bearstonem--openmohaa-vr/code/client/snd_codec_mp3.c

/*
===========================================================================
Copyright (C) 1999-2005 Id Software, Inc.
Copyright (C) 2005 Stuart Dalton (badcdev@gmail.com)
Copyright (C) 2005-2006 Joerg Dietrich <dietrich_joerg@gmx.de>
Copyright (C) 2006 Thilo Schulz <arny@ats.s.bawue.de>

This file is part of Quake III Arena source code.

Quake III Arena source code is free software; you can redistribute it
and/or modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2 of the License,
or (at your option) any later version.

Quake III Arena source code is distributed in the hope that it will be
useful, but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with Quake III Arena source code; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
===========================================================================
*/

// MP3 support is enabled by this define
#if USE_CODEC_MP3

// includes for the Q3 sound system
#include "client.h"
#include "snd_codec.h"

// includes for the MP3 codec
#if USE_SOUND_NEW
#   include "mad.h"
#else
#   include "../libmad-0.15.1b/synth.h"
#endif

#define MP3_SAMPLE_WIDTH		2
#define MP3_PCMSAMPLES_PERSLICE		32

// buffer size used when reading through the mp3
#define MP3_DATA_BUFSIZ			128*1024

// undefine this if you don't want any dithering.
#define MP3_DITHERING

// Q3 MP3 codec
snd_codec_t mp3_codec =
{
    "mp3",
    S_MP3_CodecLoad,
    S_MP3_CodecOpenStream,
    S_MP3_CodecReadStream,
    S_MP3_CodecCloseStream,
    NULL
};

// structure used for info purposes
struct snd_codec_mp3_info
{
    byte encbuf[MP3_DATA_BUFSIZ];	// left over bytes not consumed
    // by the decoder.
    struct mad_stream madstream;	// uses encbuf as buffer.
    struct mad_frame madframe;	// control structures for libmad.
    struct mad_synth madsynth;

    byte* pcmbuf;			// buffer for not-used samples.
    int buflen;			// length of buffer data.
    int pcmbufsize;			// amount of allocated memory for
    // pcmbuf. This should have at least
    // the size of a decoded mp3 frame.	

    byte* dest;			// copy decoded data here.
    int destlen;			// amount of already copied data.
    int destsize;			// amount of bytes we must decode.
};

/*************** MP3 utility functions ***************/

/*
=================
S_MP3_ReadData
=================
*/

// feed libmad with data
int S_MP3_ReadData(snd_stream_t* stream, struct mad_stream* madstream, byte* encbuf, unsigned long encbufsize)
{
    unsigned long retval;
    unsigned long leftover;

    if (!stream)
        return -1;

    leftover = madstream->bufend - madstream->next_frame;
    if (leftover > 0)
        memmove(encbuf, madstream->this_frame, leftover);


    // Fill the buffer right to the end

    retval = FS_Read(&encbuf[leftover], encbufsize - leftover, stream->file);

    if (retval <= 0)
    {
        // EOF reached, that's ok.
        return 0;
    }

    mad_stream_buffer(madstream, encbuf, retval + leftover);

    return retval;
}


/*
=================
S_MP3_Scanfile

to determine the samplecount, we apparently must get *all* headers :(
I basically used the xmms-mad plugin source to see how this stuff works.

returns a value < 0 on error.
=================
*/

int S_MP3_Scanfile(snd_stream_t* stream)
{
    struct mad_stream madstream;
    struct mad_header madheader;
    int retval;
    int samplecount;
    byte encbuf[MP3_DATA_BUFSIZ];

    // error out on invalid input.
    if (!stream)
        return -1;

    mad_stream_init(&madstream);
    mad_header_init(&madheader);

    while (1)
    {
        retval = S_MP3_ReadData(stream, &madstream, encbuf, sizeof(encbuf));
        if (retval < 0)
            return -1;
        else if (retval == 0)
            break;

        // Start decoding the headers.
        while (1)
        {
            if ((retval = mad_header_decode(&madheader, &madstream)) < 0)
            {
                if (madstream.error == MAD_ERROR_BUFLEN)
                {
                    // We need to read more data
                    break;
                }

                if (!MAD_RECOVERABLE(madstream.error))
                {
                    // unrecoverable error... we must bail out.
                    return retval;
                }

                mad_stream_skip(&madstream, madstream.skiplen);
                continue;
            }

            // we got a valid header.

            if (madheader.layer != MAD_LAYER_III)
            {
                // we don't support non-mp3s
                return -1;
            }

            if (!stream->info.samples)
            {
                // This here is the very first frame. Set initial values now,
                // that we expect to stay constant throughout the whole mp3.

                stream->info.rate = madheader.samplerate;
                stream->info.width = MP3_SAMPLE_WIDTH;
                stream->info.channels = MAD_NCHANNELS(&madheader);
                stream->info.samples = 0;
                stream->info.size = 0;				// same here.
                stream->info.dataofs = 0;
            }
            else
            {
                // Check whether something changed that shouldn't.

                if (stream->info.rate != madheader.samplerate ||
                    stream->info.channels != MAD_NCHANNELS(&madheader))
                    return -1;
            }

            // Update the counters
            samplecount = MAD_NSBSAMPLES(&madheader) * MP3_PCMSAMPLES_PERSLICE;
            stream->info.samples += samplecount;
            stream->info.size += samplecount * stream->info.channels * stream->info.width;
        }
    }

    // Reset the file pointer so we can do the real decoding.
    FS_Seek(stream->file, 0, FS_SEEK_SET);

    return 0;
}

/************************ dithering functions ***************************/

#ifdef MP3_DITHERING

// All dithering done here is taken from the GPL'ed xmms-mad plugin.

/* Copyright (C) 1997 Makoto Matsumoto and Takuji Nishimura.       */
/* Any feedback is very welcome. For any question, comments,       */
/* see http://www.math.keio.ac.jp/matumoto/emt.html or email       */
/* matumoto@math.keio.ac.jp                                        */

/* Period parameters */
#define MP3_DITH_N 624
#define MP3_DITH_M 397
#define MATRIX_A 0x9908b0df   /* constant vector a */
#define UPPER_MASK 0x80000000 /* most significant w-r bits */
#define LOWER_MASK 0x7fffffff /* least significant r bits */

/* Tempering parameters */
#define TEMPERING_MASK_B 0x9d2c5680
#define TEMPERING_MASK_C 0xefc60000
#define TEMPERING_SHIFT_U(y)  (y >> 11)
#define TEMPERING_SHIFT_S(y)  (y << 7)
#define TEMPERING_SHIFT_T(y)  (y << 15)
#define TEMPERING_SHIFT_L(y)  (y >> 18)

static unsigned long mt[MP3_DITH_N]; /* the array for the state vector  */
static int mti = MP3_DITH_N + 1; /* mti==MP3_DITH_N+1 means mt[MP3_DITH_N] is not initialized */

/* initializing the array with a NONZERO seed */
void sgenrand(unsigned long seed)
{
    /* setting initial seeds to mt[MP3_DITH_N] using         */
    /* the generator Line 25 of Table 1 in          */
    /* [KNUTH 1981, The Art of Computer Programming */
    /*    Vol. 2 (2nd Ed.), pp102]                  */
    mt[0] = seed & 0xffffffff;
    for (mti = 1; mti < MP3_DITH_N; mti++)
        mt[mti] = (69069 * mt[mti - 1]) & 0xffffffff;
}

unsigned long genrand(void)
{
    unsigned long y;
    static unsigned long mag01[2] = { 0x0, MATRIX_A };
    /* mag01[x] = x * MATRIX_A  for x=0,1 */

    if (mti >= MP3_DITH_N) { /* generate MP3_DITH_N words at one time */
        int kk;

        if (mti == MP3_DITH_N + 1)   /* if sgenrand() has not been called, */
            sgenrand(4357); /* a default initial seed is used   */

        for (kk = 0; kk < MP3_DITH_N - MP3_DITH_M; kk++) {
            y = (mt[kk] & UPPER_MASK) | (mt[kk + 1] & LOWER_MASK);
            mt[kk] = mt[kk + MP3_DITH_M] ^ (y >> 1) ^ mag01[y & 0x1];
        }
        for (; kk < MP3_DITH_N - 1; kk++) {
            y = (mt[kk] & UPPER_MASK) | (mt[kk + 1] & LOWER_MASK);
            mt[kk] = mt[kk + (MP3_DITH_M - MP3_DITH_N)] ^ (y >> 1) ^ mag01[y & 0x1];
        }
        y = (mt[MP3_DITH_N - 1] & UPPER_MASK) | (mt[0] & LOWER_MASK);
        mt[MP3_DITH_N - 1] = mt[MP3_DITH_M - 1] ^ (y >> 1) ^ mag01[y & 0x1];

        mti = 0;
    }

    y = mt[mti++];
    y ^= TEMPERING_SHIFT_U(y);
    y ^= TEMPERING_SHIFT_S(y) & TEMPERING_MASK_B;
    y ^= TEMPERING_SHIFT_T(y) & TEMPERING_MASK_C;
    y ^= TEMPERING_SHIFT_L(y);

    return y;
}

long triangular_dither_noise(int nbits) {
    // parameter nbits : the peak-to-peak amplitude desired (in bits)
    //  use with nbits set to    2 + nber of bits to be trimmed.
    // (because triangular is made from two uniformly distributed processes,
    // it starts at 2 bits peak-to-peak amplitude)
    // see The Theory of Dithered Quantization by Robert Alexander Wannamaker
    // for complete proof of why that's optimal

    long v = (genrand() / 2 - genrand() / 2); // in ]-2^31, 2^31[
    //int signe = (v>0) ? 1 : -1;
    long P = 1 << (32 - nbits); // the power of 2
    v /= P;
    // now v in ]-2^(nbits-1), 2^(nbits-1) [

    return v;
}

#endif // MP3_DITHERING

/************************ decoder functions ***************************/

/*
=================
S_MP3_Scale

Converts the signal to 16 bit LE-PCM data and does dithering.

- borrowed from xmms-mad plugin source.
=================
*/

/*
 * xmms-mad - mp3 plugin for xmms
 * Copyright (C) 2001-2002 Sam Clegg
 */

signed int S_MP3_Scale(mad_fixed_t sample)
{
    int n_bits_to_loose = MAD_F_FRACBITS + 1 - 16;
#ifdef MP3_DITHERING
    int dither;
#endif

    // round
    sample += (1L << (n_bits_to_loose - 1));

#ifdef MP3_DITHERING
    dither = triangular_dither_noise(n_bits_to_loose + 1);
    sample += dither;
#endif

    /* clip */
    if (sample >= MAD_F_ONE)
        sample = MAD_F_ONE - 1;
    else if (sample < -MAD_F_ONE)
        sample = -MAD_F_ONE;

    /* quantize */
    return sample >> n_bits_to_loose;
}



/*
=================
S_MP3_PCMCopy

Copy and convert pcm data until bytecount bytes have been written.
return the position in pcm->samples.
indicate the amount of actually written bytes in wrotecnt.
=================
*/

int S_MP3_PCMCopy(byte* buf, struct mad_pcm* pcm, int bufofs,
    int sampleofs, int bytecount, int* wrotecnt)
{
    int written = 0;
    signed int sample;
    int framesize = pcm->channels * MP3_SAMPLE_WIDTH;

    // add new pcm data.
    while (written < bytecount && sampleofs < pcm->length)
    {
        sample = S_MP3_Scale(pcm->samples[0][sampleofs]);

#ifdef Q3_BIG_ENDIAN
        // output to 16 bit big endian PCM
        buf[bufofs++] = (sample >> 8) & 0xff;
        buf[bufofs++] = sample & 0xff;
#else
        // output to 16 bit little endian PCM
        buf[bufofs++] = sample & 0xff;
        buf[bufofs++] = (sample >> 8) & 0xff;
#endif

        if (pcm->channels == 2)
        {
            sample = S_MP3_Scale(pcm->samples[1][sampleofs]);

#ifdef Q3_BIG_ENDIAN
            buf[bufofs++] = (sample >> 8) & 0xff;
            buf[bufofs++] = sample & 0xff;
#else
            buf[bufofs++] = sample & 0xff;
            buf[bufofs++] = (sample >> 8) & 0xff;
#endif
        }

        sampleofs++;
        written += framesize;
    }

    if (wrotecnt)
        *wrotecnt = written;

    return sampleofs;
}


/*
=================
S_MP3_Decode
=================
*/

// gets executed for every decoded frame.
int S_MP3_Decode(snd_stream_t* stream)
{
    struct snd_codec_mp3_info* mp3info;
    struct mad_stream* madstream;
    struct mad_frame* madframe;
    struct mad_synth* madsynth;
    struct mad_pcm* pcm;
    int cursize;
    int samplecount;
    int needcount;
    int wrote;
    int retval;

    if (!stream)
        return -1;

    mp3info = stream->ptr;
    madstream = &mp3info->madstream;
    madframe = &mp3info->madframe;

    if (mad_frame_decode(madframe, madstream))
    {
        if (madstream->error == MAD_ERROR_BUFLEN)
        {
            // we need more data. Read another chunk.
            retval = S_MP3_ReadData(stream, madstream, mp3info->encbuf, sizeof(mp3info->encbuf));

            // call myself again now that buffer is full.
            if (retval > 0)
                retval = S_MP3_Decode(stream);
        }
        else if (MAD_RECOVERABLE(madstream->error))
        {
            mad_stream_skip(madstream, madstream->skiplen);
            return S_MP3_Decode(stream);
        }
        else
            retval = -1;

        return retval;
    }

    // check whether this really is an mp3
    if (madframe->header.layer != MAD_LAYER_III)
        return -1;

    // generate pcm data
    madsynth = &mp3info->madsynth;
    mad_synth_frame(madsynth, madframe);

    pcm = &madsynth->pcm;

    // perform a few checks to see whether something changed that shouldn't.

    if (stream->info.rate != pcm->samplerate ||
        stream->info.channels != pcm->channels)
    {
        return -1;
    }
    // see whether we have got enough data now.
    cursize = pcm->length * pcm->channels * stream->info.width;
    needcount = mp3info->destsize - mp3info->destlen;

    // Copy exactly as many samples as required.
    samplecount = S_MP3_PCMCopy(mp3info->dest, pcm,
        mp3info->destlen, 0, needcount, &wrote);
    mp3info->destlen += wrote;

    if (samplecount < pcm->length)
    {
        // Not all samples got copied. Copy the rest into the pcm buffer.
        samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
            mp3info->buflen,
            samplecount,
            mp3info->pcmbufsize - mp3info->buflen,
            &wrote);
        mp3info->buflen += wrote;


        if (samplecount < pcm->length)
        {
            // The pcm buffer was not large enough. Make it bigger.
            byte* newbuf = Z_Malloc(cursize);

            if (mp3info->pcmbuf)
            {
                memcpy(newbuf, mp3info->pcmbuf, mp3info->buflen);
                Z_Free(mp3info->pcmbuf);
            }

            mp3info->pcmbuf = newbuf;
            mp3info->pcmbufsize = cursize;

            samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
                mp3info->buflen,
                samplecount,
                mp3info->pcmbufsize - mp3info->buflen,
                &wrote);
            mp3info->buflen += wrote;
        }

        // we're definitely done.
        retval = 0;
    }
    else if (mp3info->destlen >= mp3info->destsize)
        retval = 0;
    else
        retval = 1;

    return retval;
}

/*************** Callback functions for quake3 ***************/

/*
=================
S_MP3_CodecOpenStream
=================
*/

snd_stream_t* S_MP3_CodecOpenStream(const char* filename)
{
    snd_stream_t* stream;
    struct snd_codec_mp3_info* mp3info;

    // Open the stream
    stream = S_CodecUtilOpen(filename, &mp3_codec);
    if (!stream || stream->length <= 0)
        return NULL;

    // We have to scan through the MP3 to determine the important mp3 info.
    if (S_MP3_Scanfile(stream) < 0)
    {
        // scanning didn't work out...
        S_CodecUtilClose(&stream);
        return NULL;
    }

    // Initialize the mp3 info structure we need for streaming
    mp3info = Z_Malloc(sizeof(*mp3info));
    if (!mp3info)
    {
        S_CodecUtilClose(&stream);
        return NULL;
    }

    stream->ptr = mp3info;

    // initialize the libmad control structures.
    mad_stream_init(&mp3info->madstream);
    mad_frame_init(&mp3info->madframe);
    mad_synth_init(&mp3info->madsynth);

    if (S_MP3_ReadData(stream, &mp3info->madstream, mp3info->encbuf, sizeof(mp3info->encbuf)) <= 0)
    {
        // we didnt read anything, that's bad.
        S_MP3_CodecCloseStream(stream);
        return NULL;
    }

    return stream;
}

/*
=================
S_MP3_CodecCloseStream
=================
*/

// free all memory we allocated.
void S_MP3_CodecCloseStream(snd_stream_t* stream)
{
    struct snd_codec_mp3_info* mp3info;

    if (!stream)
        return;

    // free all data in our mp3info tree

    if (stream->ptr)
    {
        mp3info = stream->ptr;

        if (mp3info->pcmbuf)
            Z_Free(mp3info->pcmbuf);

        mad_synth_finish(&mp3info->madsynth);
        mad_frame_finish(&mp3info->madframe);
        mad_stream_finish(&mp3info->madstream);

        Z_Free(stream->ptr);
    }

    S_CodecUtilClose(&stream);
}

/*
=================
S_MP3_CodecReadStream
=================
*/
int S_MP3_CodecReadStream(snd_stream_t* stream, int bytes, void* buffer)
{
    struct snd_codec_mp3_info* mp3info;
    int retval;

    if (!stream)
        return -1;

    mp3info = stream->ptr;

    // Make sure we get complete frames all the way through.
    bytes -= bytes % (stream->info.channels * stream->info.width);

    if (mp3info->buflen)
    {
        if (bytes < mp3info->buflen)
        {
            // we still have enough bytes in our decoded pcm buffer
            memcpy(buffer, mp3info->pcmbuf, bytes);

            // remove the portion from our buffer.
            mp3info->buflen -= bytes;
            memmove(mp3info->pcmbuf, &mp3info->pcmbuf[bytes], mp3info->buflen);
            return bytes;
        }
        else
        {
            // copy over the samples we already have.
            memcpy(buffer, mp3info->pcmbuf, mp3info->buflen);
            mp3info->destlen = mp3info->buflen;
            mp3info->buflen = 0;
        }
    }
    else
        mp3info->destlen = 0;

    mp3info->dest = buffer;
    mp3info->destsize = bytes;

    do
    {
        retval = S_MP3_Decode(stream);
    } while (retval > 0);

    // if there was an error return nothing.
    if (retval < 0)
        return 0;

    return mp3info->destlen;
}

/*
=====================================================================
S_MP3_CodecLoad

We handle S_MP3_CodecLoad as a special case of the streaming functions
where we read the whole stream at once.
======================================================================
*/
void* S_MP3_CodecLoad(const char* filename, snd_info_t* info)
{
    snd_stream_t* stream;
    byte* pcmbuffer;

    // check if input is valid
    if (!filename)
        return NULL;

    stream = S_MP3_CodecOpenStream(filename);

    if (!stream)
        return NULL;

    // copy over the info
    info->rate = stream->info.rate;
    info->width = stream->info.width;
    info->channels = stream->info.channels;
    info->samples = stream->info.samples;
    info->dataofs = stream->info.dataofs;

    // allocate enough buffer for all pcm data
    pcmbuffer = Z_Malloc(stream->info.size);
    if (!pcmbuffer)
    {
        S_MP3_CodecCloseStream(stream);
        return NULL;
    }

    info->size = S_MP3_CodecReadStream(stream, stream->info.size, pcmbuffer);

    if (info->size <= 0)
    {
        // we didn't read anything at all. darn.
        Z_Free(pcmbuffer);
        pcmbuffer = NULL;
    }

    S_MP3_CodecCloseStream(stream);

    return pcmbuffer;
}

#endif