audiooutputca.cpp

Go to the documentation of this file.

/*****************************************************************************
 * = NAME
 * audiooutputca.cpp
 *
 * = DESCRIPTION
 * Core Audio glue for Mac OS X.
 *
 * = REVISION
 * $Id$
 *
 * = AUTHORS
 * Jeremiah Morris, Andrew Kimpton, Nigel Pearson, Jean-Yves Avenard
 *****************************************************************************/

#include <CoreServices/CoreServices.h>
#include <CoreAudio/CoreAudio.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioFormat.h>

using namespace std;

#include "mythcorecontext.h"
#include "audiooutputca.h"
#include "config.h"
#include "SoundTouch.h"

#define LOC QString("CoreAudio: ")

#define CHANNELS_MIN 1
#define CHANNELS_MAX 8

#define OSS_STATUS(x) UInt32ToFourCC((UInt32*)&x)
char* UInt32ToFourCC(UInt32* pVal)
{
    UInt32 inVal = *pVal;
    char* pIn = (char*)&inVal;
    static char fourCC[5];
    fourCC[4] = 0;
    fourCC[3] = pIn[0];
    fourCC[2] = pIn[1];
    fourCC[1] = pIn[2];
    fourCC[0] = pIn[3];
    return fourCC;
}

QString StreamDescriptionToString(AudioStreamBasicDescription desc)
{
    UInt32 formatId = desc.mFormatID;
    char* fourCC = UInt32ToFourCC(&formatId);
    QString str;

    switch (desc.mFormatID)
    {
        case kAudioFormatLinearPCM:
            str = QString("[%1] %2%3 Channel %4-bit %5 %6 (%7Hz)")
            .arg(fourCC)
            .arg((desc.mFormatFlags & kAudioFormatFlagIsNonMixable) ? "" : "Mixable ")
            .arg(desc.mChannelsPerFrame)
            .arg(desc.mBitsPerChannel)
            .arg((desc.mFormatFlags & kAudioFormatFlagIsFloat) ? "Floating Point" : "Signed Integer")
            .arg((desc.mFormatFlags & kAudioFormatFlagIsBigEndian) ? "BE" : "LE")
            .arg((UInt32)desc.mSampleRate);
            break;
        case kAudioFormatAC3:
            str = QString("[%1] AC-3/DTS (%2Hz)")
            .arg(fourCC)
            .arg((UInt32)desc.mSampleRate);
            break;
        case kAudioFormat60958AC3:
            str = QString("[%1] AC-3/DTS for S/PDIF %2 (%3Hz)")
            .arg(fourCC)
            .arg((desc.mFormatFlags & kAudioFormatFlagIsBigEndian) ? "BE" : "LE")
            .arg((UInt32)desc.mSampleRate);

            break;
        default:
            str = QString("[%1]").arg(fourCC);
            break;
    }
    return str;
}

class CoreAudioData {
public:
    CoreAudioData(AudioOutputCA *parent);
    CoreAudioData(AudioOutputCA *parent, AudioDeviceID deviceID);
    CoreAudioData(AudioOutputCA *parent, QString deviceName);
    void Initialise();

    AudioDeviceID GetDefaultOutputDevice();
    int  GetTotalOutputChannels();
    QString *GetName();
    AudioDeviceID GetDeviceWithName(QString deviceName);

    bool OpenDevice();
    int  OpenAnalog();
    void CloseAnalog();
    bool OpenSPDIF ();
    void CloseSPDIF();

    void SetAutoHogMode(bool enable);
    bool GetAutoHogMode();
    pid_t GetHogStatus();
    bool SetHogStatus(bool hog);
    bool SetMixingSupport(bool mix);
    bool GetMixingSupport();

    bool FindAC3Stream();
    void ResetAudioDevices();
    void ResetStream(AudioStreamID s);
    int *RatesList(AudioDeviceID d);
    bool *ChannelsList(AudioDeviceID d, bool passthru);

    // Helpers for iterating. Returns a malloc'd array
    AudioStreamID               *StreamsList(AudioDeviceID d);
    AudioStreamBasicDescription *FormatsList(AudioStreamID s);

    int  AudioStreamChangeFormat(AudioStreamID               s,
                                 AudioStreamBasicDescription format);

    // TODO: Convert these to macros!
    void  Debug(QString msg)
    {   LOG(VB_AUDIO, LOG_INFO,      "CoreAudioData::" + msg);   }

    void  Error(QString msg)
    {    LOG(VB_GENERAL, LOG_ERR, "CoreAudioData Error:" + msg);   }

    void  Warn (QString msg)
    {    LOG(VB_GENERAL, LOG_WARNING, "CoreAudioData Warning:" + msg);   }

    AudioOutputCA  *mCA;    // We could subclass, but this ends up tidier

    // Analog output specific
    AudioUnit      mOutputUnit;

    // SPDIF mode specific
    bool           mDigitalInUse;   // Is the digital (SPDIF) output in use?
    pid_t          mHog;
    int            mMixerRestore;
    AudioDeviceID  mDeviceID;
    AudioStreamID  mStreamID;       // StreamID that has a cac3 streamformat
    int            mStreamIndex;    // Index of mStreamID in an AudioBufferList
    UInt32         mBytesPerPacket;
    AudioStreamBasicDescription
    mFormatOrig,     // The original format the stream
    mFormatNew;      // The format we changed the stream to
    bool           mRevertFormat;   // Do we need to revert the stream format?
    bool           mIoProc;
    bool           mInitialized;
    bool           mStarted;
    bool           mWasDigital;
};

// These callbacks communicate with Core Audio.
static OSStatus RenderCallbackAnalog(void                       *inRefCon,
                                     AudioUnitRenderActionFlags *ioActionFlags,
                                     const AudioTimeStamp       *inTimeStamp,
                                     UInt32                     inBusNumber,
                                     UInt32                     inNumberFrames,
                                     AudioBufferList            *ioData);
static OSStatus RenderCallbackSPDIF(AudioDeviceID        inDevice,
                                    const AudioTimeStamp *inNow,
                                    const void           *inInputData,
                                    const AudioTimeStamp *inInputTime,
                                    AudioBufferList      *outOutputData,
                                    const AudioTimeStamp *inOutputTime,
                                    void                 *threadGlobals);

AudioOutputCA::AudioOutputCA(const AudioSettings &settings) :
AudioOutputBase(settings)
{
    main_device.remove(0, 10);
    VBAUDIO(QString("AudioOutputCA::AudioOutputCA searching %1").arg(main_device));
    d = new CoreAudioData(this, main_device);

    InitSettings(settings);
    if (settings.init)
        Reconfigure(settings);
}

AudioOutputCA::~AudioOutputCA()
{
    KillAudio();

    delete d;
}

AudioOutputSettings* AudioOutputCA::GetOutputSettings(bool digital)
{
    AudioOutputSettings *settings = new AudioOutputSettings();

    // Seek hardware sample rate available
    int rate;
    int *rates = d->RatesList(d->mDeviceID);

    if (rates == NULL)
    {
        // Error retrieving rates, assume 48kHz
        settings->AddSupportedRate(48000);
    }
    else
    {
        while ((rate = settings->GetNextRate()))
        {
                        int *p_rates = rates;
                        while (*p_rates > 0)
                        {
                                if (*p_rates == rate)
                                {
                                        settings->AddSupportedRate(*p_rates);
                                }
                p_rates++;
            }
        }
        free(rates);
    }

    // Supported format: 16 bits audio or float
    settings->AddSupportedFormat(FORMAT_S16);
    settings->AddSupportedFormat(FORMAT_FLT);

    bool *channels = d->ChannelsList(d->mDeviceID, digital);
    if (channels == NULL)
    {
        // Error retrieving list of supported channels, assume stereo only
        settings->AddSupportedChannels(2);
    }
    else
    {
        for (int i = CHANNELS_MIN; i <= CHANNELS_MAX; i++)
        {
            if (channels[i])
            {
                Debug(QString("Support %1 channels").arg(i));
                // In case 8 channels are supported but not 6, fake 6
                if (i == 8 && !channels[6])
                    settings->AddSupportedChannels(6);
                settings->AddSupportedChannels(i);
            }
        }
        free(channels);
    }

    if (d->FindAC3Stream())
    {
        settings->setPassthrough(1); // yes passthrough
    }
    return settings;
}

bool AudioOutputCA::OpenDevice()
{
    bool deviceOpened = false;

    if (d->mWasDigital) 
    {
    }
    Debug("OpenDevice: Entering");
    if (passthru || enc)
    {
        Debug("OpenDevice() Trying Digital.");
        if (!(deviceOpened = d->OpenSPDIF()))
            d->CloseSPDIF();
    }

    if (!deviceOpened)
    {
        Debug("OpenDevice() Trying Analog.");
        int result = -1;
        //for (int i=0; result < 0 && i < 10; i++)
        {
            result = d->OpenAnalog();
            Debug(QString("OpenDevice: OpenAnalog = %1").arg(result));
            if (result < 0)
            {
                d->CloseAnalog();
                usleep(1000 * 1000);
            }
        }
        deviceOpened = (result > 0);
    }

    if (!deviceOpened)
    {
        Error("Couldn't open any audio device!");
        d->CloseAnalog();
        return false;
    }

    if (internal_vol && set_initial_vol)
    {
        QString controlLabel = gCoreContext->GetSetting("MixerControl", "PCM");
        controlLabel += "MixerVolume";
        SetCurrentVolume(gCoreContext->GetNumSetting(controlLabel, 80));
    }

    return true;
}

void AudioOutputCA::CloseDevice()
{
    VBAUDIO(QString("CloseDevice [%1]: Entering")
            .arg(d->mDigitalInUse ? "SPDIF" : "Analog"));
    if (d->mDigitalInUse)
        d->CloseSPDIF();
    else
        d->CloseAnalog();
}

template <class AudioDataType>
static inline void _ReorderSmpteToCA(AudioDataType *buf, uint frames)
{
    AudioDataType tmpLS, tmpRS, tmpRLs, tmpRRs, *buf2;
    for (uint i = 0; i < frames; i++)
    {
        buf = buf2 = buf + 4;
        tmpRLs = *buf++;
        tmpRRs = *buf++;
        tmpLS = *buf++;
        tmpRS = *buf++;

        *buf2++ = tmpLS;
        *buf2++ = tmpRS;
        *buf2++ = tmpRLs;
        *buf2++ = tmpRRs;
    }
}

static inline void ReorderSmpteToCA(void *buf, uint frames, AudioFormat format)
{
    switch(AudioOutputSettings::FormatToBits(format))
    {
        case  8: _ReorderSmpteToCA((uchar *)buf, frames); break;
        case 16: _ReorderSmpteToCA((short *)buf, frames); break;
        default: _ReorderSmpteToCA((int   *)buf, frames); break;
    }
}

bool AudioOutputCA::RenderAudio(unsigned char *aubuf,
                                int size, unsigned long long timestamp)
{
    if (pauseaudio || killaudio)
    {
        actually_paused = true;
        return false;
    }

    /* This callback is called when the sound system requests
     data.  We don't want to block here, because that would
     just cause dropouts anyway, so we always return whatever
     data is available.  If we haven't received enough, either
     because we've finished playing or we have a buffer
     underrun, we play silence to fill the unused space.  */

    int written_size = GetAudioData(aubuf, size, false);
    if (written_size && (size > written_size))
    {
        // play silence on buffer underrun
        memset(aubuf + written_size, 0, size - written_size);
    }

    //Audio received is in SMPTE channel order, reorder to CA unless passthru
    if (!passthru && channels == 8)
    {
        ReorderSmpteToCA(aubuf, size / output_bytes_per_frame, output_format);
    }

    /* update audiotime (bufferedBytes is read by GetBufferedOnSoundcard) */
    UInt64 nanos = AudioConvertHostTimeToNanos(timestamp -
                                               AudioGetCurrentHostTime());
    bufferedBytes = (int)((nanos / 1000000000.0) *    // secs
                          (effdsp / 100.0) *          // frames/sec
                          output_bytes_per_frame);    // bytes/frame

    return (written_size > 0);
}

void AudioOutputCA::WriteAudio(unsigned char *aubuf, int size)
{
    (void)aubuf;
    (void)size;
    return;     // unneeded and unused in CA
}

int AudioOutputCA::GetBufferedOnSoundcard(void) const
{
    return bufferedBytes;
}

int64_t AudioOutputCA::GetAudiotime(void)
{
    int audbuf_timecode = GetBaseAudBufTimeCode();

    if (!audbuf_timecode)
        return 0;

    int totalbuffer = audioready() + GetBufferedOnSoundcard();

    return audbuf_timecode - (int)(totalbuffer * 100000.0 /
                                   (output_bytes_per_frame *
                                    effdsp * stretchfactor));
}

/* This callback provides converted audio data to the default output device. */
OSStatus RenderCallbackAnalog(void                       *inRefCon,
                              AudioUnitRenderActionFlags *ioActionFlags,
                              const AudioTimeStamp       *inTimeStamp,
                              UInt32                     inBusNumber,
                              UInt32                     inNumberFrames,
                              AudioBufferList            *ioData)
{
    (void)inBusNumber;
    (void)inNumberFrames;

    AudioOutputCA *inst = (static_cast<CoreAudioData *>(inRefCon))->mCA;

    if (!inst->RenderAudio((unsigned char *)(ioData->mBuffers[0].mData),
                           ioData->mBuffers[0].mDataByteSize,
                           inTimeStamp->mHostTime))
    {
        // play silence if RenderAudio returns false
        memset(ioData->mBuffers[0].mData, 0, ioData->mBuffers[0].mDataByteSize);
        *ioActionFlags = kAudioUnitRenderAction_OutputIsSilence;
    }
    return noErr;
}

int AudioOutputCA::GetVolumeChannel(int channel) const
{
    // FIXME: this only returns global volume
    (void)channel;
    Float32 volume;

    if (!AudioUnitGetParameter(d->mOutputUnit,
                               kHALOutputParam_Volume,
                               kAudioUnitScope_Global, 0, &volume))
        return (int)lroundf(volume * 100.0f);

    return 0;    // error case
}

void AudioOutputCA::SetVolumeChannel(int channel, int volume)
{
    // FIXME: this only sets global volume
    (void)channel;
    AudioUnitSetParameter(d->mOutputUnit, kHALOutputParam_Volume,
                          kAudioUnitScope_Global, 0, (volume * 0.01f), 0);
}

// IOProc style callback for SPDIF audio output
static OSStatus RenderCallbackSPDIF(AudioDeviceID        inDevice,
                                    const AudioTimeStamp *inNow,
                                    const void           *inInputData,
                                    const AudioTimeStamp *inInputTime,
                                    AudioBufferList      *outOutputData,
                                    const AudioTimeStamp *inOutputTime,
                                    void                 *inRefCon)
{
    CoreAudioData    *d = static_cast<CoreAudioData *>(inRefCon);
    AudioOutputCA    *a = d->mCA;
    int           index = d->mStreamIndex;

    (void)inDevice;     // unused
    (void)inNow;        // unused
    (void)inInputData;  // unused
    (void)inInputTime;  // unused

    /*
     * HACK: No idea why this would be the case, but after the second run, we get
     * incorrect value
     */
    if (d->mBytesPerPacket > 0 &&
        outOutputData->mBuffers[index].mDataByteSize > d->mBytesPerPacket)
    {
        outOutputData->mBuffers[index].mDataByteSize = d->mBytesPerPacket;
    }
    if (!a->RenderAudio((unsigned char *)(outOutputData->mBuffers[index].mData),
                        outOutputData->mBuffers[index].mDataByteSize,
                        inOutputTime->mHostTime))
    {
        // play silence if RenderAudio returns false
        memset(outOutputData->mBuffers[index].mData, 0,
              outOutputData->mBuffers[index].mDataByteSize);
    }
    return noErr;
}

void CoreAudioData::Initialise()
{
    // Initialise private data
    mOutputUnit     = NULL;
    mDeviceID       = 0;
    mDigitalInUse   = false;
    mRevertFormat   = false;
    mHog            = -1;
    mMixerRestore   = -1;
    mStreamIndex    = -1;
    mIoProc         = false;
    mInitialized    = false;
    mStarted        = false;
    mBytesPerPacket = -1;
    mWasDigital     = false;
}

CoreAudioData::CoreAudioData(AudioOutputCA *parent) : mCA(parent)
{
    Initialise();
    // Reset all the devices to a default 'non-hog' and mixable format.
    // If we don't do this we may be unable to find the Default Output device.
    // (e.g. if we crashed last time leaving it stuck in AC-3 mode)
    ResetAudioDevices();

    mDeviceID = GetDefaultOutputDevice();
}

CoreAudioData::CoreAudioData(AudioOutputCA *parent, AudioDeviceID deviceID) :
    mCA(parent)
{
    Initialise();
    ResetAudioDevices();
    mDeviceID = deviceID;
}

CoreAudioData::CoreAudioData(AudioOutputCA *parent, QString deviceName) :
    mCA(parent)
{
    Initialise();
    ResetAudioDevices();
    mDeviceID = GetDeviceWithName(deviceName);
    if (!mDeviceID)
    {
        // Didn't find specified device, use default one
        mDeviceID = GetDefaultOutputDevice();
        if (deviceName != "Default Output Device")
        {
            Warn(QString("CoreAudioData: \"%1\" not found, using default device %2.")
                 .arg(deviceName).arg(mDeviceID));
        }
    }
    Debug(QString("CoreAudioData: device number is %1")
          .arg(mDeviceID));
}

AudioDeviceID CoreAudioData::GetDeviceWithName(QString deviceName)
{
    UInt32 size = 0;
    AudioDeviceID deviceID = 0;

    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &size, NULL);
    UInt32 deviceCount = size / sizeof(AudioDeviceID);
    AudioDeviceID* pDevices = new AudioDeviceID[deviceCount];

    OSStatus err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &size, pDevices);
    if (err)
    {
        Warn(QString("GetDeviceWithName: Unable to retrieve the list of available devices. "
                     "Error [%1]")
             .arg(err));
    }
    else
    {
        for (UInt32 dev = 0; dev < deviceCount; dev++)
        {
            CoreAudioData device(NULL, pDevices[dev]);
            if (device.GetTotalOutputChannels() == 0)
                continue;
            QString *name = device.GetName();
            if (name && name == deviceName)
            {
                Debug(QString("GetDeviceWithName: Found: %1").arg(*name));
                deviceID = pDevices[dev];
                delete name;
            }
            if (deviceID)
                break;
        }
    }
    delete[] pDevices;
    return deviceID;
}

AudioDeviceID CoreAudioData::GetDefaultOutputDevice()
{
    UInt32        paramSize;
    OSStatus      err;
    AudioDeviceID deviceId = 0;

    // Find the ID of the default Device
    paramSize = sizeof(deviceId);
    err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
                                   &paramSize, &deviceId);
    if (err == noErr)
        Debug(QString("GetDefaultOutputDevice: default device ID = %1").arg(deviceId));
    else
    {
        Warn(QString("GetDefaultOutputDevice: could not get default audio device: [%1]")
             .arg(OSS_STATUS(err)));
        deviceId = 0;
    }
    return deviceId;
}

int CoreAudioData::GetTotalOutputChannels()
{
    if (!mDeviceID)
        return 0;
    UInt32 channels = 0;
    UInt32 size = 0;
    AudioDeviceGetPropertyInfo(mDeviceID, 0, false,
                               kAudioDevicePropertyStreamConfiguration,
                               &size, NULL);
    AudioBufferList *pList = (AudioBufferList *)malloc(size);
    OSStatus err = AudioDeviceGetProperty(mDeviceID, 0, false,
                                          kAudioDevicePropertyStreamConfiguration,
                                          &size, pList); 
    if (!err)
    {
        for (UInt32 buffer = 0; buffer < pList->mNumberBuffers; buffer++)
            channels += pList->mBuffers[buffer].mNumberChannels;
    }
    else
    {
        Warn(QString("GetTotalOutputChannels: Unable to get "
                     "total device output channels - id: %1 Error = [%2]")
             .arg(mDeviceID)
             .arg(err));
    }
    Debug(QString("GetTotalOutputChannels: Found %1 channels in %2 buffers")
          .arg(channels).arg(pList->mNumberBuffers));
    free(pList);
    return channels;
}

QString *CoreAudioData::GetName()
{
    if (!mDeviceID)
        return NULL;
    UInt32 propertySize;
    AudioObjectPropertyAddress propertyAddress;

    CFStringRef name;
    propertySize = sizeof(CFStringRef);
    propertyAddress.mSelector = kAudioObjectPropertyName;
    propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
    propertyAddress.mElement = kAudioObjectPropertyElementMaster;
    OSStatus err = AudioObjectGetPropertyData(mDeviceID, &propertyAddress,
                                              0, NULL, &propertySize, &name);
    if (err)
    {
        Error(QString("AudioObjectGetPropertyData for kAudioObjectPropertyName error: [%1]")
              .arg(err));
        return NULL;
    }
    char *cname = new char[CFStringGetLength(name) + 1];
    CFStringGetCString(name, cname, CFStringGetLength(name) + 1, kCFStringEncodingUTF8);
    QString *qname = new QString(cname);
    delete[] cname;
    return qname;
}

bool CoreAudioData::GetAutoHogMode()
{
    UInt32 val = 0;
    UInt32 size = sizeof(val);
    OSStatus err = AudioHardwareGetProperty(kAudioHardwarePropertyHogModeIsAllowed,
                                            &size, &val);
    if (err)
    {
        Warn(QString("GetAutoHogMode: Unable to get auto 'hog' mode. Error = [%1]")
             .arg(err));
        return false;
    }
    return (val == 1);
}

void CoreAudioData::SetAutoHogMode(bool enable)
{
    UInt32 val = enable ? 1 : 0;
    OSStatus err = AudioHardwareSetProperty(kAudioHardwarePropertyHogModeIsAllowed,
                                            sizeof(val), &val);
    if (err)
    {
        Warn(QString("SetAutoHogMode: Unable to set auto 'hog' mode. Error = [%1]")
             .arg(err));
    }
}

pid_t CoreAudioData::GetHogStatus()
{
    OSStatus err;
    pid_t PID;
    UInt32 PIDsize = sizeof(PID);

    err = AudioDeviceGetProperty(mDeviceID, 0, FALSE,
                                 kAudioDevicePropertyHogMode,
                                 &PIDsize, &PID);
    if (err != noErr)
    {
        // This is not a fatal error.
        // Some drivers simply don't support this property
        Debug(QString("GetHogStatus: unable to check: [%1]")
              .arg(err));
        return -1;
    }
    return PID;
}

bool CoreAudioData::SetHogStatus(bool hog)
{
    // According to Jeff Moore (Core Audio, Apple), Setting kAudioDevicePropertyHogMode
    // is a toggle and the only way to tell if you do get hog mode is to compare
    // the returned pid against getpid, if the match, you have hog mode, if not you don't.
    if (!mDeviceID)
        return false;

    if (hog)
    {
        if (mHog == -1) // Not already set
        {
            Debug(QString("SetHogStatus: Setting 'hog' status on device %1")
                  .arg(mDeviceID));
            OSStatus err = AudioDeviceSetProperty(mDeviceID, NULL, 0, false,
                                                  kAudioDevicePropertyHogMode,
                                                  sizeof(mHog), &mHog);
            if (err || mHog != getpid())
            {
                Warn(QString("SetHogStatus: Unable to set 'hog' status. Error = [%1]")
                     .arg(OSS_STATUS(err)));
                return false;
            }
            Debug(QString("SetHogStatus: Successfully set 'hog' status on device %1")
                  .arg(mDeviceID));
        }
    }
    else
    {
        if (mHog > -1) // Currently Set
        {
            Debug(QString("SetHogStatus: Releasing 'hog' status on device %1")
                  .arg(mDeviceID));
            pid_t hogPid = -1;
            OSStatus err = AudioDeviceSetProperty(mDeviceID, NULL, 0, false,
                                                  kAudioDevicePropertyHogMode,
                                                  sizeof(hogPid), &hogPid);
            if (err || hogPid == getpid())
            {
                Warn(QString("SetHogStatus: Unable to release 'hog' status. Error = [%1]")
                     .arg(OSS_STATUS(err)));
                return false;
            }
            mHog = hogPid; // Reset internal state
        }
    }
    return true;
}

bool CoreAudioData::SetMixingSupport(bool mix)
{
    if (!mDeviceID)
        return false;
    int restore = -1;
    if (mMixerRestore == -1) // This is our first change to this setting. Store the original setting for restore
        restore = (GetMixingSupport() ? 1 : 0);
    UInt32 mixEnable = mix ? 1 : 0;
    Debug(QString("SetMixingSupport: %1abling mixing for device %2")
          .arg(mix ? "En" : "Dis")
          .arg(mDeviceID));
    OSStatus err = AudioDeviceSetProperty(mDeviceID, NULL, 0, false,
                                          kAudioDevicePropertySupportsMixing,
                                          sizeof(mixEnable), &mixEnable);
    if (err)
    {
        Warn(QString("SetMixingSupport: Unable to set MixingSupport to %1. Error = [%2]")
             .arg(mix ? "'On'" : "'Off'")
             .arg(OSS_STATUS(err)));
        return false;
    }
    if (mMixerRestore == -1) 
        mMixerRestore = restore;
    return true;
}

bool CoreAudioData::GetMixingSupport()
{
    if (!mDeviceID)
        return false;
    UInt32 val = 0;
    UInt32 size = sizeof(val);
    OSStatus err = AudioDeviceGetProperty(mDeviceID, 0, false,
                                          kAudioDevicePropertySupportsMixing,
                                          &size, &val);
    if (err)
        return false;
    return (val > 0);
}

AudioStreamID *CoreAudioData::StreamsList(AudioDeviceID d)
{
    OSStatus       err;
    UInt32         listSize;
    AudioStreamID  *list;


    err = AudioDeviceGetPropertyInfo(d, 0, FALSE,
                                     kAudioDevicePropertyStreams,
                                     &listSize, NULL);
    if (err != noErr)
    {
        Error(QString("StreamsList: could not get list size: [%1]")
              .arg(OSS_STATUS(err)));
        return NULL;
    }

    // Space for a terminating ID:
    listSize += sizeof(AudioStreamID);
    list      = (AudioStreamID *)malloc(listSize);

    if (list == NULL)
    {
        Error("StreamsList(): out of memory?");
        return NULL;
    }

    err = AudioDeviceGetProperty(d, 0, FALSE,
                                 kAudioDevicePropertyStreams,
                                 &listSize, list);
    if (err != noErr)
    {
        Error(QString("StreamsList: could not get list: [%1]")
              .arg(OSS_STATUS(err)));
        return NULL;
    }
    // Add a terminating ID:
    list[listSize/sizeof(AudioStreamID)] = kAudioHardwareBadStreamError;

    return list;
}

AudioStreamBasicDescription *CoreAudioData::FormatsList(AudioStreamID s)
{
    OSStatus                     err;
    AudioStreamBasicDescription  *list;
    UInt32                       listSize;
    AudioDevicePropertyID        p;


    // This is deprecated for kAudioStreamPropertyAvailablePhysicalFormats,
    // but compiling on 10.3 requires the older constant
    p = kAudioStreamPropertyPhysicalFormats;

    // Retrieve all the stream formats supported by this output stream
    err = AudioStreamGetPropertyInfo(s, 0, p, &listSize, NULL);
    if (err != noErr)
    {
        Warn(QString("FormatsList(): couldn't get list size: [%1]")
             .arg(OSS_STATUS(err)));
        return NULL;
    }

    // Space for a terminating ID:
    listSize += sizeof(AudioStreamBasicDescription);
    list      = (AudioStreamBasicDescription *)malloc(listSize);

    if (list == NULL)
    {
        Error("FormatsList(): out of memory?");
        return NULL;
    }

    err = AudioStreamGetProperty(s, 0, p, &listSize, list);
    if (err != noErr)
    {
        Warn(QString("FormatsList: couldn't get list: [%1]")
             .arg(OSS_STATUS(err)));
        free(list);
        return NULL;
    }

    // Add a terminating ID:
    list[listSize/sizeof(AudioStreamBasicDescription)].mFormatID = 0;

    return list;
}

static UInt32   sNumberCommonSampleRates = 15;
static Float64  sCommonSampleRates[] = {
    8000.0,   11025.0,  12000.0,
    16000.0,  22050.0,  24000.0,
    32000.0,  44100.0,  48000.0,
    64000.0,  88200.0,  96000.0,
    128000.0, 176400.0, 192000.0 };

static bool IsRateCommon(Float64 inRate)
{
    bool theAnswer = false;
    for(UInt32 i = 0; !theAnswer && (i < sNumberCommonSampleRates); i++)
    {
        theAnswer = inRate == sCommonSampleRates[i];
    }
    return theAnswer;
}

int *CoreAudioData::RatesList(AudioDeviceID d)
{
    OSStatus                    err;
    AudioValueRange             *list;
    int                         *finallist;
    UInt32                      listSize;
    UInt32                      nbitems = 0;

    // retrieve size of rate list
    err = AudioDeviceGetPropertyInfo(d, 0, 0,
                                     kAudioDevicePropertyAvailableNominalSampleRates,
                                     &listSize, NULL);
    if (err != noErr)
    {
        Warn(QString("RatesList(): couldn't get data rate list size: [%1]")
             .arg(err));
        return NULL;
    }

    list      = (AudioValueRange *)malloc(listSize);
    if (list == NULL)
    {
        Error("RatesList(): out of memory?");
        return NULL;
    }

    err = AudioDeviceGetProperty(
        d, 0, 0, kAudioDevicePropertyAvailableNominalSampleRates,
        &listSize, list);
    if (err != noErr)
    {
        Warn(QString("RatesList(): couldn't get list: [%1]")
             .arg(err));
        free(list);
        return NULL;
    }

    finallist = (int *)malloc(((listSize / sizeof(AudioValueRange)) + 1)
                              * sizeof(int));
    if (finallist == NULL)
    {
        Error("RatesList(): out of memory?");
        free(list);
        return NULL;
    }

    // iterate through the ranges and add the minimum, maximum, and common rates in between
    UInt32 theFirstIndex = 0, theLastIndex = 0;
    for(UInt32 i = 0; i < listSize / sizeof(AudioValueRange); i++)
    {
        theFirstIndex = theLastIndex;
        // find the index of the first common rate greater than or equal to the minimum
        while((theFirstIndex < sNumberCommonSampleRates) &&  (sCommonSampleRates[theFirstIndex] < list[i].mMinimum))
            theFirstIndex++;

        if (theFirstIndex >= sNumberCommonSampleRates)
            break;

        theLastIndex = theFirstIndex;
        // find the index of the first common rate greater than or equal to the maximum
        while((theLastIndex < sNumberCommonSampleRates) && (sCommonSampleRates[theLastIndex] < list[i].mMaximum))
        {
            finallist[nbitems++] = sCommonSampleRates[theLastIndex];
            theLastIndex++;
        }
        if (IsRateCommon(list[i].mMinimum))
            finallist[nbitems++] = list[i].mMinimum;
        else if (IsRateCommon(list[i].mMaximum))
            finallist[nbitems++] = list[i].mMaximum;
    }
    free(list);

    // Add a terminating ID
    finallist[nbitems] = -1;

    return finallist;
}

bool *CoreAudioData::ChannelsList(AudioDeviceID d, bool passthru)
{
    AudioStreamID               *streams;
    AudioStreamBasicDescription *formats;
    bool                        founddigital = false;
    bool                        *list;

    if ((list = (bool *)malloc((CHANNELS_MAX+1) * sizeof(bool))) == NULL)
        return NULL;

    memset(list, 0, (CHANNELS_MAX+1) * sizeof(bool));

    streams = StreamsList(mDeviceID);
    if (!streams)
    {
        free(list);
        return NULL;
    }

    if (passthru)
    {
        for (int i = 0; streams[i] != kAudioHardwareBadStreamError; i++)
        {
            formats = FormatsList(streams[i]);
            if (!formats)
                continue;

            // Find a stream with a cac3 stream
            for (int j = 0; formats[j].mFormatID != 0; j++)
            {
                if (formats[j].mFormatID == 'IAC3' ||
                    formats[j].mFormatID == kAudioFormat60958AC3)
                {
                    list[formats[j].mChannelsPerFrame] = true;
                    founddigital = true;
                }
            }
            free(formats);
        }
    }

    if (!founddigital)
    {
        for (int i = 0; streams[i] != kAudioHardwareBadStreamError; i++)
        {
            formats = FormatsList(streams[i]);
            if (!formats)
                continue;
            for (int j = 0; formats[j].mFormatID != 0; j++)
                if (formats[j].mChannelsPerFrame <= CHANNELS_MAX)
                    list[formats[j].mChannelsPerFrame] = true;
            free(formats);
        }
    }
    return list;
}

int CoreAudioData::OpenAnalog()
{
    ComponentDescription         desc;
    AudioStreamBasicDescription  DeviceFormat;
    AudioChannelLayout          *layout;
    AudioChannelLayout           new_layout;
    AudioDeviceID                defaultDevice = GetDefaultOutputDevice();

    Debug("OpenAnalog: Entering");

    desc.componentType = kAudioUnitType_Output;
    if (defaultDevice == mDeviceID)
    {
        desc.componentSubType = kAudioUnitSubType_DefaultOutput;
    }
    else
    {
        desc.componentSubType = kAudioUnitSubType_HALOutput;
    }
    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
    desc.componentFlags = 0;
    desc.componentFlagsMask = 0;
    mDigitalInUse = false;

    Component comp = FindNextComponent(NULL, &desc);
    if (comp == NULL)
    {
        Error("OpenAnalog: AudioComponentFindNext failed");
        return false;
    }

    OSErr err = OpenAComponent(comp, &mOutputUnit);
    if (err)
    {
        Error(QString("OpenAnalog: AudioComponentInstanceNew returned %1")
              .arg(err));
        return false;
    }

    // Check if we have IO
    UInt32 hasIO      = 0;
    UInt32 size_hasIO = sizeof(hasIO);
    err = AudioUnitGetProperty(mOutputUnit,
                               kAudioOutputUnitProperty_HasIO,
                               kAudioUnitScope_Output,
                               0,
                               &hasIO, &size_hasIO);
    Debug(QString("OpenAnalog: HasIO (output) = %1").arg(hasIO));
    if (!hasIO)
    {
        UInt32 enableIO = 1;
        err = AudioUnitSetProperty(mOutputUnit,
                                   kAudioOutputUnitProperty_EnableIO,
                                   kAudioUnitScope_Global,
                                   0,
                                   &enableIO, sizeof(enableIO));
        if (err)
        {
            Warn(QString("OpenAnalog: failed enabling IO: %1")
                 .arg(err));
        }
        hasIO = 0;
        err = AudioUnitGetProperty(mOutputUnit,
                                   kAudioOutputUnitProperty_HasIO,
                                   kAudioUnitScope_Output,
                                   0,
                                   &hasIO, &size_hasIO);
        Debug(QString("HasIO = %1").arg(hasIO));
    }

    /*
     * We shouldn't have to do this distinction, however for some unknown reasons
     * assigning device to AudioUnit fail when switching from SPDIF mode
     */
    if (defaultDevice != mDeviceID)
    {
        err = AudioUnitSetProperty(mOutputUnit,
                                   kAudioOutputUnitProperty_CurrentDevice,
                                   kAudioUnitScope_Global,
                                   0,
                                   &mDeviceID, sizeof(mDeviceID));
        if (err)
        { 
            Error(QString("OpenAnalog: Unable to set current device to %1. Error = %2")
                  .arg(mDeviceID)
                  .arg(err));
            return -1;
        }  
    }
    /* Get the current format */
    UInt32 param_size = sizeof(AudioStreamBasicDescription);

    err = AudioUnitGetProperty(mOutputUnit,
                               kAudioUnitProperty_StreamFormat,
                               kAudioUnitScope_Input,
                               0,
                               &DeviceFormat,
                               &param_size );
    if (err)
    {
        Warn(QString("OpenAnalog: Unable to retrieve current stream format: [%1]")
              .arg(err));
    }
    else
    {
        Debug(QString("OpenAnalog: current format is: %1")
              .arg(StreamDescriptionToString(DeviceFormat)));
    }
    /* Get the channel layout of the device side of the unit */
    err = AudioUnitGetPropertyInfo(mOutputUnit,
                                   kAudioDevicePropertyPreferredChannelLayout,
                                   kAudioUnitScope_Output,
                                   0,
                                   &param_size,
                                   NULL);

    if(!err)
    {
        layout = (AudioChannelLayout *) malloc(param_size);

        err = AudioUnitGetProperty(mOutputUnit,
                                   kAudioDevicePropertyPreferredChannelLayout,
                                   kAudioUnitScope_Output,
                                   0,
                                   layout,
                                   &param_size);

        /* We need to "fill out" the ChannelLayout, because there are multiple ways that it can be set */
        if(layout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap)
        {
            /* bitmap defined channellayout */
            err = AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForBitmap,
                                         sizeof(UInt32), &layout->mChannelBitmap,
                                         &param_size,
                                         layout);
            if (err)
                Warn("OpenAnalog: Can't retrieve current channel layout");
        }
        else if(layout->mChannelLayoutTag != kAudioChannelLayoutTag_UseChannelDescriptions )
        {
            /* layouttags defined channellayout */
            err = AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForTag,
                                         sizeof(AudioChannelLayoutTag),
                                         &layout->mChannelLayoutTag,
                                         &param_size,
                                         layout);
            if (err)
                Warn("OpenAnalog: Can't retrieve current channel layout");
        }

        Debug(QString("OpenAnalog: Layout of AUHAL has %1 channels")
              .arg(layout->mNumberChannelDescriptions));

        int channels_found = 0;
        for(UInt32 i = 0; i < layout->mNumberChannelDescriptions; i++)
        {
            Debug(QString("OpenAnalog: this is channel: %1")
                  .arg(layout->mChannelDescriptions[i].mChannelLabel));

            switch( layout->mChannelDescriptions[i].mChannelLabel)
            {
                case kAudioChannelLabel_Left:
                case kAudioChannelLabel_Right:
                case kAudioChannelLabel_Center:
                case kAudioChannelLabel_LFEScreen:
                case kAudioChannelLabel_LeftSurround:
                case kAudioChannelLabel_RightSurround:
                case kAudioChannelLabel_RearSurroundLeft:
                case kAudioChannelLabel_RearSurroundRight:
                case kAudioChannelLabel_CenterSurround:
                    channels_found++;
                    break;
                default:
                    Debug(QString("unrecognized channel form provided by driver: %1")
                          .arg(layout->mChannelDescriptions[i].mChannelLabel));
            }
        }
        if(channels_found == 0)
        {
            Warn("Audio device is not configured. "
                 "You should configure your speaker layout with "
                 "the \"Audio Midi Setup\" utility in /Applications/"
                 "Utilities.");
        }
        free(layout);
    }
    else
    {
        Warn("this driver does not support kAudioDevicePropertyPreferredChannelLayout.");
    }

    memset (&new_layout, 0, sizeof(new_layout));
    switch(mCA->channels)
    {
        case 1:
            new_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
            break;
        case 2:
            new_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
            break;
        case 6:
            //  3F2-LFE        L   R   C    LFE  LS   RS
            new_layout.mChannelLayoutTag = kAudioChannelLayoutTag_AudioUnit_5_1;
        case 8:
            // We need
            // 3F4-LFE        L   R   C    LFE  Rls  Rrs  LS   RS
            // but doesn't exist, so we'll swap channels later
            new_layout.mChannelLayoutTag = kAudioChannelLayoutTag_MPEG_7_1_A; // L R C LFE Ls Rs Lc Rc
            break;
    }
    /* Set new_layout as the layout */
    err = AudioUnitSetProperty(mOutputUnit,
                               kAudioUnitProperty_AudioChannelLayout,
                               kAudioUnitScope_Input,
                               0,
                               &new_layout, sizeof(new_layout));
    if (err)
    {
        Warn(QString("OpenAnalog: couldn't set channels layout [%1]")
             .arg(err));
    }

    if(new_layout.mNumberChannelDescriptions > 0)
        free(new_layout.mChannelDescriptions);

    // Set up the audio output unit
    int formatFlags;
    switch (mCA->output_format)
    {
        case FORMAT_S16:
            formatFlags = kLinearPCMFormatFlagIsSignedInteger;
            break;
        case FORMAT_FLT:
            formatFlags = kLinearPCMFormatFlagIsFloat;
            break;
        default:
            formatFlags = kLinearPCMFormatFlagIsSignedInteger;
            break;
    }

    AudioStreamBasicDescription conv_in_desc;
    memset(&conv_in_desc, 0, sizeof(AudioStreamBasicDescription));
    conv_in_desc.mSampleRate       = mCA->samplerate;
    conv_in_desc.mFormatID         = kAudioFormatLinearPCM;
    conv_in_desc.mFormatFlags      = formatFlags |
        kAudioFormatFlagsNativeEndian |
        kLinearPCMFormatFlagIsPacked;
    conv_in_desc.mBytesPerPacket   = mCA->output_bytes_per_frame;
    // This seems inefficient, does it hurt if we increase this?
    conv_in_desc.mFramesPerPacket  = 1;
    conv_in_desc.mBytesPerFrame    = mCA->output_bytes_per_frame;
    conv_in_desc.mChannelsPerFrame = mCA->channels;
    conv_in_desc.mBitsPerChannel   =
        AudioOutputSettings::FormatToBits(mCA->output_format);

    /* Set AudioUnit input format */
    err = AudioUnitSetProperty(mOutputUnit,
                               kAudioUnitProperty_StreamFormat,
                               kAudioUnitScope_Input,
                               0,
                               &conv_in_desc,
                               sizeof(AudioStreamBasicDescription));
    if (err)
    {
        Error(QString("OpenAnalog: AudioUnitSetProperty returned [%1]")
              .arg(err));
        return false;
    }
    Debug(QString("OpenAnalog: set format as %1")
          .arg(StreamDescriptionToString(conv_in_desc)));
    /* Retrieve actual format */
    err = AudioUnitGetProperty(mOutputUnit,
                               kAudioUnitProperty_StreamFormat,
                               kAudioUnitScope_Input,
                               0,
                               &DeviceFormat,
                               &param_size);

    Debug(QString("OpenAnalog: the actual set AU format is %1")
          .arg(StreamDescriptionToString(DeviceFormat)));

    // Attach callback to default output
    AURenderCallbackStruct input;
    input.inputProc = RenderCallbackAnalog;
    input.inputProcRefCon = this;

    err = AudioUnitSetProperty(mOutputUnit,
                               kAudioUnitProperty_SetRenderCallback,
                               kAudioUnitScope_Input,
                               0, &input, sizeof(input));
    if (err)
    {
        Error(QString("OpenAnalog: AudioUnitSetProperty (callback) returned [%1]")
              .arg(err));
        return false;
    }
    mIoProc = true;

    // We're all set up - start the audio output unit
    ComponentResult res = AudioUnitInitialize(mOutputUnit);
    if (res)
    {
        Error(QString("OpenAnalog: AudioUnitInitialize error: [%1]")
              .arg(res));
        return false;
    }
    mInitialized = true;

    err = AudioOutputUnitStart(mOutputUnit);
    if (err)
    {
        Error(QString("OpenAnalog: AudioOutputUnitStart error: [%1]")
              .arg(err));
        return false;
    }
    mStarted = true;
    return true;
}

void CoreAudioData::CloseAnalog()
{
    OSStatus err;

    Debug(QString("CloseAnalog: Entering: %1")
          .arg((long)mOutputUnit));
    if (mOutputUnit)
    {
        if (mStarted)
        {
            err = AudioOutputUnitStop(mOutputUnit);
            Debug(QString("CloseAnalog: AudioOutputUnitStop %1")
                  .arg(err));
        }
        if (mInitialized)
        {
            err = AudioUnitUninitialize(mOutputUnit);
            Debug(QString("CloseAnalog: AudioUnitUninitialize %1")
                  .arg(err));
        }
        err = CloseComponent(mOutputUnit);
        Debug(QString("CloseAnalog: CloseComponent %1")
              .arg(err));
        mOutputUnit = NULL;
    }
    mIoProc = false;
    mInitialized = false;
    mStarted = false;
    mWasDigital = false;
}

bool CoreAudioData::OpenSPDIF()
{
    OSStatus       err;
    AudioStreamID  *streams;
    AudioStreamBasicDescription outputFormat = {0};

    Debug("OpenSPDIF: Entering");

    streams = StreamsList(mDeviceID);
    if (!streams)
    {
        Warn("OpenSPDIF: Couldn't retrieve list of streams");
        return false;
    }

    for (int i = 0; streams[i] != kAudioHardwareBadStreamError; ++i)
    {
        AudioStreamBasicDescription *formats = FormatsList(streams[i]);
        if (!formats)
            continue;

        // Find a stream with a cac3 stream
        for (int j = 0; formats[j].mFormatID != 0; j++)
        {
            Debug(QString("OpenSPDIF: Considering Physical Format: %1")
                  .arg(StreamDescriptionToString(formats[j])));
            if ((formats[j].mFormatID == 'IAC3' ||
                 formats[j].mFormatID == kAudioFormat60958AC3) &&
                formats[j].mSampleRate == mCA->samplerate)
            {
                Debug("OpenSPDIF: Found digital format");
                mStreamIndex  = i;
                mStreamID     = streams[i];
                outputFormat  = formats[j];
                break;
            }
        }
        free(formats);

        if (outputFormat.mFormatID)
            break;
    }
    free(streams);

    if (!outputFormat.mFormatID)
    {
        Error(QString("OpenSPDIF: Couldn't find suitable output"));
        return false;
    }

    if (mRevertFormat == false)
    {
        // Retrieve the original format of this stream first
        // if not done so already
        UInt32 paramSize = sizeof(mFormatOrig);
        err = AudioStreamGetProperty(mStreamID, 0,
                                     kAudioStreamPropertyPhysicalFormat,
                                     &paramSize, &mFormatOrig);
        if (err != noErr)
        {
            Warn(QString("OpenSPDIF - could not retrieve the original streamformat: [%1]")
                 .arg(OSS_STATUS(err)));
        }
        else
        {
            mRevertFormat = true;
        }
    }

    mDigitalInUse = true;

    SetAutoHogMode(false);
    bool autoHog = GetAutoHogMode();
    if (!autoHog)
    {
        // Hog the device
        SetHogStatus(true);
        // Set mixable to false if we are allowed to
        SetMixingSupport(false);
    }

    mFormatNew = outputFormat;
    if (!AudioStreamChangeFormat(mStreamID, mFormatNew))
    {
        return false;
    }
    mBytesPerPacket = mFormatNew.mBytesPerPacket;

    // Add IOProc callback
    err = AudioDeviceAddIOProc(mDeviceID,
                               (AudioDeviceIOProc)RenderCallbackSPDIF,
                               (void *)this);
    if (err != noErr)
    {
        Error(QString("OpenSPDIF: AudioDeviceAddIOProc failed: [%1]")
              .arg(OSS_STATUS(err)));
        return false;
    }
    mIoProc = true;

    // Start device
    err = AudioDeviceStart(mDeviceID, (AudioDeviceIOProc)RenderCallbackSPDIF);
    if (err != noErr)
    {
        Error(QString("OpenSPDIF: AudioDeviceStart failed: [%1]")
              .arg(OSS_STATUS(err)));
        return false;
    }
    mStarted = true;
    return true;
}

void CoreAudioData::CloseSPDIF()
{
    OSStatus  err;

    Debug(QString("CloseSPDIF: Entering [%1]").arg(mDigitalInUse));;
    if (!mDigitalInUse)
        return;

    // Stop device
    if (mStarted)
    {
        err = AudioDeviceStop(mDeviceID, (AudioDeviceIOProc)RenderCallbackSPDIF);
        if (err != noErr)
            Error(QString("CloseSPDIF: AudioDeviceStop failed: [%1]")
                  .arg(OSS_STATUS(err)));
        mStarted = false;
    }

    // Remove IOProc callback
    if (mIoProc)
    {
        err = AudioDeviceRemoveIOProc(mDeviceID,
                                      (AudioDeviceIOProc)RenderCallbackSPDIF);
        if (err != noErr)
            Error(QString("CloseSPDIF: AudioDeviceRemoveIOProc failed: [%1]")
                  .arg(OSS_STATUS(err)));
        mIoProc = false;
    }

    if (mRevertFormat)
    {
        AudioStreamChangeFormat(mStreamID, mFormatOrig);
        mRevertFormat = false;
    }

    SetHogStatus(false);
    if (mMixerRestore > -1) // We changed the mixer status
        SetMixingSupport((mMixerRestore ? true : false));
    AudioHardwareUnload();
    mMixerRestore = -1;
    mBytesPerPacket = -1;
    mStreamIndex = -1;
    mWasDigital = true;
}

int CoreAudioData::AudioStreamChangeFormat(AudioStreamID               s,
                                           AudioStreamBasicDescription format)
{
    Debug(QString("AudioStreamChangeFormat: %1 -> %2")
          .arg(s)
          .arg(StreamDescriptionToString(format)));

    OSStatus err = AudioStreamSetProperty(s, 0, 0,
                                          kAudioStreamPropertyPhysicalFormat,
                                          sizeof(format), &format);
    if (err != noErr)
    {
        Error(QString("AudioStreamChangeFormat couldn't set stream format: [%1]")
              .arg(OSS_STATUS(err)));
        return false;
    }
    return true;
}

bool CoreAudioData::FindAC3Stream()
{
    bool           foundAC3Stream = false;
    AudioStreamID  *streams;


    // Get a list of all the streams on this device
    streams = StreamsList(mDeviceID);
    if (!streams)
        return false;

    for (int i = 0; !foundAC3Stream &&
         streams[i] != kAudioHardwareBadStreamError; ++i)
    {
        AudioStreamBasicDescription *formats = FormatsList(streams[i]);
        if (!formats)
            continue;

        // Find a stream with a cac3 stream
        for (int j = 0; formats[j].mFormatID != 0; j++)
            if (formats[j].mFormatID == 'IAC3' ||
                formats[j].mFormatID == kAudioFormat60958AC3)
            {
                Debug("FindAC3Stream: found digital format");
                foundAC3Stream = true;
                break;
            }

        free(formats);
    }
    free(streams);

    return foundAC3Stream;
}

void CoreAudioData::ResetAudioDevices()
{
    AudioDeviceID  *devices;
    int            numDevices;
    UInt32         size;


    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &size, NULL);
    devices    = (AudioDeviceID*)malloc(size);
    if (!devices)
    {
        Error("ResetAudioDevices: out of memory?");
        return;
    }
    numDevices = size / sizeof(AudioDeviceID);
    AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &size, devices);

    for (int i = 0; i < numDevices; i++)
    {
        AudioStreamID  *streams;

        streams = StreamsList(devices[i]);
        if (!streams)
            continue;
        for (int j = 0; streams[j] != kAudioHardwareBadStreamError; j++)
            ResetStream(streams[j]);

        free(streams);
    }
    free(devices);
}

void CoreAudioData::ResetStream(AudioStreamID s)
{
    AudioStreamBasicDescription  currentFormat;
    OSStatus                     err;
    UInt32                       paramSize;

    // Find the streams current physical format
    paramSize = sizeof(currentFormat);
    AudioStreamGetProperty(s, 0, kAudioStreamPropertyPhysicalFormat,
                           &paramSize, &currentFormat);

    // If it's currently AC-3/SPDIF then reset it to some mixable format
    if (currentFormat.mFormatID == 'IAC3' ||
        currentFormat.mFormatID == kAudioFormat60958AC3)
    {
        AudioStreamBasicDescription *formats    = FormatsList(s);
        bool                        streamReset = false;


        if (!formats)
            return;

        for (int i = 0; !streamReset && formats[i].mFormatID != 0; i++)
            if (formats[i].mFormatID == kAudioFormatLinearPCM)
            {
                err = AudioStreamSetProperty(s, NULL, 0,
                                             kAudioStreamPropertyPhysicalFormat,
                                             sizeof(formats[i]), &(formats[i]));
                if (err != noErr)
                {
                    Warn(QString("ResetStream: could not set physical format: [%1]")
                         .arg(OSS_STATUS(err)));
                    continue;
                }
                else
                {
                    streamReset = true;
                    sleep(1);   // For the change to take effect
                }
            }

        free(formats);
    }
}

QMap<QString, QString> *AudioOutputCA::GetDevices(const char *type)
{
    QMap<QString, QString> *devs = new QMap<QString, QString>();

    // Obtain a list of all available audio devices
    UInt32 size = 0;
    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &size, NULL);
    UInt32 deviceCount = size / sizeof(AudioDeviceID);
    AudioDeviceID* pDevices = new AudioDeviceID[deviceCount];
    OSStatus err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices,
                                            &size, pDevices);
    if (err)
        VBAUDIO(QString("AudioOutputCA::GetDevices: Unable to retrieve the list of "
                        "available devices. Error [%1]")
                .arg(err));
    else
    {
        VBAUDIO(QString("GetDevices: Number of devices: %1").arg(deviceCount));

        for (UInt32 dev = 0; dev < deviceCount; dev++)
        {
            CoreAudioData device(NULL, pDevices[dev]);
            if (device.GetTotalOutputChannels() == 0)
                continue;
            QString *name = device.GetName();
            if (name)
            {
                devs->insert(*name, QString());
                delete name;
            }
        }
    }
    return devs;
}

---