - http://jsbin.com/olugOri/4/edit Edit : (more visibility for the important discussion in the