Audio tracks play back wrong audio
at some point in my mix sessions , in different sessions , one of my audio tracks starts playing back audio from both its channel and another channel simultaineously. there not always adjacent tracks , not assigned to the same voice and have seemingly nothing in common. if i change voices i can sometimes allieviate the problem or it will start playing a different track audio simultaineously. i have two records pending and need some immediate help please...
D,
Just to warn you (or throw some experience in). Logic 7 has for me, operated poorly with sessions created from older versions of Logic. It is often fine if you just want to duck in and grab something but the sessions do go weird if you just keep using them they go batty and can often get kind of self destructive.
Updating/merging/replacing the environment is one of the first steps to take, just creating new updated objects helps, even using presets and channel strips from older versions seem to give problems. Copying objects and configurations over can give you different results, but basically the more you replace and create as new the less problems you'll have.
So you know you have alot of reworking to do if you really want to do alot of work with these in the current version. Determine where a good transition point would be and try to follow with that. Often mixing or sooner than later but plan on taking some time reworking things. It's almost easiest to just grab everything and paste it over into a 7 built autoload-without the plugins. Audio Objects and plugin instances seem to give alot of problems. I usually end up redoing a bunch of stuff if I have a ways to go and export away. Things really get fixed along the way but it is a pain and better to do it sooner. Unless you're coming up on a mixing stage or something soon I'd just do it (and keep a backup of the older versions just in case I want to retrieve something.
I am working on an album that was created on a 6 autoload and 12 of 13 tracks had to be reformatted in 7 from forward INcompatabilities-some of them twice because they did not get a 100% clean autoload (as in the instruments and/or audio objects were opied over). Maybe I'm just weird, don't take only my word on this because it is alot of work for many projects to just redo things like this but you should be aware of what some people have encountered and be prepared to take some of those steps if there are more people saying their experience has been like this.
Best of Luck
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Hi there. Posted on here before...I'm having a problem with my JMF program. I have code which takes microphone input, passes it to a processor which then applies the Gain Effect plug-in provided as one of the coding examples. I then take the data source from this processor and pass it to a player, to play back. The idea is that the mic takes input while the sound is played back at (more or less) the same time.
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Below is my code. Does anyone know what I'm doing wrong?
//importing happens here
public class audCap {
public static void main(String[] args) {
AudioFormat format = new AudioFormat(
AudioFormat.LINEAR,
44100,
16,
2,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray);
Format[] alinear=new AudioFormat[]{new AudioFormat(
AudioFormat.LINEAR,
44100,
16,
2,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray)};
Vector devices= CaptureDeviceManager.getDeviceList(format);
CaptureDeviceInfo di = null;
if (devices.size() > 0) {
di = (CaptureDeviceInfo) devices.elementAt( 0); //the mic
else {
System.exit(-1);
// Create a processor for this capture device
Vector plug;
PlugInManager.addPlugIn("GainEffect", alinear, alinear, 3);
plug = PlugInManager.getPlugInList(null, null, 3);
int vectorSize = plug.size();
if(plug.elementAt(vectorSize - 1).equals("GainEffect")){
plug.removeElementAt(vectorSize - 1);
plug.insertElementAt("GainEffect", 0);
PlugInManager.setPlugInList(plug, 3);
try {
PlugInManager.commit();
} catch (IOException e) {
e.printStackTrace();
Processor processor = null;
try {
processor = Manager.createProcessor(di.getLocator());
} catch (IOException e1) {
System.exit(-1);
} catch (NoProcessorException e) {
System.exit(-1);
// configure the processor
processor.configure();
while (processor.getState() != Processor.Configured){
try {
Thread.sleep(100);
} catch (InterruptedException e) {
e.printStackTrace();
TrackControl track[] = processor.getTrackControls();
boolean encodingOk = false;
for (int i = 0; i < track.length; i++) {
if (!encodingOk && track[i] instanceof FormatControl) {
if (((FormatControl)track).
setFormat( new AudioFormat(AudioFormat.LINEAR,
44100,
16,
2,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray)) == null) {
track[i].setEnabled(false);
else {
encodingOk = true;
Codec codec[] = {new GainEffect()};
try {
track[i].setCodecChain(codec);
} catch (UnsupportedPlugInException e) {
// TODO Auto-generated catch block
e.printStackTrace();
} catch (NotConfiguredError e) {
// TODO Auto-generated catch block
e.printStackTrace();
} else {
track[i].setEnabled(false);
// realize the processor
if (encodingOk) {
processor.realize();
while (processor.getState() != Processor.Realized){
try {
Thread.sleep(100);
} catch (InterruptedException e) {
// TODO Auto-generated catch block
e.printStackTrace();
// get the output datasource of the processor and exit
// if fails
DataSource ds = null;
try {
ds = processor.getDataOutput();
} catch (NotRealizedError e) {
System.exit(-1);
processor.setContentDescriptor(null);
processor.start();
Player player = null;
try {
player = Manager.createPlayer(ds);
} catch (NoPlayerException e) {
System.err.println("Error:" + e);
System.exit(-1);
} catch (MalformedURLException e) {
System.err.println("Error:" + e);
System.exit(-1);
} catch (IOException e) {
System.err.println("Error:" + e);
System.exit(-1);
if (player != null) {
System.out.println("Player created.");
player.realize();
// wait for realizing
while (player.getState() != Player.Realized){
try {
Thread.sleep(10);
} catch (InterruptedException e) {
e.printStackTrace();
player.start();
} else {
System.err.println("Player not created.");
System.exit(-1);
The GainEffect.java class is below
import javax.media.*;
import javax.media.format.*;
//import javax.media.format.audio.*;
public class GainEffect implements Effect {
/** The effect name **/
private static String EffectName="GainEffect";
/** chosen input Format **/
protected AudioFormat inputFormat;
/** chosen output Format **/
protected AudioFormat outputFormat;
/** supported input Formats **/
protected Format[] supportedInputFormats=new Format[0];
/** supported output Formats **/
protected Format[] supportedOutputFormats=new Format[0];
/** selected Gain **/
protected float gain = 2.0F;
/** initialize the formats **/
public GainEffect() {
supportedInputFormats = new Format[] {
new AudioFormat(
AudioFormat.LINEAR,
44100,
16,
2,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray)
supportedOutputFormats = new Format[] {
new AudioFormat(
AudioFormat.LINEAR,
44100,
16,
2,
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray)
/** get the resources needed by this effect **/
public void open() throws ResourceUnavailableException {
/** free the resources allocated by this codec **/
public void close() {
/** reset the codec **/
public void reset() {
/** no controls for this simple effect **/
public Object[] getControls() {
return (Object[]) new Control[0];
* Return the control based on a control type for the effect.
public Object getControl(String controlType) {
try {
Class cls = Class.forName(controlType);
Object cs[] = getControls();
for (int i = 0; i < cs.length; i++) {
if (cls.isInstance(cs)){
return cs[i];
return null;
} catch (Exception e) { // no such controlType or such control
return null;
/************** format methods *************/
/** set the input format **/
public Format setInputFormat(Format input) {
// the following code assumes valid Format
inputFormat = (AudioFormat)input;
return (Format)inputFormat;
/** set the output format **/
public Format setOutputFormat(Format output) {
// the following code assumes valid Format
outputFormat = (AudioFormat)output;
return (Format)outputFormat;
/** get the input format **/
protected Format getInputFormat() {
return inputFormat;
/** get the output format **/
protected Format getOutputFormat() {
return outputFormat;
/** supported input formats **/
public Format [] getSupportedInputFormats() {
return supportedInputFormats;
/** output Formats for the selected input format **/
public Format [] getSupportedOutputFormats(Format in) {
if (! (in instanceof AudioFormat) ){
return new Format[0];
AudioFormat iaf=(AudioFormat) in;
if (!iaf.matches(supportedInputFormats[0])){
return new Format[0];
AudioFormat oaf= new AudioFormat(
AudioFormat.LINEAR,
iaf.getSampleRate(),
16,
iaf.getChannels(),
AudioFormat.LITTLE_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray
return new Format[] {oaf};
/** gain accessor method **/
public void setGain(float newGain){
gain=newGain;
/** return effect name **/
public String getName() {
return EffectName;
/** do the processing **/
public int process(Buffer inputBuffer, Buffer outputBuffer){
// == prolog
byte[] inData = (byte[])inputBuffer.getData();
int inLength = inputBuffer.getLength();
int inOffset = inputBuffer.getOffset();
byte[] outData = validateByteArraySize(outputBuffer, inLength);
int outOffset = outputBuffer.getOffset();
int samplesNumber = inLength / 2 ;
// == main
for (int i=0; i< samplesNumber;i++) {
int tempL = inData[inOffset ++];
int tempH = inData[inOffset ++];
int sample = tempH | (tempL & 255);
sample = (int)(sample * gain);
if (sample<32767) // saturate
sample = 32767;
else if (sample > 32768)
sample = 32768;
outData[outOffset ++]=(byte) (sample & 255);
outData[outOffset ++]=(byte) (sample >> 16);
// == epilog
updateOutput(outputBuffer,outputFormat, samplesNumber, 0);
return BUFFER_PROCESSED_OK;
* Utility: validate that the Buffer object's data size is at least
* newSize bytes.
* @return array with sufficient capacity
protected byte[] validateByteArraySize(Buffer buffer,int newSize) {
Object objectArray=buffer.getData();
byte[] typedArray;
if (objectArray instanceof byte[]) { // is correct type AND not null
typedArray=(byte[])objectArray;
if (typedArray.length >= newSize ) { // is sufficient capacity
return typedArray;
System.out.println(getClass().getName()+
" : allocating byte["+newSize+"] ");
typedArray = new byte[newSize];
buffer.setData(typedArray);
return typedArray;
/** utility: update the output buffer fields **/
protected void updateOutput(Buffer outputBuffer,
Format format,int length, int offset) {
outputBuffer.setFormat(format);
outputBuffer.setLength(length);
outputBuffer.setOffset(offset);
Thank-you.Thanks for that last answer. Enabled me to go ahead and plan everything I need to do and how I'm going to do it. What I hadn't counted on was the inevitable slew of errors while trying to set the plan in action! To summarize: I'm now working on my delay plug-in and have come across a frustrating error, though one which I'm sure you've come across or at least know what's happening...
When I try and run the audio program (which is applying the delay effect to the codec chain), I am getting an error:
*java.lang.ClassCastException: [S cannot be cast to [B*
at com.sun.media.renderer.audio.AudioRenderer.doProcessData(AudioRenderer.java:169)
at com.sun.media.renderer.audio.DirectAudioRenderer.processData(DirectAudioRenderer.java:150)
at com.sun.media.renderer.audio.AudioRenderer.process(AudioRenderer.java:130)
at com.sun.media.BasicRendererModule.processBuffer(BasicRendererModule.java:727)
at com.sun.media.BasicRendererModule.scheduleBuffer(BasicRendererModule.java:499)
at com.sun.media.BasicRendererModule.doProcess(BasicRendererModule.java:400)
at com.sun.media.RenderThread.process(BasicRendererModule.java:1114)
at com.sun.media.util.LoopThread.run(LoopThread.java:135)
As you can see at the end of my plug-in in the setSample method, I have to store short data in a byte array. I am going about this in the correct way...according to Google. Yet, when I run the program the test Sys.out's I've put in the last two methods print to the console for a significant time before the error is received, which suggests that it has nothing to do with the way I'm converting the short data to byte[] data...right?
import javax.media.Buffer;
import javax.media.Control;
import javax.media.Effect;
import javax.media.Format;
import javax.media.ResourceUnavailableException;
import javax.media.format.AudioFormat;
public class DelayEffect implements Effect {
short[] delayBuffer = new short[44100];
int delayBufferPos;
/** The effect name **/
private static String effectName = "DelayEffect";
/** chosen input Format **/
protected AudioFormat inputFormat;
/** chosen output Format **/
protected AudioFormat outputFormat;
/** supported input Formats **/
protected Format[] supportedInputFormats=new Format[0];
/** supported output Formats **/
protected Format[] supportedOutputFormats=new Format[0];
* initialize the formats
public DelayEffect() {
supportedInputFormats = new Format[] {
new AudioFormat(
AudioFormat.LINEAR,
Format.NOT_SPECIFIED,
16,
Format.NOT_SPECIFIED,
AudioFormat.BIG_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray
supportedOutputFormats = new Format[] {
new AudioFormat(
AudioFormat.LINEAR,
Format.NOT_SPECIFIED,
16,
Format.NOT_SPECIFIED,
AudioFormat.BIG_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray
* get the resources needed by this effect
public void open() throws ResourceUnavailableException {
* free the resources allocated by this codec
public void close() {
* reset the codec
public void reset() {
* no controls for this simple effect
public Object[] getControls() {
return (Object[]) new Control[0];
* Return the control based on a control type for the effect.
*@param controlType The type of control.
public Object getControl(String controlType) {
try {
Class cls = Class.forName(controlType);
Object cs[] = getControls();
for (int i = 0; i < cs.length; i++) {
if (cls.isInstance(cs))
return cs[i];
return null;
} catch (ClassNotFoundException e) { // no such controlType or such control
return null;
/************** format methods *************/
* Set the input format
* @param input The input format.
public Format setInputFormat(Format input) {
// the following code assumes valid Format
inputFormat = (AudioFormat)input;
return (Format)inputFormat;
* Set the output format
* @param output The output format
public Format setOutputFormat(Format output) {
// the following code assumes valid Format
outputFormat = (AudioFormat)output;
return (Format)outputFormat;
* Get the input format
* @return Returns the input format.
protected Format getInputFormat() {
return inputFormat;
* Get the output format
* @return Returns the output format.
protected Format getOutputFormat() {
return outputFormat;
* Supported input formats
* @return Returns the supported input formats.
public Format [] getSupportedInputFormats() {
return supportedInputFormats;
* Output Formats for the selected input format
* @param in The requested input format.
* @return Returns the supported output formats.
public Format [] getSupportedOutputFormats(Format in) {
if (! (in instanceof AudioFormat) )
return new Format[0];
AudioFormat iaf=(AudioFormat) in;
if (!iaf.matches(supportedInputFormats[0]))
return new Format[0];
AudioFormat oaf= new AudioFormat(
AudioFormat.LINEAR,
iaf.getSampleRate(),
16,
iaf.getChannels(),
AudioFormat.BIG_ENDIAN,
AudioFormat.SIGNED,
16,
Format.NOT_SPECIFIED,
Format.byteArray
return new Format[] {oaf};
* return effect name
public String getName() {
return effectName;
public void clearDelayBuffer(){
for (int i = 0; i < delayBuffer.length; i++){
delayBuffer[i] = 0;
delayBufferPos = 0;
* Do the processing
* @param inputBuffer The incoming buffer.
* @param outputBuffer The processed buffer.
* @return A status code..
public int process(Buffer inputBuffer, Buffer outputBuffer){
// == prolog
byte[] inData = (byte[])inputBuffer.getData();
int inLength = inputBuffer.getLength();
int inOffset = inputBuffer.getOffset();
byte[] outData = validateByteArraySize(outputBuffer, inLength);
int outOffset = outputBuffer.getOffset();
int j = outOffset;
int outLength = inLength;
double decay = 0.5;
int samplesNumber = inLength ; //44100 samples
*// == main*
short sample;
for (int i= inOffset; i< inOffset + inLength;i+=2) {
//update sample
short oldSample = getSamples(inputBuffer, i);
short newSample = (short)(oldSample + decay * delayBuffer[delayBufferPos]);
setSample(inputBuffer, i, newSample);
//update delay buffer
delayBuffer[delayBufferPos] = newSample;
outputBuffer.setData(delayBuffer);
delayBufferPos++;
if(delayBufferPos == delayBuffer.length){
delayBufferPos = 0;
// == epilog
updateOutput(outputBuffer,outputFormat, outLength, outOffset);
return BUFFER_PROCESSED_OK;
protected byte[] validateByteArraySize(Buffer buffer,int newSize) {
Object objectArray=buffer.getData();
byte[] typedArray;
if (objectArray instanceof byte[]) { // is correct type AND not null
typedArray=(byte[])objectArray;
if (typedArray.length >= newSize ) { // is sufficient capacity
return typedArray;
typedArray = new byte[newSize];
buffer.setData(typedArray);
return typedArray;
protected void updateOutput(Buffer outputBuffer,
Format format,int length, int offset) {
outputBuffer.setFormat(format);
outputBuffer.setLength(length);
outputBuffer.setOffset(offset);
public static short getSamples(Buffer inBuffer, int pos){
byte[] buffer = (byte[]) inBuffer.getData();
System.out.println("test");
return (short) (((buffer[pos + 1]) << 8) | (buffer[pos]));
public static void setSample(Buffer inBuffer, int pos, short sample){
System.out.println("test2");
byte[] buffer = (byte[]) inBuffer.getData();
buffer[pos] = (byte) (sample & 0x00FF);
buffer[pos + 1] = (byte)((sample>>8)); -
Mpeg audio not playing back correctly
I just upgraded to CS6 and am doing my first project in it.
I have a Sony HVR-Z1U camera which i recorded HDV to and have captured this into PPro using HDV capture settings
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Source Audio Format: 48000 Hz - compressed - Stereo
Project Audio Format: 48000 Hz - 32 bit floating point - Stereo
Total Duration: 00:51:54:21
Pixel Aspect Ratio: 1.3333
When i try and play back the clip in CS6, the video comes out ok but the audio sounds like a chirp.
If i right clip the clip and say to edit in Audition, it sounds fine
If i play the file back from Explorer in Windows Media play it looks and sounds fine
If i import the same file into CS5.5 it plays both video and audio fine.
What am i doing wrong here?Try clearing your media cache files and see if that fixes the issue. You can do so by going up top inside premiere and going to "edit>prefrences>media then click where it says "clean" under media cache datebase. This might fix the problem.
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DVD Player audio track plays slower than Video and at transposed pitch
I have a strange problem with DVD player. I have an opera DVD (not a commercial one - it is from a personal performance and is on a DVD-R disk)) which I put in my Mac Book Pro (OSX 10.5.6)and DVD Player played it fine. I wanted to try to convert the info on the DVD to extract a short section of the DVD. I went to Hand Brake and tried that, it told me I needed to install VLC which I did, Hand Brake then recoded the Files. Now when I insert the DVD though DVD Player starts automatically but plays the DVD audio at a slower speed and transposed lower. I have no idea why this is! Needless to say I do not want it playing slower. VLC also did not work very well with the audio so I have uninstalled the VLC program as it played the VOB files but the sound track was very wonky with the pitch shifting all over the place. How do I return to having DVD Player working properly?
okay, I can help you a little bit. your dvd player (standalone) plays mpeg-2, I think, so
in order for a disk to work in it, both the audio and video have to be combined into a whole package, which i think is video_ts.Using Toast, choose video, then dvd-video then drag n' drop, and it may work. As far as i know, disk utility cannot make video dvd's, but I may be wrong, and you may be right. I used iDVD, but there are probably other programs that may do as good a job of burning. Oh, and this is most important: Some blanks will work ok, and others won't. Verbatim, maxell ( I think) and Taiyo Yuden will work fine. the other ones are a gamble. I think what it's supposed to do is called muxing. I'm sure someone will correct me on this. You could try using "Burn" which is relatively cheap or Handbrake and see if that will do what you want to do. not that it's super important, but the amount of Ram in your machine is helpful too
anyways, good luck
John b -
Audio tracks playing audio when there are no objects...!?
Hi. I would be really grateful if anyone could help me with this problem as I have spent days and nights tearing my hair out so far.
Basically, all my LSO's were created in Logic 6.4.3 and were working fine until (I think) I upgraded to 10.4.8. Since then I have just bought the upgrade to 7.2.3 to try and rectify this problem but still to no avail.
What is happening is that in all my songs, I have tracks playing audio there is no audio object there. I also have audio objects that are not playing at all.
The strange thing is that I have tried in Logic Express and there are no problems with the same songs.
I have been using Logic since version 4 so am a relatively experienced user; I have tried all the obvious stuff (trashing prefs, permissions etc) as well as checking all my inputs, outputs etc in Logic itself. However, what has happened has ultimately changed all my songs in one go and has not been caused by any mistake in use.
As I said, I am really desperate here and would be so grateful if someone can suggest how to overcome these issues.
many thanksthanks for your reply.
Everything is in order in the Mixer and Arrange window. I have checked the Environment and even tried building an Environment from scratch.
I have removed all plugins to see if that is causing it but still nothing.
As I say, exactly the same arrangements are playing fine (just as they used to!) in Logic Express but if I try in either 7.2.3 or go back to 6.4.3, I have this problem...
Please keep the suggestions coming as I really don't know how I can sort this out and it affects all of my work (3 albums +)
thanks again -
Audio Tracks playing but instrument tracks almost mute - anybody had this?
I have two audio tracks which are playing correctly but the instrment tracks are playing almost mute. I don' tthink I changed anything to cause this to happen... Any ideas? thanks
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