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mirror of https://github.com/mstorsjo/fdk-aac.git synced 2024-12-11 07:55:17 +01:00
fdk-aac/libAACenc/src/bandwidth.cpp
Jean-Michel Trivi 46ba3676b8 AAC/SBR encoder improvements
* AAC-Encoder

  - AAC-ELD core encoder audio quality tuning. Update tuning tables, configure
    bitreservoir size and adapt afterburner iteration value.
     Modified file(s):
        libAACenc/src/aacenc.h
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/adj_thr.cpp
        libAACenc/src/adj_thr.h
        libAACenc/src/adj_thr_data.h
        libAACenc/src/bandwidth.cpp
        libAACenc/src/pnsparam.cpp
        libAACenc/src/qc_main.cpp

   - Introduze dead zone quantizer for ELD to improve audio quality at certain
     configurations.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/adj_thr.cpp
        libAACenc/src/adj_thr.h
        libAACenc/src/qc_data.h
        libAACenc/src/qc_main.cpp
        libAACenc/src/quantize.cpp
        libAACenc/src/quantize.h
        libAACenc/src/sf_estim.cpp
        libAACenc/src/sf_estim.h

   - Revise TNS module to improve ELD audio quality.
   - Use new window function and separate prediction gain according TNS filters.
   - Add missing memory initilization to TNS configuration.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/aacenc_tns.cpp
        libAACenc/src/aacenc_tns.h
        libAACenc/src/psy_main.cpp
        libAACenc/src/tns_func.h

* SBR-Encoder

   - Revise frequency resolution calculation and handle differently depending
     on number of envelopes and split frames decision.
   - Add and adjust ELD SBR tuning tables.
     Modified file(s):
        libSBRenc/include/sbr_encoder.h
        libSBRenc/src/bit_sbr.h
        libSBRenc/src/env_est.cpp
        libSBRenc/src/fram_gen.cpp
        libSBRenc/src/fram_gen.h
        libSBRenc/src/mh_det.cpp
        libSBRenc/src/sbr_def.h
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/sbr_rom.cpp
        libSBRenc/src/tran_det.cpp

   - Replace ELD transient detector with fast implementation.
     Modified file(s):
        libSBRenc/src/env_est.cpp
        libSBRenc/src/env_est.h
        libSBRenc/src/fram_gen.cpp
        libSBRenc/src/sbr_def.h
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/tran_det.cpp
        libSBRenc/src/tran_det.h

* FDK-Library
   - Introduce generic compare function in tools library.
     Modified file(s):
        libFDK/include/fixpoint_math.h
        libFDK/src/FDK_core.cpp

* SBR-Encoder

  -  Revise ELD frame splitter to improve bit distribution.
     Modified file(s):
        libSBRenc/include/sbr_encoder.h
        libSBRenc/src/bit_sbr.h
        libSBRenc/src/env_est.cpp
        libSBRenc/src/fram_gen.cpp
        libSBRenc/src/fram_gen.h
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/tran_det.cpp
        libSBRenc/src/tran_det.h

   - Configure amplitude resolution according the tonality of the audio signal.
     Modified file(s):
        libSBRenc/include/sbr_encoder.h
        libSBRenc/src/bit_sbr.h
        libSBRenc/src/env_est.cpp
        libSBRenc/src/nf_est.cpp
        libSBRenc/src/nf_est.h
        libSBRenc/src/sbr_def.h
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/ton_corr.cpp
        libSBRenc/src/ton_corr.h
        libSBRenc/src/tran_det.cpp
        libSBRenc/src/tran_det.h

Change-Id: Ie0672b989a06ee63b50240616b8d1d4b790b6cb2
2016-04-08 10:52:42 -07:00

380 lines
13 KiB
C++

/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2015 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
----------------------------------------------------------------------------------------------------------- */
/************************* Fast MPEG AAC Audio Encoder **********************
Initial author: M. Schug / A. Groeschel
contents/description: bandwidth expert
******************************************************************************/
#include "channel_map.h"
#include "bandwidth.h"
#include "aacEnc_ram.h"
typedef struct{
INT chanBitRate;
INT bandWidthMono;
INT bandWidth2AndMoreChan;
} BANDWIDTH_TAB;
static const BANDWIDTH_TAB bandWidthTable[] = {
{0, 3700, 5000},
{12000, 5000, 6400},
{20000, 6900, 9640},
{28000, 9600, 13050},
{40000, 12060, 14260},
{56000, 13950, 15500},
{72000, 14200, 16120},
{96000, 17000, 17000},
{576001,17000, 17000}
};
static const BANDWIDTH_TAB bandWidthTable_LD_22050[] = {
{ 8000, 2000, 2400},
{12000, 2500, 2700},
{16000, 3300, 3100},
{24000, 6250, 7200},
{32000, 9200, 10500},
{40000, 16000, 16000},
{48000, 16000, 16000},
{360001, 16000, 16000}
};
static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = {
{ 8000, 2000, 2000},
{12000, 2000, 2300},
{16000, 2200, 2500},
{24000, 5650, 7200},
{32000, 11600, 12000},
{40000, 12000, 16000},
{48000, 16000, 16000},
{64000, 16000, 16000},
{360001, 16000, 16000}
};
static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = {
{ 8000, 2000, 2000},
{12000, 2000, 2000},
{24000, 4250, 7200},
{32000, 8400, 9000},
{40000, 9400, 11300},
{48000, 11900, 14700},
{64000, 14800, 16000},
{76000, 16000, 16000},
{360001, 16000, 16000}
};
static const BANDWIDTH_TAB bandWidthTable_LD_44100[] = {
{ 8000, 2000, 2000},
{24000, 2000, 2000},
{32000, 4400, 5700},
{40000, 7400, 8800},
{48000, 9000, 10700},
{56000, 11000, 12900},
{64000, 14400, 15500},
{80000, 16000, 16200},
{96000, 16500, 16000},
{128000, 16000, 16000},
{360001, 16000, 16000}
};
static const BANDWIDTH_TAB bandWidthTable_LD_48000[] = {
{ 8000, 2000, 2000},
{24000, 2000, 2000},
{32000, 4400, 5700},
{40000, 7400, 8800},
{48000, 9000, 10700},
{56000, 11000, 12800},
{64000, 14300, 15400},
{80000, 16000, 16200},
{96000, 16500, 16000},
{128000, 16000, 16000},
{360001, 16000, 16000}
};
typedef struct{
AACENC_BITRATE_MODE bitrateMode;
int bandWidthMono;
int bandWidth2AndMoreChan;
} BANDWIDTH_TAB_VBR;
static const BANDWIDTH_TAB_VBR bandWidthTableVBR[]= {
{AACENC_BR_MODE_CBR, 0, 0},
{AACENC_BR_MODE_VBR_1, 13050, 13050},
{AACENC_BR_MODE_VBR_2, 13050, 13050},
{AACENC_BR_MODE_VBR_3, 14260, 14260},
{AACENC_BR_MODE_VBR_4, 15500, 15500},
{AACENC_BR_MODE_VBR_5, 48000, 48000},
{AACENC_BR_MODE_SFR, 0, 0},
{AACENC_BR_MODE_FF, 0, 0}
};
static INT GetBandwidthEntry(
const INT frameLength,
const INT sampleRate,
const INT chanBitRate,
const INT entryNo)
{
INT bandwidth = -1;
const BANDWIDTH_TAB *pBwTab = NULL;
INT bwTabSize = 0;
switch (frameLength) {
case 1024:
pBwTab = bandWidthTable;
bwTabSize = sizeof(bandWidthTable)/sizeof(BANDWIDTH_TAB);
break;
case 480:
case 512:
switch (sampleRate) {
case 8000:
case 11025:
case 12000:
case 16000:
case 22050:
pBwTab = bandWidthTable_LD_22050;
bwTabSize = sizeof(bandWidthTable_LD_22050)/sizeof(BANDWIDTH_TAB);
break;
case 24000:
pBwTab = bandWidthTable_LD_24000;
bwTabSize = sizeof(bandWidthTable_LD_24000)/sizeof(BANDWIDTH_TAB);
break;
case 32000:
pBwTab = bandWidthTable_LD_32000;
bwTabSize = sizeof(bandWidthTable_LD_32000)/sizeof(BANDWIDTH_TAB);
break;
case (44100):
pBwTab = bandWidthTable_LD_44100;
bwTabSize = sizeof(bandWidthTable_LD_44100)/sizeof(BANDWIDTH_TAB);
break;
case 48000:
case 64000:
case 88200:
case 96000:
pBwTab = bandWidthTable_LD_48000;
bwTabSize = sizeof(bandWidthTable_LD_48000)/sizeof(BANDWIDTH_TAB);
break;
}
break;
default:
pBwTab = NULL;
bwTabSize = 0;
}
if (pBwTab!=NULL) {
int i;
for (i=0; i<bwTabSize-1; i++) {
if (chanBitRate >= pBwTab[i].chanBitRate &&
chanBitRate < pBwTab[i+1].chanBitRate)
{
switch (frameLength) {
case 1024:
bandwidth = (entryNo==0)
? pBwTab[i].bandWidthMono
: pBwTab[i].bandWidth2AndMoreChan;
break;
case 480:
case 512:
{
INT q_res = 0;
INT startBw = (entryNo==0) ? pBwTab[i ].bandWidthMono : pBwTab[i ].bandWidth2AndMoreChan;
INT endBw = (entryNo==0) ? pBwTab[i+1].bandWidthMono : pBwTab[i+1].bandWidth2AndMoreChan;
INT startBr = pBwTab[i].chanBitRate;
INT endBr = pBwTab[i+1].chanBitRate;
FIXP_DBL bwFac_fix = fDivNorm(chanBitRate-startBr, endBr-startBr, &q_res);
bandwidth = (INT)scaleValue(fMult(bwFac_fix, (FIXP_DBL)(endBw-startBw)),q_res) + startBw;
}
break;
default:
bandwidth = -1;
}
break;
} /* within bitrate range */
}
} /* pBwTab!=NULL */
return bandwidth;
}
AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth,
INT proposedBandWidth,
INT bitrate,
AACENC_BITRATE_MODE bitrateMode,
INT sampleRate,
INT frameLength,
CHANNEL_MAPPING* cm,
CHANNEL_MODE encoderMode)
{
AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK;
INT chanBitRate = bitrate/cm->nChannels;
/* vbr */
switch(bitrateMode){
case AACENC_BR_MODE_VBR_1:
case AACENC_BR_MODE_VBR_2:
case AACENC_BR_MODE_VBR_3:
case AACENC_BR_MODE_VBR_4:
case AACENC_BR_MODE_VBR_5:
if (proposedBandWidth != 0){
/* use given bw */
*bandWidth = proposedBandWidth;
} else {
/* take bw from table */
switch(encoderMode){
case MODE_1:
*bandWidth = bandWidthTableVBR[bitrateMode].bandWidthMono;
break;
case MODE_2:
case MODE_1_2:
case MODE_1_2_1:
case MODE_1_2_2:
case MODE_1_2_2_1:
case MODE_1_2_2_2_1:
case MODE_7_1_REAR_SURROUND:
case MODE_7_1_FRONT_CENTER:
*bandWidth = bandWidthTableVBR[bitrateMode].bandWidth2AndMoreChan;
break;
default:
return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;
}
}
break;
case AACENC_BR_MODE_CBR:
case AACENC_BR_MODE_SFR:
case AACENC_BR_MODE_FF:
/* bandwidth limiting */
if (proposedBandWidth != 0) {
*bandWidth = FDKmin(proposedBandWidth, FDKmin(20000, sampleRate>>1));
}
else { /* search reasonable bandwidth */
int entryNo = 0;
switch(encoderMode){
case MODE_1: /* mono */
entryNo = 0; /* use mono bandwith settings */
break;
case MODE_2: /* stereo */
case MODE_1_2: /* sce + cpe */
case MODE_1_2_1: /* sce + cpe + sce */
case MODE_1_2_2: /* sce + cpe + cpe */
case MODE_1_2_2_1: /* (5.1) sce + cpe + cpe + lfe */
case MODE_1_2_2_2_1: /* (7.1) sce + cpe + cpe + cpe + lfe */
case MODE_7_1_REAR_SURROUND:
case MODE_7_1_FRONT_CENTER:
entryNo = 1; /* use stereo bandwith settings */
break;
default:
return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;
}
*bandWidth = GetBandwidthEntry(
frameLength,
sampleRate,
chanBitRate,
entryNo);
if (*bandWidth==-1) {
ErrorStatus = AAC_ENC_INVALID_CHANNEL_BITRATE;
}
}
break;
default:
*bandWidth = 0;
return AAC_ENC_UNSUPPORTED_BITRATE_MODE;
}
*bandWidth = FDKmin(*bandWidth, sampleRate/2);
return ErrorStatus;;
}