add standalone ambisonics converter program ambi_converter.c

/******************************************************************************************************

   (C) 2022-2024 IVAS codec Public Collaboration with portions copyright Dolby International AB, Ericsson AB,

   Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V., Huawei Technologies Co. LTD.,

   Koninklijke Philips N.V., Nippon Telegraph and Telephone Corporation, Nokia Technologies Oy, Orange,

   Panasonic Holdings Corporation, Qualcomm Technologies, Inc., VoiceAge Corporation, and other

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   The IVAS codec Public Collaboration consisting of Dolby International AB, Ericsson AB,

   Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V., Huawei Technologies Co. LTD.,

   Koninklijke Philips N.V., Nippon Telegraph and Telephone Corporation, Nokia Technologies Oy, Orange,

   Panasonic Holdings Corporation, Qualcomm Technologies, Inc., VoiceAge Corporation, and other

   contributors to this repository retain full ownership rights in their respective contributions in

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*******************************************************************************************************/

#include <stdio.h>

#include <assert.h>

#include <math.h>

#include "tinywavein_c.h"

#include "tinywaveout_c.h"

#include "ambi_convert.h"

#define L_FRAME48k 960

#define MAX_CHANNELS 16

int main(int argc, char* argv[])

{

    WAVEFILEIN *wavFile_in;

    WAVEFILEOUT *wavFile_out;

    char* fileName_in, *fileName_out;

    uint32_t samplingRate;

    uint32_t samplesInFile;

    uint32_t numSamples = L_FRAME48k;

    uint32_t numSamplesRead32 = 0;

    int16_t bps;

    int16_t samples[L_FRAME48k*MAX_CHANNELS];

    int16_t order = 0;

    int16_t numChannels;

    AMBI_FMT in_format, out_format;

    float samples_f_in[L_FRAME48k*MAX_CHANNELS];

    float samples_f_out[L_FRAME48k*MAX_CHANNELS];

    float *in[MAX_CHANNELS], *out[MAX_CHANNELS];

    for(int16_t j=0;j<MAX_CHANNELS;j++)

    {

        in[j]=&samples_f_in[j*L_FRAME48k];

        out[j]=&samples_f_out[j*L_FRAME48k];

    }

    fileName_in = argv[1];

    fileName_out = argv[2];

    in_format = atoi(argv[3]);

    out_format = atoi(argv[4]);

    wavFile_in  = OpenWav( fileName_in, &samplingRate, &numChannels, &samplesInFile, &bps );

    wavFile_out = CreateWav( fileName_out, samplingRate, numChannels, 16 );

    order = (int16_t)sqrtf(numChannels) - 1;

    assert( order > 0 && order <=3 );

    while ( ReadWavShort( wavFile_in, samples, numSamples, &numSamplesRead32 ) != __TWI_SUCCESS )

    {

        int16_t err = 0;

        for (int16_t i=0;i<numSamples;i++)

        {

            for(int16_t j=0;j<numChannels;j++)

            {

                in[i][j]= (float) samples[i*numChannels + j];

            }

        }

        err = convert_ambi_format( in, out, order, in_format, out_format );

        for (int16_t i=0;i<numSamples;i++)

        {

            for(int16_t j=0;j<numChannels;j++)

            {

                samples[i*numChannels + j]= (int16_t) out[j][i];

            }

        }

        err = WriteWavShort( wavFile_out, samples, numSamples );

    }

    CloseWav( wavFile_out );

    CloseWavIn( wavFile_in );

    return 0;

}