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freeswitch/libs/libsndfile/src/ALAC/alac_encoder.c

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/*
* Copyright (c) 2011 Apple Inc. All rights reserved.
* Copyright (C) 2012-2013 Erik de Castro Lopo <erikd@mega-nerd.com>
*
* @APPLE_APACHE_LICENSE_HEADER_START@
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*
* @APPLE_APACHE_LICENSE_HEADER_END@
*/
/*
File: ALACEncoder.cpp
*/
// build stuff
#define VERBOSE_DEBUG 0
#define DebugMsg printf
// headers
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "sfendian.h"
#include "alac_codec.h"
#include "aglib.h"
#include "dplib.h"
#include "matrixlib.h"
#include "ALACBitUtilities.h"
#include "ALACAudioTypes.h"
#include "EndianPortable.h"
typedef enum
{
false = 0,
true = 1
} bool ;
static void GetConfig(ALAC_ENCODER *p, ALACSpecificConfig * config );
static int32_t EncodeStereo(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * input, uint32_t stride, uint32_t channelIndex, uint32_t numSamples );
static int32_t EncodeStereoFast(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * input, uint32_t stride, uint32_t channelIndex, uint32_t numSamples );
static int32_t EncodeStereoEscape(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * input, uint32_t stride, uint32_t numSamples );
static int32_t EncodeMono(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * input, uint32_t stride, uint32_t channelIndex, uint32_t numSamples );
// Note: in C you can't typecast to a 2-dimensional array pointer but that's what we need when
// picking which coefs to use so we declare this typedef b/c we *can* typecast to this type
typedef int16_t (*SearchCoefs)[kALACMaxCoefs];
// defines/constants
const uint32_t kALACEncoderMagic = MAKE_MARKER ('d', 'p', 'g', 'e');
const uint32_t kMaxSampleSize = 32; // max allowed bit width is 32
const uint32_t kDefaultMixBits = 2;
const uint32_t kDefaultMixRes = 0;
const uint32_t kMaxRes = 4;
const uint32_t kDefaultNumUV = 8;
const uint32_t kMinUV = 4;
const uint32_t kMaxUV = 8;
// static functions
#if VERBOSE_DEBUG
static void AddFiller( BitBuffer * bits, int32_t numBytes );
#endif
/*
Map Format: 3-bit field per channel which is the same as the "element tag" that should be placed
at the beginning of the frame for that channel. Indicates whether SCE, CPE, or LFE.
Each particular field is accessed via the current channel indx. Note that the channel
indx increments by two for channel pairs.
For example:
C L R 3-channel input = (ID_CPE << 3) | (ID_SCE)
indx 0 value = (map & (0x7ul << (0 * 3))) >> (0 * 3)
indx 1 value = (map & (0x7ul << (1 * 3))) >> (1 * 3)
C L R Ls Rs LFE 5.1-channel input = (ID_LFE << 15) | (ID_CPE << 9) | (ID_CPE << 3) | (ID_SCE)
indx 0 value = (map & (0x7ul << (0 * 3))) >> (0 * 3)
indx 1 value = (map & (0x7ul << (1 * 3))) >> (1 * 3)
indx 3 value = (map & (0x7ul << (3 * 3))) >> (3 * 3)
indx 5 value = (map & (0x7ul << (5 * 3))) >> (5 * 3)
indx 7 value = (map & (0x7ul << (7 * 3))) >> (7 * 3)
*/
static const uint32_t sChannelMaps[kALACMaxChannels] =
{
ID_SCE,
ID_CPE,
(ID_CPE << 3) | (ID_SCE),
(ID_SCE << 9) | (ID_CPE << 3) | (ID_SCE),
(ID_CPE << 9) | (ID_CPE << 3) | (ID_SCE),
(ID_SCE << 15) | (ID_CPE << 9) | (ID_CPE << 3) | (ID_SCE),
(ID_SCE << 18) | (ID_SCE << 15) | (ID_CPE << 9) | (ID_CPE << 3) | (ID_SCE),
(ID_SCE << 21) | (ID_CPE << 15) | (ID_CPE << 9) | (ID_CPE << 3) | (ID_SCE)
};
static const uint32_t sSupportediPodSampleRates[] =
{
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
};
#if PRAGMA_MARK
#pragma mark -
#endif
void
alac_set_fastmode (ALAC_ENCODER * p, int32_t fast )
{
p->mFastMode = fast;
}
/*
HEADER SPECIFICATION
For every segment we adopt the following header:
1 byte reserved (always 0)
1 byte flags (see below)
[4 byte frame length] (optional, see below)
---Next, the per-segment ALAC parameters---
1 byte mixBits (middle-side parameter)
1 byte mixRes (middle-side parameter, interpreted as signed char)
1 byte shiftU (4 bits modeU, 4 bits denShiftU)
1 byte filterU (3 bits pbFactorU, 5 bits numU)
(numU) shorts (signed DP coefficients for V channel)
---Next, 2nd-channel ALAC parameters in case of stereo mode---
1 byte shiftV (4 bits modeV, 4 bits denShiftV)
1 byte filterV (3 bits pbFactorV, 5 bits numV)
(numV) shorts (signed DP coefficients for V channel)
---After this come the shift-off bytes for (>= 24)-bit data (n-byte shift) if indicated---
---Then comes the AG-compressor bitstream---
FLAGS
-----
The presence of certain flag bits changes the header format such that the parameters might
not even be sent. The currently defined flags format is:
0000psse
where 0 = reserved, must be 0
p = 1-bit field "partial frame" flag indicating 32-bit frame length follows this byte
ss = 2-bit field indicating "number of shift-off bytes ignored by compression"
e = 1-bit field indicating "escape"
The "partial frame" flag means that the following segment is not equal to the frame length specified
in the out-of-band decoder configuration. This allows the decoder to deal with end-of-file partial
segments without incurring the 32-bit overhead for each segment.
The "shift-off" field indicates the number of bytes at the bottom of the word that were passed through
uncompressed. The reason for this is that the entropy inherent in the LS bytes of >= 24-bit words
quite often means that the frame would have to be "escaped" b/c the compressed size would be >= the
uncompressed size. However, by shifting the input values down and running the remaining bits through
the normal compression algorithm, a net win can be achieved. If this field is non-zero, it means that
the shifted-off bytes follow after the parameter section of the header and before the compressed
bitstream. Note that doing this also allows us to use matrixing on 32-bit inputs after one or more
bytes are shifted off the bottom which helps the eventual compression ratio. For stereo channels,
the shifted off bytes are interleaved.
The "escape" flag means that this segment was not compressed b/c the compressed size would be
>= uncompressed size. In that case, the audio data was passed through uncompressed after the header.
The other header parameter bytes will not be sent.
PARAMETERS
----------
If the segment is not a partial or escape segment, the total header size (in bytes) is given exactly by:
4 + (2 + 2 * numU) (mono mode)
4 + (2 + 2 * numV) + (2 + 2 * numV) (stereo mode)
where the ALAC filter-lengths numU, numV are bounded by a
constant (in the current source, numU, numV <= NUMCOEPAIRS), and
this forces an absolute upper bound on header size.
Each segment-decode process loads up these bytes from the front of the
local stream, in the above order, then follows with the entropy-encoded
bits for the given segment.
To generalize middle-side, there are various mixing modes including middle-side, each lossless,
as embodied in the mix() and unmix() functions. These functions exploit a generalized middle-side
transformation:
u := [(rL + (m-r)R)/m];
v := L - R;
where [ ] denotes integer floor. The (lossless) inverse is
L = u + v - [rV/m];
R = L - v;
In the segment header, m and r are encoded in mixBits and mixRes.
Classical "middle-side" is obtained with m = 2, r = 1, but now
we have more generalized mixes.
NOTES
-----
The relevance of the ALAC coefficients is explained in detail
in patent documents.
*/
/*
EncodeStereo()
- encode a channel pair
*/
static int32_t
EncodeStereo(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * inputBuffer, uint32_t stride, uint32_t channelIndex, uint32_t numSamples )
{
BitBuffer workBits;
BitBuffer startBits = *bitstream; // squirrel away copy of current state in case we need to go back and do an escape packet
AGParamRec agParams;
uint32_t bits1, bits2;
uint32_t dilate;
int32_t mixBits, mixRes, maxRes;
uint32_t minBits, minBits1, minBits2;
uint32_t numU, numV;
uint32_t mode;
uint32_t pbFactor;
uint32_t chanBits;
uint8_t bytesShifted;
SearchCoefs coefsU;
SearchCoefs coefsV;
uint32_t indx;
uint8_t partialFrame;
uint32_t escapeBits;
bool doEscape;
int32_t status = ALAC_noErr;
int32_t bestRes;
uint32_t numUV, converge;
// make sure we handle this bit-depth before we get going
RequireAction( (p->mBitDepth == 16) || (p->mBitDepth == 20) || (p->mBitDepth == 24) || (p->mBitDepth == 32), return kALAC_ParamError; );
// reload coefs pointers for this channel pair
// - note that, while you might think they should be re-initialized per block, retaining state across blocks
// actually results in better overall compression
// - strangely, re-using the same coefs for the different passes of the "mixRes" search loop instead of using
// different coefs for the different passes of "mixRes" results in even better compression
coefsU = (SearchCoefs) p->mCoefsU[channelIndex];
coefsV = (SearchCoefs) p->mCoefsV[channelIndex];
// matrix encoding adds an extra bit but 32-bit inputs cannot be matrixed b/c 33 is too many
// so enable 16-bit "shift off" and encode in 17-bit mode
// - in addition, 24-bit mode really improves with one byte shifted off
if ( p->mBitDepth == 32 )
bytesShifted = 2;
else if ( p->mBitDepth >= 24 )
bytesShifted = 1;
else
bytesShifted = 0;
chanBits = p->mBitDepth - (bytesShifted * 8) + 1;
// flag whether or not this is a partial frame
partialFrame = (numSamples == p->mFrameSize) ? 0 : 1;
// brute-force encode optimization loop
// - run over variations of the encoding params to find the best choice
mixBits = kDefaultMixBits;
maxRes = kMaxRes;
numU = numV = kDefaultNumUV;
mode = 0;
pbFactor = 4;
dilate = 8;
minBits = minBits1 = minBits2 = 1ul << 31;
bestRes = p->mLastMixRes[channelIndex];
for ( mixRes = 0; mixRes <= maxRes; mixRes++ )
{
// mix the stereo inputs
switch ( p->mBitDepth )
{
case 16:
mix16( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples/dilate, mixBits, mixRes );
break;
case 20:
mix20( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples/dilate, mixBits, mixRes );
break;
case 24:
// includes extraction of shifted-off bytes
mix24( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples/dilate,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
case 32:
// includes extraction of shifted-off bytes
mix32( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples/dilate,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
}
BitBufferInit( &workBits, p->mWorkBuffer, p->mMaxOutputBytes );
// run the dynamic predictors
pc_block( p->mMixBufferU, p->mPredictorU, numSamples/dilate, coefsU[numU - 1], numU, chanBits, DENSHIFT_DEFAULT );
pc_block( p->mMixBufferV, p->mPredictorV, numSamples/dilate, coefsV[numV - 1], numV, chanBits, DENSHIFT_DEFAULT );
// run the lossless compressor on each channel
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples/dilate, numSamples/dilate, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorU, &workBits, numSamples/dilate, chanBits, &bits1 );
RequireNoErr( status, goto Exit; );
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples/dilate, numSamples/dilate, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorV, &workBits, numSamples/dilate, chanBits, &bits2 );
RequireNoErr( status, goto Exit; );
// look for best match
if ( (bits1 + bits2) < minBits1 )
{
minBits1 = bits1 + bits2;
bestRes = mixRes;
}
}
p->mLastMixRes[channelIndex] = (int16_t)bestRes;
// mix the stereo inputs with the current best mixRes
mixRes = p->mLastMixRes[channelIndex];
switch ( p->mBitDepth )
{
case 16:
mix16( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples, mixBits, mixRes );
break;
case 20:
mix20( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples, mixBits, mixRes );
break;
case 24:
// also extracts the shifted off bytes into the shift buffers
mix24( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
case 32:
// also extracts the shifted off bytes into the shift buffers
mix32( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
}
// now it's time for the predictor coefficient search loop
numU = numV = kMinUV;
minBits1 = minBits2 = 1ul << 31;
for ( numUV = kMinUV; numUV <= kMaxUV; numUV += 4 )
{
BitBufferInit( &workBits, p->mWorkBuffer, p->mMaxOutputBytes );
dilate = 32;
// run the predictor over the same data multiple times to help it converge
for ( converge = 0; converge < 8; converge++ )
{
pc_block( p->mMixBufferU, p->mPredictorU, numSamples/dilate, coefsU[numUV-1], numUV, chanBits, DENSHIFT_DEFAULT );
pc_block( p->mMixBufferV, p->mPredictorV, numSamples/dilate, coefsV[numUV-1], numUV, chanBits, DENSHIFT_DEFAULT );
}
dilate = 8;
set_ag_params( &agParams, MB0, (pbFactor * PB0)/4, KB0, numSamples/dilate, numSamples/dilate, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorU, &workBits, numSamples/dilate, chanBits, &bits1 );
if ( (bits1 * dilate + 16 * numUV) < minBits1 )
{
minBits1 = bits1 * dilate + 16 * numUV;
numU = numUV;
}
set_ag_params( &agParams, MB0, (pbFactor * PB0)/4, KB0, numSamples/dilate, numSamples/dilate, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorV, &workBits, numSamples/dilate, chanBits, &bits2 );
if ( (bits2 * dilate + 16 * numUV) < minBits2 )
{
minBits2 = bits2 * dilate + 16 * numUV;
numV = numUV;
}
}
// test for escape hatch if best calculated compressed size turns out to be more than the input size
minBits = minBits1 + minBits2 + (8 /* mixRes/maxRes/etc. */ * 8) + ((partialFrame == true) ? 32 : 0);
if ( bytesShifted != 0 )
minBits += (numSamples * (bytesShifted * 8) * 2);
escapeBits = (numSamples * p->mBitDepth * 2) + ((partialFrame == true) ? 32 : 0) + (2 * 8); /* 2 common header bytes */
doEscape = (minBits >= escapeBits) ? true : false;
if ( doEscape == false )
{
// write bitstream header and coefs
BitBufferWrite( bitstream, 0, 12 );
BitBufferWrite( bitstream, (partialFrame << 3) | (bytesShifted << 1), 4 );
if ( partialFrame )
BitBufferWrite( bitstream, numSamples, 32 );
BitBufferWrite( bitstream, mixBits, 8 );
BitBufferWrite( bitstream, mixRes, 8 );
//Assert( (mode < 16) && (DENSHIFT_DEFAULT < 16) );
//Assert( (pbFactor < 8) && (numU < 32) );
//Assert( (pbFactor < 8) && (numV < 32) );
BitBufferWrite( bitstream, (mode << 4) | DENSHIFT_DEFAULT, 8 );
BitBufferWrite( bitstream, (pbFactor << 5) | numU, 8 );
for ( indx = 0; indx < numU; indx++ )
BitBufferWrite( bitstream, coefsU[numU - 1][indx], 16 );
BitBufferWrite( bitstream, (mode << 4) | DENSHIFT_DEFAULT, 8 );
BitBufferWrite( bitstream, (pbFactor << 5) | numV, 8 );
for ( indx = 0; indx < numV; indx++ )
BitBufferWrite( bitstream, coefsV[numV - 1][indx], 16 );
// if shift active, write the interleaved shift buffers
if ( bytesShifted != 0 )
{
uint32_t bitShift = bytesShifted * 8;
//Assert( bitShift <= 16 );
for ( indx = 0; indx < (numSamples * 2); indx += 2 )
{
uint32_t shiftedVal;
shiftedVal = ((uint32_t) p->mShiftBufferUV[indx + 0] << bitShift) | (uint32_t) p->mShiftBufferUV[indx + 1];
BitBufferWrite( bitstream, shiftedVal, bitShift * 2 );
}
}
// run the dynamic predictor and lossless compression for the "left" channel
// - note: to avoid allocating more buffers, we're mixing and matching between the available buffers instead
// of only using "U" buffers for the U-channel and "V" buffers for the V-channel
if ( mode == 0 )
{
pc_block( p->mMixBufferU, p->mPredictorU, numSamples, coefsU[numU - 1], numU, chanBits, DENSHIFT_DEFAULT );
}
else
{
pc_block( p->mMixBufferU, p->mPredictorV, numSamples, coefsU[numU - 1], numU, chanBits, DENSHIFT_DEFAULT );
pc_block( p->mPredictorV, p->mPredictorU, numSamples, NULL, 31, chanBits, 0 );
}
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples, numSamples, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorU, bitstream, numSamples, chanBits, &bits1 );
RequireNoErr( status, goto Exit; );
// run the dynamic predictor and lossless compression for the "right" channel
if ( mode == 0 )
{
pc_block( p->mMixBufferV, p->mPredictorV, numSamples, coefsV[numV - 1], numV, chanBits, DENSHIFT_DEFAULT );
}
else
{
pc_block( p->mMixBufferV, p->mPredictorU, numSamples, coefsV[numV - 1], numV, chanBits, DENSHIFT_DEFAULT );
pc_block( p->mPredictorU, p->mPredictorV, numSamples, NULL, 31, chanBits, 0 );
}
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples, numSamples, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorV, bitstream, numSamples, chanBits, &bits2 );
RequireNoErr( status, goto Exit; );
/* if we happened to create a compressed packet that was actually bigger than an escape packet would be,
chuck it and do an escape packet
*/
minBits = BitBufferGetPosition( bitstream ) - BitBufferGetPosition( &startBits );
if ( minBits >= escapeBits )
{
*bitstream = startBits; // reset bitstream state
doEscape = true;
printf( "compressed frame too big: %u vs. %u \n", minBits, escapeBits );
}
}
if ( doEscape == true )
{
/* escape */
status = EncodeStereoEscape(p, bitstream, inputBuffer, stride, numSamples );
#if VERBOSE_DEBUG
DebugMsg( "escape!: %u vs %u\n", minBits, escapeBits );
#endif
}
Exit:
return status;
}
/*
EncodeStereoFast()
- encode a channel pair without the search loop for maximum possible speed
*/
static int32_t
EncodeStereoFast(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * inputBuffer, uint32_t stride, uint32_t channelIndex, uint32_t numSamples )
{
BitBuffer startBits = *bitstream; // squirrel away current bit position in case we decide to use escape hatch
AGParamRec agParams;
uint32_t bits1, bits2;
int32_t mixBits, mixRes;
uint32_t minBits, minBits1, minBits2;
uint32_t numU, numV;
uint32_t mode;
uint32_t pbFactor;
uint32_t chanBits;
uint8_t bytesShifted;
SearchCoefs coefsU;
SearchCoefs coefsV;
uint32_t indx;
uint8_t partialFrame;
uint32_t escapeBits;
bool doEscape;
int32_t status;
// make sure we handle this bit-depth before we get going
RequireAction( (p->mBitDepth == 16) || (p->mBitDepth == 20) || (p->mBitDepth == 24) || (p->mBitDepth == 32), return kALAC_ParamError; );
// reload coefs pointers for this channel pair
// - note that, while you might think they should be re-initialized per block, retaining state across blocks
// actually results in better overall compression
// - strangely, re-using the same coefs for the different passes of the "mixRes" search loop instead of using
// different coefs for the different passes of "mixRes" results in even better compression
coefsU = (SearchCoefs) p->mCoefsU[channelIndex];
coefsV = (SearchCoefs) p->mCoefsV[channelIndex];
// matrix encoding adds an extra bit but 32-bit inputs cannot be matrixed b/c 33 is too many
// so enable 16-bit "shift off" and encode in 17-bit mode
// - in addition, 24-bit mode really improves with one byte shifted off
if ( p->mBitDepth == 32 )
bytesShifted = 2;
else if ( p->mBitDepth >= 24 )
bytesShifted = 1;
else
bytesShifted = 0;
chanBits = p->mBitDepth - (bytesShifted * 8) + 1;
// flag whether or not this is a partial frame
partialFrame = (numSamples == p->mFrameSize) ? 0 : 1;
// set up default encoding parameters for "fast" mode
mixBits = kDefaultMixBits;
mixRes = kDefaultMixRes;
numU = numV = kDefaultNumUV;
mode = 0;
pbFactor = 4;
minBits = minBits1 = minBits2 = 1ul << 31;
// mix the stereo inputs with default mixBits/mixRes
switch ( p->mBitDepth )
{
case 16:
mix16( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples, mixBits, mixRes );
break;
case 20:
mix20( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples, mixBits, mixRes );
break;
case 24:
// also extracts the shifted off bytes into the shift buffers
mix24( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
case 32:
// also extracts the shifted off bytes into the shift buffers
mix32( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples,
mixBits, mixRes, p->mShiftBufferUV, bytesShifted );
break;
}
/* speculatively write the bitstream assuming the compressed version will be smaller */
// write bitstream header and coefs
BitBufferWrite( bitstream, 0, 12 );
BitBufferWrite( bitstream, (partialFrame << 3) | (bytesShifted << 1), 4 );
if ( partialFrame )
BitBufferWrite( bitstream, numSamples, 32 );
BitBufferWrite( bitstream, mixBits, 8 );
BitBufferWrite( bitstream, mixRes, 8 );
//Assert( (mode < 16) && (DENSHIFT_DEFAULT < 16) );
//Assert( (pbFactor < 8) && (numU < 32) );
//Assert( (pbFactor < 8) && (numV < 32) );
BitBufferWrite( bitstream, (mode << 4) | DENSHIFT_DEFAULT, 8 );
BitBufferWrite( bitstream, (pbFactor << 5) | numU, 8 );
for ( indx = 0; indx < numU; indx++ )
BitBufferWrite( bitstream, coefsU[numU - 1][indx], 16 );
BitBufferWrite( bitstream, (mode << 4) | DENSHIFT_DEFAULT, 8 );
BitBufferWrite( bitstream, (pbFactor << 5) | numV, 8 );
for ( indx = 0; indx < numV; indx++ )
BitBufferWrite( bitstream, coefsV[numV - 1][indx], 16 );
// if shift active, write the interleaved shift buffers
if ( bytesShifted != 0 )
{
uint32_t bitShift = bytesShifted * 8;
//Assert( bitShift <= 16 );
for ( indx = 0; indx < (numSamples * 2); indx += 2 )
{
uint32_t shiftedVal;
shiftedVal = ((uint32_t) p->mShiftBufferUV[indx + 0] << bitShift) | (uint32_t) p->mShiftBufferUV[indx + 1];
BitBufferWrite( bitstream, shiftedVal, bitShift * 2 );
}
}
// run the dynamic predictor and lossless compression for the "left" channel
// - note: we always use mode 0 in the "fast" path so we don't need the code for mode != 0
pc_block( p->mMixBufferU, p->mPredictorU, numSamples, coefsU[numU - 1], numU, chanBits, DENSHIFT_DEFAULT );
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples, numSamples, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorU, bitstream, numSamples, chanBits, &bits1 );
RequireNoErr( status, goto Exit; );
// run the dynamic predictor and lossless compression for the "right" channel
pc_block( p->mMixBufferV, p->mPredictorV, numSamples, coefsV[numV - 1], numV, chanBits, DENSHIFT_DEFAULT );
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples, numSamples, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorV, bitstream, numSamples, chanBits, &bits2 );
RequireNoErr( status, goto Exit; );
// do bit requirement calculations
minBits1 = bits1 + (numU * sizeof(int16_t) * 8);
minBits2 = bits2 + (numV * sizeof(int16_t) * 8);
// test for escape hatch if best calculated compressed size turns out to be more than the input size
minBits = minBits1 + minBits2 + (8 /* mixRes/maxRes/etc. */ * 8) + ((partialFrame == true) ? 32 : 0);
if ( bytesShifted != 0 )
minBits += (numSamples * (bytesShifted * 8) * 2);
escapeBits = (numSamples * p->mBitDepth * 2) + ((partialFrame == true) ? 32 : 0) + (2 * 8); /* 2 common header bytes */
doEscape = (minBits >= escapeBits) ? true : false;
if ( doEscape == false )
{
/* if we happened to create a compressed packet that was actually bigger than an escape packet would be,
chuck it and do an escape packet
*/
minBits = BitBufferGetPosition( bitstream ) - BitBufferGetPosition( &startBits );
if ( minBits >= escapeBits )
{
doEscape = true;
printf( "compressed frame too big: %u vs. %u\n", minBits, escapeBits );
}
}
if ( doEscape == true )
{
/* escape */
// reset bitstream position since we speculatively wrote the compressed version
*bitstream = startBits;
// write escape frame
status = EncodeStereoEscape(p, bitstream, inputBuffer, stride, numSamples );
#if VERBOSE_DEBUG
DebugMsg( "escape!: %u vs %u\n", minBits, (numSamples * p->mBitDepth * 2) );
#endif
}
Exit:
return status;
}
/*
EncodeStereoEscape()
- encode stereo escape frame
*/
static int32_t
EncodeStereoEscape(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * inputBuffer, uint32_t stride, uint32_t numSamples )
{
uint8_t partialFrame;
uint32_t indx;
// flag whether or not this is a partial frame
partialFrame = (numSamples == p->mFrameSize) ? 0 : 1;
// write bitstream header
BitBufferWrite( bitstream, 0, 12 );
BitBufferWrite( bitstream, (partialFrame << 3) | 1, 4 ); // LSB = 1 means "frame not compressed"
if ( partialFrame )
BitBufferWrite( bitstream, numSamples, 32 );
// just copy the input data to the output buffer
switch ( p->mBitDepth )
{
case 16:
for ( indx = 0; indx < (numSamples * stride); indx += stride )
{
BitBufferWrite( bitstream, inputBuffer[indx + 0] >> 16, 16 );
BitBufferWrite( bitstream, inputBuffer[indx + 1] >> 16, 16 );
}
break;
case 20:
for ( indx = 0; indx < (numSamples * stride); indx += stride )
{
BitBufferWrite( bitstream, inputBuffer[indx + 0] >> 12, 16 );
BitBufferWrite( bitstream, inputBuffer[indx + 1] >> 12, 16 );
}
break;
case 24:
// mix24() with mixres param = 0 means de-interleave so use it to simplify things
mix24( inputBuffer, stride, p->mMixBufferU, p->mMixBufferV, numSamples, 0, 0, p->mShiftBufferUV, 0 );
for ( indx = 0; indx < numSamples; indx++ )
{
BitBufferWrite( bitstream, p->mMixBufferU[indx] >> 8, 24 );
BitBufferWrite( bitstream, p->mMixBufferV[indx] >> 8, 24 );
}
break;
case 32:
for ( indx = 0; indx < (numSamples * stride); indx += stride )
{
BitBufferWrite( bitstream, inputBuffer[indx + 0], 32 );
BitBufferWrite( bitstream, inputBuffer[indx + 1], 32 );
}
break;
}
return ALAC_noErr;
}
/*
EncodeMono()
- encode a mono input buffer
*/
static int32_t
EncodeMono(ALAC_ENCODER *p, struct BitBuffer * bitstream, int32_t * inputBuffer, uint32_t stride, uint32_t channelIndex, uint32_t numSamples )
{
BitBuffer startBits = *bitstream; // squirrel away copy of current state in case we need to go back and do an escape packet
AGParamRec agParams;
uint32_t bits1;
uint32_t numU;
SearchCoefs coefsU;
uint32_t dilate;
uint32_t minBits, bestU;
uint32_t minU, maxU;
uint32_t indx, indx2;
uint8_t bytesShifted;
uint32_t shift;
uint32_t mask;
uint32_t chanBits;
uint8_t pbFactor;
uint8_t partialFrame;
uint32_t escapeBits;
bool doEscape;
int32_t status = ALAC_noErr;
uint32_t converge;
// make sure we handle this bit-depth before we get going
RequireAction( (p->mBitDepth == 16) || (p->mBitDepth == 20) || (p->mBitDepth == 24) || (p->mBitDepth == 32), return kALAC_ParamError; );
// reload coefs array from previous frame
coefsU = (SearchCoefs) p->mCoefsU[channelIndex];
// pick bit depth for actual encoding
// - we lop off the lower byte(s) for 24-/32-bit encodings
if ( p->mBitDepth == 32 )
bytesShifted = 2;
else if ( p->mBitDepth >= 24 )
bytesShifted = 1;
else
bytesShifted = 0;
shift = bytesShifted * 8;
mask = (1ul << shift) - 1;
chanBits = p->mBitDepth - (bytesShifted * 8);
// flag whether or not this is a partial frame
partialFrame = (numSamples == p->mFrameSize) ? 0 : 1;
// convert N-bit data to 32-bit for predictor
switch ( p->mBitDepth )
{
case 16:
// convert 16-bit data to 32-bit for predictor
for ( indx = 0, indx2 = 0; indx < numSamples; indx++, indx2 += stride )
p->mMixBufferU[indx] = inputBuffer[indx2] >> 16;
break;
case 20:
// convert 20-bit data to 32-bit for predictor
for ( indx = 0, indx2 = 0; indx < numSamples; indx++, indx2 += stride )
p->mMixBufferU[indx] = inputBuffer[indx2] >> 12;
break;
case 24:
// convert 24-bit data to 32-bit for the predictor and extract the shifted off byte(s)
for ( indx = 0, indx2 = 0; indx < numSamples; indx++, indx2 += stride )
{
p->mMixBufferU[indx] = inputBuffer[indx2] >> 8;
p->mShiftBufferUV[indx] = (uint16_t)(p->mMixBufferU[indx] & mask);
p->mMixBufferU[indx] >>= shift;
}
break;
case 32:
// just copy the 32-bit input data for the predictor and extract the shifted off byte(s)
for ( indx = 0, indx2 = 0; indx < numSamples; indx++, indx2 += stride )
{
p->mShiftBufferUV[indx] = (uint16_t)(inputBuffer[indx2] & mask);
p->mMixBufferU[indx] = inputBuffer[indx2] >> shift;
}
break;
}
// brute-force encode optimization loop (implied "encode depth" of 0 if comparing to cmd line tool)
// - run over variations of the encoding params to find the best choice
minU = 4;
maxU = 8;
minBits = 1ul << 31;
pbFactor = 4;
bestU = minU;
for ( numU = minU; numU <= maxU; numU += 4 )
{
BitBuffer workBits;
uint32_t numBits;
BitBufferInit( &workBits, p->mWorkBuffer, p->mMaxOutputBytes );
dilate = 32;
for ( converge = 0; converge < 7; converge++ )
pc_block( p->mMixBufferU, p->mPredictorU, numSamples/dilate, coefsU[numU-1], numU, chanBits, DENSHIFT_DEFAULT );
dilate = 8;
pc_block( p->mMixBufferU, p->mPredictorU, numSamples/dilate, coefsU[numU-1], numU, chanBits, DENSHIFT_DEFAULT );
set_ag_params( &agParams, MB0, (pbFactor * PB0) / 4, KB0, numSamples/dilate, numSamples/dilate, MAX_RUN_DEFAULT );
status = dyn_comp( &agParams, p->mPredictorU, &workBits, numSamples/dilate, chanBits, &bits1 );
RequireNoErr( status, goto Exit; );
numBits = (dilate * bits1) + (16 * numU);
if ( numBits < minBits )
{
bestU = numU;
minBits = numBits;
}
}
// test for escape hatch if best calculated compressed size turns out to be more than the input size
// - first, add bits for the header bytes mixRes/maxRes/shiftU/filterU
minBits += (4 /* mixRes/maxRes/etc. */ * 8) + ((partialFrame == true) ? 32 : 0);
if ( bytesShifted != 0 )
minBits += (numSamples * (bytesShifted * 8));
escapeBits = (numSamples * p->mBitDepth) + ((partialFrame == true) ? 32 : 0) + (2 * 8); /* 2 common header bytes */
doEscape = (minBits >= escapeBits) ? true : false;
if ( doEscape == false )
{
// write bitstream header
BitBufferWrite( bitstream, 0, 12 );
BitBufferWrite( bitstream, (partialFrame << 3) | (bytesShifted << 1), 4 );
if ( partialFrame )
BitBufferWrite( bitstream, numSamples, 32 );
BitBufferWrite( bitstream, 0, 16 ); // mixBits = mixRes = 0
// write the params and predictor coefs
numU = bestU;
BitBufferWrite( bitstream, (0 << 4) | DENSHIFT_DEFAULT, 8 ); // modeU = 0
BitBufferWrite( bitstream, (pbFactor << 5) | numU, 8 );
for ( indx = 0; indx < numU; indx++ )
BitBufferWrite( bitstream, coefsU[numU-1][indx], 16 );
// if shift active, write the interleaved shift buffers
if ( bytesShifted != 0 )
{
for ( indx = 0; indx < numSamples; indx++ )
BitBufferWrite( bitstream, p->mShiftBufferUV[indx], shift );
}
// run the dynamic predictor with the best result
pc_block( p->mMixBufferU, p->mPredictorU, numSamples, coefsU[numU-1], numU, chanBits, DENSHIFT_DEFAULT );
// do lossless compression
set_standard_ag_params( &agParams, numSamples, numSamples );
status = dyn_comp( &agParams, p->mPredictorU, bitstream, numSamples, chanBits, &bits1 );
//AssertNoErr( status );
/* if we happened to create a compressed packet that was actually bigger than an escape packet would be,
chuck it and do an escape packet
*/
minBits = BitBufferGetPosition( bitstream ) - BitBufferGetPosition( &startBits );
if ( minBits >= escapeBits )
{
*bitstream = startBits; // reset bitstream state
doEscape = true;
printf( "compressed frame too big: %u vs. %u\n", minBits, escapeBits );
}
}
if ( doEscape == true )
{
// write bitstream header and coefs
BitBufferWrite( bitstream, 0, 12 );
BitBufferWrite( bitstream, (partialFrame << 3) | 1, 4 ); // LSB = 1 means "frame not compressed"
if ( partialFrame )
BitBufferWrite( bitstream, numSamples, 32 );
// just copy the input data to the output buffer
switch ( p->mBitDepth )
{
case 16:
for ( indx = 0; indx < (numSamples * stride); indx += stride )
BitBufferWrite( bitstream, inputBuffer[indx] >> 16, 16 );
break;
case 20:
// convert 20-bit data to 32-bit for simplicity
for ( indx = 0; indx < (numSamples * stride); indx += stride )
BitBufferWrite( bitstream, inputBuffer[indx] >> 12, 20 );
break;
case 24:
// convert 24-bit data to 32-bit for simplicity
for ( indx = 0, indx2 = 0; indx < numSamples; indx++, indx2 += stride )
{
p->mMixBufferU[indx] = inputBuffer[indx2] >> 8;
BitBufferWrite( bitstream, p->mMixBufferU[indx], 24 );
}
break;
case 32:
for ( indx = 0; indx < (numSamples * stride); indx += stride )
BitBufferWrite( bitstream, inputBuffer[indx], 32 );
break;
}
#if VERBOSE_DEBUG
DebugMsg( "escape!: %u vs %u\n", minBits, (numSamples * p->mBitDepth) );
#endif
}
Exit:
return status;
}
#if PRAGMA_MARK
#pragma mark -
#endif
/*
Encode()
- encode the next block of samples
*/
int32_t
alac_encode(ALAC_ENCODER *p, uint32_t numChannels, uint32_t numSamples,
int32_t * theReadBuffer, unsigned char * theWriteBuffer, uint32_t * ioNumBytes)
{
uint32_t outputSize;
BitBuffer bitstream;
int32_t status;
// create a bit buffer structure pointing to our output buffer
BitBufferInit( &bitstream, theWriteBuffer, p->mMaxOutputBytes );
if ( numChannels == 2 )
{
// add 3-bit frame start tag ID_CPE = channel pair & 4-bit element instance tag = 0
BitBufferWrite( &bitstream, ID_CPE, 3 );
BitBufferWrite( &bitstream, 0, 4 );
// encode stereo input buffer
if ( p->mFastMode == false )
status = EncodeStereo(p, &bitstream, theReadBuffer, 2, 0, numSamples );
else
status = EncodeStereoFast(p, &bitstream, theReadBuffer, 2, 0, numSamples );
RequireNoErr( status, goto Exit; );
}
else if ( numChannels == 1 )
{
// add 3-bit frame start tag ID_SCE = mono channel & 4-bit element instance tag = 0
BitBufferWrite( &bitstream, ID_SCE, 3 );
BitBufferWrite( &bitstream, 0, 4 );
// encode mono input buffer
status = EncodeMono(p, &bitstream, theReadBuffer, 1, 0, numSamples );
RequireNoErr( status, goto Exit; );
}
else
{
int32_t * inputBuffer;
uint32_t tag;
uint32_t channelIndex;
uint32_t inputIncrement;
uint8_t stereoElementTag;
uint8_t monoElementTag;
uint8_t lfeElementTag;
inputBuffer = theReadBuffer;
inputIncrement = ((p->mBitDepth + 7) / 8);
stereoElementTag = 0;
monoElementTag = 0;
lfeElementTag = 0;
for ( channelIndex = 0; channelIndex < numChannels; )
{
tag = (sChannelMaps[numChannels - 1] & (0x7ul << (channelIndex * 3))) >> (channelIndex * 3);
BitBufferWrite( &bitstream, tag, 3 );
switch ( tag )
{
case ID_SCE:
// mono
BitBufferWrite( &bitstream, monoElementTag, 4 );
status = EncodeMono(p, &bitstream, inputBuffer, numChannels, channelIndex, numSamples );
inputBuffer += inputIncrement;
channelIndex++;
monoElementTag++;
break;
case ID_CPE:
// stereo
BitBufferWrite( &bitstream, stereoElementTag, 4 );
status = EncodeStereo(p,&bitstream, inputBuffer, numChannels, channelIndex, numSamples );
inputBuffer += (inputIncrement * 2);
channelIndex += 2;
stereoElementTag++;
break;
case ID_LFE:
// LFE channel (subwoofer)
BitBufferWrite( &bitstream, lfeElementTag, 4 );
status = EncodeMono(p, &bitstream, inputBuffer, numChannels, channelIndex, numSamples );
inputBuffer += inputIncrement;
channelIndex++;
lfeElementTag++;
break;
default:
printf( "That ain't right! (%u)\n", tag );
status = kALAC_ParamError;
goto Exit;
}
RequireNoErr( status, goto Exit; );
}
}
#if VERBOSE_DEBUG
{
// if there is room left in the output buffer, add some random fill data to test decoder
int32_t bitsLeft;
int32_t bytesLeft;
bitsLeft = BitBufferGetPosition( &bitstream ) - 3; // - 3 for ID_END tag
bytesLeft = bitstream.byteSize - ((bitsLeft + 7) / 8);
if ( (bytesLeft > 20) && ((bytesLeft & 0x4u) != 0) )
AddFiller( &bitstream, bytesLeft );
}
#endif
// add 3-bit frame end tag: ID_END
BitBufferWrite( &bitstream, ID_END, 3 );
// byte-align the output data
BitBufferByteAlign( &bitstream, true );
outputSize = BitBufferGetPosition( &bitstream ) / 8;
//Assert( outputSize <= mMaxOutputBytes );
// all good, let iTunes know what happened and remember the total number of input sample frames
*ioNumBytes = outputSize;
//mEncodedFrames += encodeMsg->numInputSamples;
// gather encoding stats
p->mTotalBytesGenerated += outputSize;
p->mMaxFrameBytes = MAX( p->mMaxFrameBytes, outputSize );
status = ALAC_noErr;
Exit:
return status;
}
#if PRAGMA_MARK
#pragma mark -
#endif
/*
GetConfig()
*/
void
GetConfig(ALAC_ENCODER *p, ALACSpecificConfig * config )
{
config->frameLength = Swap32NtoB(p->mFrameSize);
config->compatibleVersion = (uint8_t) kALACCompatibleVersion;
config->bitDepth = (uint8_t) p->mBitDepth;
config->pb = (uint8_t) PB0;
config->kb = (uint8_t) KB0;
config->mb = (uint8_t) MB0;
config->numChannels = (uint8_t) p->mNumChannels;
config->maxRun = Swap16NtoB((uint16_t) MAX_RUN_DEFAULT);
config->maxFrameBytes = Swap32NtoB(p->mMaxFrameBytes);
config->avgBitRate = Swap32NtoB(p->mAvgBitRate);
config->sampleRate = Swap32NtoB(p->mOutputSampleRate);
}
uint32_t
alac_get_magic_cookie_size(uint32_t inNumChannels)
{
if (inNumChannels > 2)
{
return sizeof(ALACSpecificConfig) + kChannelAtomSize + sizeof(ALACAudioChannelLayout);
}
else
{
return sizeof(ALACSpecificConfig);
}
}
void
alac_get_magic_cookie(ALAC_ENCODER *p, void * outCookie, uint32_t * ioSize )
{
ALACSpecificConfig theConfig = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};
ALACAudioChannelLayout theChannelLayout = {0, 0, 0};
uint8_t theChannelAtom[kChannelAtomSize] = {0, 0, 0, 0, 'c', 'h', 'a', 'n', 0, 0, 0, 0};
uint32_t theCookieSize = sizeof(ALACSpecificConfig);
uint8_t * theCookiePointer = (uint8_t *)outCookie;
GetConfig(p, &theConfig);
if (theConfig.numChannels > 2)
{
theChannelLayout.mChannelLayoutTag = Swap32NtoB(ALACChannelLayoutTags[theConfig.numChannels - 1]);
theCookieSize += (sizeof(ALACAudioChannelLayout) + kChannelAtomSize);
}
if (*ioSize >= theCookieSize)
{
memcpy(theCookiePointer, &theConfig, sizeof(ALACSpecificConfig));
theChannelAtom[3] = (sizeof(ALACAudioChannelLayout) + kChannelAtomSize);
if (theConfig.numChannels > 2)
{
theCookiePointer += sizeof(ALACSpecificConfig);
memcpy(theCookiePointer, theChannelAtom, kChannelAtomSize);
theCookiePointer += kChannelAtomSize;
memcpy(theCookiePointer, &theChannelLayout, sizeof(ALACAudioChannelLayout));
}
*ioSize = theCookieSize;
}
else
{
*ioSize = 0; // no incomplete cookies
}
}
/*
alac_encoder_init()
- initialize the encoder component with the current config
*/
int32_t
alac_encoder_init (ALAC_ENCODER *p, uint32_t samplerate, uint32_t channels, uint32_t format_flags, uint32_t frameSize)
{
int32_t status;
uint32_t indx;
int32_t channel, search;
p->mFrameSize = (frameSize > 0 && frameSize <= ALAC_FRAME_LENGTH) ? frameSize : ALAC_FRAME_LENGTH ;
p->mOutputSampleRate = samplerate;
p->mNumChannels = channels;
switch (format_flags)
{
case 1:
p->mBitDepth = 16;
break;
case 2:
p->mBitDepth = 20;
break;
case 3:
p->mBitDepth = 24;
break;
case 4:
p->mBitDepth = 32;
break;
default:
break;
}
// set up default encoding parameters and state
// - note: mFrameSize is set in the constructor or via alac_set_frame_size() which must be called before this routine
for ( indx = 0; indx < kALACMaxChannels; indx++ )
p->mLastMixRes[indx] = kDefaultMixRes;
// the maximum output frame size can be no bigger than (samplesPerBlock * numChannels * ((10 + sampleSize)/8) + 1)
// but note that this can be bigger than the input size!
// - since we don't yet know what our input format will be, use our max allowed sample size in the calculation
p->mMaxOutputBytes = p->mFrameSize * p->mNumChannels * ((10 + kMaxSampleSize) / 8) + 1;
status = ALAC_noErr;
// initialize coefs arrays once b/c retaining state across blocks actually improves the encode ratio
for ( channel = 0; channel < (int32_t) p->mNumChannels; channel++ )
{
for ( search = 0; search < kALACMaxSearches; search++ )
{
init_coefs( p->mCoefsU[channel][search], DENSHIFT_DEFAULT, kALACMaxCoefs );
init_coefs( p->mCoefsV[channel][search], DENSHIFT_DEFAULT, kALACMaxCoefs );
}
}
return status;
}
/*
alac_get_source_format()
- given the input format, return one of our supported formats
*/
void
alac_get_source_format(ALAC_ENCODER *p, const AudioFormatDescription * source, AudioFormatDescription * output )
{
(void) output ;
// default is 16-bit native endian
// - note: for float input we assume that's coming from one of our decoders (mp3, aac) so it only makes sense
// to encode to 16-bit since the source was lossy in the first place
// - note: if not a supported bit depth, find the closest supported bit depth to the input one
if ( (source->mFormatID != kALACFormatLinearPCM) || ((source->mFormatFlags & kALACFormatFlagIsFloat) != 0) ||
( source->mBitsPerChannel <= 16 ) )
p->mBitDepth = 16;
else if ( source->mBitsPerChannel <= 20 )
p->mBitDepth = 20;
else if ( source->mBitsPerChannel <= 24 )
p->mBitDepth = 24;
else
p->mBitDepth = 32;
// we support 16/20/24/32-bit integer data at any sample rate and our target number of channels
// and sample rate were specified when we were configured
/*
MakeUncompressedAudioFormat( mNumChannels, (float) mOutputSampleRate, mBitDepth, kAudioFormatFlagsNativeIntegerPacked, output );
*/
}
#if VERBOSE_DEBUG
#if PRAGMA_MARK
#pragma mark -
#endif
/*
AddFiller()
- add fill and data stream elements to the bitstream to test the decoder
*/
static void AddFiller( BitBuffer * bits, int32_t numBytes )
{
uint8_t tag;
int32_t indx;
// out of lameness, subtract 6 bytes to deal with header + alignment as required for fill/data elements
numBytes -= 6;
if ( numBytes <= 0 )
return;
// randomly pick Fill or Data Stream Element based on numBytes requested
tag = (numBytes & 0x8) ? ID_FIL : ID_DSE;
BitBufferWrite( bits, tag, 3 );
if ( tag == ID_FIL )
{
// can't write more than 269 bytes in a fill element
numBytes = (numBytes > 269) ? 269 : numBytes;
// fill element = 4-bit size unless >= 15 then 4-bit size + 8-bit extension size
if ( numBytes >= 15 )
{
uint16_t extensionSize;
BitBufferWrite( bits, 15, 4 );
// 8-bit extension count field is "extra + 1" which is weird but I didn't define the syntax
// - otherwise, there's no way to represent 15
// - for example, to really mean 15 bytes you must encode extensionSize = 1
// - why it's not like data stream elements I have no idea
extensionSize = (numBytes - 15) + 1;
//Assert( extensionSize <= 255 );
BitBufferWrite( bits, extensionSize, 8 );
}
else
BitBufferWrite( bits, numBytes, 4 );
BitBufferWrite( bits, 0x10, 8 ); // extension_type = FILL_DATA = b0001 or'ed with fill_nibble = b0000
for ( indx = 0; indx < (numBytes - 1); indx++ )
BitBufferWrite( bits, 0xa5, 8 ); // fill_byte = b10100101 = 0xa5
}
else
{
// can't write more than 510 bytes in a data stream element
numBytes = (numBytes > 510) ? 510 : numBytes;
BitBufferWrite( bits, 0, 4 ); // element instance tag
BitBufferWrite( bits, 1, 1 ); // byte-align flag = true
// data stream element = 8-bit size unless >= 255 then 8-bit size + 8-bit size
if ( numBytes >= 255 )
{
BitBufferWrite( bits, 255, 8 );
BitBufferWrite( bits, numBytes - 255, 8 );
}
else
BitBufferWrite( bits, numBytes, 8 );
BitBufferByteAlign( bits, true ); // byte-align with zeros
for ( indx = 0; indx < numBytes; indx++ )
BitBufferWrite( bits, 0x5a, 8 );
}
}
#endif /* VERBOSE_DEBUG */
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