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Happy notes for sound buffs: As you'll see from the sample code provided on the Developer CD Series disc, you can make your Macintosh play and record sounds at the same time, simply by using double buffering to record into one buffer while playing a second buffer, and then flipping between the buffers. If you want to take things a few steps further, pull out elements of this code and tailor them to suit your own acoustic needs.

We all know that the Macintosh is a sound machine, so to speak, but with a little clever programming you can turn it into an echo box as well. The sample 2BufRecordToBufCmd included on theDeveloper CD Series disc is just a small application (sans interface) that demonstrates one way to record sounds at the same time that you're playing them. There are other ways to achieve the same goal, but my purpose is to educate you about the Sound Manager, not to lead you down the definitive road to becoming your own recording studio.

In addition to the main routine, 2BufRecordToBufCmd includes various setup routines and a completion routine. For easy reading, I've left out any unnecessary code out of this article.


Before I get into the sample code itself, here are a few of the constants you'll run into in the application.

The kMilliSecondsOfSound constant is used to declare how many milliseconds of sound the application should record before it starts to play back. The smaller the number of milliseconds, the more quickly the sound is played back. This constant is used to calculate the size of the 'snd ' buffer handles (just the data). Depending on the sound effect you're after, kMilliSecondsOfSound can range from 50 milliseconds to 400,000 or so. If you set it below 50, you risk problems: there may not be enough time for the completion routine to finish executing before it's called again. On the high end of the range, only the application's available memory limits the size. The smaller the value, of course, the faster the buffers fill up and play back, and the faster an echo effect you'll get. A millisecond value of 1000 provides a one-second delay between record and echo, which I've found is good for general use. You'll want to experiment to find the effect you like. (Beware of feedback, both from your machine and from anyone who's in close enough proximity to "enjoy" the experimentation secondhand.)

The next three constants (kBaseHeaderSize, kSynthSize, and kCmdSize) are used to parse the sound header buffers in the routine FindHeaderSize. kBaseHeaderSize is the number of bytes at the top of all 'snd ' headers that aren't needed in the application itself. While the number of bytes isn't really ofinterest here, you need to parse the header in order to find the part of the sound header that you'll pass to the bufferCmd. How much you parse off the top is determined by the format of the header and the type of file; for the purposes of this code, however, all you need to be concerned with are the 'snd ' resources. The second constant, kSynthSize, is the size of one 'snth'. In the calculations of the header, I find out how many 'snth's there are, and multiply that number by kSynthSize. The last constant, kCmdSize, is the size of one command, which is used in the same way as kSynthSize. (These equations are derived fromInside Macintosh Volume VI, page 22-20.)

2BufRecordToBufCmd includes error checking, but only as a placeholder for future commercialization of the product. If the present code detects an error, it calls the ExitWithMessage routine, which displays a dialog box that tells you more or less where the error occurred and what the error was. Closing this dialog box quits the application, at which point you have to start over again. Note that calling ExitWithMessage at interrupt time could be fatal, since it uses routines that might move memory. For errors that could occur at interrupt time, DebugStr is used instead.


Use of the sound input driver is fairly well documented inInside Macintosh Volume VI, Chapter 22 (pages 22-58 through 22-68 and 22-92 through 22-99), but here's a little overview of what 2BufRecordToBufCmd does at this point in the routine, and why. When you use sound input calls at the low level (not using SndRecord or SndRecordToFile), you need to open the sound input driver. This section of the code just opens the driver, which the user selects via the sound cdev.

gError = SPBOpenDevice (kDefaultDriver, siWritePermission,

To open the driver, you call SPBOpenDevice and pass in a couple of simple parameters. The first parameter is a driver name. It doesn't really matter what the name of the driver is; it simply needs to be the user-selected driver, so the code passes in nil (which is what kDefaultDriver translates into). The constant siWritePermission tells the driver you'd like read/write permission to the sound input driver. This will enable the application to actually use the recording calls. The last parameter is the gSoundRefNum. This parameter is needed later in the sample so that you can ask specific questions about the driver that's open. The error checking is just to make sure that nothing went wrong; if something did go wrong, the code goes to ExitWithMessage, and then the sample quits.

gError = SPBSetDeviceInfo (gSoundRefNum, siContinuous,
           (Ptr) &contOnOff);

Continuous recording is activated here to avoid a "feature" of the new Macintosh Quadra 700 and 900 that gives you a slowly increasing ramp of the sound input levels to their normal levels each time you call SPBRecord. The result in 2BufRecordToBufCmd is a pause and gradual increase in the sound volume between buffers as the buffers are being played. Continuous recording gives you this ramp only on the first buffer, where it's almost unnoticeable.


Now that the sound input driver is open, the code can get the information it needs to build the 'snd ' buffers. As its name implies, 2BufRecordToBufCmd uses two buffers. The reason is sound (no pun intended): The code basically uses a double-buffer method to record and play the buffers. The code doesn't tell the machine to start to play the sound until the recording completion routine has been called, so you don't have to worry about playing a buffer before it has been filled with recorded data. The code also does not restart the recording until the previous buffer has started to play.

To build the sound headers, you need to get some information from the sound input driver about how the sound data will be recorded and stored. That's the function of the GetSoundDeviceInfo routine, which looks for information about the SampleRate (the number of samples per second at which the sound is recorded), the SampleSize (the sample size of the sound being recorded--8 bits per sample is normal), the CompressionType (see "Putting on the Squeeze"), the NumberChannels(the number of sound input channels, normally 1), and the DeviceBufferInfo (the size of the internal buffers).

This code (minus the error checking) extracts these values from the sound input driver.

gError = SPBGetDeviceInfo (gSoundRefNum, siSampleRate,
        (Ptr) &gSampleRate);

gError = SPBGetDeviceInfo (gSoundRefNum, siSampleSize,
        (Ptr) &gSampleSize);

gError = SPBGetDeviceInfo (gSoundRefNum, siCompressionType,
        (Ptr) &gCompression);

gError = SPBGetDeviceInfo (gSoundRefNum, siNumberChannels,
        (Ptr) &gNumberOfChannels);

gError = SPBGetDeviceInfo (gSoundRefNum, siDeviceBufferInfo,
        (Ptr) &gInternalBuffer);

value = kMilliSecondsOfSound;
gError = SPBMillisecondsToBytes (gSoundRefNum, &value);
gSampleAreaSize = (value / gInternalBuffer) * gInternalBuffer;

Opening the sound input driver gives you the gSoundRefNum. The values siSampleRate, siSampleSize, siCompressionType, siNumberChannels, and siDeviceBufferInfo are constants defined in the SoundInput.h file; these constants tell the SPBGetDeviceInfo call what information you want. The last parameter is a pointer to a global variable. The SPBGetDeviceInfo call uses this parameter to return the requested information.

The last bit of work the code needs to do before it's ready to start building the 'snd ' headers is to convert the constant kMilliSecondsOfSound to the sample size of the buffer. To do this, the routine needs to call SPBMillisecondsToBytes and then round down the resulting value to a multiple of the size of the internal sound buffer. This is to bypass a bug connected with the continuous recording feature of Apple's built-in sound input device, which will collect garbage rather than audio data if the recording buffer is not a multiple of the device's internal buffer. Creating a buffer of the right size not only avoids this problem, but also enables the input device to more efficiently record data into your buffer.

Now the code has the information it needs to build the sound buffers. To save code space, I've made a short routine that builds the buffers and their headers. All the code has to do is call this routine for each of the buffers it needs and pass in the appropriate data.

The first line of code in the SetupSounds routine is fairly obvious. It simply calls the Memory Manager to allocate the requested handles, based on the known size of the data buffer and an estimated maximum size for the header, and does some error checking (see the code itself). Then, if the handle is good, the routine builds the 'snd ' header. Setting up the sound buffer requires building the header by making a simple call, SetupSndHeader, to the Sound Manager. There's a small problem with calling SetupSndHeader only once, however: When you call it, you don't know how big the sound header is, so you just give the call the buffer, along with a 0 value for the buffer size. When the call returns with the header built, one of the values in the header--the one that's the number of bytes in the sample--will be wrong. (The header size will be correct, but the data in the header will not be.) To correct this, you simply wait until your recording is complete and then put the correct number of bytes directly into the header, at which time you'll know how much data there is to play back. The misinformation in the header won't affect your recording, only the playback. Once the header's built, the code resets the size of the handle, moves the handle high (to avoid fragmentation of the heap), and locks it down. It's important to lock down the handles in this way; otherwise the Sound Manager will move the sound buffers it's working with out from under itself.

*bufferHandle = NewHandle (gSampleAreaSize + kEstimatedHeaderSize);

gError = SetupSndHeader (*bufferHandle, gNumberOfChannels,
    gSampleRate, gSampleSize, gCompression, kMiddleC, 0, headerSize);

SetHandleSize (*bufferHandle, (Size) *headerSize + gSampleAreaSize);
MoveHHi (*bufferHandle);
HLock (*bufferHandle);


The next part of the program allocates and initializes a sound input parameter block, gRecordStruct. This structure tells the sound input call how to do what the code wants it to do.

The first instruction is obvious: it simply creates a new pointer into which the structure can be stored.

gRecordStruct = (SPBPtr) NewPtr (sizeof (SPB));

The recording call will need to know where it can find the open sound input driver, so next it needs the reference number to the driver (gSoundRefNum). The subsequent three lines of code inform the recording call how much buffer space it has to record into. Here, you could either give the call a count value, tell it how many milliseconds are available for recording, or give it the size of the sound buffer. For this code, it's easiest to just make the bufferLength the same as the count and ignore the milliseconds value. The code then tells the recording call where to put the sound data as it's recorded.

gRecordStruct->inRefNum = gSoundRefNum;
gRecordStruct->count = gSampleAreaSize;
gRecordStruct->milliseconds = 0;
gRecordStruct->bufferLength = gSampleAreaSize;
gRecordStruct->bufferPtr = (Ptr) ((*bufferHandle) + gHeaderLength);
gRecordStruct->completionRoutine = (ProcPtr) MyRecComp;
gRecordStruct->interruptRoutine = nil;
gRecordStruct->userLong = SetCurrentA5();
gRecordStruct->error = 0;
gRecordStruct->unused1 = 0;

The recording call also needs to know what to do when it's finished recording. Since the call is done asynchronously, it needs a completion routine. (I'll talk more about this routine later on.) Youcould leave out the completion routine and just poll the driver periodically to see if it's finished recording. To do that, you'd repeatedly call the routine SPBGetRecordStatus, and when the status routine informed you that recording was finished, you'd restart the recording and play the buffer that had just been filled. For this code, however, it's better to know as soon as possible when the recording is done because the more quickly you can restart the recording, the more likely you are to prevent pauses between recordings.

The userLong field is a good place to store 2BufRecordToBufCmd's A5 value, which you'll need in order to have access to the application's global variables from the completion routine. As you can see, the rest of the fields are set to 0. The code doesn't need an interrupt routine. There's also no point in passing an error back or using the unused1 field.

You'd need to use an interrupt routine if you wanted to change the recorded sound before compression, or before the completion routine was called (see "Routine Interruptions").

Just before the code jumps into the main loop, it needs to open a sound channel. This generally is not a big deal, but for 2BufRecordToBufCmd, I initialized the channel to use no interpolation.

gError = SndNewChannel (&gChannel, sampledSynth, initNoInterp, nil);

Interpolation causes clicks between the sound buffers when they're played back to back, which can be a rather annoying addition to your recording (unless, of course, you're going for that samba beat).

To start recording, all the code needs to do now is call the low-level recording routine, pass in gRecordStruct, and tell it that it wants the recording to occur asynchronously.

gError = SPBRecord (gRecordStruct, true);


The main loop of this code is a simple while loop that waits until the mouse button is pressed or an error occurs in the recording, at which time the application quits.

/* main loop of the app */
while (!Button() || (gRecordStruct->error < noErr));

You don't want a completion routine to do much, generally, since it's run at interrupt time and keeps your system locked up while it's running. There are three parts to this completion routine, one of which has four parts to itself.

The first part of the completion routine sets its A5 value to be the same as the A5 value of the application. This gives you access to the application's global variables from the completion routine.

storeA5 = SetA5 (inParamPtr->userLong);

If the completion routine weren't broken into two parts here, the MPW C compiler optimization scheme would cause a problem at this point: access to global arrays would be pointed to in an address register as an offset of A5 before you had a chance to set A5 to your application's A5 value, and you'd get garbage information. Therefore, it's necessary to restore your A5 value (part 1 of the completion routine) and then call the secondary completion routine to actually do all the work.

Before the routine does any work, it needs to make sure that there have not been any problems with the recording. If there were errors, the code drops out of the completion routine without doing anything.

if (gRecordStruct->error < 0)

Next the routine prepares the header of the buffer, which has just been filled, by correcting the header's length field. This field needs to be set to the count field of gRecordStruct, which now contains the actual number of bytes recorded.

header = (SoundHeaderPtr)(*(gBufferHandle[gWhichRecordBuffer]) +
header->length = gRecordStruct->count;

Once the header's been fixed, the code just sends the buffer handle off to the play routine to play the sound. (See "Play Time" for a full explanation of the play routine.)

PlayBuffer (gBufferHandle[gWhichRecordBuffer]);
The last part of the real completion routine prepares gRecordStruct to start the next recording. To do this, the code needs to select the correct buffer to record to and rebuild gRecordStruct to reflect any changes. The macro NextBuffer performs an XOR on the variable gWhichRecordBuffer to make it either 1 or 0. The changes include setting the correct buffer to record to and checking to see that the bufferLength is correct. Once the structure is reset, the code makes the next call to SPBRecord to restart the recording.
#define NextBuffer(x) (x ^= 1)

gWhichRecordBuffer = NextBuffer (gWhichRecordBuffer);
gRecordStruct->bufferPtr = (*(gBufferHandle[gWhichRecordBuffer]) +
gRecordStruct->milliseconds = 0;
gRecordStruct->count = gSampleAreaSize;
gRecordStruct->bufferLength = gSampleAreaSize;

err = SPBRecord (gRecordStruct, true);

The last piece of the completion routine resets A5 to what its value was when the routine started.

storeA5 = SetA5 (storeA5);

The code in the PlayBuffer routine is very simple Sound Manager code. All it does is set up the command parameters and call SndDoCommand. The routine needs to know what channel to play into and what buffer to play, so the code sets up the local sound structure by telling it which buffer to play, and sends that local structure to SndDoCommand along with the necessary channel information (gChannel). SndDoCommand then plays the sound. The last parameter in the SndDoCommand call, false, basically tells the Sound Manager to always insert the command in the channel's queue: if the queue is full, SndDoCommand will wait until there's space to insert the command before returning.

localSndCmd.cmd = bufferCmd;
localSndCmd.param1 = 0;
localSndCmd.param2 = (long) ((*bufferHandle) + gHeaderSize);
gError = SndDoCommand (gChannel, &localSndCmd, false);

If you wanted to send the sounds to a different machine to be played, you could simply replace the code in the the PlayBuffer routine with IPC or Communications Toolbox calls telling a second machine to play the buffers.


Once the code finds the mouse button down or discovers that an error occurred in the recording and exits the main loop, there's only one last thing to do: clean up. The first part of cleaning up is to close the sound input driver. Before you can close the driver, you need to make sure it's not in use; the routine SPBStopRecording stops the recording.

gError = SPBStopRecording (gSoundRefNum);
SPBCloseDevice (gSoundRefNum);

Next you need to dispose of the handles and pointers you've been using. Before sending them on their way, however, you have to make sure that they have been allocated, so the code checks to see whether or not the handles and pointer are nil.

for (index = 0; index < kNumberOfBuffers; ++index)
    DisposeHandle (gBufferHandle[index]);
DisposePtr ((Ptr) gRecordStruct);
Last but not least, the code disposes of the sound channel for you. Setting the quitNow flag clears the sound queue before the channel is closed.

gError = SndDisposeChannel (gChannel, true);


So now you know a little bit more about doing basic sound input at a low level. I've fielded many questions about clicks, pauses between buffers, and so on, which I've resolved and built into 2BufRecordToBufCmd. The specific techniques I've outlined here may not apply to what you're interested in doing right now, but if you're using the sound input driver or are interested in continuous recording, parts of this sample may be useful to you in some other application. You've heard the saying "take what you like and leave the rest"? Sound advice (so to speak).


You do need to check two rather critical sound attributes for 2BufRecordToBufCmd. First of all, your machine must have a sound input driver. There's very little point in trying to record sounds if the sample is being run on a machine that doesn't have sound input capabilities. Checking bit 5 of the returned feature variable with the Gestalt Manager will give you this handy bit of information.

Second, your hardware needs to support stereo sound, since you need one channel for sound input and one for sound output. Check for this attribute by checking bit 0 of the returned feature variable.

The following code shows how you can test all of the bits returned in the feature variable. (I didn't use this code in my sample.)

err = Gestalt (gestaltSoundAttr, &feature);
if (!err) {
    if (feature & (1 << gestaltStereoCapability))
        //This Macintosh Supports Stereo (test bit 0)
    if (feature & (1 << gestaltStereoMixing))
        //This Macintosh Supports Stereo Mixing (test bit 1)
    if (feature & (1 << gestaltSoundIOMgrPresent))
        //This Macintosh Has the New Sound Manager (test bit 3)
    if (feature & (1 << gestaltBuiltInSoundInput))
        //This Macintosh Has Built-in Sound Input (test bit 4)
    if (feature & (1 << gestaltHasSoundInputDevice))
        //This Macintosh Supports Sound Input (test bit 5)


If you want to use compression for 2BufRecordToBufCmd, keep in mind that the Sound Manager basically supports three types of sound compression: none at all, which is what I'm using, and MAC3 and MAC6, which are Mace compression types for 3:1 and 6:1 compression, respectively.

If you set the compression, the sound data is compressed after the interrupt routine is called (if you have one) and before the Sound Manager internal buffers are moved to the application's sound buffers.

You have a couple of options for playing back a compressed sound. Either the bufferCmd or SndPlay will decompress the sounds on the fly. If you need to decompress a sound yourself, you'll want to call the Sound Manager routine Exp1to3 or Exp1to6 (depending on the compression you were using).


The interrupt routine gives you a chance to manipulate the sound data before any sound compression is done. For some of the operations that you may want to carry out inside the interrupt routine, you'll need access to the A5 world of the application, which is why I stored 2BufRecordToBufCmd's A5 value in the userLong field of gRecordStruct.For more information about sound interrupt routines, take a look at Inside Macintosh  Volume VI, page 22-63.

Warning:  Don't try to accomplish too much in an interrupt routine. In general, you'll want interrupts to be minimal, and possibly written in assembly language, to avoid unnecessary compiler-generated code.

RICH COLLYER is just your run-of-the-mill three-year Developer Technical Support veteran: He's often heard screaming at his computer to the soothing accompaniment of Blazy and Bob on KOME radio, he's honed his archery skills to a fine point dodging (and casting) the slings and arrows at Apple, and he actually admits to a degree from Cal Poly with a specialty in computational fluid dynamics. We let you in on his outdoor adventures last time he wrote for us and he claims most of his indoor adventures aren't appropriate develop  material, but we have it on good authority that he lives with carnivorous animals, if that's any clue. He's also a confirmed laserdisc and CD addict; he keeps promising to start a recovery program for those of us with the same affliction just as soon as he finishes writing that next sample . . . *

THANKS TO OUR TECHNICAL REVIEWERS Neil Day, Kip Olson, and Jim Reekes, who burned the midnight oil ripping this code to shreds and putting it back together again.*


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