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https://github.com/mstorsjo/fdk-aac.git
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4f0d97057c
- Update disclaimer in all source files. - Increment library version in each FDK sub-module. Bug 9428126 Change-Id: I490b96d4ee472246b01483202b0bb4f1e9c2a5d7
883 lines
27 KiB
C++
883 lines
27 KiB
C++
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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/***************************** MPEG-4 AAC Encoder **************************
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Author(s):
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Description:
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******************************************************************************/
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#include "tpenc_latm.h"
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#include "genericStds.h"
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static const short celpFrameLengthTable[64] = {
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154, 170, 186, 147, 156, 165, 114, 120,
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186, 126, 132, 138, 142, 146, 154, 166,
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174, 182, 190, 198, 206, 210, 214, 110,
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114, 118, 120, 122, 218, 230, 242, 254,
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266, 278, 286, 294, 318, 342, 358, 374,
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390, 406, 422, 136, 142, 148, 154, 160,
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166, 170, 174, 186, 198, 206, 214, 222,
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230, 238, 216, 160, 280, 338, 0, 0
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};
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/*******
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write value to transport stream
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first two bits define the size of the value itself
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then the value itself, with a size of 0-3 bytes
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*******/
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static
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UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value)
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{
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UCHAR valueBytes = 4;
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unsigned int bitsWritten = 0;
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int i;
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if ( value < (1<<8) ) {
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valueBytes = 1;
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} else if ( value < (1<<16) ) {
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valueBytes = 2;
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} else if ( value < (1<<24) ) {
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valueBytes = 3;
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} else {
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valueBytes = 4;
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}
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FDKwriteBits(hBs, valueBytes-1, 2 ); /* size of value in Bytes */
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for (i=0; i<valueBytes; i++) {
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/* write most significant Byte first */
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FDKwriteBits(hBs, (UCHAR)(value>>((valueBytes-1-i)<<3)), 8);
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}
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bitsWritten = (valueBytes<<3)+2;
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return bitsWritten;
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}
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static
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UINT transportEnc_LatmCountFixBitDemandHeader ( HANDLE_LATM_STREAM hAss )
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{
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int bitDemand = 0;
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int insertSetupData = 0 ;
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/* only if start of new latm frame */
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if (hAss->subFrameCnt==0)
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{
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/* AudioSyncStream */
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if (hAss->tt == TT_MP4_LOAS) {
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bitDemand += 11 ; /* syncword */
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bitDemand += 13 ; /* audioMuxLengthBytes */
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}
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/* AudioMuxElement*/
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/* AudioMuxElement::Stream Mux Config */
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if (hAss->muxConfigPeriod > 0) {
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insertSetupData = (hAss->latmFrameCounter == 0);
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} else {
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insertSetupData = 0;
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}
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if (hAss->tt != TT_MP4_LATM_MCP0) {
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/* AudioMuxElement::useSameStreamMux Flag */
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bitDemand+=1;
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if( insertSetupData ) {
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bitDemand += hAss->streamMuxConfigBits;
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}
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}
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/* AudioMuxElement::otherDataBits */
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bitDemand += 8*hAss->otherDataLenBytes;
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/* AudioMuxElement::ByteAlign */
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if ( bitDemand % 8 ) {
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hAss->fillBits = 8 - (bitDemand % 8);
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bitDemand += hAss->fillBits ;
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} else {
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hAss->fillBits = 0;
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}
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}
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return bitDemand ;
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}
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static
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UINT transportEnc_LatmCountVarBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
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{
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int bitDemand = 0;
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int prog, layer;
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/* Payload Length Info*/
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if( hAss->allStreamsSameTimeFraming ) {
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for( prog=0; prog<hAss->noProgram; prog++ ) {
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for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
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LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
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if( p_linfo->streamID >= 0 ) {
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switch( p_linfo->frameLengthType ) {
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case 0:
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if ( streamDataLength > 0 ) {
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streamDataLength -= bitDemand ;
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while( streamDataLength >= (255<<3) ) {
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bitDemand+=8;
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streamDataLength -= (255<<3);
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}
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bitDemand += 8;
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}
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break;
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case 1:
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case 4:
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case 6:
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bitDemand += 2;
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break;
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default:
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return 0;
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}
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}
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}
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}
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} else {
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/* there are many possibilities to use this mechanism. */
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switch( hAss->varMode ) {
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case LATMVAR_SIMPLE_SEQUENCE: {
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/* Use the sequence generated by the encoder */
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//int streamCntPosition = transportEnc_SetWritePointer( hAss->hAssemble, 0 );
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//int streamCntPosition = FDKgetValidBits( hAss->hAssemble );
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bitDemand+=4;
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hAss->varStreamCnt = 0;
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for( prog=0; prog<hAss->noProgram; prog++ ) {
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for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
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LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
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if( p_linfo->streamID >= 0 ) {
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bitDemand+=4; /* streamID */
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switch( p_linfo->frameLengthType ) {
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case 0:
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streamDataLength -= bitDemand ;
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while( streamDataLength >= (255<<3) ) {
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bitDemand+=8;
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streamDataLength -= (255<<3);
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}
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bitDemand += 8;
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break;
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/*bitDemand += 1; endFlag
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break;*/
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case 1:
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case 4:
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case 6:
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break;
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default:
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return 0;
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}
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hAss->varStreamCnt++;
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}
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}
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}
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bitDemand+=4;
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//transportEnc_UpdateBitstreamField( hAss->hAssemble, streamCntPosition, hAss->varStreamCnt-1, 4 );
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//UINT pos = streamCntPosition-FDKgetValidBits(hAss->hAssemble);
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//FDKpushBack( hAss->hAssemble, pos);
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//FDKwriteBits( hAss->hAssemble, hAss->varStreamCnt-1, 4);
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//FDKpushFor( hAss->hAssemble, pos-4);
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}
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break;
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default:
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return 0;
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}
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}
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return bitDemand ;
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}
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TRANSPORTENC_ERROR
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CreateStreamMuxConfig(
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HANDLE_LATM_STREAM hAss,
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HANDLE_FDK_BITSTREAM hBs,
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int bufferFullness,
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CSTpCallBacks *cb
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)
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{
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INT streamIDcnt, tmp;
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int layer, prog;
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USHORT coreFrameOffset=0;
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hAss->audioMuxVersionA = 0; /* for future extensions */
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hAss->streamMuxConfigBits = 0;
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FDKwriteBits( hBs, hAss->audioMuxVersion, 1 ); /* audioMuxVersion */
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hAss->streamMuxConfigBits += 1;
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if ( hAss->audioMuxVersion == 1 ) {
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FDKwriteBits( hBs, hAss->audioMuxVersionA, 1 ); /* audioMuxVersionA */
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hAss->streamMuxConfigBits+=1;
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}
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if ( hAss->audioMuxVersionA == 0 )
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{
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if ( hAss->audioMuxVersion == 1 ) {
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hAss->streamMuxConfigBits+= transportEnc_LatmWriteValue( hBs, hAss->taraBufferFullness );/* taraBufferFullness */
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}
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FDKwriteBits( hBs, hAss->allStreamsSameTimeFraming ? 1:0, 1 ); /* allStreamsSameTimeFraming */
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FDKwriteBits( hBs, hAss->noSubframes-1, 6 ); /* Number of Subframes */
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FDKwriteBits( hBs, hAss->noProgram-1, 4 ); /* Number of Programs */
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hAss->streamMuxConfigBits+=11;
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streamIDcnt = 0;
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for( prog=0; prog<hAss->noProgram; prog++ ) {
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int transLayer = 0;
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FDKwriteBits( hBs, hAss->noLayer[prog]-1, 3 );
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hAss->streamMuxConfigBits+=3;
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for( layer=0; layer<LATM_MAX_LAYERS; layer++ ) {
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LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
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CODER_CONFIG *p_lci = hAss->config[prog][layer];
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p_linfo->streamID = -1;
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if( hAss->config[prog][layer] != NULL ) {
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int useSameConfig = 0;
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if( transLayer > 0 ) {
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FDKwriteBits( hBs, useSameConfig ? 1 : 0, 1 );
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hAss->streamMuxConfigBits+=1;
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}
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if( (useSameConfig == 0) || (transLayer==0) ) {
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UINT bits;
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if ( hAss->audioMuxVersion == 1 ) {
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FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
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}
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bits = FDKgetValidBits( hBs );
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transportEnc_writeASC(
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hBs,
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hAss->config[prog][layer],
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cb
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);
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bits = FDKgetValidBits( hBs ) - bits;
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if ( hAss->audioMuxVersion == 1 ) {
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FDKpushBack(hBs, bits+2);
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hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
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transportEnc_writeASC(
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hBs,
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hAss->config[prog][layer],
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cb
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);
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}
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hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */
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}
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transLayer++;
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if( !hAss->allStreamsSameTimeFraming ) {
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if( streamIDcnt >= LATM_MAX_STREAM_ID )
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return TRANSPORTENC_INVALID_CONFIG;
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}
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p_linfo->streamID = streamIDcnt++;
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switch( p_lci->aot ) {
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case AOT_AAC_MAIN :
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case AOT_AAC_LC :
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case AOT_AAC_SSR :
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case AOT_AAC_LTP :
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case AOT_AAC_SCAL :
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case AOT_ER_AAC_LD :
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case AOT_ER_AAC_ELD :
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case AOT_USAC:
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case AOT_RSVD50:
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p_linfo->frameLengthType = 0;
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FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
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FDKwriteBits( hBs, bufferFullness, 8 ); /* bufferFullness */
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hAss->streamMuxConfigBits+=11;
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if ( !hAss->allStreamsSameTimeFraming ) {
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CODER_CONFIG *p_lci_prev = hAss->config[prog][layer-1];
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if ( ((p_lci->aot == AOT_AAC_SCAL) || (p_lci->aot == AOT_ER_AAC_SCAL)) &&
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((p_lci_prev->aot == AOT_CELP) || (p_lci_prev->aot == AOT_ER_CELP)) ) {
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FDKwriteBits( hBs, coreFrameOffset, 6 ); /* coreFrameOffset */
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hAss->streamMuxConfigBits+=6;
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}
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}
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break;
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case AOT_TWIN_VQ:
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p_linfo->frameLengthType = 1;
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tmp = ( (p_lci->bitsFrame+7) >> 3 ) - 20; /* transmission frame length in bytes */
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if( (tmp < 0) ) {
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return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH;
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}
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FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
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FDKwriteBits( hBs, tmp, 9 );
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hAss->streamMuxConfigBits+=12;
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p_linfo->frameLengthBits = (tmp+20) << 3;
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break;
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case AOT_CELP:
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p_linfo->frameLengthType = 4;
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FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
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hAss->streamMuxConfigBits+=3;
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{
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int i;
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for( i=0; i<62; i++ ) {
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if( celpFrameLengthTable[i] == p_lci->bitsFrame )
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break;
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}
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if( i>=62 ) {
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return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH;
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}
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FDKwriteBits( hBs, i, 6 ); /* CELPframeLengthTabelIndex */
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hAss->streamMuxConfigBits+=6;
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}
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p_linfo->frameLengthBits = p_lci->bitsFrame;
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break;
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case AOT_HVXC:
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p_linfo->frameLengthType = 6;
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FDKwriteBits( hBs, p_linfo->frameLengthType, 3 ); /* frameLengthType */
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hAss->streamMuxConfigBits+=3;
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{
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int i;
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if( p_lci->bitsFrame == 40 ) {
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i = 0;
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} else if( p_lci->bitsFrame == 80 ) {
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i = 1;
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} else {
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return TRANSPORTENC_INVALID_FRAME_BITS;
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}
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FDKwriteBits( hBs, i, 1 ); /* HVXCframeLengthTableIndex */
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hAss->streamMuxConfigBits+=1;
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}
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p_linfo->frameLengthBits = p_lci->bitsFrame;
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break;
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case AOT_NULL_OBJECT:
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default:
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return TRANSPORTENC_INVALID_AOT;
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}
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}
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}
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}
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FDKwriteBits( hBs, (hAss->otherDataLenBytes>0) ? 1:0, 1 ); /* otherDataPresent */
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hAss->streamMuxConfigBits+=1;
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if( hAss->otherDataLenBytes > 0 ) {
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INT otherDataLenTmp = hAss->otherDataLenBytes;
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INT escCnt = 0;
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INT otherDataLenEsc = 1;
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while(otherDataLenTmp) {
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otherDataLenTmp >>= 8;
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escCnt ++;
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}
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do {
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otherDataLenTmp = (hAss->otherDataLenBytes>>(escCnt*8)) & 0xFF;
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escCnt--;
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otherDataLenEsc = escCnt>0;
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FDKwriteBits( hBs, otherDataLenEsc, 1 );
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FDKwriteBits( hBs, otherDataLenTmp, 8 );
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hAss->streamMuxConfigBits+=9;
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} while(otherDataLenEsc);
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}
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{
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USHORT crcCheckPresent=0;
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USHORT crcCheckSum=0;
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FDKwriteBits( hBs, crcCheckPresent, 1 ); /* crcCheckPresent */
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hAss->streamMuxConfigBits+=1;
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if ( crcCheckPresent ){
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FDKwriteBits( hBs, crcCheckSum, 8 ); /* crcCheckSum */
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|
hAss->streamMuxConfigBits+=8;
|
|
}
|
|
}
|
|
|
|
} else { /* if ( audioMuxVersionA == 0 ) */
|
|
|
|
/* for future extensions */
|
|
|
|
}
|
|
|
|
return TRANSPORTENC_OK;
|
|
}
|
|
|
|
|
|
static TRANSPORTENC_ERROR
|
|
WriteAuPayloadLengthInfo( HANDLE_FDK_BITSTREAM hBitStream, int AuLengthBits )
|
|
{
|
|
int restBytes;
|
|
|
|
if( AuLengthBits % 8 )
|
|
return TRANSPORTENC_INVALID_AU_LENGTH;
|
|
|
|
while( AuLengthBits >= 255*8 ) {
|
|
FDKwriteBits( hBitStream, 255, 8 ); /* 255 shows incomplete AU */
|
|
AuLengthBits -= (255*8);
|
|
}
|
|
|
|
restBytes = (AuLengthBits) >> 3;
|
|
FDKwriteBits( hBitStream, restBytes, 8 );
|
|
|
|
return TRANSPORTENC_OK;
|
|
}
|
|
|
|
static
|
|
TRANSPORTENC_ERROR transportEnc_LatmSetNrOfSubframes( HANDLE_LATM_STREAM hAss,
|
|
INT noSubframes_next) /* nr of access units / payloads within a latm frame */
|
|
{
|
|
/* sanity chk */
|
|
if (noSubframes_next < 1 || noSubframes_next > MAX_NR_OF_SUBFRAMES) {
|
|
return TRANSPORTENC_LATM_INVALID_NR_OF_SUBFRAMES;
|
|
}
|
|
|
|
hAss->noSubframes_next = noSubframes_next;
|
|
|
|
/* if at start then we can take over the value immediately, otherwise we have to wait for the next SMC */
|
|
if ( (hAss->subFrameCnt == 0) && (hAss->latmFrameCounter == 0) ) {
|
|
hAss->noSubframes = noSubframes_next;
|
|
}
|
|
|
|
return TRANSPORTENC_OK;
|
|
}
|
|
|
|
static
|
|
int allStreamsSameTimeFraming( HANDLE_LATM_STREAM hAss, UCHAR noProgram, UCHAR noLayer[] /* return */ )
|
|
{
|
|
int prog, layer;
|
|
|
|
signed int lastNoSamples = -1;
|
|
signed int minFrameSamples = FDK_INT_MAX;
|
|
signed int maxFrameSamples = 0;
|
|
|
|
signed int highestSamplingRate = -1;
|
|
|
|
for( prog=0; prog<noProgram; prog++ ) {
|
|
noLayer[prog] = 0;
|
|
|
|
for( layer=0; layer<LATM_MAX_LAYERS; layer++ )
|
|
{
|
|
if( hAss->config[prog][layer] != NULL )
|
|
{
|
|
INT hsfSamplesFrame;
|
|
|
|
noLayer[prog]++;
|
|
|
|
if( highestSamplingRate < 0 )
|
|
highestSamplingRate = hAss->config[prog][layer]->samplingRate;
|
|
|
|
hsfSamplesFrame = hAss->config[prog][layer]->samplesPerFrame * highestSamplingRate / hAss->config[prog][layer]->samplingRate;
|
|
|
|
if( hsfSamplesFrame <= minFrameSamples ) minFrameSamples = hsfSamplesFrame;
|
|
if( hsfSamplesFrame >= maxFrameSamples ) maxFrameSamples = hsfSamplesFrame;
|
|
|
|
if( lastNoSamples == -1 ) {
|
|
lastNoSamples = hsfSamplesFrame;
|
|
} else {
|
|
if( hsfSamplesFrame != lastNoSamples ) {
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
/**
|
|
* Initialize LATM/LOAS Stream and add layer 0 at program 0.
|
|
*/
|
|
static
|
|
TRANSPORTENC_ERROR transportEnc_InitLatmStream( HANDLE_LATM_STREAM hAss,
|
|
int fractDelayPresent,
|
|
signed int muxConfigPeriod, /* insert setup data every muxConfigPeriod frames */
|
|
UINT audioMuxVersion,
|
|
TRANSPORT_TYPE tt
|
|
)
|
|
{
|
|
TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
|
|
|
|
if (hAss == NULL)
|
|
return TRANSPORTENC_INVALID_PARAMETER;
|
|
|
|
hAss->tt = tt;
|
|
|
|
hAss->noProgram = 1;
|
|
|
|
hAss->audioMuxVersion = audioMuxVersion;
|
|
|
|
/* Fill noLayer array using hAss->config */
|
|
hAss->allStreamsSameTimeFraming = allStreamsSameTimeFraming( hAss, hAss->noProgram, hAss->noLayer );
|
|
/* Only allStreamsSameTimeFraming==1 is supported */
|
|
FDK_ASSERT(hAss->allStreamsSameTimeFraming);
|
|
|
|
hAss->fractDelayPresent = fractDelayPresent;
|
|
hAss->otherDataLenBytes = 0;
|
|
|
|
hAss->varMode = LATMVAR_SIMPLE_SEQUENCE;
|
|
|
|
/* initialize counters */
|
|
hAss->subFrameCnt = 0;
|
|
hAss->noSubframes = DEFAULT_LATM_NR_OF_SUBFRAMES;
|
|
hAss->noSubframes_next = DEFAULT_LATM_NR_OF_SUBFRAMES;
|
|
|
|
/* sync layer related */
|
|
hAss->audioMuxLengthBytes = 0;
|
|
|
|
hAss->latmFrameCounter = 0;
|
|
hAss->muxConfigPeriod = muxConfigPeriod;
|
|
|
|
return ErrorStatus;
|
|
}
|
|
|
|
|
|
/**
|
|
*
|
|
*/
|
|
UINT transportEnc_LatmCountTotalBitDemandHeader ( HANDLE_LATM_STREAM hAss , unsigned int streamDataLength )
|
|
{
|
|
UINT bitDemand = 0;
|
|
|
|
switch (hAss->tt) {
|
|
case TT_MP4_LOAS:
|
|
case TT_MP4_LATM_MCP0:
|
|
case TT_MP4_LATM_MCP1:
|
|
if (hAss->subFrameCnt == 0) {
|
|
bitDemand = transportEnc_LatmCountFixBitDemandHeader ( hAss );
|
|
}
|
|
bitDemand += transportEnc_LatmCountVarBitDemandHeader ( hAss , streamDataLength /*- bitDemand*/);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return bitDemand;
|
|
}
|
|
|
|
static TRANSPORTENC_ERROR
|
|
AdvanceAudioMuxElement (
|
|
HANDLE_LATM_STREAM hAss,
|
|
HANDLE_FDK_BITSTREAM hBs,
|
|
int auBits,
|
|
int bufferFullness,
|
|
CSTpCallBacks *cb
|
|
)
|
|
{
|
|
TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
|
|
int insertMuxSetup;
|
|
|
|
/* Insert setup data to assemble Buffer */
|
|
if (hAss->subFrameCnt == 0)
|
|
{
|
|
if (hAss->muxConfigPeriod > 0) {
|
|
insertMuxSetup = (hAss->latmFrameCounter == 0);
|
|
} else {
|
|
insertMuxSetup = 0;
|
|
}
|
|
|
|
if (hAss->tt != TT_MP4_LATM_MCP0) {
|
|
if( insertMuxSetup ) {
|
|
FDKwriteBits( hBs, 0, 1 ); /* useSameStreamMux useNewStreamMux */
|
|
CreateStreamMuxConfig(hAss, hBs, bufferFullness, cb);
|
|
if (ErrorStatus != TRANSPORTENC_OK)
|
|
return ErrorStatus;
|
|
} else {
|
|
FDKwriteBits( hBs, 1, 1 ); /* useSameStreamMux */
|
|
}
|
|
}
|
|
}
|
|
|
|
/* PayloadLengthInfo */
|
|
{
|
|
int prog, layer;
|
|
|
|
for (prog = 0; prog < hAss->noProgram; prog++) {
|
|
for (layer = 0; layer < hAss->noLayer[prog]; layer++) {
|
|
ErrorStatus = WriteAuPayloadLengthInfo( hBs, auBits );
|
|
if (ErrorStatus != TRANSPORTENC_OK)
|
|
return ErrorStatus;
|
|
}
|
|
}
|
|
}
|
|
/* At this point comes the access unit. */
|
|
|
|
return TRANSPORTENC_OK;
|
|
}
|
|
|
|
TRANSPORTENC_ERROR
|
|
transportEnc_LatmWrite (
|
|
HANDLE_LATM_STREAM hAss,
|
|
HANDLE_FDK_BITSTREAM hBs,
|
|
int auBits,
|
|
int bufferFullness,
|
|
CSTpCallBacks *cb
|
|
)
|
|
{
|
|
TRANSPORTENC_ERROR ErrorStatus;
|
|
|
|
if (hAss->subFrameCnt == 0) {
|
|
/* Start new frame */
|
|
FDKresetBitbuffer(hBs, BS_WRITER);
|
|
}
|
|
|
|
hAss->latmSubframeStart = FDKgetValidBits(hBs);
|
|
|
|
/* Insert syncword and syncword distance
|
|
- only if loas
|
|
- we must update the syncword distance (=audiomuxlengthbytes) later
|
|
*/
|
|
if( hAss->tt == TT_MP4_LOAS && hAss->subFrameCnt == 0)
|
|
{
|
|
/* Start new LOAS frame */
|
|
FDKwriteBits( hBs, 0x2B7, 11 );
|
|
hAss->audioMuxLengthBytes = 0;
|
|
hAss->audioMuxLengthBytesPos = FDKgetValidBits( hBs ); /* store read pointer position */
|
|
FDKwriteBits( hBs, hAss->audioMuxLengthBytes, 13 );
|
|
}
|
|
|
|
ErrorStatus = AdvanceAudioMuxElement(
|
|
hAss,
|
|
hBs,
|
|
auBits,
|
|
bufferFullness,
|
|
cb
|
|
);
|
|
|
|
if (ErrorStatus != TRANSPORTENC_OK)
|
|
return ErrorStatus;
|
|
|
|
return ErrorStatus;
|
|
}
|
|
|
|
void transportEnc_LatmAdjustSubframeBits(HANDLE_LATM_STREAM hAss,
|
|
int *bits)
|
|
{
|
|
/* Substract bits from possible previous subframe */
|
|
*bits -= hAss->latmSubframeStart;
|
|
/* Add fill bits */
|
|
if (hAss->subFrameCnt == 0)
|
|
*bits += hAss->fillBits;
|
|
}
|
|
|
|
|
|
void transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss,
|
|
HANDLE_FDK_BITSTREAM hBs,
|
|
int *bytes)
|
|
{
|
|
|
|
hAss->subFrameCnt++;
|
|
if (hAss->subFrameCnt >= hAss->noSubframes)
|
|
{
|
|
|
|
/* Add LOAS frame length if required. */
|
|
if (hAss->tt == TT_MP4_LOAS)
|
|
{
|
|
int latmBytes;
|
|
|
|
latmBytes = (FDKgetValidBits(hBs)+7) >> 3;
|
|
|
|
/* write length info into assembler buffer */
|
|
hAss->audioMuxLengthBytes = latmBytes - 3; /* 3=Syncword + length */
|
|
{
|
|
FDK_BITSTREAM tmpBuf;
|
|
|
|
FDKinitBitStream( &tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0, BS_WRITER ) ;
|
|
FDKpushFor( &tmpBuf, hAss->audioMuxLengthBytesPos );
|
|
FDKwriteBits( &tmpBuf, hAss->audioMuxLengthBytes, 13 );
|
|
FDKsyncCache( &tmpBuf );
|
|
}
|
|
}
|
|
|
|
/* Write AudioMuxElement byte alignment fill bits */
|
|
FDKwriteBits(hBs, 0, hAss->fillBits);
|
|
|
|
FDK_ASSERT( (FDKgetValidBits(hBs) % 8) == 0);
|
|
|
|
hAss->subFrameCnt = 0;
|
|
|
|
FDKsyncCache(hBs);
|
|
*bytes = (FDKgetValidBits(hBs) + 7)>>3;
|
|
//FDKfetchBuffer(hBs, buffer, (UINT*)bytes);
|
|
|
|
if (hAss->muxConfigPeriod > 0)
|
|
{
|
|
hAss->latmFrameCounter++;
|
|
|
|
if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) {
|
|
hAss->latmFrameCounter = 0;
|
|
hAss->noSubframes = hAss->noSubframes_next;
|
|
}
|
|
}
|
|
} else {
|
|
/* No data this time */
|
|
*bytes = 0;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Init LATM/LOAS
|
|
*/
|
|
TRANSPORTENC_ERROR transportEnc_Latm_Init(
|
|
HANDLE_LATM_STREAM hAss,
|
|
HANDLE_FDK_BITSTREAM hBs,
|
|
CODER_CONFIG *layerConfig,
|
|
UINT audioMuxVersion,
|
|
TRANSPORT_TYPE tt,
|
|
CSTpCallBacks *cb
|
|
)
|
|
{
|
|
TRANSPORTENC_ERROR ErrorStatus;
|
|
int fractDelayPresent = 0;
|
|
int prog, layer;
|
|
|
|
int setupDataDistanceFrames = layerConfig->headerPeriod;
|
|
|
|
FDK_ASSERT(setupDataDistanceFrames>=0);
|
|
|
|
for (prog=0; prog<LATM_MAX_PROGRAMS; prog++) {
|
|
for (layer=0; layer<LATM_MAX_LAYERS; layer++) {
|
|
hAss->config[prog][layer] = NULL;
|
|
hAss->m_linfo[prog][layer].streamID = -1;
|
|
}
|
|
}
|
|
|
|
hAss->config[0][0] = layerConfig;
|
|
hAss->m_linfo[0][0].streamID = 0;
|
|
|
|
ErrorStatus = transportEnc_InitLatmStream( hAss,
|
|
fractDelayPresent,
|
|
setupDataDistanceFrames,
|
|
(audioMuxVersion)?1:0,
|
|
tt
|
|
);
|
|
if (ErrorStatus != TRANSPORTENC_OK)
|
|
goto bail;
|
|
|
|
ErrorStatus = transportEnc_LatmSetNrOfSubframes(
|
|
hAss,
|
|
layerConfig->nSubFrames
|
|
);
|
|
if (ErrorStatus != TRANSPORTENC_OK)
|
|
goto bail;
|
|
|
|
/* Get the size of the StreamMuxConfig somehow */
|
|
AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb);
|
|
//CreateStreamMuxConfig(hAss, hBs, 0);
|
|
|
|
bail:
|
|
return ErrorStatus;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|