/*

* Copyright (c) 2010 Ixonos Plc.

* All rights reserved.

* This component and the accompanying materials are made available

* under the terms of the "Eclipse Public License v1.0"

* which accompanies this distribution, and is available

* at the URL "http://www.eclipse.org/legal/epl-v10.html".

*

* Initial Contributors:

* Nokia Corporation - Initial contribution

*

* Contributors:

* Ixonos Plc

*

* Description:  

*

*/

#include "ProcDecoder.h"

#include "audconstants.h"

#include    <MmfDatabuffer.h>

#include    <mmfcontrollerpluginresolver.h>

#include    <mmf/plugin/mmfCodecImplementationUIDs.hrh>

#include    <MMFCodec.h>

// CONSTANTS

// MACROS

// Debug print macro

#if defined _DEBUG 

#include <e32svr.h>

#define PRINT(x) RDebug::Print x;

#else

#define PRINT(x)

#endif

CProcDecoder* CProcDecoder::NewL()

    {

    CProcDecoder* self = NewLC();

    CleanupStack::Pop(self);

    return self;

    }

CProcDecoder* CProcDecoder::NewLC()

    {

    CProcDecoder* self = new (ELeave) CProcDecoder();

    CleanupStack::PushL(self);

    self->ConstructL();

    return self;

    }

CProcDecoder::~CProcDecoder()

    {

    delete iSourceInputBuffer;

    // Input buffer for encoder (alternative output buffer for decoder).

    delete  iDestInputBuffer;

    // Codec used in decoding input audio to PCM16

    delete iSourceCodec;

    delete iSampleRateChannelBuffer;

    delete iRateConverter;

    }

TBool CProcDecoder::InitL(TAudFileProperties aProperties, TInt aTargetSamplingRate, TChannelMode aChannelMode)

    {

    PRINT((_L("CProcDecoder::InitL() in")));

    if (aTargetSamplingRate == 0)

        {

        // we are just adding a clip now, check if the source codec is available

        iProperties = aProperties;    

        SetSourceCodecL();

        return ETrue;

        }

    iDecimFactor = 0; // by default

    iDoDecoding = ETrue;

    if (aProperties.iAudioType == EAudWAV)

        {

        // if input is PCM, no decoding is required, but possibly sample rate conversion

        iDoDecoding = EFalse;

        }

    // set output params

    iToSampleRate = aTargetSamplingRate;

    iToChannels = 1;

    if (aChannelMode == EAudStereo)

        {

        iToChannels = 2;    

        }

    //set input params

    iProperties = aProperties;

    iFromSampleRate = iProperties.iSamplingRate;

    iFromChannels = 1;

    if (aProperties.iChannelMode == EAudStereo)

        {

        iFromChannels = 2;    

        }

    iDoSampleRateChannelConversion = EFalse;

    if (iFromSampleRate != iToSampleRate || iProperties.iChannelMode != aChannelMode)

        {

        // sample rate or channel conversion is needed

        iDoSampleRateChannelConversion = ETrue;

        }

    PRINT((_L("CProcDecoder::PrepareConverterL() in")));

    TInt destInputBufferSize = 0;

    TInt sourceInputBufferSize = 0;

    // buffer sizes for input and output

    if ( iProperties.iAudioType == EAudAMR )

        {

        destInputBufferSize = KAedSizeAMRBuffer;

        sourceInputBufferSize = KAedMaxAMRFrameLength;

        }

    else if ( iProperties.iAudioType == EAudAMRWB )

        {

        destInputBufferSize = KAedSizeAWBBuffer;

        sourceInputBufferSize = KAedMaxAWBFrameLength;

        }

    else if (iProperties.iAudioType == EAudAAC_MPEG4 &&

             iProperties.iAudioTypeExtension == EAudExtensionTypeNoExtension)

        {

        if ( aProperties.iChannelMode == EAudSingleChannel )

            {

            destInputBufferSize = KAedSizeAACBuffer;

            sourceInputBufferSize = KAedMaxAACFrameLengthPerChannel;

            }

        else

            {

            destInputBufferSize = KAedSizeAACStereoBuffer;

            sourceInputBufferSize = 2* KAedMaxAACFrameLengthPerChannel;

            }

        }

    else if (iProperties.iAudioType == EAudAAC_MPEG4 &&

             iProperties.iAudioTypeExtension != EAudExtensionTypeNoExtension)

        {

        if ( iProperties.iAudioTypeExtension == EAudExtensionTypeEnhancedAACPlusParametricStereo )

            {

            // output is parametric stereo (mono channel, but decoded to stereo)

            destInputBufferSize = KAedSizeAACStereoBuffer*2; // 2 for eAAC+

            sourceInputBufferSize = KAedMaxAACFrameLengthPerChannel;

            }

        else

            {

            // output is normal stereo

            destInputBufferSize = KAedSizeAACBuffer*2; // 2 for eAAC+

            sourceInputBufferSize = 2* KAedMaxAACFrameLengthPerChannel;

            }

        }

    else if ( iProperties.iAudioType == EAudMP3 )

        {

        // mp3 can be directly decoded to mono if necessary

        if (iToChannels == 1)

            {

            iFromChannels = 1;

            }

        // check if decimation would be useful    

        if (iFromSampleRate == iToSampleRate * 4)

            {

            iFromSampleRate /= 4;

            iDecimFactor = 4;

            }

        if (iFromSampleRate == iToSampleRate * 2)

            {

            iFromSampleRate /= 2;

            iDecimFactor = 2;

            }

        if ((iToSampleRate == 8000) && ((iFromSampleRate == 48000) || (iFromSampleRate == 44100)))

            {

            iFromSampleRate /= 2;

            iDecimFactor = 2;

            }        

        const TReal KMP3FrameDurationSec = 0.026;

        destInputBufferSize =  (iFromSampleRate * KMP3FrameDurationSec + 1);

        destInputBufferSize *= 2; //16 bit samples

        if (iFromChannels == 2)

            {

            destInputBufferSize *= 2; // stereo

            }

        sourceInputBufferSize = KAedMaxMP3FrameLength;

        }

    else if (iProperties.iAudioType == EAudWAV)

        {

        // frame duration in WAV case is always 20 ms, that is 50 frames per sec

        const TInt KNumberOfFramesInSecond = 50;

        destInputBufferSize = iFromSampleRate / KNumberOfFramesInSecond;    // Number of samples

        destInputBufferSize *= iFromChannels;

        if ((destInputBufferSize % 2) != 0) destInputBufferSize++;          // Must be even

        destInputBufferSize *= 2;                                           // 16-bit samples

        sourceInputBufferSize = iProperties.iFrameLen;

        TInt sourceBufferSizeAfterExpansion = sourceInputBufferSize;

        if (iProperties.iNumberOfBitsPerSample == 8)

            {

            // 8-bit samples are expanded to 16-bit so twice as big buffer is needed

            sourceBufferSizeAfterExpansion *= 2;

            }

        if (sourceBufferSizeAfterExpansion > destInputBufferSize)

            {

            // Make sure there's no overflow

            destInputBufferSize = sourceBufferSizeAfterExpansion;

            }

        }

    PRINT((_L("CProcDecoder::InitL() source size: %d, dest size %d"), sourceInputBufferSize, destInputBufferSize));

    if ( iSourceInputBuffer )

        {

        delete iSourceInputBuffer;

        iSourceInputBuffer = NULL;

        }

    // create buffer for input data

    iSourceInputBuffer = CMMFDataBuffer::NewL(sourceInputBufferSize);

    if ( iDestInputBuffer )

        {

        delete iDestInputBuffer;

        iDestInputBuffer = NULL;

        }

    TInt errC = KErrNone;

    if (iDoDecoding)

        {

        SetSourceCodecL();

        // create buffer for output data if necessary

        iDestInputBuffer = CMMFDataBuffer::NewL( destInputBufferSize);

        }

    TInt err = KErrNone;

    if (iProperties.iAudioType == EAudAAC_MPEG4)

        {

        // configure AAC plus decoder, used also for AAC

        TRAP( err, ConfigureAACPlusDecoderL());

        }

    else if ( iProperties.iAudioType == EAudMP3 )

        {

        // configure mp3 decoder

        TRAP( err, ConfigureMP3DecoderL());

        }

    if (err != KErrNone || errC != KErrNone)

        {

        User::Leave(KErrNotSupported);

        }

    if ( iDoSampleRateChannelConversion )

        {

        if (iRateConverter)

            {

            delete iRateConverter;

            iRateConverter = NULL;

            }

        iRateConverter = CRateConverter::NewL(iFromSampleRate, iToSampleRate, iFromChannels, iToChannels);

        if( !iRateConverter->InitL(destInputBufferSize / (2 * iFromChannels)) ) // 16-bit samples

            {

            User::Leave(KErrNotSupported);

            }

        TInt sampleRateBufferSize = iRateConverter->GetOutputBufferSize() * 2 * iToChannels;    // 16-bit samples

        PRINT((_L("CProcDecoder::InitL() sample rate buffer size %d"), sampleRateBufferSize));

        if ( iSampleRateChannelBuffer )

            {

            delete iSampleRateChannelBuffer;

            iSampleRateChannelBuffer = NULL;

            }

        // Buffer for sample rate conversion output

        iSampleRateChannelBuffer = CMMFDataBuffer::NewL(sampleRateBufferSize);

        }

    iReady = ETrue;

    PRINT((_L("CProcDecoder::InitL() out")));

    return ETrue;

    }

TBool CProcDecoder::FillDecBufferL(const HBufC8* aEncFrame, HBufC8*& aDecBuffer)

    {

    PRINT((_L("CProcDecoder::FillDecBufferL() in")));

    iDecBuffer = 0;

    if (iProperties.iAudioType == EAudWAV && !iDoSampleRateChannelConversion)

        {

        // if we don't have to do anything for input data ->

        aDecBuffer = HBufC8::NewL(aEncFrame->Size());

        aDecBuffer->Des().Append(*aEncFrame);

        PRINT((_L("CProcDecoder::FillDecBufferL() out from Wav branch with ETrue")));

        return ETrue;

        }

    if (!iReady)

        {

        User::Leave(KErrNotReady);

        }

    if ( aEncFrame == 0 || !aEncFrame->Length() )

        {

        // no data in input buffer

        PRINT((_L("CProcDecoder::FillDecBufferL() no input data, out with EFalse")));

        return EFalse;

        }

    if ( (TInt)(aEncFrame->Length() + iSourceInputBuffer->Position() ) > iSourceInputBuffer->Data().MaxLength() )

        {

        ReAllocBufferL( iSourceInputBuffer, aEncFrame->Length() + iSourceInputBuffer->Position() );

        }

    iSourceInputBuffer->Data().SetLength( 0 );

    iSourceInputBuffer->SetPosition( 0 );

    iSourceInputBuffer->Data().Append( *aEncFrame );

    iSourceInputBuffer->Data().SetLength( aEncFrame->Length() );

    PRINT((_L("CProcDecoder::FillDecBufferL(), iSourceInputBuffer length = %d"),aEncFrame->Length()));

    if (iDoDecoding)

        {

        iDestInputBuffer->Data().SetLength(0);

        iDestInputBuffer->SetPosition(0); 

        }

    if ( iDoSampleRateChannelConversion )

        {

        iSampleRateChannelBuffer->Data().SetLength(0);

        iSampleRateChannelBuffer->SetPosition(0);

        }

    FeedCodecL( iSourceCodec, iSourceInputBuffer, iDestInputBuffer);

    if ( aDecBuffer )

        {

        // in case of EDstNotFilled from decoder, you may end up looping and to avoid memory leaks, you need to 

        // delete the previous allocated buffer. Alternative might be to not allocate a new one in this case.

        delete aDecBuffer;

        aDecBuffer = NULL;

        }

    aDecBuffer = iDecBuffer;    

    if (iDoDecoding)

        {

        iDestInputBuffer->Data().SetLength(0);

        iDestInputBuffer->SetPosition(0); 

        }

    if (iDecBuffer != 0 && iDecBuffer->Size() > 0)

        {

        PRINT((_L("CProcDecoder::FillDecBufferL() out with ETrue")));

        return ETrue;

        }

    PRINT((_L("CProcDecoder::FillDecBufferL() out with EFalse")));

    return EFalse;

    }

void CProcDecoder::ConstructL()

    {

    }

CProcDecoder::CProcDecoder()

    {

    }

void CProcDecoder::ConfigureAACPlusDecoderL()

    {

    RArray<TInt> config;

    if (iProperties.iAudioTypeExtension == EAudExtensionTypeEnhancedAACPlusParametricStereo)

        {

        // for sample rate and channel converter, the output from parametric stereo

        // is stereo, only the AAC part is incoded as mono

        iFromChannels = 2;

        }

    config.Append( iFromSampleRate);

    config.Append( iFromChannels );

    if (iProperties.iAACObjectType ==  EAudAACObjectTypeLC)

        {

        config.Append( 1 );     // {1 - LC, 3 - LTP}    

        }

    else if (iProperties.iAACObjectType ==  EAudAACObjectTypeLTP)

        {

        config.Append( 3 );     // {1 - LC, 3 - LTP}    

        }

    else

        {

        User::Leave(KErrNotSupported);

        }

    config.Append( 8192 );  //Size of PCM Samples generated by decoder

    config.Append( 1024 );  //Number of PCM Samples generated by decoder per frame

    config.Append( iFromSampleRate);        //Sampling freq of AAC Code decoder

    config.Append( 0 );     // not used ??

    config.Append( 0 );     // down sampled mode

    config.Append( 16 );    // Sample resolution: 16-bit resolution

    // NOTE!: for some reason, the sample rate of the output from AACPlus decoder used

    // to be a half of the proper output sampling rate. 

    // eAAC+ data consists of AAC band + enhancement band. The enhancement band is not included

    // in the rate indicated in the header (iFromSampleRate)

    // There was a workaround for the 

    // earlier problem not to increase the sampling rate.

    // Now, with 2006 releases it seems to work, and the sampling rate need to be doubled

    // for the sampling rate converter.

    if (iProperties.iAudioTypeExtension == EAudExtensionTypeEnhancedAACPlus)

        {

        iFromSampleRate *= 2; // for sample rate converter ->

        config.Append(iFromSampleRate);// Output sampling frequency

        config.Append( 5 ); //Type of extended object (5=SBR/HE AAC profile; 6=PS is present)

        }

    else if (iProperties.iAudioTypeExtension == EAudExtensionTypeEnhancedAACPlusParametricStereo)

        {

        iFromSampleRate *= 2; // for sample rate converter ->

        config.Append(iFromSampleRate);// Output sampling frequency

        config.Append( 6 ); //Type of extended object (5=SBR/HE AAC profile; 6=PS is present)

        }

    else

        {

        // AAC, in&out samplerates are equal, and extended object type is 0

        config.Append(iFromSampleRate);// Output sampling frequency

        config.Append( 0 );

        }

    TUid uid ={KUidMmfCodecAudioSettings}; // Use Uid reserved for codec configurations

    TRAPD( err, iSourceCodec->ConfigureL( uid, reinterpret_cast<TDesC8&>(config)));

    if ( err != KErrNone )

        {