sevengo8378 / as3wavsound Goto Github PK

Hi, first tank you for your help and your great work to provide a class that 
hellp lot of people.

What steps will reproduce the problem?

All work great, but when i use
SoundMixer.stopAll();
after that, i cant play the sounds again.

What version of the product are you using? On what operating system?
v0.9, windows 7 64 bit, flash CS6,

The attached WAV file (the Devanagari letter "ka" with a horrible goraa accent) 
plays fine in VLC Media Player. With AWS, it makes me sound like a 
chipmunk---if chipmunks spoke Hindi.

Here's the code I'm using in ActionScript3 in an SWF on IE9 on Windows 7 
Professional 64-bit:

    var urlRequest:URLRequest = new URLRequest(soundURI);
    var wav:URLLoader = new URLLoader();
    wav.dataFormat = "binary";
    wav.addEventListener(Event.COMPLETE, playWAV);
    wav.load(urlRequest);  

...

private final function playWAV(e:Event):void
{  
    var tts:WavSound = new WavSound(e.target.data as ByteArray);  
    tts.play();
}   

What steps will reproduce the problem?
1. Load a wav file from attachment into a demo example.
File was recorded with WAVWriter library:
http://code.google.com/p/speechapi/source/browse/trunk/src/flash/com/adobe/audio
/format/WAVWriter.as?r=15
2. Try to play it (on my machine file is playing correct).
3. I also tried (it didn't helped) to use specific AudioSettings:
new WavSound(new SoundClass() as ByteArray, new AudioSetting(1, 44100, 16);


What is the expected output? What do you see instead?
File is not playing. Error:
Error: Error #2030: End of file was encountered.
    at flash.utils::ByteArray/readBytes()
    at org.as3wavsound.sazameki.format.riff::LIST/splitList()[D:\Projects\Adobe Flash Builder 4.6\Kanarek\src\org\as3wavsound\sazameki\format\riff\LIST.as:65]
    at org.as3wavsound.sazameki.format.wav::Wav/decode()[D:\Projects\Adobe Flash Builder 4.6\Kanarek\src\org\as3wavsound\sazameki\format\wav\Wav.as:38]
    at org.as3wavsound::WavSound/load()[D:\Projects\Adobe Flash Builder 4.6\Kanarek\src\org\as3wavsound\WavSound.as:115]
    at org.as3wavsound::WavSound()[D:\Projects\Adobe Flash Builder 4.6\Kanarek\src\org\as3wavsound\WavSound.as:101]


What version of the product are you using? On what operating system?
version:0.9
operating system: Android 4.0.4 with Adobe AIR/flex

1. I'm trying to synchronize two sounds - an MP3 song and a recorded microphone 
output. But when I play the recorded microphone output  through 
WavSoundChannel, it starts with a short delay of about a second(I assume this 
delay is different on different computers).

I tried to use the position of the soundchannel to catch the start of the real 
sound through the speakers, thinking that if the position is greater than 0, it 
should mean that the sound has started playing. But the position is greater 
than 0 and I don't hear the sound yet. I think it returns the right position 
but there is some kind of latency due to encoding. So I cannot use the position 
method to catch the start of the real sound through the speakers.

Is there a listener that I can use to get the actual sound start of the output, 
so i can play both sounds(the recorded output and the MP3) synchronized?


I'm using version 0.8 on Windows 7

Thanks

Kiril

What steps will reproduce the problem?
1. use the code I have provided
2. run SoundManager.playSound(<name>);
3. there is a 1 second delay for me, i have traced to confirm this. The trace 
in playSound traces 1 second before i hear the sound. The sound file is 
confirmed to have no blank space in the spectrum. and it is only 6kb.

<code>: SoundManager.as
public function SoundManager() 
        {
            buildSounds();
            trace("sound manager loaded");
        }

        public function playSound(name:String)
        {
            trace("playing " + name);
            sounds[name][1] = sounds[name][0].play();
            trace("played " + name);
            sounds[name][1].addEventListener( Event.SOUND_COMPLETE, looper );
        }

private function loadSound(path:String, name:String)
        {
            sounds[name] = new Array("", "", path, false, false, 0, 100, 0, 0);
            var wavLoader:URLLoader = new URLLoader();
            wavLoader.dataFormat = "binary";
            wavLoader.load(new URLRequest(File.applicationDirectory.resolvePath(path).url));
            wavLoader.addEventListener( Event.COMPLETE, function(e:Event){makeSound(e,name)} );
        }

        private function makeSound(e:Event, name:String):void
        {
            sounds[name][0] = new WavSound(e.target.data as ByteArray);
        }

        private function buildSounds()
        {
            loadSound("sound/menu/itemDown.wav", "menuItemDown")
            loadSound("sound/menu/itemUp.wav","menuItemUp")
        }
</code>

What is the expected output? What do you see instead?
Instant play, but instead i have a 1 second delayed play.

What version of the product are you using? On what operating system?
v0.9, windows 7 64 bit

Please provide any additional information below.
run in adobe air 3.2 in flash professional cs6

1. play a WavSound and keep the returned SoundChannel in a variable
2. using a timer, try to access the SoundChannel.postion during playback
3. the resulting value is always 0. 

Exected output is the current position in ms that the sound has been playing, 
i.e. as with the normal Sound class. I am using the as3wavsound 0.6 version on 
MacOS 10.5.8. with flash CS5.

I added a trace of the current duration (in ms) in the wavSoundChannel class to 
see if I could work around the problem myself:

trace(phase / (_wavSound.samples.length / _wavSound.length));

But, for some reason the actual sound hitting the speakers is lagging with 
about 1 second when I compare it with this trace. I don't mind a delay, but 
since I can't calculate it I can't sync the sound with other sounds. I would 
really appreciate any help on this.

line 56 in WavSoundChannel looks like:

var maxPositionInMillis:Number = Math.floor(length); 

but should probably refer to the _wavSound.length like:

var maxPositionInMillis:Number = Math.floor(_wavSound.length); 

Can you add documentation on how to create a .wav file from byte array data?  
Thanks.

In my case i have 10000 wav (G.711) files for play depending upon contentID. 
can you please let me know how i can accomplish this task. 

After adding AS3WavSoundDemo.swf  expressInstall.swf  index.html files at my 
tomcat server i am able to listen rain_loop file. Can you please tell me how i 
can pass diffrent media file names for play to AS3WavSoundDemo.swf


What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?


Please provide any additional information below.

What steps will reproduce the problem?
1. Added function to measure volume: 
(100 * ((channel.leftPeak + channel.rightPeak) / 2.0))

2. Value was always very low (or 0), so I decided to measure each value in 
function buffer.

Recorded sample was soft, then loud, then soft. The number seem all over the 
place.

Sample output attached.

What steps will reproduce the problem?
1. add an soundComplete event listener to the channel returned by the play 
method of the class WavSound
2. run the code

What is the expected output? What do you see instead?
should call the callback when the sound finished playing.
but callback is called when sound starts.


What version of the product are you using? On what operating system?
v0.4

Please provide any additional information below.

The current version of as3wavsound doesn't support playback 16 kHz audio, which 
is really required in many cases, especially for speech recognition.

Even we manually remove the limitation for 16kHz in the audio setting, the 
quality of the current upsample impl in Wav.as is not acceptable (the noise).

So in our case, we implemented an upsample replacement:
https://github.com/englishtown/as3wavsound/blob/feature/upsample/src/org/as3wavs
ound/sazameki/format/wav/Wav.as#L81

Out impl is copied from the following stackoverflow ticket:
http://stackoverflow.com/questions/7995470/how-can-i-resample-wav-file

I suggest you guys could consider use this algorithm to replace your current 
one.

var drumLoop:WavSound = new WavSound(new DrumLoop() as ByteArray);
sound.play(); // must be drumLoop.play()

For large .wav files, there can be a long delay reading them into a bytearray 
before beginning playback.  Similar to flash.media.Sound, WavSound should 
support loading a wav as a stream and not just a ByteArray.

What steps will reproduce the problem?
1. Load 2 wave audio files and play them. The files should have around 5-10 
seconds, to have time to notice the problem.


What is the expected output? What do you see instead?
I taught that the files will be played one after another, but unfortunately 
they are played asynchronous, and you can hear both of them at the same time.


What version of the product are you using? On what operating system?
0.8

Is it possible somehow to synchronize the files to be played one after another 
??

What steps will reproduce the problem?
1. Record wav with sampling rate of 11025.
2. Load wav file into WavSound object:
var mySound:WavSound = new WavSound(fileContent, new AudioSetting(1, 11025));
3. Play mySound file: mySound.play();

What is the expected output? What do you see instead?
Expected the audio file to playback properly, but instead it was very 
fast-paced like it was not using the proper sampling rate.

What version of the product are you using? On what operating system?


Please provide any additional information below.

What steps will reproduce the problem?
I have a ByteArray of sound data captured from the microphone called 
"soundBytes".
var snd:Sound =  new WavSound(soundBytes);
var mySoundChannel = new SoundChannel();
var mySoundTransform = new SoundTransform;
var playbackTimer:Timer;

private function doPlay():void
{
    mySoundTransform.volume = 1;
    mySoundChannel = snd.play(0,0,mySoundTransform);
    playbackTimer = new Timer(100);
    playbackTimer.addEventListener(TimerEvent.TIMER, playbacklevel);
    playbackTimer.start();
}

private function playbacklevel(event:TimerEvent):void 
{
    myProgressBar.width = (mySoundChannel.leftPeak + mySoundChannel.rightPeak)/2 * 100;
}



One would expect the rightPeak and leftPeak on the SoundChannel object to have 
some value other than zero as long there is sound playing, and it does if I'm 
using the standard Sound object, rather than the WavSound object.

This is not the case here...  the rightPeak and leftPeak values are always zero.

To reproduce:

1. Download or create a WAV file with 8-bit samples (Ex: 
http://simplythebest.net/sounds/WAV/sound_effects_WAV/sound_effect_WAV_files/bub
bles_2.wav)
2. Create a project either with the sound file embedded or downloaded and play 
the sound using the WavSound class.

Behavior:

Instead of the sound being reproduced accurately, a garbled sound of the same 
duration is played.

Constructing a WavSound instance with odd-length WAV data throws an exception.

Here is sample code that will reproduce the problem:

var wavLoader:URLLoader = new URLLoader();
wavLoader.dataFormat = URLLoaderDataFormat.BINARY;
wavLoader.load(new URLRequest(
    "http://media.merriam-webster.com/soundc11/b/bootes01.wav"));
wavLoader.addEventListener(Event.COMPLETE, function(evt:*):void {
    var sound:WavSound = new WavSound(wavLoader.data as ByteArray);
});

Expected: the constructor returns and the WavSound object is playable.

Instead: an exception is thrown:

    Error: Error #2030: End of file was encountered.

All G.711 files are encoded with a background noise. 
I am using the latest trunk version of library. Example of G.711 wav file is 
attached. 

Is it planned to support sample rate 8000?