Stream wav file in Flex

I am trying to use the 'Sound' object to play wav file. It
seems that it only works with mp3 not wav. I am using Flex 2.0 beta
2. The documentation is not very clear on what formats can be used
with Sound class.

The Flash Player will only play mp3 file. You will need to
convert your wav
into an mp3. Google mp3 converter to find a utility to do
this.
The documentation for flash.media.Sound mentions in several
places that an
mp3 is required.
From the docs:
The Sound class lets you work with sound in an application.
The Sound class
lets you create a new Sound object, load and play an external
MP3 file into
that object, close the sound stream, and access data about
the sound, such
as information about the number of bytes in the stream and
ID3 metadata.
Can you tell us where you looked in the documentation?
Perhaps we are not
mentioning the mp3 requirement in a more obvious place. If
so, we can update
the documentation.
Thanks,
Jason Szeto
"liaod" <[email protected]> wrote in message
news:e20q73$jqv$[email protected]..
>I am trying to use the 'Sound' object to play wav file.
It seems that it
>only
> works with mp3 not wav. I am using Flex 2.0 beta 2. The
documentation is
> not
> very clear on what formats can be used with Sound class.
>

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Thanks for the input however I had already tried a variety of combinations. But I did go through it again.
The Flash Player supports 8-bit or 16-bit audio at sample rates of 5.512, 11.025, 22.05, or 44.1 kHz. I have tried all permutations of wav from 8 bit to 16bit and the all sampling rates from 11.025. The original file is attached at it was at 32000hz. Nothing I can do will make these play in flash once edited. I am using soundbooth howeve I have used audacity and wavepad. I have removed the metadata tags as that was suggested in one of the forums. I have attached the original files and the shortened file that I cannot get to work. I went into Soundbooth and went through all the preferences and checked the box where it asked for FLash compaticble at 16bit 44 hz etc. This is really frustrating. I know wav can be streamed.

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Jon, I read. But the links to websites are not playing on a Windows7 computer. Can you tell me why they are not?
I believe that with a new dawn comes new understanding of the problem.
I do not have access to a Windows 7 computer nor do I run Windows on my Mac.
I am not a web coding programmer. However...
Neither of the original poster's links (http://openmap.bbn.com/~mthome/quicktimeaudio.html and http://openmap.bbn.com/~mthome/wavtest.wav) work because they both point to the same HTML (quicktimeaudio.html) file.
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When I click on either link, this is what I see:
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return duration;
public PullBufferStream[] getStreams()
if (streams == null)
streams = new MyPullBufferStream[1];
stream = streams[0] = new MyPullBufferStream(data);
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import java.io.ByteArrayInputStream;
import java.io.IOException;
import javax.media.Buffer;
import javax.media.Control;
import javax.media.Format;
import javax.media.format.AudioFormat;
import javax.media.protocol.ContentDescriptor;
import javax.media.protocol.PullBufferStream;
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private static final int BLOCK_SIZE = 500;
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Format.byteArray);
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protected Control[] controls = new Control[0];
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bais = new ByteArrayInputStream(data);
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return audioFormat;
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Object outdata = buffer.getData();
if (outdata == null || !(outdata.getClass() == Format.byteArray)
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buffer.setData(outdata);
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buffer.setLength(BLOCK_SIZE);
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try
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Here's some additional information. After making the following changes to MyPullBufferStream class, I can play a wav file with gsm-ms encoding with one issue: the wav file is played many times faster.
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public void read(Buffer buffer) throws IOException
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How to use jmf convert the rtp packet (captured by jpcap) in to wav file?

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pedrorp wrote:
Hi Captfoss!
I fixed it but now I have another problem. My application send me this message:
Cannot initialize audio renderer with format: LINEAR, Unknown Sample Rate, 16-bit, Mono, LittleEndian, Signed
Unable to handle format: ALAW/rtp, Unknown Sample Rate, 8-bit, Mono, FrameSize=8 bits
Failed to prefetch: com.sun.media.PlaybackEngine@1b45ddc
Error: Unable to prefetch com.sun.media.PlaybackEngine@1b45ddc
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java.lang.IllegalArgumentException:
stream is null
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Thanks,
Paulo

Hi
If your stream is null, then the wav file cannot be reached. Try this code:
stream = getClass().getResourceAsStream("/newalert.wav");If this doesn't work either, make sure your wav file is directly in the res directory and not in another subdirectory of it. I suggets not running your application directly from the phone until it works from the WTK emulator.
Mihai

Corrupt wav files in Adobe but play fine in other apps

I have some old projects that have been backed up several times. The problem is some of the sound files appear corrupt when I load them into Adobe Audition. The sound plays all distorted with VERY LOUD clicks/pops in it. However, I can play the same files just fine in any standard media player (Winamp, WMP, VLC player)
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http://home.comcast.net/~chicken3/Track17.zip
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Format                           : Wave
File size                        : 36.8 MiB
Duration                         : 1mn 49s
Overall bit rate                 : 2 822 Kbps
Audio
ID                               : 0
Format                           : PCM
Codec ID                         : 1
Codec ID/Hint                    : Microsoft
Duration                         : 1mn 49s
Bit rate                         : 2 822 Kbps
Channel(s)                       : 2 channels
Sampling rate                    : 44.1 KHz
Resolution                       : 32 bits
Stream size                      : 36.8 MiB (100%)
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I've found the source files on each attendants PC (C:\Program Files\Cisco\Call Manager Attendant Console\audio), copied in a replacement & renamed it to Cisco's original file name.
However, on startup AC is copying the original files back from CCM.
If I changed the files on CCM then this would solve my problem but I can't find them.
How can I change the audio file source ?
Any help appreciated,
Paul.

The current capabilities and features of MoH include:
MoH multicast and unicast streaming service
Music streaming service for "user" hold and "network" hold
51 sources per media convergence server (MCS)
Fifty continuously looping .wav file sources
One real-time streaming source
Each source configurable as either unicast or multi-cast stream
Support for audio streaming to selected devices
Gateways (multicast only):
DT-24+No
6608Not until Seaview 3.3
VG200 (H.323)Yes with Cisco IOS. Software Release 12.2(11)T and later, enable the ccm-manager music-on-hold command
VG200 (MGCP)Yes, same as VG200 (H.323)
VG248Yes
Gateways (unicast only)AT-2/-4/-8, AS-2/-4/-8, and all other Cisco IOS. VoIP gateways, including Cisco 1750, 2600/3600, 5300, 58xx, and 72xx
Cisco IP phones (unicast, multicast)7910, 7940, and 7960
Cisco IP phones (unicast only)Cisco 7935, 12 SP+, and 30 VIP phones
Cisco IP SoftPhone (unicast only)
Maximum 250 simultaneous on-hold streaming sessions per server
Multiple server instances for application scalability
Multiple server instances for server load balancing and redundancy
G.711, G.729A, and wide-band audio codec support
Off-line audio translation utility
This URL should help you:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a0080094766.shtml

Stream wav file in Flex

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