/*
 * CDDL HEADER START
 *
 * The contents of this file are subject to the terms of the
 * Common Development and Distribution License, Version 1.0 only
 * (the "License").  You may not use this file except in compliance
 * with the License.
 *
 * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
 * or http://www.opensolaris.org/os/licensing.
 * See the License for the specific language governing permissions
 * and limitations under the License.
 *
 * When distributing Covered Code, include this CDDL HEADER in each
 * file and include the License file at usr/src/OPENSOLARIS.LICENSE.
 * If applicable, add the following below this CDDL HEADER, with the
 * fields enclosed by brackets "[]" replaced with your own identifying
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/*
 * Copyright (c) 1992-2001 by Sun Microsystems, Inc.
 * All rights reserved.
 *
 * Copyright 2019 RackTop Systems.
 */

#include <stdlib.h>
#include <memory.h>
#include <math.h>

#include <AudioDebug.h>
#include <AudioTypeSampleRate.h>

// This is the first stab at a conversion class for Sample Rate conversions

// class AudioTypeSampleRate methods

// Constructor
AudioTypeSampleRate::
AudioTypeSampleRate(int inrate, int outrate) :
        resampler(inrate, outrate), input_rate(inrate), output_rate(outrate)
{
}

// Destructor
AudioTypeSampleRate::
~AudioTypeSampleRate()
{
}

// Test conversion possibilities.
// Return TRUE if conversion to/from the specified type is possible.
Boolean AudioTypeSampleRate::
CanConvert(
        AudioHdr        h) const                // target header
{
        if ((input_rate <= 0) || (output_rate <= 0))
                return (FALSE);
        if ((h.encoding != LINEAR) ||
            ((h.sample_rate != output_rate) && (h.sample_rate != input_rate)) ||
            (h.bytes_per_unit != 2) ||
            (h.channels != 1)) {
                return (FALSE);
        }
        return (TRUE);
}


// Convert buffer to the specified type
// May replace the buffer with a new one, if necessary
AudioError AudioTypeSampleRate::
Convert(
        AudioBuffer*&   inbuf,                  // data buffer to process
        AudioHdr        outhdr)                 // target header
{
        AudioBuffer*    outbuf;
        AudioHdr        inhdr;
        Double          length;
        int             i;
        size_t          nsamps;
#ifdef DEBUG
        size_t          insamps;
#endif
        AudioError      err;

        inhdr = inbuf->GetHeader();
        length = inbuf->GetLength();

        if (Undefined(length)) {
                return (AUDIO_ERR_BADARG);
        }

        // Make sure we're not being asked to do the impossible
        // XXX - need a better error code
        if ((err = inhdr.Validate()) || (err = outhdr.Validate())) {
                return (err);
        }

        // If the requested conversion is different than what was initially
        // established, then return an error.
        // XXX - Maybe one day flush and re-init the filter
        if ((inhdr.sample_rate != input_rate) ||
            (outhdr.sample_rate != output_rate)) {
                return (AUDIO_ERR_BADARG);
        }

        // If conversion is a no-op, just return success
        if (inhdr.sample_rate == outhdr.sample_rate) {
                return (AUDIO_SUCCESS);
        }

        // If nothing in the buffer, do the simple thing
        if (length == 0.) {
                inbuf->SetHeader(outhdr);
                return (AUDIO_SUCCESS);
        }

        // Add some padding to the output buffer
        i = 4 * ((input_rate / output_rate) + (output_rate / input_rate));
        length += outhdr.Samples_to_Time(i);

        // Allocate a new buffer
        outbuf = new AudioBuffer(length, "(SampleRate conversion buffer)");
        if (outbuf == 0)
                return (AUDIO_UNIXERROR);
        err = outbuf->SetHeader(outhdr);
        if (err != AUDIO_SUCCESS) {
                delete outbuf;
                return (err);
        }

        // here's where the guts go ...
        nsamps = resampler.filter((short *)inbuf->GetAddress(),
                    (int)inbuf->GetHeader().Time_to_Samples(inbuf->GetLength()),
                    (short *)outbuf->GetAddress());

#ifdef DEBUG
        // do a sanity check. did we write more bytes then we had
        // available in the output buffer?
        insamps = (unsigned int)
                outbuf->GetHeader().Time_to_Samples(outbuf->GetSize());

        AUDIO_DEBUG((2, "TypeResample: after filter, insamps=%d, outsamps=%d\n",
                    insamps, nsamps));
#endif

        if (nsamps > outbuf->GetHeader().Time_to_Samples(outbuf->GetSize())) {
                AudioStderrMsg(outbuf, AUDIO_NOERROR, Fatal,
                    (char *)"resample filter corrupted the heap");
        }

        // set output size appropriately
        outbuf->SetLength(outbuf->GetHeader().Samples_to_Time(nsamps));

        // This will delete the buffer
        inbuf->Reference();
        inbuf->Dereference();

        inbuf = outbuf;
        return (AUDIO_SUCCESS);
}

AudioError AudioTypeSampleRate::
Flush(
        AudioBuffer*&   outbuf)
{
        AudioHdr        h;
        Double          pos;
        int             nsamp;
        size_t          cnt;
        AudioError      err;
        unsigned char   *tmpbuf;

        if (outbuf == NULL)
                return (AUDIO_SUCCESS);
        h = outbuf->GetHeader();

        nsamp = resampler.getFlushSize();
        if (nsamp > 0) {
                cnt = (size_t)nsamp * h.bytes_per_unit;
                tmpbuf = new unsigned char[cnt];

                // this does a flush
                nsamp = resampler.filter(NULL, 0, (short *)tmpbuf);

                // Copy to the supplied buffer
                if (nsamp > 0) {
                        cnt = (size_t)nsamp * h.bytes_per_unit;
                        pos = outbuf->GetLength();
                        err = outbuf->AppendData(tmpbuf, cnt, pos);
                        if (err)
                                return (err);
                }
                delete[] tmpbuf;
        }
        return (AUDIO_SUCCESS);
}