I'm trying to play sound from ByteArray captured from the microphone and I'm expecting to hear the sound from the microphone but what I get is only random, distorted sound. This is the code that I'm using now :
var playBa:ByteArray;
var player:Sound = new Sound();
sound.addEventListener(SampleDataEvent.SAMPLE_DATA, playMic);
sound.play();
var mic:Microphone = Microphone.getMicrophone();
mic.gain = 100;
mic.rate = 44;
mic.setSilenceLevel(0, 4000);
mic.addEventListener(SampleDataEvent.SAMPLE_DATA, onMicSample);
function playMic(e:SampleDataEvent):void
{
if (playBa != null) e.data.writeBytes(playBa, 0, playBa.length);
}
function onMicSample(e:SampleDataEvent):void
{
playBa = e.data;
playBa.position = 0;
player.play();
}
And after a few seconds of distorted sounds, this is what I got on the Output Window:
RangeError: Error #2004: One of the parameters is invalid.
at flash.media::Sound/play()
at vclass_fla::MainTimeline/onMicSample()
Please help me. I'm a totally new to AS3 and here. Any help are highly appreciated. Thanks.
Some Microphones are very sensitive. You need to adjust moderate settings like shown below. Also, if speakers are closer to microphone the sound get looped.
var myMic:Microphone = Microphone.getMicrophone(); // detect microphone
myMic.gain = 50;
myMic.setUseEchoSuppression(true);
myMic.setLoopBack(true);
myMic.setSilenceLevel(50, 1000);
To more about Sound Capturing try this,
Capturing microphone sound data
The Problem is that sample data event requires blocks of 2048 to 8192 samples. If the samples recorded by microphone at the call of "playMic" the runtime throw "RangeError: Error #2004...".
function playMic(event:SampleDataEvent):void
{
trace("sample event");
for (var ii:uint = 0; ii < 8192 && playBa.bytesAvailable > 0 ; ii++)
{
var n1:Number = playBa.readFloat();
//trace(n1);
event.data.writeFloat(n1);
event.data.writeFloat(n1);
if(playBa.bytesAvailable == 0)
{
trace("data_finished");
break;
}
}
}
Related
I would like to know is there a way we can detect audio frequency from microphone in html 5 web audio. I wish to make an online guitar tuner, and I need to have the audio frequency in hertz, from the sound input. I've seen some EQ and filters effects, but I didn't see anything about frequency recognition.
EDIT:
I found this: http://www.smartjava.org/content/exploring-html5-web-audio-visualizing-sound
The 2nd point (analyser node) is really interesting. I seen his source code, but I can't figure how to connect the analyser to the microphone input.
He calls a playSound() function when the mp3 file starts to play, and there he draws his canvas. But I do not have a playSound() like function...
I wrote a web audio library which, among other things, can detect frequency from mic input. Check it out at https://github.com/rserota/wad#pitch-detection
var voice = new Wad({source : 'mic' });
var tuner = new Wad.Poly();
tuner.add(voice);
voice.play();
tuner.updatePitch() // The tuner is now calculating the pitch and note name of its input 60 times per second. These values are stored in tuner.pitch and tuner.noteName.
var logPitch = function(){
console.log(tuner.pitch, tuner.noteName)
requestAnimationFrame(logPitch)
};
logPitch();
// If you sing into your microphone, your pitch will be logged to the console in real time.
tuner.stopUpdatingPitch(); // Stop calculating the pitch if you don't need to know it anymore.
You should be able to use BiquadFilterNode.
Example code from the link:
var audioCtx = new AudioContext();
var biquadFilter = audioCtx.createBiquadFilter();
biquadfilter.getFrequencyResponse(myFrequencyArray,magResponseOutput,phaseResponseOutput);
You can use the following code to get the frequencies from the mic.
navigator.mediaDevices.getUserMedia({audio:true}).then(function(localStream){
var audioContext = new(window.AudioContext || window.webkitAudioContext)();
var input = audioContext.createMediaStreamSource(localStream);
var analyser = audioContext.createAnalyser();
var scriptProcessor = audioContext.createScriptProcessor();
// Some analyser setup
analyser.smoothingTimeConstant = 0;
analyser.fftSize = 64;
input.connect(analyser);
analyser.connect(scriptProcessor);
scriptProcessor.connect(audioContext.destination);
var getAverageVolume = function( array){
var length = array.length;
var values = 0;
var i = 0;
for (; i < length; i++) {
values += array[i];
}
return values / length;
};
var onAudio = function(){
var tempArray = new window.Uint8Array(analyser.frequencyBinCount);
analyser.getByteFrequencyData(tempArray);
var latestFrequency = (getAverageVolume(tempArray));
//use latestFrequency
};
scriptProcessor.onaudioprocess = onAudio;
})
.catch(function(){
//Handle error
});
I would like to use present time of video to manage all video. For example, I have some cue points, I choose it and video is now playing from this cue point, and after 10 sec video goes to other part and everything depends on present time of video. I do not use it on web so my video isn't loading (I mean that property of VideoProgressEvent like bytesLoaded will not help me). Is it possible to do it in action script 3.0 ? Another question is if I can add some transitions between cue points.
import fl.video.*;
// Video component instance name
var flvControl:FLVPlayback = display;
var flvSource:String = "myMovie.flv";
// Set video
flvControl.source = flvSource;
var myTextFormat:TextFormat = new TextFormat();
myTextFormat.size = 20
btn1.setStyle("textFormat", myTextFormat);
btn2.setStyle("textFormat", myTextFormat);
btn3.setStyle("textFormat", myTextFormat);
btn4.setStyle("textFormat", myTextFormat);
display.autoPlay = false;
// Add seek to time code
function seekToTimeHandler1(event:MouseEvent):void
{
var sec:Number = 15;
flvControl.seek(sec);
}
btn1.addEventListener(MouseEvent.CLICK, seekToTimeHandler1);
// Add seek to time code
function seekToTimeHandler2(event:MouseEvent):void
{
var sec:Number = 61;
flvControl.seek(sec);
}
btn2.addEventListener(MouseEvent.CLICK, seekToTimeHandler2);
// Add seek to time code
function seekToTimeHandler3(event:MouseEvent):void
{
var sec:Number = 63;
flvControl.seek(sec);
}
btn3.addEventListener(MouseEvent.CLICK, seekToTimeHandler3);
// Add seek to time code
function seekToTimeHandler4(event:MouseEvent):void
{
var sec:Number = 80;
flvControl.seek(sec);
}
btn4.addEventListener(MouseEvent.CLICK, seekToTimeHandler4);
Now I have a code like this:
soundData = new ByteArray();
microphone = Microphone.getMicrophone();
microphone.codec = SoundCodec.SPEEX;
microphone.rate = 8;
microphone.gain = 100;
microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, micSampleDataHandler);
function micSampleDataHandler(event:SampleDataEvent):void {
while (event.data.bytesAvailable) {
var sample:Number = event.data.readFloat();
soundData.writeFloat(sample);
}
}
The raw data is recorded from the microphone. How do I go about casting it to a ByteArray after using SPEEX codec compression? Note that the converted data must play back.
refer a this code.
soundData.position=0;
var soundOutput:Sound = new Sound();
soundOutput.addEventListener(SampleDataEvent.SAMPLE_DATA, playSound);
soundOutput.play();
function playSound(soundOutput:SampleDataEvent):void {
if (! soundData.bytesAvailable>0)
{
return;
}
for (var i:int = 0; i < 8192; i++)
{
var sample:Number=0;
if (soundData.bytesAvailable>0)
{
sample=soundData.readFloat();
}
soundOutput.data.writeFloat(sample);
soundOutput.data.writeFloat(sample);
}
}
using a SoundCodec.SPEEX above code playrate not is 1x you should correct playSound function. maybe you tested. if you remove microphone.codec = SoundCodec.SPEEX; know.
More information: Adobe Official Capturing sound input
have a some problem when recorded in speex.
refer a follow artice.
http://forums.adobe.com/message/3571251#3571251
http://forums.adobe.com/message/3584747
If the SoundFormat indicates Speex, the audio is compressed mono sampled at 16 kHz. In flash, a sound object plays at 44khz. Since you're sampling at 16khz(Speex), you're sending data through the SampleDataEvent Event handler 2.75 faster then you are getting that data.
so, you must changed the playSound for(or while) loop.
I recommend following site. this article is 'how to playrate adjust?' great tutorial.
http://www.kelvinluck.com/2008/11/first-steps-with-flash-10-audio-programming/
I am trying to play a flv video file in flex 4.5 with netStream byteArray. What I am doing is below:
Creating a netStream and video object
Attaching a netStream with video
Reading flv file in byteArray
Append byteArray in netStream using "appendBytes" method
Playing video
In this scenario Play, Pause, Stop functionalities are working fine with video.
But when I am trying to seeking in video then it is not working.
You can follow the code what I am doing by clicking on the link http://pastebin.com/fZp0mKDs
Can anybody tell me, where am I am going wrong to implement seeking.
Any code sample or any kind of help would be appreciated.
I got, the code below worked in my case
// onmetadata function get all timestamp and corresponding fileposition..
function onMetaData(informationObject:Object):void
{
for (var propertyName:String in informationObject)
{
if (propertyName == "keyframes")
{
var kfObject:Object = informationObject[propertyName];
var timeArray:Array = kfObject["times"];
var filePositionArray:Array = kfObject["filepositions"];
for(var i:int=0;i<timeArray.length;i++)
{
var tagPosition:int = filePositionArray[i];//Read the tag size;
var timestamp:Number = timeArray[i];//read the timestamp;
tags.push({timestamp:timestamp,tagPosition:tagPosition});
}
}
}
}
// onseek click get approximate timestamp and its fileposition
protected function seek_click(seektime:Number):void
{
var currentTime:Number = 0;
var previousTime:Number = 0;
for (var i:int=1; i<tags.length; i++)
{
currentTime = tags[i].timestamp;
previousTime = tags[i-1].timestamp;
if(previousTime < seektime)
{
if(seektime < currentTime)
{
seekPos = tags[i-1].tagPosition;
stream.seek(previousTime);
break;
}
}
}
}
// append bytes on seekposition
private function netStatusHandler(event:NetStatusEvent):void
{
switch (event.info.code)
{
case "NetStream.Seek.Notify" :
stream.appendBytesAction(NetStreamAppendBytesAction.RESET_SEEK);
totalfilePositionArray.position = seekPos;
var bytes:filePositionArray = new filePositionArray();
totalfilePositionArray.readBytes(bytes);
stream.appendBytes(bytes);
stream.resume();
break;
}
}
For inject MetaData keyframes into flv file.Use some injector tool, fe. FLV MetaData Injector
http://www.buraks.com/flvmdi/
I think there is a problem in seeking of byteArray constructed after reading file. Just play you netStream directly, it works:
var fileName:String = "dummy-video.flv";
ns.play(fileName);
Suppose there is a live WAV stream that can be reached at a certain URL, and we need to stream it with as little latency as possible. Using HTML5 <audio> for this task is a no-go, because browsers attempt to pre-buffer several seconds of the stream, and the latency goes up accordingly. That's the reason behind using Flash for this task. However, due to my inexperience with this technology, I only managed to get occasional clicks and white noise. What's wrong in the code below? Thanks.
var soundBuffer: ByteArray = new ByteArray();
var soundStream: URLStream = new URLStream();
soundStream.addEventListener(ProgressEvent.PROGRESS, readSound);
soundStream.load(new URLRequest(WAV_FILE_URL));
var sound = new Sound();
sound.addEventListener(SampleDataEvent.SAMPLE_DATA,playSound);
sound.play();
function readSound(event:ProgressEvent):void {
soundStream.readBytes(soundBuffer, 0, soundStream.bytesAvailable);
}
function playSound(event:SampleDataEvent):void {
/* The docs say that if we send too few samples,
Sound will consider it an EOF */
var samples:int = (soundBuffer.length - soundBuffer.position) / 4
var toadd:int = 4096 - samples;
try {
for (var c: int=0; c < samples; c++) {
var n:Number = soundBuffer.readFloat();
event.data.writeFloat(n);
event.data.writeFloat(n);
}
} catch(e:Error) {
ExternalInterface.call("errorReport", e.message);
}
for (var d: int = 0; d < toadd; d++) {
event.data.writeFloat(0);
event.data.writeFloat(0);
}
}
Like The_asMan pointed out, playing a wav file is not that easy. See as3wavsound for an example.
If your goal is low latency, the best option would be to convert to MP3, so you can use just use a SoundLoaderContext.