omxacmp3_synthpqmf_s32_s16.c

The OpenMAX DL (Development Layer) APIs contain a comprehensive set of audio, video, signal processi

/**
 * 
 * File Name:  omxACMP3_SynthPQMF_S32_S16.c
 * OpenMAX DL: v1.0.2
 * Revision:   10586
 * Date:       Wednesday, March 5, 2008
 * 
 * (c) Copyright 2007-2008 ARM Limited. All Rights Reserved.
 * 
 * 
 * Description:
 * This function is used to do synthesis PQMF filtering
 */

#include <math.h>
#include "omxtypes.h"
#include "armOMX.h"
#include "omxAC.h"
#include "armCOMM_Bitstream.h"
#include "armCOMM.h"
#include "armACMP3_Tables.h"
#include "armACMP3_CommonTables.h"
#include "armAC.h"

#define ARM_MP3_VBUFFER_SZ_MASK (0x1ff)

/**
 * Function:  omxACMP3_SynthPQMF_S32_S16   (3.1.3.4.2)
 *
 * Description:
 * Stage 2 of the hybrid synthesis filter bank; a critically-sampled 
 * 32-channel PQMF synthesis bank that generates 32 time-domain output samples 
 * for each 32-sample input block of IMDCT outputs. For each input block, the 
 * PQMF generates an output sequence of 16-bit signed little-endian PCM 
 * samples in the vector pointed to by pDstAudioOut. If mode equals 2, the 
 * left and right channel output samples are interleaved (i.e., LRLRLR), such 
 * that the left channel data is organized as follows: pDstAudioOut [2*i], i=0 
 * to 31. If mode equals 1, then the left and right channel outputs are not 
 * interleaved. A workspace buffer of size 512 x Number of Channels must be 
 * preallocated (pVBuffer). This buffer is referenced by the pointer pVBuffer, 
 * and its elements should be initialized to zero prior to the first call. 
 * During subsequent calls, the pVBuffer input for the current call should 
 * contain the pVbuffer output generated by the previous call. The state 
 * variable pVPosition should be initialized to zero and preserved between 
 * calls. The values contained in pVBuffer or pVPosition should be modified 
 * only during decoder reset, and the reset values should always be zero. 
 * Reference [ISO13818-3], sub-clause 2.5.3.3.2 
 *
 * Input Arguments:
 *   
 *   pSrcY - pointer to the block of 32 IMDCT sub-band input samples, in 
 *            Q7.24 format 
 *   pVBuffer - pointer to the input workspace buffer containing Q7.24 data. 
 *            The elements of this buffer should be initialized to zero during 
 *            decoder reset. During decoder operation, the values contained in 
 *            this buffer should be modified only by the PQMF function. 
 *   pVPosition - pointer to the internal workspace index; should be 
 *            initialized to zero during decoder reset. During decoder 
 *            operation, the value of this index should be preserved between 
 *            PQMF calls and should be modified only by the function. 
 *   mode - flag that indicates whether or not the PCM audio output channels 
 *            should be interleaved 1 - not interleaved 2 - interleaved 
 *
 * Output Arguments:
 *   
 *   pDstAudioOut - pointer to a block of 32 reconstructed PCM output samples 
 *            in 16-bit signed Q0.15 format (little-endian); left and right 
 *            channels are interleaved according to the mode flag. This should 
 *            be aligned on a 4-byte boundary 
 *   pVBuffer - pointer to the updated internal workspace buffer containing 
 *            Q7.24 data; see usage notes under input argument discussion 
 *   pVPosition - pointer to the updated internal workspace index; see usage 
 *            notes under input argument discussion 
 *
 * Return Value:
 *    
 *    OMX_Sts_NoErr - No errors 
 *    OMX_Sts_BadArgErr - bad arguments detected; one or more of the 
 *              following conditions is true: mode<1 or mode>2 
 *    -   *pVPosition is outside the range [0, 15] at least one of the 
 *              following pointers is NULL: pSrcY, pDstAudioOut, pVBuffer, 
 *              and/or pVPosition. 
 *
 */
OMXResult omxACMP3_SynthPQMF_S32_S16(
     OMX_S32 *pSrcY,
     OMX_S16 *pDstAudioOut,
     OMX_S32 *pVBuffer,
     OMX_INT *pVPosition,
     OMX_INT mode
)
{
    OMX_F64     acc, Angle;
    OMX_F64     VPrime [ARM_MP3_VBUFFER_SZ], Src [ARM_MP3_PQMF_SZ];
    OMX_F64     U [ARM_MP3_VBUFFER_SZ];
    OMX_S32     i, j, k, J;

    /* Arguments check */
    armRetArgErrIf(pSrcY == NULL, OMX_Sts_BadArgErr)
    armRetArgErrIf(pDstAudioOut == NULL, OMX_Sts_BadArgErr)
    armRetArgErrIf(pVBuffer == NULL, OMX_Sts_BadArgErr)
    armRetArgErrIf(pVPosition == NULL, OMX_Sts_BadArgErr)
    armRetArgErrIf(mode < 1, OMX_Sts_BadArgErr)
    armRetArgErrIf(mode > 2, OMX_Sts_BadArgErr)

    /* Validate Arguments */
    armRetArgErrIf(*pVPosition < 0, OMX_Sts_BadArgErr)
    armRetArgErrIf(*pVPosition > 15, OMX_Sts_BadArgErr)

    /* Get input */
    for (i = 0; i < ARM_MP3_PQMF_SZ; i++)
    {
        /* Convert Q7.24 to float */
        Src [i] = ((OMX_F64)pSrcY [i]) / (1 << 24);
    }

    for (i = 0; i < ARM_MP3_VBUFFER_SZ; i++)
    {
        /* Convert Q7.24 to float */
        VPrime[i] = ((OMX_F64)pVBuffer [i]) / (1 << 24);
    }

    /* 
     *  Following equation given in MP3 standard (ISO/IEC 11172-3) is modified 
     *  to use a 32 point DCT and symmetry.
     *
     *                           31             
     *              V[j*64+i] = sum  N[i][k] * S[k]
     *                           k=0                   
     *                                  for 0 =< i =< 63, 0 =< j =< 15
     *                          where N[i][k] = cos((16+i)*(2k+1)*PI/64)
     *                                   S[k] = 32 sub-band samples
     * 
     *  The implementation uses 32 point DCT followed by copy as given below.
     *  This change is needed because OMX Spec uses a circular V buffer of size
     *  512, and pointer to the current update position. 32 DCT outputs for the 
     *  current position will be stored after the function call. 
     *                           31             
     *             V' [j*32+i] = sum  S[k] * cos(i*(2k + 1)*PI/64)
     *                           k=0                   
     *                  where 0 =< k =< 31, 0 =< i =< 31, 0 =< j =< 15
     *             
     *             V[j*64+i+0]  =  V'[j*32+i+16]
     *             V[j*64+i+17] = -V'[j*32+31-i]
     *             V[j*64+i+32] = -V'[j*32+16-i]
     *             V[j*64+i+48] = -V'[j*32+i]
     *             V[j*64+16] = 0
     *                  where 0 =< i =< 15, 0 =< j =< 15
     */   
     

    /* 
     *  Variables used for implementing:
     *
     *  V        -> ISO defined buffer of size 1024, each call will update 64 values
     *  VPrime   -> V'. Contains 32 point DCT values. V can be obtained from V'.
     *  pVBuffer -> Same as VPrime, but with data type of function API, instead of float.
     *  pVPointer-> Index to VPrime buffer for updating. Implements circular buffer.
     *  U        -> ISO defined array of 512, created as from VPrime, to be multiplied by DWindow
     *
     */

    /* 
     *  Indices used for implementing:
     *
     *  i        -> 32 point DCT output index, 0=<i=<32 
     *  k        -> ISO defined, 32 samples of one sub-band are indexed using k. 0=<k=<32 
     *  j        -> Sub-band index, VPrime contains 16 sub-bands
     */

    /* calculate 32pt DCT of Vi */
    
    j = *pVPosition;
    
    for (i = 0; i < ARM_MP3_PQMF_SZ; i++)
    {
        Angle = (PI / 64.0) * i;
        acc = 0.0;
        for (k = 0; k < ARM_MP3_PQMF_SZ; k++)
        {
            acc += cos (Angle * (2.0 * k + 1.0)) * Src [k];
        }
        VPrime [(ARM_MP3_PQMF_SZ * j) + i] = acc;
        pVBuffer[(ARM_MP3_PQMF_SZ * j) + i] = 
            armSatRoundFloatToS32 (acc * (1 << 24));

    }

    /* update pVPosition */
    k = ARM_MP3_PQMF_SZ * *pVPosition;
    *pVPosition -= 1;
    if (*pVPosition < 0)
    {
        *pVPosition = 15;
    }
    
    /*
     * ISO defines calculation for U[512] as,
     *     
     *     U[j*64+i] = V[j*128+i]
     *     U[j*64+32+i] = V[j*128+96+i]
     *           for 0 =< i =< 31, 0 =< j =< 7
     *    
     *    Where V is ISO defined arry of size [16*64] (64 elements per DCT32+Add Matrixing operation)
     * This calculation of U[512] is implemented using V' of size [16*32] (32 elements DCT32 elements)
     *    
     */
     
    /* Copy 64 element at a time to fill U of size [512] */
    for (j = 0; j < 8; j++)
    {
        for (i = 0; i < 16; i++)
        {
            /* Copy first 32 elements */
            /* Make U values from DCT32 values stored in V' buffer using equation
             *   Relationship of U, V & V'   
             *      U [j*64+i] = V[j*128+i+0]  =  V'[k+i+16]
             *      U [j*64+i+17] = V[j*128+i+17] = -V'[k+31-i]
             *          Where 0 =< i =< 15, 0 =< j =< 7,  0 =< k-VPosition =< 15
             */
            U [j * 64 + i] = VPrime [k + i + 16];
            U [j * 64 + i + 17] = -VPrime [k + 31 - i];
        }
        
        /* k wraps at the v-buffer boundary  */
        k = (k + ARM_MP3_PQMF_SZ) & ARM_MP3_VBUFFER_SZ_MASK;

        for (i = 0; i < 16; i++)
        {
            /* Copy next 32 elements */
            /* Make U values from DCT32 values stored in V using equation
             *   Relationship of U, V & V'   
             *      U [j*64+i+32] = V[j*128+96+i] = V'[k+16-i]
             *      U [j*64+i+48] = V[j*128+96+i+17] = -V'[k+32+i]
             *          Where 0 =< i =< 15, 0 =< j =< 7,,  0 =< k-VPosition-1 =< 15  
             */
            U [j * 64 + i + 32] = -VPrime [k + 16 - i];
            U [j * 64 + i + 48] = -VPrime [k + i];
        }
        U [j * 64 + 16] = 0;

        /* k wraps at the v-buffer boundary  */
        k = (k + ARM_MP3_PQMF_SZ) & ARM_MP3_VBUFFER_SZ_MASK;
    }

    for (i = 0; i < ARM_MP3_PQMF_SZ; i++)
    {
        acc = 0.0;
        for (J = 0; J < 16; J++)
        {
            /* i+32*J */
            k = i + J * ARM_MP3_PQMF_SZ; 
            acc += U [k] * armACMP3_DWindow [k];
        }
        /* Convert float to Q16.15 */
        pDstAudioOut [i * mode] = armSatRoundFloatToS16 (acc * (1 << 15));
    }

    return OMX_Sts_NoErr;
}

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 *                              END OF FILE
 *****************************************************************************/