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+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/******************* MPEG transport format encoder library *********************
+
+ Author(s):
+
+ Description:
+
+*******************************************************************************/
+
+#include "tpenc_latm.h"
+
+#include "genericStds.h"
+
+static const short celpFrameLengthTable[64] = {
+ 154, 170, 186, 147, 156, 165, 114, 120, 186, 126, 132, 138, 142,
+ 146, 154, 166, 174, 182, 190, 198, 206, 210, 214, 110, 114, 118,
+ 120, 122, 218, 230, 242, 254, 266, 278, 286, 294, 318, 342, 358,
+ 374, 390, 406, 422, 136, 142, 148, 154, 160, 166, 170, 174, 186,
+ 198, 206, 214, 222, 230, 238, 216, 160, 280, 338, 0, 0};
+
+/*******
+ write value to transport stream
+ first two bits define the size of the value itself
+ then the value itself, with a size of 0-3 bytes
+*******/
+static UINT transportEnc_LatmWriteValue(HANDLE_FDK_BITSTREAM hBs, int value) {
+ UCHAR valueBytes = 4;
+ unsigned int bitsWritten = 0;
+ int i;
+
+ if (value < (1 << 8)) {
+ valueBytes = 1;
+ } else if (value < (1 << 16)) {
+ valueBytes = 2;
+ } else if (value < (1 << 24)) {
+ valueBytes = 3;
+ } else {
+ valueBytes = 4;
+ }
+
+ FDKwriteBits(hBs, valueBytes - 1, 2); /* size of value in Bytes */
+ for (i = 0; i < valueBytes; i++) {
+ /* write most significant Byte first */
+ FDKwriteBits(hBs, (UCHAR)(value >> ((valueBytes - 1 - i) << 3)), 8);
+ }
+
+ bitsWritten = (valueBytes << 3) + 2;
+
+ return bitsWritten;
+}
+
+static UINT transportEnc_LatmCountFixBitDemandHeader(HANDLE_LATM_STREAM hAss) {
+ int bitDemand = 0;
+ int insertSetupData = 0;
+
+ /* only if start of new latm frame */
+ if (hAss->subFrameCnt == 0) {
+ /* AudioSyncStream */
+
+ if (hAss->tt == TT_MP4_LOAS) {
+ bitDemand += 11; /* syncword */
+ bitDemand += 13; /* audioMuxLengthBytes */
+ }
+
+ /* AudioMuxElement*/
+
+ /* AudioMuxElement::Stream Mux Config */
+ if (hAss->muxConfigPeriod > 0) {
+ insertSetupData = (hAss->latmFrameCounter == 0);
+ } else {
+ insertSetupData = 0;
+ }
+
+ if (hAss->tt != TT_MP4_LATM_MCP0) {
+ /* AudioMuxElement::useSameStreamMux Flag */
+ bitDemand += 1;
+
+ if (insertSetupData) {
+ bitDemand += hAss->streamMuxConfigBits;
+ }
+ }
+
+ /* AudioMuxElement::otherDataBits */
+ bitDemand += hAss->otherDataLenBits;
+
+ /* AudioMuxElement::ByteAlign */
+ if (bitDemand % 8) {
+ hAss->fillBits = 8 - (bitDemand % 8);
+ bitDemand += hAss->fillBits;
+ } else {
+ hAss->fillBits = 0;
+ }
+ }
+
+ return bitDemand;
+}
+
+static UINT transportEnc_LatmCountVarBitDemandHeader(
+ HANDLE_LATM_STREAM hAss, unsigned int streamDataLength) {
+ int bitDemand = 0;
+ int prog, layer;
+
+ /* Payload Length Info*/
+ if (hAss->allStreamsSameTimeFraming) {
+ for (prog = 0; prog < hAss->noProgram; prog++) {
+ for (layer = 0; layer < LATM_MAX_LAYERS; layer++) {
+ LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
+
+ if (p_linfo->streamID >= 0) {
+ switch (p_linfo->frameLengthType) {
+ case 0:
+ if (streamDataLength > 0) {
+ streamDataLength -= bitDemand;
+ while (streamDataLength >= (255 << 3)) {
+ bitDemand += 8;
+ streamDataLength -= (255 << 3);
+ }
+ bitDemand += 8;
+ }
+ break;
+
+ case 1:
+ case 4:
+ case 6:
+ bitDemand += 2;
+ break;
+
+ default:
+ return 0;
+ }
+ }
+ }
+ }
+ } else {
+ /* there are many possibilities to use this mechanism. */
+ switch (hAss->varMode) {
+ case LATMVAR_SIMPLE_SEQUENCE: {
+ /* Use the sequence generated by the encoder */
+ // int streamCntPosition = transportEnc_SetWritePointer(
+ // hAss->hAssemble, 0 ); int streamCntPosition = FDKgetValidBits(
+ // hAss->hAssemble );
+ bitDemand += 4;
+
+ hAss->varStreamCnt = 0;
+ for (prog = 0; prog < hAss->noProgram; prog++) {
+ for (layer = 0; layer < LATM_MAX_LAYERS; layer++) {
+ LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
+
+ if (p_linfo->streamID >= 0) {
+ bitDemand += 4; /* streamID */
+ switch (p_linfo->frameLengthType) {
+ case 0:
+ streamDataLength -= bitDemand;
+ while (streamDataLength >= (255 << 3)) {
+ bitDemand += 8;
+ streamDataLength -= (255 << 3);
+ }
+
+ bitDemand += 8;
+ break;
+ /*bitDemand += 1; endFlag
+ break;*/
+
+ case 1:
+ case 4:
+ case 6:
+
+ break;
+
+ default:
+ return 0;
+ }
+ hAss->varStreamCnt++;
+ }
+ }
+ }
+ bitDemand += 4;
+ // transportEnc_UpdateBitstreamField( hAss->hAssemble,
+ // streamCntPosition, hAss->varStreamCnt-1, 4 ); UINT pos =
+ // streamCntPosition-FDKgetValidBits(hAss->hAssemble); FDKpushBack(
+ // hAss->hAssemble, pos); FDKwriteBits( hAss->hAssemble,
+ // hAss->varStreamCnt-1, 4); FDKpushFor( hAss->hAssemble, pos-4);
+ } break;
+
+ default:
+ return 0;
+ }
+ }
+
+ return bitDemand;
+}
+
+TRANSPORTENC_ERROR
+CreateStreamMuxConfig(HANDLE_LATM_STREAM hAss, HANDLE_FDK_BITSTREAM hBs,
+ int bufferFullness, CSTpCallBacks *cb) {
+ INT streamIDcnt, tmp;
+ int layer, prog;
+
+ USHORT coreFrameOffset = 0;
+
+ hAss->taraBufferFullness = 0xFF;
+ hAss->audioMuxVersionA = 0; /* for future extensions */
+ hAss->streamMuxConfigBits = 0;
+
+ FDKwriteBits(hBs, hAss->audioMuxVersion, 1); /* audioMuxVersion */
+ hAss->streamMuxConfigBits += 1;
+
+ if (hAss->audioMuxVersion == 1) {
+ FDKwriteBits(hBs, hAss->audioMuxVersionA, 1); /* audioMuxVersionA */
+ hAss->streamMuxConfigBits += 1;
+ }
+
+ if (hAss->audioMuxVersionA == 0) {
+ if (hAss->audioMuxVersion == 1) {
+ hAss->streamMuxConfigBits += transportEnc_LatmWriteValue(
+ hBs, hAss->taraBufferFullness); /* taraBufferFullness */
+ }
+ FDKwriteBits(hBs, hAss->allStreamsSameTimeFraming ? 1 : 0,
+ 1); /* allStreamsSameTimeFraming */
+ FDKwriteBits(hBs, hAss->noSubframes - 1, 6); /* Number of Subframes */
+ FDKwriteBits(hBs, hAss->noProgram - 1, 4); /* Number of Programs */
+
+ hAss->streamMuxConfigBits += 11;
+
+ streamIDcnt = 0;
+ for (prog = 0; prog < hAss->noProgram; prog++) {
+ int transLayer = 0;
+
+ FDKwriteBits(hBs, hAss->noLayer[prog] - 1, 3);
+ hAss->streamMuxConfigBits += 3;
+
+ for (layer = 0; layer < LATM_MAX_LAYERS; layer++) {
+ LATM_LAYER_INFO *p_linfo = &(hAss->m_linfo[prog][layer]);
+ CODER_CONFIG *p_lci = hAss->config[prog][layer];
+
+ p_linfo->streamID = -1;
+
+ if (hAss->config[prog][layer] != NULL) {
+ int useSameConfig = 0;
+
+ if (transLayer > 0) {
+ FDKwriteBits(hBs, useSameConfig ? 1 : 0, 1);
+ hAss->streamMuxConfigBits += 1;
+ }
+ if ((useSameConfig == 0) || (transLayer == 0)) {
+ const UINT alignAnchor = FDKgetValidBits(hBs);
+
+ if (0 !=
+ (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) {
+ return TRANSPORTENC_UNKOWN_ERROR;
+ }
+
+ if (hAss->audioMuxVersion == 1) {
+ UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
+ FDKbyteAlign(hBs, alignAnchor);
+ ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
+ FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
+
+ transportEnc_LatmWriteValue(hBs, ascLen);
+
+ if (0 !=
+ (transportEnc_writeASC(hBs, hAss->config[prog][layer], cb))) {
+ return TRANSPORTENC_UNKOWN_ERROR;
+ }
+
+ FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */
+ }
+
+ hAss->streamMuxConfigBits +=
+ FDKgetValidBits(hBs) -
+ alignAnchor; /* add asc length to smc summary */
+ }
+ transLayer++;
+
+ if (!hAss->allStreamsSameTimeFraming) {
+ if (streamIDcnt >= LATM_MAX_STREAM_ID)
+ return TRANSPORTENC_INVALID_CONFIG;
+ }
+ p_linfo->streamID = streamIDcnt++;
+
+ switch (p_lci->aot) {
+ case AOT_AAC_MAIN:
+ case AOT_AAC_LC:
+ case AOT_AAC_SSR:
+ case AOT_AAC_LTP:
+ case AOT_AAC_SCAL:
+ case AOT_ER_AAC_LD:
+ case AOT_ER_AAC_ELD:
+ case AOT_USAC:
+ p_linfo->frameLengthType = 0;
+
+ FDKwriteBits(hBs, p_linfo->frameLengthType,
+ 3); /* frameLengthType */
+ FDKwriteBits(hBs, bufferFullness, 8); /* bufferFullness */
+ hAss->streamMuxConfigBits += 11;
+
+ if (!hAss->allStreamsSameTimeFraming) {
+ CODER_CONFIG *p_lci_prev = hAss->config[prog][layer - 1];
+ if (((p_lci->aot == AOT_AAC_SCAL) ||
+ (p_lci->aot == AOT_ER_AAC_SCAL)) &&
+ ((p_lci_prev->aot == AOT_CELP) ||
+ (p_lci_prev->aot == AOT_ER_CELP))) {
+ FDKwriteBits(hBs, coreFrameOffset, 6); /* coreFrameOffset */
+ hAss->streamMuxConfigBits += 6;
+ }
+ }
+ break;
+
+ case AOT_TWIN_VQ:
+ p_linfo->frameLengthType = 1;
+ tmp = ((p_lci->bitsFrame + 7) >> 3) -
+ 20; /* transmission frame length in bytes */
+ if ((tmp < 0)) {
+ return TRANSPORTENC_INVALID_TRANSMISSION_FRAME_LENGTH;
+ }
+ FDKwriteBits(hBs, p_linfo->frameLengthType,
+ 3); /* frameLengthType */
+ FDKwriteBits(hBs, tmp, 9);
+ hAss->streamMuxConfigBits += 12;
+
+ p_linfo->frameLengthBits = (tmp + 20) << 3;
+ break;
+
+ case AOT_CELP:
+ p_linfo->frameLengthType = 4;
+ FDKwriteBits(hBs, p_linfo->frameLengthType,
+ 3); /* frameLengthType */
+ hAss->streamMuxConfigBits += 3;
+ {
+ int i;
+ for (i = 0; i < 62; i++) {
+ if (celpFrameLengthTable[i] == p_lci->bitsFrame) break;
+ }
+ if (i >= 62) {
+ return TRANSPORTENC_INVALID_CELP_FRAME_LENGTH;
+ }
+
+ FDKwriteBits(hBs, i, 6); /* CELPframeLengthTabelIndex */
+ hAss->streamMuxConfigBits += 6;
+ }
+ p_linfo->frameLengthBits = p_lci->bitsFrame;
+ break;
+
+ case AOT_HVXC:
+ p_linfo->frameLengthType = 6;
+ FDKwriteBits(hBs, p_linfo->frameLengthType,
+ 3); /* frameLengthType */
+ hAss->streamMuxConfigBits += 3;
+ {
+ int i;
+
+ if (p_lci->bitsFrame == 40) {
+ i = 0;
+ } else if (p_lci->bitsFrame == 80) {
+ i = 1;
+ } else {
+ return TRANSPORTENC_INVALID_FRAME_BITS;
+ }
+ FDKwriteBits(hBs, i, 1); /* HVXCframeLengthTableIndex */
+ hAss->streamMuxConfigBits += 1;
+ }
+ p_linfo->frameLengthBits = p_lci->bitsFrame;
+ break;
+
+ case AOT_NULL_OBJECT:
+ default:
+ return TRANSPORTENC_INVALID_AOT;
+ }
+ }
+ }
+ }
+
+ FDKwriteBits(hBs, (hAss->otherDataLenBits > 0) ? 1 : 0,
+ 1); /* otherDataPresent */
+ hAss->streamMuxConfigBits += 1;
+
+ if (hAss->otherDataLenBits > 0) {
+ FDKwriteBits(hBs, 0, 1);
+ FDKwriteBits(hBs, hAss->otherDataLenBits, 8);
+ hAss->streamMuxConfigBits += 9;
+ }
+
+ FDKwriteBits(hBs, 0, 1); /* crcCheckPresent=0 */
+ hAss->streamMuxConfigBits += 1;
+
+ } 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->otherDataLenBits = 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 */
+ if (TRANSPORTENC_OK != (ErrorStatus = CreateStreamMuxConfig(
+ hAss, hBs, bufferFullness, cb))) {
+ 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->otherDataLenBits;
+ *bits += hAss->fillBits;
+ }
+}
+
+TRANSPORTENC_ERROR transportEnc_LatmGetFrame(HANDLE_LATM_STREAM hAss,
+ HANDLE_FDK_BITSTREAM hBs,
+ int *pBytes) {
+ TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
+
+ hAss->subFrameCnt++;
+ if (hAss->subFrameCnt >= hAss->noSubframes) {
+ /* Add LOAS frame length if required. */
+ if (hAss->tt == TT_MP4_LOAS) {
+ FDK_BITSTREAM tmpBuf;
+
+ /* Determine frame length info */
+ hAss->audioMuxLengthBytes =
+ ((FDKgetValidBits(hBs) + hAss->otherDataLenBits + 7) >> 3) -
+ 3; /* 3=Syncword + length */
+
+ /* Check frame length info */
+ if (hAss->audioMuxLengthBytes >= (1 << 13)) {
+ ErrorStatus = TRANSPORTENC_INVALID_AU_LENGTH;
+ goto bail;
+ }
+
+ /* Write length info into assembler buffer */
+ FDKinitBitStream(&tmpBuf, hBs->hBitBuf.Buffer, hBs->hBitBuf.bufSize, 0,
+ BS_WRITER);
+ FDKpushFor(&tmpBuf, hAss->audioMuxLengthBytesPos);
+ FDKwriteBits(&tmpBuf, hAss->audioMuxLengthBytes, 13);
+ FDKsyncCache(&tmpBuf);
+ }
+
+ /* Write AudioMuxElement other data bits */
+ FDKwriteBits(hBs, 0, hAss->otherDataLenBits);
+
+ /* Write AudioMuxElement byte alignment fill bits */
+ FDKwriteBits(hBs, 0, hAss->fillBits);
+
+ FDK_ASSERT((FDKgetValidBits(hBs) % 8) == 0);
+
+ hAss->subFrameCnt = 0;
+
+ FDKsyncCache(hBs);
+ *pBytes = (FDKgetValidBits(hBs) + 7) >> 3;
+
+ if (hAss->muxConfigPeriod > 0) {
+ hAss->latmFrameCounter++;
+
+ if (hAss->latmFrameCounter >= hAss->muxConfigPeriod) {
+ hAss->latmFrameCounter = 0;
+ hAss->noSubframes = hAss->noSubframes_next;
+ }
+ }
+ } else {
+ /* No data this time */
+ *pBytes = 0;
+ }
+
+bail:
+ return ErrorStatus;
+}
+
+/**
+ * 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 */
+ if (TRANSPORTENC_OK !=
+ (ErrorStatus = AdvanceAudioMuxElement(hAss, hBs, 0, 0, cb))) {
+ goto bail;
+ }
+
+ // CreateStreamMuxConfig(hAss, hBs, 0);
+
+bail:
+ return ErrorStatus;
+}
+
+TRANSPORTENC_ERROR transportEnc_LatmAddOtherDataBits(HANDLE_LATM_STREAM hAss,
+ const int otherDataBits) {
+ TRANSPORTENC_ERROR ErrorStatus = TRANSPORTENC_OK;
+
+ if ((hAss->otherDataLenBits != 0) || (otherDataBits % 8 != 0)) {
+ /* This implementation allows to add other data bits only once.
+ To keep existing alignment only whole bytes are allowed. */
+ ErrorStatus = TRANSPORTENC_UNKOWN_ERROR;
+ } else {
+ /* Ensure correct addional bits in payload. */
+ if (hAss->tt == TT_MP4_LATM_MCP0) {
+ hAss->otherDataLenBits = otherDataBits;
+ } else {
+ hAss->otherDataLenBits = otherDataBits - 9;
+ hAss->streamMuxConfigBits += 9;
+ }
+ }
+
+ return ErrorStatus;
+}