16 bit WAV file

Hi everyone. Im trying to understand something about WAV files (or rather about how 16 bit audio works).
a 16 bit sample has values from -32768 to 32767.
If 0 is the zero crossing then what do you do with this asymmetry?
There are 32767 "pos" values but 32768 "neg" values.
Wouldnt it have been better for them to throw out that last 32768th value?
Sorry if i have this all wrong. Im just imaging a waveform and wondering why the values above and below the zero crossing arent equal.

no worries Jon I was dragging the destination into the window when there was no need cheers

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//set of import statements in the file.
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.TargetDataLine;
import javax.sound.sampled.Mixer.Info;
private AudioFormat format;
*_The Java Program console output is..._*9-17-2009 10:54 - Listener initiated. Listening Now...
--------------AFTER I CALL-------------
9-17-2009 10:55 - Incoming Call...
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.30:5060;rport=5060;branch=z9hG4bK-d8754z-f24ed87f010f565d-1---d8754z-;received=10.0.2.30
To: "Java Program"<sip:[email protected]:5060>;tag=402319264639678169
From: "Kris" <sip:[email protected]:5060>;tag=ff7a573a
Call-ID: MjYxYjNhOTdkYzcxY2U2Y2EzYjllZmYxYTg2ZDkzOTg.
CSeq: 1 INVITE
Content-Length: 0
Call Event: 1: CALL_INCOMING_INVITE, CALL_NORMAL
Open log file: F:\logs\jmf.log
9-17-2009 10:55 - Created RTP session: 10.0.2.58 18216
Call Event: 1: CALL_CONNECTED, CALL_NORMAL
9-17-2009 10:55 - Media End Of File Event Received
9-17-2009 10:55 - Created RTP session: 10.0.2.58 18216
*_JMF LOG OUTPUT - START_*#
# JMF Version 2.1.1e
## Platform: Windows XP, x86, 5.1
## Java VM: Sun Microsystems Inc., 1.6.0_07
## DataSource created: com.sun.media.protocol.file.DataSource@1b48197
$$ Profile: instantiation: 16 ms
## Processor created: com.sun.media.processor.unknown.Handler@1385660
## using DataSource: com.sun.media.protocol.file.DataSource@1b48197
$$ Profile: parsing: 15 ms
## Building flow graph for: file:///F:\audio_message.wav
## Building Track: 0
## Input: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed, 16000.0 frame rate, FrameSize=16 bits
## Custom options specified.
## A custom codec is specified: com.sun.media.codec.audio.ulaw.Packetizer@203c31
## An output format is specified: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
## An output content type is specified: RAW/RTP
## Here's the completed flow graph:
com.sun.media.parser.audio.WavParser@16925b0
connects to: com.ibm.media.codec.audio.ulaw.JavaEncoder@297ffb
format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed, 16000.0 frame rate, FrameSize=16 bits
com.ibm.media.codec.audio.ulaw.JavaEncoder@297ffb
connects to: com.sun.media.codec.audio.ulaw.Packetizer@203c31
format: ULAW, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
com.sun.media.codec.audio.ulaw.Packetizer@203c31
connects to: com.sun.media.multiplexer.RTPSyncBufferMux@914f6a
format: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
$$ Profile: graph building: 188 ms
$$ Profile: realize, post graph building: 93 ms
$$ Profile: prefetch: 47 ms
## Computed latency for video: 20 ms
## DataSource created: com.sun.media.protocol.file.DataSource@1402d5a
$$ Profile: instantiation: 0 ms
## Processor created: com.sun.media.processor.unknown.Handler@9cfec1
## using DataSource: com.sun.media.protocol.file.DataSource@1402d5a
$$ Profile: parsing: 0 ms
## Building flow graph for: file:///F:\audio_message.wav
## Building Track: 0
## Input: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed, 16000.0 frame rate, FrameSize=16 bits
## Custom options specified.
## A custom codec is specified: com.sun.media.codec.audio.ulaw.Packetizer@edf389
## An output format is specified: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
## An output content type is specified: RAW/RTP
## Here's the completed flow graph:
com.sun.media.parser.audio.WavParser@21e554
connects to: com.ibm.media.codec.audio.ulaw.JavaEncoder@15718f2
format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed, 16000.0 frame rate, FrameSize=16 bits
com.ibm.media.codec.audio.ulaw.JavaEncoder@15718f2
connects to: com.sun.media.codec.audio.ulaw.Packetizer@edf389
format: ULAW, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
com.sun.media.codec.audio.ulaw.Packetizer@edf389
connects to: com.sun.media.multiplexer.RTPSyncBufferMux@126f827
format: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
$$ Profile: graph building: 47 ms
$$ Profile: realize, post graph building: 0 ms
XX Error: Unable to prefetch com.sun.media.ProcessEngine@af72d8
$$ Profile: prefetch: 15 ms
*_JMF LOG OUTPUT - END_*
*_....contd..... in the next post_*
Edited by: codebuster on Sep 16, 2009 10:59 PM

_....contd of the post above...._
CASE 2:
BUT, when i play which is a big file(20MB suppose), wherein - AudioFormat=PCM, Audio Sample Rate = 44100Hz, Audio Sample Size = 16-bit, Channel = 2(Stereo), BitRate=1411Kbps.
I did the following changes in the parameters in the AudioFormat function -
format = new AudioFormat(44100, 16, 2, true, true);The Java Program console output is...
9-17-2009 10:54 - Listener initiated. Listening Now...
--------------AFTER I CALL-------------
9-17-2009 11:08 - Incoming Call...
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.30:5060;rport=5060;branch=z9hG4bK-d8754z-04215a630a579a34-1---d8754z-;received=10.0.2.30
To: "Java Program"<sip:[email protected]:5060>;tag=402319264639678169
From: "Kris" <sip:[email protected]:5060>;tag=ff7a573a
Call-ID: NGQ2ZjJjZjI1NDE4YTE3MmMxYTViZjJkOGE0ZTM2MzY.
CSeq: 1 INVITE
Content-Length: 0
Open log file: F:\logs\jmf.log
An error has occurred. Check jmf.log for details.
Exception in thread "Thread-20" javax.media.NotRealizedError: getDataOutput cannot be called before realized
at com.sun.media.ProcessEngine.getDataOutput(ProcessEngine.java:379)
at com.sun.media.MediaProcessor.getDataOutput(MediaProcessor.java:138)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.createProcessor(Transmitter.java:392)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.start(Transmitter.java:127)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.<init>(Transmitter.java:118)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.JmfPlugin.initializeRtpSend(JmfPlugin.java:117)
at com.sipresponse.onlinevoipcallmgr.internal.actions.AnswerCallAction.startMediaSend(AnswerCallAction.java:228)
at com.sipresponse.onlinevoipcallmgr.internal.actions.AnswerCallAction.run(AnswerCallAction.java:99)
The process of decoding is not completed i think.
JMF LOG OUTPUT - START*
# JMF Version 2.1.1e
## Platform: Windows XP, x86, 5.1
## Java VM: Sun Microsystems Inc., 1.6.0_07
## DataSource created: com.sun.media.protocol.file.DataSource@105b99f
$$ Profile: instantiation: 15 ms
## Processor created: com.sun.media.processor.unknown.Handler@ac6a45
## using DataSource: com.sun.media.protocol.file.DataSource@105b99f
$$ Profile: parsing: 15 ms
## Getting the supported output formats for:
## LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, 176400.0 frame rate, FrameSize=32 bits
## # of nodes visited: 1016
## # of formats supported: 308
$$ Profile: getSupportedOutputFormats: 1109 ms
## Building flow graph for: file:///F:\audio_message.wav
## Building Track: 0
## Input: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, 176400.0 frame rate, FrameSize=32 bits
## Custom options specified.
## A custom codec is specified: com.sun.media.codec.audio.ulaw.Packetizer@3ef810
## An output format is specified: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
## An output content type is specified: RAW/RTP
XX Failed to build a graph for the given custom options.
XX Failed to realize: com.sun.media.ProcessEngine@12452e8
XX Cannot build a flow graph with the customized options:
XX Unable to transcode format: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, 176400.0 frame rate, FrameSize=32 bits
XX to: ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits
XX outputting to: RAW/RTP
XX Unable to add customed codecs:
XX com.sun.media.codec.audio.ulaw.Packetizer@3ef810
XX Error: Unable to realize com.sun.media.ProcessEngine@12452e8
javax.media.NotRealizedError: getDataOutput cannot be called before realized
at com.sun.media.ProcessEngine.getDataOutput(ProcessEngine.java:379)
at com.sun.media.MediaProcessor.getDataOutput(MediaProcessor.java:138)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.createProcessor(Transmitter.java:392)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.start(Transmitter.java:127)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.Transmitter.<init>(Transmitter.java:118)
at com.sipresponse.onlinevoipcallmgr.plugin.jmf.JmfPlugin.initializeRtpSend(JmfPlugin.java:117)
at com.sipresponse.onlinevoipcallmgr.internal.actions.AnswerCallAction.startMediaSend(AnswerCallAction.java:228)
at com.sipresponse.onlinevoipcallmgr.internal.actions.AnswerCallAction.run(AnswerCallAction.java:99)
JMF LOG OUTPUT - END*
Am I making any mistake in the set of parameters I have provided? Can you suggest me something to get this working?How do i give the support to the program to read huge .wav files? Is there anything i need to focus on?
Thanks in advance
Edited by: codebuster on Sep 16, 2009 11:04 PM

Converting wav files to a different bit rate

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I can't get my iPod to accept the 9216 kps song files. I went to set iTunes / Preferences / Advanced / Importing to WAV but the options for sample rate only go up to 48.0 kps and the bit rate up to 16-bit.
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Hi,
I'm from germany so no 'best buy' but I think I will lock around sleeve/bag for a different product with same measures. Any tip?
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Hi all, I've searched everywhere and can't find an answer, so hopefully someone out there will have mercy and help me!
I have Canon 7D footage (h.264 codec) and H4N Zoom audio (24bit 48khz .wav files) loaded into Premiere Pro CS 5. Editing on a Macbook Pro OS 10.6.8. I haven't added timecode.
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Yes, it has always worked in the past. The past projects audio clips do not change names. I obviously did something different in this current project, but unfortunately I have no idea what.

16 bit WAV file

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