Merge remote-tracking branch 'aosp/master'

Clang warns about this being deprecated, when building without specifying
-std=c++98.

This doesn't give any measurable encoding speed impact on ARM or AArch64,
on neither GCC nor Clang.

Change-Id: I63a0bbd1dccb97547522da188ee585d4d8127c29

.gitignore

@@ -26,3 +26,4 @@ m4/lt~obsolete.m4
 missing
 stamp-h1
 aac-enc
+compile

Android.mk

@@ -1,31 +1,31 @@
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
-aacdec_sources := $(wildcard $(LOCAL_PATH)/libAACdec/src/*.cpp)
+aacdec_sources := $(sort $(wildcard $(LOCAL_PATH)/libAACdec/src/*.cpp))
 aacdec_sources := $(aacdec_sources:$(LOCAL_PATH)/libAACdec/src/%=%)
 
-aacenc_sources := $(wildcard $(LOCAL_PATH)/libAACenc/src/*.cpp)
+aacenc_sources := $(sort $(wildcard $(LOCAL_PATH)/libAACenc/src/*.cpp))
 aacenc_sources := $(aacenc_sources:$(LOCAL_PATH)/libAACenc/src/%=%)
 
-pcmutils_sources := $(wildcard $(LOCAL_PATH)/libPCMutils/src/*.cpp)
+pcmutils_sources := $(sort $(wildcard $(LOCAL_PATH)/libPCMutils/src/*.cpp))
 pcmutils_sources := $(pcmutils_sources:$(LOCAL_PATH)/libPCMutils/src/%=%)
 
-fdk_sources := $(wildcard $(LOCAL_PATH)/libFDK/src/*.cpp)
+fdk_sources := $(sort $(wildcard $(LOCAL_PATH)/libFDK/src/*.cpp))
 fdk_sources := $(fdk_sources:$(LOCAL_PATH)/libFDK/src/%=%)
 
-sys_sources := $(wildcard $(LOCAL_PATH)/libSYS/src/*.cpp)
+sys_sources := $(sort $(wildcard $(LOCAL_PATH)/libSYS/src/*.cpp))
 sys_sources := $(sys_sources:$(LOCAL_PATH)/libSYS/src/%=%)
 
-mpegtpdec_sources := $(wildcard $(LOCAL_PATH)/libMpegTPDec/src/*.cpp)
+mpegtpdec_sources := $(sort $(wildcard $(LOCAL_PATH)/libMpegTPDec/src/*.cpp))
 mpegtpdec_sources := $(mpegtpdec_sources:$(LOCAL_PATH)/libMpegTPDec/src/%=%)
 
-mpegtpenc_sources := $(wildcard $(LOCAL_PATH)/libMpegTPEnc/src/*.cpp)
+mpegtpenc_sources := $(sort $(wildcard $(LOCAL_PATH)/libMpegTPEnc/src/*.cpp))
 mpegtpenc_sources := $(mpegtpenc_sources:$(LOCAL_PATH)/libMpegTPEnc/src/%=%)
 
-sbrdec_sources := $(wildcard $(LOCAL_PATH)/libSBRdec/src/*.cpp)
+sbrdec_sources := $(sort $(wildcard $(LOCAL_PATH)/libSBRdec/src/*.cpp))
 sbrdec_sources := $(sbrdec_sources:$(LOCAL_PATH)/libSBRdec/src/%=%)
 
-sbrenc_sources := $(wildcard $(LOCAL_PATH)/libSBRenc/src/*.cpp)
+sbrenc_sources := $(sort $(wildcard $(LOCAL_PATH)/libSBRenc/src/*.cpp))
 sbrenc_sources := $(sbrenc_sources:$(LOCAL_PATH)/libSBRenc/src/%=%)
 
 LOCAL_SRC_FILES := \
@@ -39,8 +39,8 @@ LOCAL_SRC_FILES := \
         $(sbrdec_sources:%=libSBRdec/src/%) \
         $(sbrenc_sources:%=libSBRenc/src/%)
 
-LOCAL_CFLAGS := -DANDROID
 LOCAL_CFLAGS += -Wno-sequence-point -Wno-extra
+LOCAL_CFLAGS += "-Wno-\#warnings" -Wno-constant-logical-operand -Wno-self-assign
 
 LOCAL_C_INCLUDES := \
         $(LOCAL_PATH)/libAACdec/include \
@@ -53,6 +53,7 @@ LOCAL_C_INCLUDES := \
         $(LOCAL_PATH)/libSBRdec/include \
         $(LOCAL_PATH)/libSBRenc/include
 
+
 LOCAL_MODULE:= libFraunhoferAAC
 
 include $(BUILD_STATIC_LIBRARY)

ChangeLog

@@ -1,3 +1,12 @@
+0.1.4
+ - Updated upstream sources, with minor changes to the decoder API
+   breaking the ABI. (Calling code using AUDIO_CHANNEL_TYPE may need to
+   be updated. A new option AAC_PCM_LIMITER_ENABLE has been added, enabled
+   by default, which incurs extra decoding delay.)
+ - PowerPC optimizations, fixes for building on AIX
+ - Support for reading streamed wav files in the encoder example
+ - Fix VBR encoding of sample rates over 64 kHz
+
 0.1.3
  - Updated upstream sources, with a number of crash fixes and new features
    (including support for encoding 7.1)

Makefile.am

@@ -74,7 +74,6 @@ AACENC_SRC = \
     libAACenc/src/pre_echo_control.cpp \
     libAACenc/src/quantize.cpp \
     libAACenc/src/tonality.cpp \
-    libAACenc/src/aacenc_hcr.cpp \
     libAACenc/src/aacEnc_rom.cpp \
     libAACenc/src/bandwidth.cpp \
     libAACenc/src/channel_map.cpp \
@@ -119,6 +118,7 @@ MPEGTPDEC_SRC = \
     libMpegTPDec/src/tpdec_adif.cpp \
     libMpegTPDec/src/tpdec_adts.cpp \
     libMpegTPDec/src/tpdec_asc.cpp \
+    libMpegTPDec/src/tpdec_drm.cpp \
     libMpegTPDec/src/tpdec_latm.cpp \
     libMpegTPDec/src/tpdec_lib.cpp
 
@@ -130,6 +130,7 @@ MPEGTPENC_SRC = \
     libMpegTPEnc/src/tpenc_lib.cpp
 
 PCMUTILS_SRC = \
+    libPCMutils/src/limiter.cpp \
     libPCMutils/src/pcmutils_lib.cpp
 
 SBRDEC_SRC = \
@@ -185,9 +186,11 @@ libfdk_aac_la_SOURCES = \
 
 EXTRA_DIST = \
     $(top_srcdir)/autogen.sh \
+    $(top_srcdir)/MODULE_LICENSE_FRAUNHOFER \
     $(top_srcdir)/NOTICE \
     $(top_srcdir)/Android.mk \
     $(top_srcdir)/fdk-aac.sym \
+    $(top_srcdir)/Makefile.vc \
     $(top_srcdir)/documentation/*.pdf \
     $(top_srcdir)/libAACdec/src/*.h \
     $(top_srcdir)/libAACdec/src/arm/*.cpp \
@@ -208,9 +211,12 @@ EXTRA_DIST = \
     $(top_srcdir)/libMpegTPDec/src/*.h \
     $(top_srcdir)/libMpegTPDec/src/version \
     $(top_srcdir)/libFDK/include/*.h \
+    $(top_srcdir)/libFDK/include/aarch64/*.h \
     $(top_srcdir)/libFDK/include/arm/*.h \
     $(top_srcdir)/libFDK/include/mips/*.h \
+    $(top_srcdir)/libFDK/include/ppc/*.h \
     $(top_srcdir)/libFDK/include/x86/*.h \
     $(top_srcdir)/libFDK/src/arm/*.cpp \
-    $(top_srcdir)/libFDK/src/mips/*.cpp
+    $(top_srcdir)/libFDK/src/mips/*.cpp \
+    $(top_srcdir)/win32/*.h
 

Makefile.vc

@@ -0,0 +1,250 @@
+#
+# Options:
+# prefix=\path\to\install
+#
+# Compiling:  nmake -f Makefile.vc
+# Installing: nmake -f Makefile.vc prefix=\path\to\x install
+#
+
+VERSION=0.1.5
+
+# Linker and librarian commands
+LD = link
+AR = lib
+
+!IFDEF HOME
+# In case we are using a cross compiler shell.
+MKDIR_FLAGS = -p
+!ENDIF
+
+AM_CPPFLAGS = \
+    -Iwin32 \
+    -IlibAACdec/include \
+    -IlibAACenc/include \
+    -IlibSBRdec/include \
+    -IlibSBRenc/include \
+    -IlibMpegTPDec/include \
+    -IlibMpegTPEnc/include \
+    -IlibSYS/include \
+    -IlibFDK/include \
+    -IlibPCMutils/include
+
+AACDEC_SRC = \
+    libAACdec/src/aacdec_drc.cpp \
+    libAACdec/src/aacdec_hcr.cpp \
+    libAACdec/src/aacdecoder.cpp \
+    libAACdec/src/aacdec_pns.cpp \
+    libAACdec/src/aac_ram.cpp \
+    libAACdec/src/block.cpp \
+    libAACdec/src/channelinfo.cpp \
+    libAACdec/src/ldfiltbank.cpp \
+    libAACdec/src/rvlcbit.cpp \
+    libAACdec/src/rvlc.cpp \
+    libAACdec/src/aacdec_hcr_bit.cpp \
+    libAACdec/src/aacdec_hcrs.cpp \
+    libAACdec/src/aacdecoder_lib.cpp \
+    libAACdec/src/aacdec_tns.cpp \
+    libAACdec/src/aac_rom.cpp \
+    libAACdec/src/channel.cpp \
+    libAACdec/src/conceal.cpp \
+    libAACdec/src/pulsedata.cpp \
+    libAACdec/src/rvlcconceal.cpp \
+    libAACdec/src/stereo.cpp
+
+AACENC_SRC = \
+    libAACenc/src/aacenc.cpp \
+    libAACenc/src/aacEnc_ram.cpp \
+    libAACenc/src/band_nrg.cpp \
+    libAACenc/src/block_switch.cpp \
+    libAACenc/src/grp_data.cpp \
+    libAACenc/src/metadata_main.cpp \
+    libAACenc/src/pre_echo_control.cpp \
+    libAACenc/src/quantize.cpp \
+    libAACenc/src/tonality.cpp \
+    libAACenc/src/aacEnc_rom.cpp \
+    libAACenc/src/bandwidth.cpp \
+    libAACenc/src/channel_map.cpp \
+    libAACenc/src/intensity.cpp \
+    libAACenc/src/ms_stereo.cpp \
+    libAACenc/src/psy_configuration.cpp \
+    libAACenc/src/sf_estim.cpp \
+    libAACenc/src/transform.cpp \
+    libAACenc/src/aacenc_lib.cpp \
+    libAACenc/src/aacenc_tns.cpp \
+    libAACenc/src/bit_cnt.cpp \
+    libAACenc/src/chaosmeasure.cpp \
+    libAACenc/src/line_pe.cpp \
+    libAACenc/src/noisedet.cpp \
+    libAACenc/src/psy_main.cpp \
+    libAACenc/src/spreading.cpp \
+    libAACenc/src/aacenc_pns.cpp \
+    libAACenc/src/adj_thr.cpp \
+    libAACenc/src/bitenc.cpp \
+    libAACenc/src/dyn_bits.cpp \
+    libAACenc/src/metadata_compressor.cpp \
+    libAACenc/src/pnsparam.cpp \
+    libAACenc/src/qc_main.cpp
+
+FDK_SRC = \
+    libFDK/src/autocorr2nd.cpp \
+    libFDK/src/dct.cpp \
+    libFDK/src/FDK_bitbuffer.cpp \
+    libFDK/src/FDK_core.cpp \
+    libFDK/src/FDK_crc.cpp \
+    libFDK/src/FDK_hybrid.cpp \
+    libFDK/src/FDK_tools_rom.cpp \
+    libFDK/src/FDK_trigFcts.cpp \
+    libFDK/src/fft.cpp \
+    libFDK/src/fft_rad2.cpp \
+    libFDK/src/fixpoint_math.cpp \
+    libFDK/src/mdct.cpp \
+    libFDK/src/qmf.cpp \
+    libFDK/src/scale.cpp \
+
+MPEGTPDEC_SRC = \
+    libMpegTPDec/src/tpdec_adif.cpp \
+    libMpegTPDec/src/tpdec_adts.cpp \
+    libMpegTPDec/src/tpdec_asc.cpp \
+    libMpegTPDec/src/tpdec_drm.cpp \
+    libMpegTPDec/src/tpdec_latm.cpp \
+    libMpegTPDec/src/tpdec_lib.cpp
+
+MPEGTPENC_SRC = \
+    libMpegTPEnc/src/tpenc_adif.cpp \
+    libMpegTPEnc/src/tpenc_adts.cpp \
+    libMpegTPEnc/src/tpenc_asc.cpp \
+    libMpegTPEnc/src/tpenc_latm.cpp \
+    libMpegTPEnc/src/tpenc_lib.cpp
+
+PCMUTILS_SRC = \
+    libPCMutils/src/limiter.cpp \
+    libPCMutils/src/pcmutils_lib.cpp
+
+SBRDEC_SRC = \
+    libSBRdec/src/env_calc.cpp \
+    libSBRdec/src/env_dec.cpp \
+    libSBRdec/src/env_extr.cpp \
+    libSBRdec/src/huff_dec.cpp \
+    libSBRdec/src/lpp_tran.cpp \
+    libSBRdec/src/psbitdec.cpp \
+    libSBRdec/src/psdec.cpp \
+    libSBRdec/src/psdec_hybrid.cpp \
+    libSBRdec/src/sbr_crc.cpp \
+    libSBRdec/src/sbr_deb.cpp \
+    libSBRdec/src/sbr_dec.cpp \
+    libSBRdec/src/sbrdec_drc.cpp \
+    libSBRdec/src/sbrdec_freq_sca.cpp \
+    libSBRdec/src/sbrdecoder.cpp \
+    libSBRdec/src/sbr_ram.cpp \
+    libSBRdec/src/sbr_rom.cpp
+
+SBRENC_SRC = \
+    libSBRenc/src/bit_sbr.cpp \
+    libSBRenc/src/env_bit.cpp \
+    libSBRenc/src/fram_gen.cpp \
+    libSBRenc/src/mh_det.cpp \
+    libSBRenc/src/ps_bitenc.cpp \
+    libSBRenc/src/ps_encode.cpp \
+    libSBRenc/src/resampler.cpp \
+    libSBRenc/src/sbr_encoder.cpp \
+    libSBRenc/src/sbr_ram.cpp \
+    libSBRenc/src/ton_corr.cpp \
+    libSBRenc/src/code_env.cpp \
+    libSBRenc/src/env_est.cpp \
+    libSBRenc/src/invf_est.cpp \
+    libSBRenc/src/nf_est.cpp \
+    libSBRenc/src/ps_main.cpp \
+    libSBRenc/src/sbrenc_freq_sca.cpp \
+    libSBRenc/src/sbr_misc.cpp \
+    libSBRenc/src/sbr_rom.cpp \
+    libSBRenc/src/tran_det.cpp
+
+SYS_SRC = \
+    libSYS/src/cmdl_parser.cpp \
+    libSYS/src/conv_string.cpp \
+    libSYS/src/genericStds.cpp \
+    libSYS/src/wav_file.cpp
+
+libfdk_aac_SOURCES = \
+    $(AACDEC_SRC) $(AACENC_SRC) \
+    $(MPEGTPDEC_SRC) $(MPEGTPENC_SRC) \
+    $(SBRDEC_SRC) $(SBRENC_SRC) \
+    $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC)
+
+
+aac_enc_SOURCES = aac-enc.c wavreader.c
+
+prefix = \usr\local
+prefix_win = $(prefix:/=\)	# In case we are using MSYS or MinGW.
+
+CFLAGS   = /nologo /W3 /Ox /MT /EHsc /Dinline=__inline $(TARGET_FLAGS) $(AM_CPPFLAGS) $(XCFLAGS)
+CXXFLAGS = $(CFLAGS)
+CPPFLAGS = $(CFLAGS)
+LDFLAGS  = -nologo $(XLDFLAGS)
+ARFLAGS  = -nologo
+
+incdir = $(prefix_win)\include\fdk-aac
+bindir = $(prefix_win)\bin
+libdir = $(prefix_win)\lib
+
+INST_DIRS = $(bindir) $(incdir) $(libdir)
+
+LIB_DEF    = fdk-aac.def
+STATIC_LIB = fdk-aac.lib
+SHARED_LIB = fdk-aac-1.dll
+IMP_LIB    = fdk-aac.dll.lib
+
+AAC_ENC_OBJS = $(aac_enc_SOURCES:.c=.obj)
+FDK_OBJS     = $(libfdk_aac_SOURCES:.cpp=.obj)
+
+PROGS = aac-enc.exe
+
+
+
+all: $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) $(PROGS)
+
+clean:
+	del /f $(LIB_DEF) $(STATIC_LIB) $(SHARED_LIB) $(IMP_LIB) $(PROGS) libfdk-aac.pc 2>NUL
+	del /f *.obj *.exp 2>NUL
+	del /f libAACdec\src\*.obj 2>NUL
+	del /f libAACenc\src\*.obj 2>NUL
+	del /f libFDK\src\*.obj 2>NUL
+	del /f libMpegTPDec\src\*.obj 2>NUL
+	del /f libMpegTPEnc\src\*.obj 2>NUL
+	del /f libPCMutils\src\*.obj 2>NUL
+	del /f libSBRdec\src\*.obj 2>NUL
+	del /f libSBRenc\src\*.obj 2>NUL
+	del /f libSYS\src\*.obj 2>NUL
+
+install: $(INST_DIRS)
+	copy libAACdec\include\aacdecoder_lib.h $(incdir)
+	copy libAACenc\include\aacenc_lib.h $(incdir)
+	copy libSYS\include\FDK_audio.h $(incdir)
+	copy libSYS\include\genericStds.h $(incdir)
+	copy libSYS\include\machine_type.h $(incdir)
+	copy $(STATIC_LIB) $(libdir)
+	copy $(IMP_LIB) $(libdir)
+	copy $(SHARED_LIB) $(bindir)
+	copy $(PROGS) $(bindir)
+	copy $(LIB_DEF) $(libdir)
+
+$(INST_DIRS):
+	@mkdir $(MKDIR_FLAGS) $@
+
+$(STATIC_LIB): $(FDK_OBJS)
+	$(AR) $(ARFLAGS) -out:$@ $(FDK_OBJS)
+
+$(IMP_LIB): $(SHARED_LIB)
+
+$(SHARED_LIB): $(FDK_OBJS)
+	$(LD) $(LDFLAGS) -OUT:$@ -DEF:$(LIB_DEF) -implib:$(IMP_LIB) -DLL $(FDK_OBJS)
+
+$(PROGS): $(AAC_ENC_OBJS)
+	$(LD) $(LDFLAGS) -out:$@ $(AAC_ENC_OBJS) $(STATIC_LIB)
+
+.cpp.obj:
+	$(CXX) $(CXXFLAGS) -c -Fo$@ $<
+
+$(LIB_DEF):
+	@echo EXPORTS > $(LIB_DEF)
+	@type fdk-aac.sym >> $(LIB_DEF)

aac-enc.c

@@ -18,7 +18,13 @@
 
 #include <stdio.h>
 #include <stdint.h>
+
+#if defined(_MSC_VER)
+#include <getopt.h>
+#else
 #include <unistd.h>
+#endif
+
 #include <stdlib.h>
 #include "libAACenc/include/aacenc_lib.h"
 #include "wavreader.h"

configure.ac

@@ -1,7 +1,7 @@
 dnl                                               -*- Autoconf -*-
 dnl Process this file with autoconf to produce a configure script.
 
-AC_INIT([fdk-aac], [0.1.3], [http://sourceforge.net/projects/opencore-amr/])
+AC_INIT([fdk-aac], [0.1.5], [http://sourceforge.net/projects/opencore-amr/])
 AC_CONFIG_AUX_DIR(.)
 AC_CONFIG_MACRO_DIR([m4])
 AM_INIT_AUTOMAKE([tar-ustar foreign])
@@ -26,7 +26,7 @@ AC_SEARCH_LIBS([sin], [m])
 dnl soname version to use
 dnl goes by ‘current[:revision[:age]]’ with the soname ending up as
 dnl current.age.revision
-FDK_AAC_VERSION=0:4:0
+FDK_AAC_VERSION=1:0:0
 
 AS_IF([test x$enable_shared = xyes], [LIBS_PRIVATE=$LIBS], [LIBS_PUBLIC=$LIBS])
 AC_SUBST(FDK_AAC_VERSION)

libAACdec/include/aacdecoder_lib.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -144,10 +144,9 @@ to allocate memory for the required structures, and the corresponding mpegFileRe
 files and to de-allocate associated structures. mpegFileRead_Open() tries to detect the bitstream format and
 in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio
 Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to
-provide this information manually (see \ref CommandLineUsage). For any other bitstream formats that are
-usually applicable in streaming applications, the decoder itself will try to synchronize and parse the given
-bitstream fragment using the FDK transport library. Hence, for streaming applications (without file access)
-this step is not necessary.
+provide this information manually. For any other bitstream formats that are usually applicable in streaming
+applications, the decoder itself will try to synchronize and parse the given bitstream fragment using the
+FDK transport library. Hence, for streaming applications (without file access) this step is not necessary.
 
 -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance.
 \dontinclude main.cpp
@@ -349,6 +348,10 @@ Where N equals to CStreamInfo::frameSize .
 
 #include "genericStds.h"
 
+#define AACDECODER_LIB_VL0 2
+#define AACDECODER_LIB_VL1 5
+#define AACDECODER_LIB_VL2 17
+
 /**
  * \brief  AAC decoder error codes.
  */
@@ -379,6 +382,7 @@ typedef enum {
                                                          not exist. */
   AAC_DEC_NEED_TO_RESTART                = 0x200B,  /*!< The decoder needs to be restarted, since the requiered configuration change cannot be
                                                          performed. */
+  AAC_DEC_OUTPUT_BUFFER_TOO_SMALL        = 0x200C,  /*!< The provided output buffer is too small. */
   aac_dec_init_error_end                 = 0x2FFF,
 
   /* Decode errors. Output buffer is valid but concealed. */
@@ -430,21 +434,68 @@ typedef enum {
 typedef enum
 {
   AAC_PCM_OUTPUT_INTERLEAVED              = 0x0000,  /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */
-  AAC_PCM_OUTPUT_CHANNELS                 = 0x0001,  /*!< Number of PCM output channels (if different from encoded audio channels, downmixing or
-                                                          upmixing is applied). \n
-                                                          -1: Disable up-/downmixing. The decoder output contains the same number of channels as the
-                                                              encoded bitstream. \n
-                                                           1: The decoder performs a mono matrix mix-down if the encoded audio channels are greater
-                                                              than one. Thus it ouputs always exact one channel. \n
-                                                           2: The decoder performs a stereo matrix mix-down if the encoded audio channels are greater
-                                                              than two. If the encoded audio channels are smaller than two the decoder duplicates the
-                                                              output. Thus it ouputs always exact two channels. \n */
-  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals:
-                                                          0: Leave both signals as they are (default).
-                                                          1: Create a dual mono output signal from channel 1.
-                                                          2: Create a dual mono output signal from channel 2.
+  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals: \n
+                                                          0: Leave both signals as they are (default). \n
+                                                          1: Create a dual mono output signal from channel 1. \n
+                                                          2: Create a dual mono output signal from channel 2. \n
                                                           3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
   AAC_PCM_OUTPUT_CHANNEL_MAPPING          = 0x0003,  /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */
+  AAC_PCM_LIMITER_ENABLE                  = 0x0004,  /*!< Enable signal level limiting. \n
+                                                          -1: Auto-config. Enable limiter for all non-lowdelay configurations by default. \n
+                                                           0: Disable limiter in general. \n
+                                                           1: Enable limiter always.
+                                                          It is recommended to call the decoder with a AACDEC_CLRHIST flag to reset all states when
+                                                          the limiter switch is changed explicitly. */
+  AAC_PCM_LIMITER_ATTACK_TIME             = 0x0005,  /*!< Signal level limiting attack time in ms.
+                                                          Default confguration is 15 ms. Adjustable range from 1 ms to 15 ms. */
+  AAC_PCM_LIMITER_RELEAS_TIME             = 0x0006,  /*!< Signal level limiting release time in ms.
+                                                          Default configuration is 50 ms. Adjustable time must be larger than 0 ms. */
+  AAC_PCM_MIN_OUTPUT_CHANNELS             = 0x0011,  /*!< Minimum number of PCM output channels. If higher than the number of encoded audio channels,
+                                                          a simple channel extension is applied. \n
+                                                          -1, 0: Disable channel extenstion feature. The decoder output contains the same number of
+                                                                 channels as the encoded bitstream. \n
+                                                           1:    This value is currently needed only together with the mix-down feature. See
+                                                                 ::AAC_PCM_MAX_OUTPUT_CHANNELS and note 2 below. \n
+                                                           2:    Encoded mono signals will be duplicated to achieve a 2/0/0.0 channel output
+                                                                 configuration. \n
+                                                           6:    The decoder trys to reorder encoded signals with less than six channels to achieve
+                                                                 a 3/0/2.1 channel output signal. Missing channels will be filled with a zero signal.
+                                                                 If reordering is not possible the empty channels will simply be appended. Only
+                                                                 available if instance is configured to support multichannel output. \n
+                                                           8:    The decoder trys to reorder encoded signals with less than eight channels to
+                                                                 achieve a 3/0/4.1 channel output signal. Missing channels will be filled with a
+                                                                 zero signal. If reordering is not possible the empty channels will simply be
+                                                                 appended. Only available if instance is configured to support multichannel output.\n
+                                                          NOTE: \n
+                                                            1. The channel signalling (CStreamInfo::pChannelType and CStreamInfo::pChannelIndices)
+                                                               will not be modified. Added empty channels will be signalled with channel type
+                                                               AUDIO_CHANNEL_TYPE::ACT_NONE. \n
+                                                            2. If the parameter value is greater than that of ::AAC_PCM_MAX_OUTPUT_CHANNELS both will
+                                                               be set to the same value. \n
+                                                            3. This parameter does not affect MPEG Surround processing. */
+  AAC_PCM_MAX_OUTPUT_CHANNELS             = 0x0012,  /*!< Maximum number of PCM output channels. If lower than the number of encoded audio channels,
+                                                          downmixing is applied accordingly. If dedicated metadata is available in the stream it
+                                                          will be used to achieve better mixing results. \n
+                                                          -1, 0: Disable downmixing feature. The decoder output contains the same number of channels
+                                                                 as the encoded bitstream. \n
+                                                           1:    All encoded audio configurations with more than one channel will be mixed down to
+                                                                 one mono output signal. \n
+                                                           2:    The decoder performs a stereo mix-down if the number encoded audio channels is
+                                                                 greater than two. \n
+                                                           6:    If the number of encoded audio channels is greater than six the decoder performs a
+                                                                 mix-down to meet the target output configuration of 3/0/2.1 channels. Only
+                                                                 available if instance is configured to support multichannel output. \n
+                                                           8:    This value is currently needed only together with the channel extension feature.
+                                                                 See ::AAC_PCM_MIN_OUTPUT_CHANNELS and note 2 below. Only available if instance is
+                                                                 configured to support multichannel output. \n
+                                                          NOTE: \n
+                                                            1. Down-mixing of any seven or eight channel configuration not defined in ISO/IEC 14496-3
+                                                               PDAM 4 is not supported by this software version. \n
+                                                            2. If the parameter value is greater than zero but smaller than ::AAC_PCM_MIN_OUTPUT_CHANNELS
+                                                               both will be set to same value. \n
+                                                            3. The operating mode of the MPEG Surround module will be set accordingly. \n
+                                                            4. Setting this param with any value will disable the binaural processing of the MPEG
+                                                               Surround module (::AAC_MPEGS_BINAURAL_ENABLE=0). */
 
   AAC_CONCEAL_METHOD                      = 0x0100,  /*!< Error concealment: Processing method. \n
                                                           0: Spectral muting. \n
@@ -485,18 +536,18 @@ typedef enum
  */
 typedef struct
 {
-  /* These three members are the only really relevant ones for the user.                                                           */
+  /* These five members are the only really relevant ones for the user.                                                            */
   INT               sampleRate;          /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing).   */
   INT               frameSize;           /*!< The frame size of the decoded PCM audio signal. \n
                                               1024 or 960 for AAC-LC \n
                                               2048 or 1920 for HE-AAC (v2) \n
                                               512 or 480 for AAC-LD and AAC-ELD                                                    */
   INT               numChannels;         /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */
-  AUDIO_CHANNEL_TYPE *pChannelType;       /*!< Audio channel type of each output audio channel.           */
-  UCHAR             *pChannelIndices;     /*!< Audio channel index for each output audio channel.
+  AUDIO_CHANNEL_TYPE *pChannelType;      /*!< Audio channel type of each output audio channel.                                     */
+  UCHAR             *pChannelIndices;    /*!< Audio channel index for each output audio channel.
                                                See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
   /* Decoder internal members. */
-  INT               aacSampleRate;       /*!< sampling rate in Hz without SBR (from configuration info).                           */
+  INT               aacSampleRate;       /*!< Sampling rate in Hz without SBR (from configuration info).                           */
   INT               profile;             /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)).               */
   AUDIO_OBJECT_TYPE aot;                 /*!< Audio Object Type (from ASC): is set to the appropriate value for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */
   INT               channelConfig;       /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ...                       */
@@ -509,7 +560,9 @@ typedef struct
   AUDIO_OBJECT_TYPE extAot;              /*!< Extension Audio Object Type (from ASC)   */
   INT               extSamplingRate;     /*!< Extension sampling rate in Hz (from ASC) */
 
-  UINT              flags;               /*!< Copy if internal flags. Only to be written by the decoder, and only to be read externally. */
+  UINT              outputDelay;         /*!< The number of samples the output is additionally delayed by the decoder. */
+
+  UINT              flags;               /*!< Copy of internal flags. Only to be written by the decoder, and only to be read externally. */
 
   SCHAR             epConfig;            /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.)  */
 
@@ -522,10 +575,25 @@ typedef struct
   UINT              numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */
   UINT              numBadAccessUnits;   /*!< This is the number of total access units that were considered with errors from numTotalBytes. */
 
+  /* Metadata */
+  SCHAR             drcProgRefLev;       /*!< DRC program reference level. Defines the reference level below full-scale.
+                                              It is quantized in steps of 0.25dB. The valid values range from 0 (0 dBFS) to 127 (-31.75 dBFS).
+                                              It is used to reflect the average loudness of the audio in LKFS accoring to ITU-R BS 1770.
+                                              If no level has been found in the bitstream the value is -1. */
+  SCHAR             drcPresMode;         /*!< DRC presentation mode. According to ETSI TS 101 154, this field indicates whether
+                                              light (MPEG-4 Dynamic Range Control tool) or heavy compression (DVB heavy compression)
+                                              dynamic range control shall take priority on the outputs.
+                                              For details, see ETSI TS 101 154, table C.33. Possible values are: \n
+                                              -1: No corresponding metadata found in the bitstream \n
+                                               0: DRC presentation mode not indicated \n
+                                               1: DRC presentation mode 1 \n
+                                               2: DRC presentation mode 2 \n
+                                               3: Reserved */
+
 } CStreamInfo;
 
 
-typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;
+typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;  /*!< Pointer to a AAC decoder instance. */
 
 #ifdef __cplusplus
 extern "C"
@@ -634,11 +702,15 @@ aacDecoder_Fill ( HANDLE_AACDECODER  self,
                   const UINT         bufferSize[],
                   UINT              *bytesValid );
 
-#define AACDEC_CONCEAL  1 /*!< Flag for aacDecoder_DecodeFrame(): do not consider new input data. Do concealment. */
-#define AACDEC_FLUSH    2 /*!< Flag for aacDecoder_DecodeFrame(): Do not consider new input data. Flush filterbanks (output delayed audio). */
-#define AACDEC_INTR     4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data discontinuity. Resync any internals as necessary. */
-#define AACDEC_CLRHIST  8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history buffers.
-                               Caution: This can cause discontinuities in the output signal. */
+#define AACDEC_CONCEAL  1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment module \
+                                 to generate a substitute signal for one lost frame. New input data will not be
+                                 considered. */
+#define AACDEC_FLUSH    2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed audio \
+                                 without having new input data. Thus new input data will not be considered.*/
+#define AACDEC_INTR     4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data discontinuity. \
+                                 Resync any internals as necessary. */
+#define AACDEC_CLRHIST  8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history buffers.\
+                                 CAUTION: This can cause discontinuities in the output signal. */
 
 /**
  * \brief            Decode one audio frame

libAACdec/src/aac_ram.cpp

@@ -108,15 +108,15 @@ C_ALLOC_MEM(AacDecoder, AAC_DECODER_INSTANCE, 1)
 /*! The structure CAacDecoderStaticChannelInfo contains the static sideinfo which is needed
     for the decoding of one aac channel. <br>
     Dimension: #AacDecoderChannels                                                      */
-C_ALLOC_MEM2(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo, 1, (6))
+C_ALLOC_MEM2(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo, 1, (8))
 
 /*! The structure CAacDecoderChannelInfo contains the dynamic sideinfo which is needed
     for the decoding of one aac channel. <br>
     Dimension: #AacDecoderChannels                                                      */
-C_ALLOC_MEM2(AacDecoderChannelInfo, CAacDecoderChannelInfo, 1, (6))
+C_AALLOC_MEM2(AacDecoderChannelInfo, CAacDecoderChannelInfo, 1, (8))
 
 /*! Overlap buffer */
-C_ALLOC_MEM2(OverlapBuffer, FIXP_DBL, OverlapBufferSize, (6))
+C_ALLOC_MEM2(OverlapBuffer, FIXP_DBL, OverlapBufferSize, (8))
 
 C_ALLOC_MEM(DrcInfo, CDrcInfo, 1)
 
@@ -128,7 +128,7 @@ C_ALLOC_MEM(DrcInfo, CDrcInfo, 1)
   Dynamic memory areas, might be reused in other algorithm sections,
   e.g. the sbr decoder
 */
-C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((6)*1024), SECT_DATA_L2, WORKBUFFER2_TAG)
+C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8)*1024), SECT_DATA_L2, WORKBUFFER2_TAG)
 
 
 C_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1, 1, SECT_DATA_L1, WORKBUFFER1_TAG)

libAACdec/src/aac_rom.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -167,6 +167,36 @@ const SCHAR ExponentTable [4][14] =
 } ;
 
 
+/* 41 scfbands */
+static const SHORT sfb_96_1024[42] =
+{
+     0,    4,    8,   12,   16,   20,   24,   28,   32,   36,   40,   44,
+    48,   52,   56,   64,   72,   80,   88,   96,  108,  120,  132,  144,
+   156,  172,  188,  212,  240,  276,  320,  384,  448,  512,  576,  640,
+   704,  768,  832,  896,  960, 1024
+};
+/* 12 scfbands */
+static const SHORT sfb_96_128[13] =
+{
+     0,    4,    8,   12,   16,   20,   24,   32,   40,   48,    64,  92,
+   128
+};
+
+/* 47 scfbands*/
+static const SHORT sfb_64_1024[48] =
+{
+     0,   4,   8,  12,  16,  20,  24,   28,  32,  36,  40,  44,  48,  52,
+    56,  64,  72,  80,  88, 100, 112,  124, 140, 156, 172, 192, 216, 240,
+   268, 304, 344, 384, 424, 464, 504,  544, 584, 624, 664, 704, 744, 784,
+   824, 864, 904, 944, 984,1024
+};
+
+/* 12 scfbands */
+static const SHORT sfb_64_128[13] =
+{
+     0,   4,   8,  12,  16,  20,  24,
+    32,  40,  48,  64,  92, 128
+};
 
 /* 49 scfbands */
 static const SHORT sfb_48_1024[50] = {
@@ -239,6 +269,35 @@ static const SHORT sfb_8_128[16] =
 };
 
 
+static const SHORT sfb_96_960[42] =
+{
+  0,      4,      8,      12,     16,     20,     24,     28,     32,     36,
+  40,     44,     48,     52,     56,     64,     72,     80,     88,     96,
+  108,    120,    132,    144,    156,    172,    188,    212,    240,    276,
+  320,    384,    448,    512,    576,    640,    704,    768,    832,    896,
+  960
+};   /* 40 scfbands */
+
+static const SHORT sfb_96_120[13] =
+{
+  0,      4,      8,      12,     16,     20,     24,     32,     40,     48,
+  64,     92,     120
+};   /* 12 scfbands */
+
+static const SHORT sfb_64_960[47] =
+{
+  0,      4,      8,      12,     16,     20,     24,     28,     32,     36,
+  40,     44,     48,     52,     56,     64,     72,     80,     88,     100,
+  112,    124,    140,    156,    172,    192,    216,    240,    268,    304,
+  344,    384,    424,    464,    504,    544,    584,    624,    664,    704,
+  744,    784,    824,    864,    904,    944,    960
+};   /* 46 scfbands */
+
+static const SHORT sfb_64_120[13] =
+{
+  0,      4,      8,      12,     16,     20,     24,     32,     40,     48,
+  64,     92,     120
+};   /* 12 scfbands */
 
 static const SHORT sfb_48_960[50] =
 {
@@ -358,9 +417,9 @@ static const SHORT sfb_24_480[31] =
 const SFB_INFO sfbOffsetTables[5][16] =
 {
   {
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
+    { sfb_96_1024, sfb_96_128, 41, 12 },
+    { sfb_96_1024, sfb_96_128, 41, 12 },
+    { sfb_64_1024, sfb_64_128, 47, 12 },
     { sfb_48_1024, sfb_48_128, 49, 14 },
     { sfb_48_1024, sfb_48_128, 49, 14 },
     { sfb_32_1024, sfb_48_128, 51, 14 },
@@ -372,9 +431,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
     { sfb_8_1024, sfb_8_128, 40, 15 },
     { sfb_8_1024, sfb_8_128, 40, 15 },
   }, {
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
+    { sfb_96_960, sfb_96_120, 40, 12 },
+    { sfb_96_960, sfb_96_120, 40, 12 },
+    { sfb_64_960, sfb_64_120, 46, 12 },
     { sfb_48_960, sfb_48_120, 49, 14 },
     { sfb_48_960, sfb_48_120, 49, 14 },
     { sfb_32_960, sfb_48_120, 49, 14 },
@@ -388,9 +447,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
   }, {
     { NULL, NULL, 0, 0 },
   }, {
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
+    { sfb_48_512,  NULL, 36, 0 },
+    { sfb_48_512,  NULL, 36, 0 },
+    { sfb_48_512,  NULL, 36, 0 },
     { sfb_48_512,  NULL, 36, 0 },
     { sfb_48_512, NULL, 36, 0},
     { sfb_32_512, NULL, 37, 0 },
@@ -402,9 +461,9 @@ const SFB_INFO sfbOffsetTables[5][16] =
     { sfb_24_512, NULL, 31, 0 },
     { sfb_24_512, NULL, 31, 0 },
   }, {
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
-    { NULL, NULL, 0, 0 },
+    { sfb_48_480, NULL, 35, 0 },
+    { sfb_48_480, NULL, 35, 0 },
+    { sfb_48_480, NULL, 35, 0 },
     { sfb_48_480, NULL, 35, 0 },
     { sfb_48_480, NULL, 35, 0 },
     { sfb_32_480, NULL, 37, 0 },
@@ -1777,42 +1836,62 @@ const FIXP_TCC FDKaacDec_tnsCoeff4 [16] =
 };
 
 /* MPEG like mapping (no change). */
-const UCHAR channelMappingTablePassthrough[8][8] =
+const UCHAR channelMappingTablePassthrough[15][8] =
 {
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  fallback  */
      { 0,  1,255,255,255,255,255,255},  /*  mono / PS */
      { 0,  1,255,255,255,255,255,255},  /*  stereo    */
      { 0,  1,  2,255,255,255,255,255},  /*  3ch       */
      { 0,  1,  2,  3,255,255,255,255},  /*  4ch       */
      { 0,  1,  2,  3,  4,255,255,255},  /*  5ch       */
      { 0,  1,  2,  3,  4,  5,255,255},  /*  5.1ch     */
-     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7ch       */
-     { 0,  1,  2,  3,  4,  5,  6,  7}   /*  7.1ch     */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7.1 front */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,255},  /*  6.1ch     */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7.1 rear  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7}   /*  7.1 top   */
 };
 
 /* WAV file like mapping (from MPEG mapping). */
-const UCHAR channelMappingTableWAV[8][8] =
+const UCHAR channelMappingTableWAV[15][8] =
 {
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  fallback  */
      { 0,  1,255,255,255,255,255,255},  /*  mono / PS */
      { 0,  1,255,255,255,255,255,255},  /*  stereo    */
      { 2,  0,  1,255,255,255,255,255},  /*  3ch       */
      { 2,  0,  1,  3,255,255,255,255},  /*  4ch       */
      { 2,  0,  1,  3,  4,255,255,255},  /*  5ch       */
      { 2,  0,  1,  4,  5,  3,255,255},  /*  5.1ch     */
-     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7ch       */
-     { 2,  0,  1,  6,  7,  4,  5,  3}   /*  7.1ch     */
+     { 2,  6,  7,  0,  1,  4,  5,  3},  /*  7.1 front */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 2,  0,  1,  4,  5,  6,  3,255},  /*  6.1ch     */
+     { 2,  0,  1,  6,  7,  4,  5,  3},  /*  7.1 rear  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 2,  0,  1,  4,  5,  3,  6,  7}   /*  7.1 top   */
 };
 
 /* Lookup tables for elements in ER bitstream */
-const MP4_ELEMENT_ID elementsTab[8][7] =
+const MP4_ELEMENT_ID elementsTab[15][7] =
 {
-  {ID_SCE, ID_EXT, ID_END, ID_NONE, ID_NONE,ID_NONE,ID_NONE }, /* 1 channel  */
-  {ID_CPE, ID_EXT, ID_END, ID_NONE, ID_NONE,ID_NONE,ID_NONE }  /* 2 channels */
-  ,
-  {ID_SCE, ID_CPE, ID_EXT, ID_END, ID_NONE,ID_NONE,ID_NONE }, /* 3 channels */
-  {ID_SCE, ID_CPE, ID_SCE, ID_EXT, ID_END, ID_NONE,ID_NONE }, /* 4 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_EXT, ID_END, ID_NONE,ID_NONE }, /* 5 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END, ID_NONE }, /* 6 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END}    /* 8 channels */
+ /*  1 */  { ID_SCE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE, ID_NONE }, /* 1 channel  */
+ /*  2 */  { ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE, ID_NONE }  /* 2 channels */
+ /*  3 */ ,{ ID_SCE,  ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE }, /* 3 channels */
+ /*  4 */  { ID_SCE,  ID_CPE,  ID_SCE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE }, /* 4 channels */
+ /*  5 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE }, /* 5 channels */
+ /*  6 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END,  ID_NONE }  /* 6 channels */
+ /*  7 */ ,{ ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END  },  /* 8 channels */
+ /*  8 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /*  9 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 10 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 11 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_SCE,  ID_LFE,  ID_EXT,  ID_END  },  /* 7 channels */
+ /* 12 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END  },  /* 8 channels */
+ /* 13 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 14 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_CPE,  ID_EXT,  ID_END  }   /* 8 channels */
 };
 
 /*! Random sign bit used for concealment

libAACdec/src/aac_rom.h

@@ -177,11 +177,12 @@ extern const USHORT randomSign[AAC_NF_NO_RANDOM_VAL/16];
 extern const FIXP_DBL pow2_div24minus1[47];
 extern const int offsetTab[2][16];
 
-/* Channel mapping indices for time domain I/O. First dimension is channel count-1. */
-extern const UCHAR channelMappingTablePassthrough[8][8];
-extern const UCHAR channelMappingTableWAV[8][8];
+/* Channel mapping indices for time domain I/O.
+   The first dimension is the channel configuration index. */
+extern const UCHAR channelMappingTablePassthrough[15][8];
+extern const UCHAR channelMappingTableWAV[15][8];
 
 /* Lookup tables for elements in ER bitstream */
-extern const MP4_ELEMENT_ID elementsTab[8][7];
+extern const MP4_ELEMENT_ID elementsTab[15][7];
 
 #endif /* #ifndef AAC_ROM_H */

libAACdec/src/aacdec_drc.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -130,7 +130,6 @@ void aacDecoder_drcInit (
   /* init control fields */
   self->enable = 0;
   self->numThreads = 0;
-  self->digitalNorm = 0;
 
   /* init params */
   pParams = &self->params;
@@ -139,12 +138,15 @@ void aacDecoder_drcInit (
   pParams->usrCut   = FL2FXCONST_DBL(0.0f);
   pParams->boost    = FL2FXCONST_DBL(0.0f);
   pParams->usrBoost = FL2FXCONST_DBL(0.0f);
-  pParams->targetRefLevel = AACDEC_DRC_DEFAULT_REF_LEVEL;
+  pParams->targetRefLevel = -1;
   pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
+  pParams->applyDigitalNorm = 0;
   pParams->applyHeavyCompression = 0;
 
   /* initial program ref level = target ref level */
   self->progRefLevel = pParams->targetRefLevel;
+  self->progRefLevelPresent = 0;
+  self->presMode = -1;
 }
 
 
@@ -222,11 +224,12 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
       return AAC_DEC_INVALID_HANDLE;
     }
     if (value < 0) {
-      self->digitalNorm = 0;
+      self->params.applyDigitalNorm = 0;
+      self->params.targetRefLevel = -1;
     }
     else {
       /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */
-      self->digitalNorm    = 1;
+      self->params.applyDigitalNorm = 1;
       if (self->params.targetRefLevel != (SCHAR)value) {
         self->params.targetRefLevel = (SCHAR)value;
         self->progRefLevel = (SCHAR)value;  /* Always set the program reference level equal to the
@@ -234,6 +237,16 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
       }
     }
     break;
+  case APPLY_NORMALIZATION:
+    if (value < 0 || value > 1) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    if (self == NULL) {
+      return AAC_DEC_INVALID_HANDLE;
+    }
+    /* Store new parameter value */
+    self->params.applyDigitalNorm = (UCHAR)value;
+    break;
   case APPLY_HEAVY_COMPRESSION:
     if (value < 0 || value > 1) {
       return AAC_DEC_SET_PARAM_FAIL;
@@ -278,7 +291,7 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
   self->enable = ( (self->params.boost > (FIXP_DBL)0)
                 || (self->params.cut   > (FIXP_DBL)0)
                 || (self->params.applyHeavyCompression != 0)
-                || (self->digitalNorm == 1) );
+                || (self->params.targetRefLevel >= 0) );
 
 
   return ErrorStatus;
@@ -539,7 +552,7 @@ static int aacDecoder_drcReadCompression (
     UINT                  payloadPosition )
 {
   int  bitCnt = 0;
-  int  dmxLevelsPresent, compressionPresent;
+  int  dmxLevelsPresent, extensionPresent, compressionPresent;
   int  coarseGrainTcPresent, fineGrainTcPresent;
 
   /* Move to the beginning of the DRC payload field */
@@ -561,8 +574,9 @@ static int aacDecoder_drcReadCompression (
     return 0;
   }
   FDKreadBits(bs, 2);                          /* dolby_surround_mode */
-  FDKreadBits(bs, 2);                          /* presentation_mode */
-  if (FDKreadBits(bs, 2) != 0) {               /* reserved, set to 0 */
+  pDrcBs->presMode = FDKreadBits(bs, 2);       /* presentation_mode */
+  FDKreadBits(bs, 1);                          /* stereo_downmix_mode */
+  if (FDKreadBits(bs, 1) != 0) {               /* reserved, set to 0 */
     return 0;
   }
 
@@ -571,9 +585,7 @@ static int aacDecoder_drcReadCompression (
     return 0;
   }
   dmxLevelsPresent = FDKreadBits(bs, 1);       /* downmixing_levels_MPEG4_status */
-  if (FDKreadBits(bs, 1) != 0) {               /* reserved, set to 0 */
-    return 0;
-  }
+  extensionPresent = FDKreadBits(bs, 1);       /* ancillary_data_extension_status; */
   compressionPresent   = FDKreadBits(bs, 1);   /* audio_coding_mode_and_compression status */
   coarseGrainTcPresent = FDKreadBits(bs, 1);   /* coarse_grain_timecode_status */
   fineGrainTcPresent   = FDKreadBits(bs, 1);   /* fine_grain_timecode_status */
@@ -631,6 +643,19 @@ static int aacDecoder_drcReadCompression (
     bitCnt += 16;
   }
 
+  /* Read extension just to get the right amount of bits. */
+  if (extensionPresent) {
+    int  extBits = 8;
+
+    FDKreadBits(bs, 1);                     /* reserved, set to 0 */
+    if (FDKreadBits(bs, 1)) extBits += 8;   /* ext_downmixing_levels_status */
+    if (FDKreadBits(bs, 1)) extBits += 16;  /* ext_downmixing_global_gains_status */
+    if (FDKreadBits(bs, 1)) extBits += 8;   /* ext_downmixing_lfe_level_status */
+
+    FDKpushFor(bs, extBits - 4);            /* skip the extension payload remainder. */
+    bitCnt += extBits;
+  }
+
   return (bitCnt);
 }
 
@@ -680,7 +705,7 @@ static int aacDecoder_drcExtractAndMap (
   }
   self->numPayloads = 0;
 
-  if (self->dvbAncDataAvailable)
+  if (self->dvbAncDataAvailable && self->numThreads < MAX_DRC_THREADS)
   { /* Append a DVB heavy compression payload thread if available. */
     int bitsParsed;
 
@@ -780,9 +805,15 @@ static int aacDecoder_drcExtractAndMap (
      */
     if (pThreadBs->progRefLevel >= 0) {
       self->progRefLevel = pThreadBs->progRefLevel;
+      self->progRefLevelPresent = 1;
       self->prlExpiryCount = 0;  /* Got a new value -> Reset counter */
     }
 
+    if (drcPayloadType == DVB_DRC_ANC_DATA) {
+      /* Announce the presentation mode of this valid thread. */
+      self->presMode = pThreadBs->presMode;
+    }
+
     /* SCE, CPE and LFE */
     for (ch = 0; ch < validChannels; ch++) {
       int mapedChannel = channelMapping[ch];
@@ -802,6 +833,7 @@ static int aacDecoder_drcExtractAndMap (
   if ( (pParams->expiryFrame > 0)
     && (self->prlExpiryCount++ > pParams->expiryFrame) )
   { /* The program reference level is too old, so set it back to the target level. */
+    self->progRefLevelPresent = 0;
     self->progRefLevel = pParams->targetRefLevel;
     self->prlExpiryCount = 0;
   }
@@ -815,6 +847,7 @@ void aacDecoder_drcApply (
         void                   *pSbrDec,
         CAacDecoderChannelInfo *pAacDecoderChannelInfo,
         CDrcChannelData        *pDrcChData,
+        FIXP_DBL               *extGain,
         int  ch,   /* needed only for SBR */
         int  aacFrameSize,
         int  bSbrPresent )
@@ -826,8 +859,8 @@ void aacDecoder_drcApply (
   FIXP_DBL max_mantissa;
   INT max_exponent;
 
-  FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.0f);
-  INT  norm_exponent = 0;
+  FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.5f);
+  INT  norm_exponent = 1;
 
   FIXP_DBL fact_mantissa[MAX_DRC_BANDS];
   INT  fact_exponent[MAX_DRC_BANDS];
@@ -849,6 +882,15 @@ void aacDecoder_drcApply (
 
   if (!self->enable) {
     sbrDecoder_drcDisable( (HANDLE_SBRDECODER)pSbrDec, ch );
+    if (extGain != NULL) {
+      INT gainScale = (INT)*extGain;
+      /* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
+      if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
+        *extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
+      } else {
+        FDK_ASSERT(0);
+      }
+    }
     return;
   }
 
@@ -864,7 +906,7 @@ void aacDecoder_drcApply (
   reduced DAC SNR (if signal is attenuated) or clipping (if signal is
   boosted) */
 
-  if (self->digitalNorm == 1)
+  if (pParams->targetRefLevel >= 0)
   {
     /* 0.5^((targetRefLevel - progRefLevel)/24) */
     norm_mantissa = fLdPow(
@@ -874,7 +916,18 @@ void aacDecoder_drcApply (
             3,
            &norm_exponent );
   }
-  else {
+  /* Always export the normalization gain (if possible). */
+  if (extGain != NULL) {
+    INT gainScale = (INT)*extGain;
+    /* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
+    if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
+      *extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
+    } else {
+      FDK_ASSERT(0);
+    }
+  }
+  if (self->params.applyDigitalNorm == 0) {
+    /* Reset normalization gain since this module must not apply it */
     norm_mantissa = FL2FXCONST_DBL(0.5f);
     norm_exponent = 1;
   }
@@ -1112,3 +1165,24 @@ int aacDecoder_drcEpilog (
   return err;
 }
 
+/*
+ * Export relevant metadata info from bitstream payload.
+ */
+void aacDecoder_drcGetInfo (
+        HANDLE_AAC_DRC  self,
+        SCHAR          *pPresMode,
+        SCHAR          *pProgRefLevel )
+{
+  if (self != NULL) {
+    if (pPresMode != NULL) {
+      *pPresMode = self->presMode;
+    }
+    if (pProgRefLevel != NULL) {
+      if (self->progRefLevelPresent) {
+        *pProgRefLevel = self->progRefLevel;
+      } else {
+        *pProgRefLevel = -1;
+      }
+    }
+  }
+}

libAACdec/src/aacdec_drc.h

@@ -98,7 +98,6 @@ amm-info@iis.fraunhofer.de
 #include "channel.h"
 #include "FDK_bitstream.h"
 
-#define AACDEC_DRC_DEFAULT_REF_LEVEL  ( 108 )   /* -27 dB below full scale (typical for movies) */
 #define AACDEC_DRC_DFLT_EXPIRY_FRAMES (  50 )   /* Default DRC data expiry time in AAC frames   */
 
 /**
@@ -111,6 +110,7 @@ typedef enum
   TARGET_REF_LEVEL,
   DRC_BS_DELAY,
   DRC_DATA_EXPIRY_FRAME,
+  APPLY_NORMALIZATION,
   APPLY_HEAVY_COMPRESSION
 
 } AACDEC_DRC_PARAM;
@@ -149,6 +149,8 @@ int aacDecoder_drcProlog (
  * \param pSbrDec pointer to SBR decoder instance
  * \param pAacDecoderChannelInfo AAC decoder channel instance to be processed
  * \param pDrcDat DRC channel data
+ * \param extGain Pointer to a FIXP_DBL where a externally applyable gain will be stored into (independently on whether it will be apply internally or not).
+ *                At function call the buffer must hold the scale (0 >= scale < DFRACT_BITS) to be applied on the gain value.
  * \param ch channel index
  * \param aacFrameSize AAC frame size
  * \param bSbrPresent flag indicating that SBR is present, in which case DRC is handed over to the SBR instance pSbrDec
@@ -158,6 +160,7 @@ void aacDecoder_drcApply (
         void                   *pSbrDec,
         CAacDecoderChannelInfo *pAacDecoderChannelInfo,
         CDrcChannelData        *pDrcDat,
+        FIXP_DBL               *extGain,
         int  ch,
         int  aacFrameSize,
         int  bSbrPresent );
@@ -170,5 +173,17 @@ int aacDecoder_drcEpilog (
         UCHAR  channelMapping[],
         int    validChannels );
 
+/**
+ * \brief Get metadata information found in bitstream.
+ * \param self DRC module instance handle.
+ * \param pPresMode Pointer to field where the presentation mode will be written to.
+ * \param pProgRefLevel Pointer to field where the program reference level will be written to.
+ * \return Nothing.
+ */
+void aacDecoder_drcGetInfo (
+        HANDLE_AAC_DRC  self,
+        SCHAR  *pPresMode,
+        SCHAR  *pProgRefLevel );
+
 
 #endif  /* AACDEC_DRC_H */

libAACdec/src/aacdec_drc_types.h

@@ -124,6 +124,7 @@ typedef struct
 {
   UINT   excludedChnsMask;
   SCHAR  progRefLevel;
+  SCHAR  presMode;     /* Presentation mode: 0 (not indicated), 1, 2, and 3 (reserved). */
   SCHAR  pceInstanceTag;
 
   CDrcChannelData channelData;
@@ -140,6 +141,7 @@ typedef struct
   UINT   expiryFrame;
   SCHAR  targetRefLevel;
   UCHAR  bsDelayEnable;
+  UCHAR  applyDigitalNorm;
   UCHAR  applyHeavyCompression;
 
 } CDrcParams;
@@ -155,9 +157,11 @@ typedef struct
   USHORT numPayloads;                        /* The number of DRC data payload elements found within frame */
   USHORT numThreads;                         /* The number of DRC data threads extracted from the found payload elements */
   SCHAR  progRefLevel;                       /* Program reference level for all channels */
+  UCHAR  progRefLevelPresent;                /* Program reference level found in bitstream */
 
   UINT   prlExpiryCount;                     /* Counter that can be used to monitor the life time of the program reference level. */
 
+  SCHAR  presMode;                            /* Presentation mode as defined in ETSI TS 101 154 */
   UCHAR  dvbAncDataAvailable;                  /* Flag that indicates whether DVB ancillary data is present or not */
   UINT   dvbAncDataPosition;                   /* Used to store the DVB ancillary data payload position in the bitstream (only one per frame) */
   UINT   drcPayloadPosition[MAX_DRC_THREADS];  /* Used to store the DRC payload positions in the bitstream */

libAACdec/src/aacdecoder.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -157,23 +157,9 @@ amm-info@iis.fraunhofer.de
 
 #include "conceal.h"
 
+  #include "FDK_crc.h"
 
 
-#define CAN_DO_PS(aot) \
-  ((aot) == AOT_AAC_LC \
-|| (aot) == AOT_SBR \
-|| (aot) == AOT_PS \
-|| (aot) == AOT_ER_BSAC \
-|| (aot) == AOT_DRM_AAC)
-
-#define IS_USAC(aot) \
-  ((aot) == AOT_USAC \
-|| (aot) == AOT_RSVD50)
-
-#define IS_LOWDELAY(aot) \
-  ((aot) == AOT_ER_AAC_LD \
-|| (aot) == AOT_ER_AAC_ELD)
-
 void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self)
 {
 
@@ -338,17 +324,22 @@ AAC_DECODER_ERROR CAacDecoder_AncDataParse (
   \return  Error code
 */
 static AAC_DECODER_ERROR CDataStreamElement_Read (
+                                                  HANDLE_AACDECODER    self,
                                                   HANDLE_FDK_BITSTREAM bs,
-                                                  CAncData *ancData,
-                                                  HANDLE_AAC_DRC hDrcInfo,
-                                                  HANDLE_TRANSPORTDEC pTp,
                                                   UCHAR    *elementInstanceTag,
                                                   UINT      alignmentAnchor )
 {
+  HANDLE_TRANSPORTDEC  pTp;
+  CAncData *ancData;
   AAC_DECODER_ERROR error = AAC_DEC_OK;
-  UINT dataStart;
+  UINT dataStart, dseBits;
   int dataByteAlignFlag, count;
 
+  FDK_ASSERT(self != NULL);
+
+  ancData = &self->ancData;
+  pTp = self->hInput;
+
   int crcReg = transportDec_CrcStartReg(pTp, 0);
 
   /* Element Instance Tag */
@@ -361,6 +352,7 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
   if (count == 255) {
     count += FDKreadBits(bs,8); /* EscCount */
   }
+  dseBits = count*8;
 
   if (dataByteAlignFlag) {
     FDKbyteAlign(bs, alignmentAnchor);
@@ -372,20 +364,30 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
   transportDec_CrcEndReg(pTp, crcReg);
 
   {
-    INT readBits, dataBits = count<<3;
-
     /* Move to the beginning of the data junk */
     FDKpushBack(bs, dataStart-FDKgetValidBits(bs));
 
     /* Read Anc data if available */
-    readBits = aacDecoder_drcMarkPayload( hDrcInfo, bs, DVB_DRC_ANC_DATA );
-
-    if (readBits != dataBits) {
-      /* Move to the end again. */
-      FDKpushBiDirectional(bs, FDKgetValidBits(bs)-dataStart+dataBits);
-    }
+    aacDecoder_drcMarkPayload( self->hDrcInfo, bs, DVB_DRC_ANC_DATA );
   }
 
+  {
+    PCMDMX_ERROR dmxErr = PCMDMX_OK;
+
+    /* Move to the beginning of the data junk */
+    FDKpushBack(bs, dataStart-FDKgetValidBits(bs));
+
+    /* Read DMX meta-data */
+    dmxErr = pcmDmx_Parse (
+                     self->hPcmUtils,
+                     bs,
+                     dseBits,
+                     0 /* not mpeg2 */ );
+    }
+
+  /* Move to the very end of the element. */
+  FDKpushBiDirectional(bs, FDKgetValidBits(bs)-dataStart+dseBits);
+
   return error;
 }
 
@@ -536,8 +538,9 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
                 previous_element,
                 elIndex,
                 self->flags & AC_INDEP );
-        /* Enable SBR for implicit SBR signalling. */
-        if (sbrError == SBRDEC_OK) {
+        /* Enable SBR for implicit SBR signalling but only if no severe error happend. */
+        if ( (sbrError == SBRDEC_OK)
+          || (sbrError == SBRDEC_PARSE_ERROR) ) {
           self->sbrEnabled = 1;
         }
       } else {
@@ -552,7 +555,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
         FDKpushBiDirectional(hBs, *count);
         *count = 0;
       } else {
-        /* If this is not a fill element with a known length, we are screwed an no further parsing makes sense. */
+        /* If this is not a fill element with a known length, we are screwed and further parsing makes no sense. */
         if (sbrError != SBRDEC_OK) {
           self->frameOK = 0;
         }
@@ -701,6 +704,12 @@ void CStreamInfoInit(CStreamInfo *pStreamInfo)
   pStreamInfo->numChannels = 0;
   pStreamInfo->sampleRate = 0;
   pStreamInfo->frameSize = 0;
+
+  pStreamInfo->outputDelay = 0;
+
+  /* DRC */
+  pStreamInfo->drcProgRefLev = -1;                           /* set program reference level to not indicated */
+  pStreamInfo->drcPresMode = -1;                             /* default: presentation mode not indicated */
 }
 
 /*!
@@ -774,7 +783,7 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self)
   if (self == NULL)
     return;
 
-  for (ch=0; ch<(6); ch++) {
+  for (ch=0; ch<(8); ch++) {
     if (self->pAacDecoderStaticChannelInfo[ch] != NULL) {
       if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer != NULL) {
         FreeOverlapBuffer (&self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer);
@@ -825,12 +834,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
   switch (asc->m_aot) {
   case AOT_AAC_LC:
     self->streamInfo.profile = 1;
-    break;
+
+  case AOT_ER_AAC_SCAL:
+    if (asc->m_sc.m_gaSpecificConfig.m_layer > 0) {
+      /* aac_scalable_extension_element() currently not supported. */
+      return AAC_DEC_UNSUPPORTED_FORMAT;
+    }
 
   case AOT_SBR:
   case AOT_PS:
   case AOT_ER_AAC_LD:
   case AOT_ER_AAC_ELD:
+  case AOT_DRM_AAC:
     break;
 
   default:
@@ -851,18 +866,19 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
         /* valid number of channels -> copy program config element (PCE) from ASC */
         FDKmemcpy(&self->pce, &asc->m_progrConfigElement, sizeof(CProgramConfig));
         /* Built element table */
-        el = CProgramConfig_GetElementTable(&asc->m_progrConfigElement, self->elements, 7);
-        for (; el<7; el++) {
+        el = CProgramConfig_GetElementTable(&asc->m_progrConfigElement, self->elements, (8), &self->chMapIndex);
+        for (; el<(8); el++) {
           self->elements[el] = ID_NONE;
         }
       } else {
         return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
       }
     } else {
+      self->chMapIndex = 0;
       if (transportDec_GetFormat(self->hInput) == TT_MP4_ADTS) {
         /* set default max_channels for memory allocation because in implicit channel mapping mode
            we don't know the actual number of channels until we processed at least one raw_data_block(). */
-        ascChannels = (6);
+        ascChannels = (8);
       } else {
         return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
       }
@@ -874,26 +890,34 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
   case 1: case 2: case 3: case 4: case 5: case 6:
     ascChannels = asc->m_channelConfiguration;
     break;
-  case 7:
+  case 11:
+    ascChannels = 7;
+    break;
+  case 7: case 12: case 14:
     ascChannels = 8;
     break;
   default:
     return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
   }
 
+  if (ascChannels > (8)) {
+    return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
+  }
+
   /* Initialize constant mappings for channel config 1-7 */
   if (asc->m_channelConfiguration > 0) {
     int el;
-    FDKmemcpy(self->elements, elementsTab[asc->m_channelConfiguration-1], sizeof(MP4_ELEMENT_ID)*FDKmin(7,7));
-    for (el=7; el<7; el++) {
+    FDKmemcpy(self->elements, elementsTab[asc->m_channelConfiguration-1], sizeof(MP4_ELEMENT_ID)*FDKmin(7,(8)));
+    for (el=7; el<(8); el++) {
       self->elements[el] = ID_NONE;
     }
     for (ch=0; ch<ascChannels; ch++) {
       self->chMapping[ch] = ch;
     }
-    for (; ch<(6); ch++) {
+    for (; ch<(8); ch++) {
       self->chMapping[ch] = 255;
     }
+    self->chMapIndex = asc->m_channelConfiguration;
   }
  #ifdef TP_PCE_ENABLE
   else {
@@ -909,9 +933,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
 
   self->streamInfo.channelConfig = asc->m_channelConfiguration;
 
-  if (ascChannels > (6)) {
-    return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
-  }
   if (self->streamInfo.aot != asc->m_aot) {
     self->streamInfo.aot = asc->m_aot;
     ascChanged = 1;
@@ -944,11 +965,20 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
 
   if (asc->m_aot == AOT_ER_AAC_ELD) {
     self->flags |=  AC_ELD;
+    self->flags |= (asc->m_sbrPresentFlag) ? AC_SBR_PRESENT : 0;  /* Need to set the SBR flag for backward-compatibility
+                                                                     reasons. Even if SBR is not supported. */
     self->flags |= (asc->m_sc.m_eldSpecificConfig.m_sbrCrcFlag) ? AC_SBRCRC : 0;
     self->flags |= (asc->m_sc.m_eldSpecificConfig.m_useLdQmfTimeAlign) ? AC_LD_MPS : 0;
   }
   self->flags |= (asc->m_aot == AOT_ER_AAC_LD) ? AC_LD : 0;
   self->flags |= (asc->m_epConfig >= 0) ? AC_ER : 0;
+  if ( asc->m_aot == AOT_DRM_AAC ) {
+    self->flags |= AC_DRM|AC_SBRCRC|AC_SCALABLE;
+  }
+  if ( (asc->m_aot == AOT_AAC_SCAL)
+    || (asc->m_aot == AOT_ER_AAC_SCAL) ) {
+    self->flags |= AC_SCALABLE;
+  }
 
 
   if (asc->m_sbrPresentFlag) {
@@ -1096,6 +1126,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
   MP4_ELEMENT_ID type = ID_NONE;            /* Current element type */
   INT aacChannels=0;                        /* Channel counter for channels found in the bitstream */
+  int chOutMapIdx;                          /* Output channel mapping index (see comment below) */
 
   INT auStartAnchor = (INT)FDKgetValidBits(bs);  /* AU start bit buffer position for AU byte alignment */
 
@@ -1119,12 +1150,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
     if (self->streamInfo.channelConfig == 0) {
       /* Init Channel/Element mapping table */
-      for (ch=0; ch<(6); ch++) {
+      for (ch=0; ch<(8); ch++) {
         self->chMapping[ch] = 255;
       }
       if (!CProgramConfig_IsValid(pce)) {
         int el;
-        for (el=0; el<7; el++) {
+        for (el=0; el<(8); el++) {
           self->elements[el] = ID_NONE;
         }
       }
@@ -1133,6 +1164,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
   /* Check sampling frequency  */
   switch ( self->streamInfo.aacSampleRate ) {
+    case 96000:
+    case 88200:
+    case 64000:
     case 16000:
     case 12000:
    case 11025:
@@ -1161,11 +1195,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       CConcealment_InitChannelData(&self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo,
                                    &self->concealCommonData,
                                     self->streamInfo.aacSamplesPerFrame );
-      /* Clear concealment buffers to get rid of the complete history */
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo.spectralCoefficient, 1024 * sizeof(FIXP_CNCL));
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo.specScale, 8 * sizeof(SHORT));
       /* Clear overlap-add buffers to avoid clicks. */
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->IMdct.overlap.freq, OverlapBufferSize*sizeof(FIXP_DBL));
+      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize*sizeof(FIXP_DBL));
      }
   }
 
@@ -1378,10 +1409,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         {
           UCHAR element_instance_tag;
 
-          CDataStreamElement_Read( bs,
-                                  &self->ancData,
-                                   self->hDrcInfo,
-                                   self->hInput,
+          CDataStreamElement_Read( self,
+                                   bs,
                                   &element_instance_tag,
                                    auStartAnchor );
 
@@ -1401,29 +1430,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
             //self->frameOK = 0;
           }
         }
-
-        {
-          UCHAR *pDvbAncData = NULL;
-          AAC_DECODER_ERROR ancErr;
-          int ancIndex;
-          int dvbAncDataSize = 0;
-
-          /* Ask how many anc data elements are in buffer */
-          ancIndex = self->ancData.nrElements - 1;
-          /* Get the last one (if available) */
-          ancErr   = CAacDecoder_AncDataGet( &self->ancData,
-                                              ancIndex,
-                                             &pDvbAncData,
-                                             &dvbAncDataSize );
-
-          if (ancErr == AAC_DEC_OK) {
-            pcmDmx_ReadDvbAncData (
-                  self->hPcmUtils,
-                  pDvbAncData,
-                  dvbAncDataSize,
-                  0 /* not mpeg2 */ );
-          }
-        }
         break;
 
 #ifdef TP_PCE_ENABLE
@@ -1442,9 +1448,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           }
           else if ( result > 1 ) {
             /* Built element table */
-            int elIdx = CProgramConfig_GetElementTable(pce, self->elements, 7);
+            int elIdx = CProgramConfig_GetElementTable(pce, self->elements, (8), &self->chMapIndex);
             /* Reset the remaining tabs */
-            for ( ; elIdx<7; elIdx++) {
+            for ( ; elIdx<(8); elIdx++) {
               self->elements[elIdx] = ID_NONE;
             }
             /* Make new number of channel persistant */
@@ -1489,7 +1495,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           /* get the remaining bits of this frame */
           bitCnt = transportDec_GetAuBitsRemaining(self->hInput, 0);
 
-          if ( (bitCnt > 0) && (self->flags & AC_SBR_PRESENT) && (self->flags & (AC_USAC|AC_RSVD50|AC_ELD)) )
+          if ( (bitCnt > 0) && (self->flags & AC_SBR_PRESENT) && (self->flags & (AC_USAC|AC_RSVD50|AC_ELD|AC_DRM)) )
           {
             SBR_ERROR err = SBRDEC_OK;
             int  elIdx, numChElements = el_cnt[ID_SCE] + el_cnt[ID_CPE];
@@ -1510,14 +1516,30 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                 break;
               }
             }
-            if (err == SBRDEC_OK) {
+            switch (err) {
+            case SBRDEC_PARSE_ERROR:
+              /* Can not go on parsing because we do not
+                 know the length of the SBR extension data. */
+              FDKpushFor(bs, bitCnt);
+              bitCnt = 0;
+              break;
+            case SBRDEC_OK:
               self->sbrEnabled = 1;
-            } else {
+              break;
+            default:
               self->frameOK = 0;
+              break;
             }
           }
 
 
+          if (self->flags & AC_DRM)
+          {
+            if ((bitCnt = (INT)FDKgetValidBits(bs)) != 0) {
+              FDKpushBiDirectional(bs, bitCnt);
+            }
+          }
+
           if ( ! (self->flags & (AC_USAC|AC_RSVD50|AC_DRM)) )
           {
             while ( bitCnt > 7 ) {
@@ -1603,13 +1625,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     self->frameOK=0;
   }
 
-  /* store or restore the number of channels */
+  /* store or restore the number of channels and the corresponding info */
   if ( self->frameOK && !(flags &(AACDEC_CONCEAL|AACDEC_FLUSH)) ) {
-    self->concealChannels = aacChannels;  /* store */
+    self->aacChannelsPrev = aacChannels;  /* store */
+    FDKmemcpy(self->channelTypePrev, self->channelType, (8)*sizeof(AUDIO_CHANNEL_TYPE));  /* store */
+    FDKmemcpy(self->channelIndicesPrev, self->channelIndices, (8)*sizeof(UCHAR));         /* store */
     self->sbrEnabledPrev = self->sbrEnabled;
   } else {
     if (self->aacChannels > 0) {
-      aacChannels = self->concealChannels;  /* restore */
+      aacChannels = self->aacChannelsPrev;  /* restore */
+      FDKmemcpy(self->channelType, self->channelTypePrev, (8)*sizeof(AUDIO_CHANNEL_TYPE));  /* restore */
+      FDKmemcpy(self->channelIndices, self->channelIndicesPrev, (8)*sizeof(UCHAR));         /* restore */
       self->sbrEnabled = self->sbrEnabledPrev;
      }
   }
@@ -1632,12 +1658,31 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     return ErrorStatus;
   }
 
+  /* Setup the output channel mapping. The table below shows the four possibilities:
+   *   # | chCfg | PCE | cChCfg | chOutMapIdx
+   *  ---+-------+-----+--------+------------------
+   *   1 |  > 0  |  no |    -   | chCfg
+   *   2 |   0   | yes |  > 0   | cChCfg
+   *   3 |   0   | yes |    0   | aacChannels || 0
+   *   4 |   0   |  no |    -   | aacChannels || 0
+   *  ---+-------+-----+--------+------------------
+   *  Where chCfg is the channel configuration index from ASC and cChCfg is a corresponding chCfg
+   *  derived from a given PCE. The variable aacChannels represents the number of channel found
+   *  during bitstream decoding. Due to the structure of the mapping table it can only be used for
+   *  mapping if its value is smaller than 7. Otherwise we use the fallback (0) which is a simple
+   *  pass-through. The possibility #4 should appear only with MPEG-2 (ADTS) streams. This is
+   *  mode is called "implicit channel mapping".
+   */
+  chOutMapIdx = ((self->chMapIndex==0) && (aacChannels<7)) ? aacChannels : self->chMapIndex;
+
   /*
     Inverse transform
   */
   {
     int stride, offset, c;
 
+    /* Turn on/off DRC modules level normalization in digital domain depending on the limiter status. */
+    aacDecoder_drcSetParam( self->hDrcInfo, APPLY_NORMALIZATION, (self->limiterEnableCurr) ? 0 : 1 );
     /* Extract DRC control data and map it to channels (without bitstream delay) */
     aacDecoder_drcProlog (
             self->hDrcInfo,
@@ -1663,13 +1708,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       /* Setup offset and stride for time buffer traversal. */
       if (interleaved) {
         stride = aacChannels;
-        offset = self->channelOutputMapping[aacChannels-1][c];
+        offset = self->channelOutputMapping[chOutMapIdx][c];
       } else {
         stride = 1;
-        offset = self->channelOutputMapping[aacChannels-1][c] * self->streamInfo.aacSamplesPerFrame;
+        offset = self->channelOutputMapping[chOutMapIdx][c] * self->streamInfo.aacSamplesPerFrame;
       }
 
 
+      if ( flags&AACDEC_FLUSH ) {
+        /* Clear pAacDecoderChannelInfo->pSpectralCoefficient because with AACDEC_FLUSH set it contains undefined data. */
+        FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, sizeof(FIXP_DBL)*self->streamInfo.aacSamplesPerFrame);
+      }
+
       /*
         Conceal defective spectral data
       */
@@ -1688,12 +1738,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         /* Reset DRC control data for this channel */
         aacDecoder_drcInitChannelData ( &self->pAacDecoderStaticChannelInfo[c]->drcData );
       }
+      /* The DRC module demands to be called with the gain field holding the gain scale. */
+      self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING;
       /* DRC processing */
       aacDecoder_drcApply (
               self->hDrcInfo,
               self->hSbrDecoder,
               pAacDecoderChannelInfo,
              &self->pAacDecoderStaticChannelInfo[c]->drcData,
+              self->extGain,
               c,
               self->streamInfo.aacSamplesPerFrame,
               self->sbrEnabled
@@ -1711,6 +1764,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                   (self->frameOK && !(flags&AACDEC_CONCEAL)),
                   self->aacCommonData.workBufferCore1->mdctOutTemp
                   );
+          self->extGainDelay = self->streamInfo.aacSamplesPerFrame;
           break;
         case AACDEC_RENDER_ELDFB:
           CBlock_FrequencyToTimeLowDelay(
@@ -1720,6 +1774,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                   self->streamInfo.aacSamplesPerFrame,
                   stride
                   );
+          self->extGainDelay = (self->streamInfo.aacSamplesPerFrame*2 -  self->streamInfo.aacSamplesPerFrame/2 - 1)/2;
           break;
         default:
           ErrorStatus = AAC_DEC_UNKNOWN;
@@ -1743,11 +1798,20 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           );
   }
 
+  /* Add additional concealment delay */
+  self->streamInfo.outputDelay += CConcealment_GetDelay(&self->concealCommonData) * self->streamInfo.aacSamplesPerFrame;
+
+  /* Map DRC data to StreamInfo structure */
+  aacDecoder_drcGetInfo (
+            self->hDrcInfo,
+           &self->streamInfo.drcPresMode,
+           &self->streamInfo.drcProgRefLev
+          );
 
   /* Reorder channel type information tables.  */
   {
-    AUDIO_CHANNEL_TYPE types[(6)];
-    UCHAR idx[(6)];
+    AUDIO_CHANNEL_TYPE types[(8)];
+    UCHAR idx[(8)];
     int c;
 
     FDK_ASSERT(sizeof(self->channelType) == sizeof(types));
@@ -1757,8 +1821,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     FDKmemcpy(idx, self->channelIndices, sizeof(idx));
 
     for (c=0; c<aacChannels; c++) {
-      self->channelType[self->channelOutputMapping[aacChannels-1][c]] = types[c];
-      self->channelIndices[self->channelOutputMapping[aacChannels-1][c]] = idx[c];
+      self->channelType[self->channelOutputMapping[chOutMapIdx][c]] = types[c];
+      self->channelIndices[self->channelOutputMapping[chOutMapIdx][c]] = idx[c];
     }
   }
 

libAACdec/src/aacdecoder.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -111,6 +111,7 @@ amm-info@iis.fraunhofer.de
 #include "aacdec_drc.h"
 
  #include "pcmutils_lib.h"
+ #include "limiter.h"
 
 
 /* Capabilities flags */
@@ -176,27 +177,31 @@ struct AAC_DECODER_INSTANCE {
 
   UINT                  flags;                       /*!< Flags for internal decoder use. DO NOT USE self::streaminfo::flags ! */
 
-  MP4_ELEMENT_ID        elements[7]; /*!< Table where the element Id's are listed          */
-  UCHAR                 elTags[7];   /*!< Table where the elements id Tags are listed      */
-  UCHAR                 chMapping[(6)];   /*!< Table of MPEG canonical order to bitstream channel order mapping. */
+  MP4_ELEMENT_ID        elements[(8)]; /*!< Table where the element Id's are listed          */
+  UCHAR                 elTags[(8)];   /*!< Table where the elements id Tags are listed      */
+  UCHAR                 chMapping[(8)];   /*!< Table of MPEG canonical order to bitstream channel order mapping. */
 
-  AUDIO_CHANNEL_TYPE    channelType[(6)];    /*!< Audio channel type of each output audio channel (from 0 upto numChannels).           */
-  UCHAR                 channelIndices[(6)]; /*!< Audio channel index for each output audio channel (from 0 upto numChannels).         */
+  AUDIO_CHANNEL_TYPE    channelType[(8)];    /*!< Audio channel type of each output audio channel (from 0 upto numChannels).           */
+  UCHAR                 channelIndices[(8)]; /*!< Audio channel index for each output audio channel (from 0 upto numChannels).         */
                                                              /* See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
 
 
   const UCHAR         (*channelOutputMapping)[8];    /*!< Table for MPEG canonical order to output channel order mapping. */
-
+  UCHAR                 chMapIndex;                  /*!< Index to access one line of the channelOutputMapping table. This is required
+                                                          because not all 8 channel configurations have the same output mapping. */
 
   CProgramConfig                pce;
   CStreamInfo                   streamInfo;         /*!< pointer to StreamInfo data (read from the bitstream) */
-  CAacDecoderChannelInfo       *pAacDecoderChannelInfo[(6)];       /*!< Temporal channel memory */
-  CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(6)]; /*!< Persistent channel memory */
+  CAacDecoderChannelInfo       *pAacDecoderChannelInfo[(8)];       /*!< Temporal channel memory */
+  CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */
 
   CAacDecoderCommonData         aacCommonData;             /*!< Temporal shared data for all channels hooked into pAacDecoderChannelInfo */
 
   CConcealParams                concealCommonData;
-  INT   concealChannels;
+
+  INT                   aacChannelsPrev;                          /*!< The amount of AAC core channels of the last successful decode call.         */
+  AUDIO_CHANNEL_TYPE    channelTypePrev[(8)];     /*!< Array holding the channelType values of the last successful decode call.    */
+  UCHAR                 channelIndicesPrev[(8)];  /*!< Array holding the channelIndices values of the last successful decode call. */
 
 
   HANDLE_SBRDECODER   hSbrDecoder;                   /*!< SBR decoder handle.                        */
@@ -214,6 +219,14 @@ struct AAC_DECODER_INSTANCE {
   CAncData      ancData;                             /*!< structure to handle ancillary data         */
 
   HANDLE_PCM_DOWNMIX  hPcmUtils;                     /*!< privat data for the PCM utils.             */
+  TDLimiterPtr hLimiter;                             /*!< Handle of time domain limiter.             */
+  UCHAR        limiterEnableUser;                    /*!< The limiter configuration requested by the library user */
+  UCHAR        limiterEnableCurr;                    /*!< The current limiter configuration.         */
+
+  FIXP_DBL     extGain[1];                           /*!< Gain that must be applied to the output signal. */
+  UINT         extGainDelay;                         /*!< Delay that must be accounted for extGain. */
+
+  INT_PCM      pcmOutputBuffer[(8)*(2048)];
 
 };
 

libAACdec/src/aacdecoder_lib.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -110,10 +110,15 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 2
 #define AACDECODER_LIB_VL1 5
-#define AACDECODER_LIB_VL2 5
+#define AACDECODER_LIB_VL2 17
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
+#ifdef __ANDROID__
+#define AACDECODER_LIB_BUILD_DATE ""
+#define AACDECODER_LIB_BUILD_TIME ""
+#else
 #define AACDECODER_LIB_BUILD_DATE __DATE__
 #define AACDECODER_LIB_BUILD_TIME __TIME__
+#endif
 
 static AAC_DECODER_ERROR
 setConcealMethod ( const HANDLE_AACDECODER  self,
@@ -176,8 +181,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_ConfigRaw (
           break;
         }
         /* if baselayer is OK we continue decoding */
-        if(layer  >= 1){
+        if(layer >= 1){
           self->nrOfLayers = layer;
+          err = AAC_DEC_OK;
         }
         break;
       }
@@ -397,12 +403,14 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
   CConcealParams  *pConcealData = NULL;
   HANDLE_AAC_DRC hDrcInfo = NULL;
   HANDLE_PCM_DOWNMIX hPcmDmx = NULL;
+  TDLimiterPtr hPcmTdl = NULL;
 
   /* check decoder handle */
   if (self != NULL) {
     pConcealData = &self->concealCommonData;
     hDrcInfo = self->hDrcInfo;
     hPcmDmx = self->hPcmUtils;
+    hPcmTdl = self->hLimiter;
   } else {
     errorStatus = AAC_DEC_INVALID_HANDLE;
   }
@@ -420,8 +428,8 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
     self->outputInterleaved = value;
     break;
 
-  case AAC_PCM_OUTPUT_CHANNELS:
-    if (value < -1 || value > (6)) {
+  case AAC_PCM_MIN_OUTPUT_CHANNELS:
+    if (value < -1 || value > (8)) {
       return AAC_DEC_SET_PARAM_FAIL;
     }
     {
@@ -429,7 +437,30 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
 
       err = pcmDmx_SetParam (
               hPcmDmx,
-              NUMBER_OF_OUTPUT_CHANNELS,
+              MIN_NUMBER_OF_OUTPUT_CHANNELS,
+              value );
+
+      switch (err) {
+      case PCMDMX_OK:
+        break;
+      case PCMDMX_INVALID_HANDLE:
+        return AAC_DEC_INVALID_HANDLE;
+      default:
+        return AAC_DEC_SET_PARAM_FAIL;
+      }
+    }
+    break;
+
+  case AAC_PCM_MAX_OUTPUT_CHANNELS:
+    if (value < -1 || value > (8)) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    {
+      PCMDMX_ERROR err;
+
+      err = pcmDmx_SetParam (
+              hPcmDmx,
+              MAX_NUMBER_OF_OUTPUT_CHANNELS,
               value );
 
       switch (err) {
@@ -449,7 +480,7 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
 
       err = pcmDmx_SetParam (
               hPcmDmx,
-              DUAL_CHANNEL_DOWNMIX_MODE,
+              DMX_DUAL_CHANNEL_MODE,
               value );
 
       switch (err) {
@@ -463,6 +494,47 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
     }
     break;
 
+
+  case AAC_PCM_LIMITER_ENABLE:
+    if (value < -1 || value > 1) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    if (self == NULL) {
+      return AAC_DEC_INVALID_HANDLE;
+    }
+    self->limiterEnableUser = value;
+    break;
+
+  case AAC_PCM_LIMITER_ATTACK_TIME:
+    if (value <= 0) {  /* module function converts value to unsigned */
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    switch (setLimiterAttack(hPcmTdl, value)) {
+    case TDLIMIT_OK:
+      break;
+    case TDLIMIT_INVALID_HANDLE:
+      return AAC_DEC_INVALID_HANDLE;
+    case TDLIMIT_INVALID_PARAMETER:
+    default:
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    break;
+
+  case AAC_PCM_LIMITER_RELEAS_TIME:
+    if (value <= 0) {  /* module function converts value to unsigned */
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    switch (setLimiterRelease(hPcmTdl, value)) {
+    case TDLIMIT_OK:
+      break;
+    case TDLIMIT_INVALID_HANDLE:
+      return AAC_DEC_INVALID_HANDLE;
+    case TDLIMIT_INVALID_PARAMETER:
+    default:
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    break;
+
   case AAC_PCM_OUTPUT_CHANNEL_MAPPING:
     switch (value) {
       case 0:
@@ -609,6 +681,14 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT
     goto bail;
   }
 
+  aacDec->hLimiter = createLimiter(TDL_ATTACK_DEFAULT_MS, TDL_RELEASE_DEFAULT_MS, SAMPLE_MAX, (8), 96000);
+  if (NULL == aacDec->hLimiter) {
+    err = -1;
+    goto bail;
+  }
+  aacDec->limiterEnableUser = (UCHAR)-1;
+  aacDec->limiterEnableCurr = 0;
+
 
 
   /* Assure that all modules have same delay */
@@ -706,8 +786,8 @@ static INT aacDecoder_EstimateNumberOfLostFrames(HANDLE_AACDECODER self)
 
 LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
         HANDLE_AACDECODER  self,
-        INT_PCM           *pTimeData,
-        const INT          timeDataSize,
+        INT_PCM           *pTimeData_extern,
+        const INT          timeDataSize_extern,
         const UINT         flags)
 {
     AAC_DECODER_ERROR ErrorStatus;
@@ -717,12 +797,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
     HANDLE_FDK_BITSTREAM hBs;
     int fTpInterruption = 0;  /* Transport originated interruption detection. */
     int fTpConceal = 0;       /* Transport originated concealment. */
+    INT_PCM *pTimeData = NULL;
+    INT timeDataSize = 0;
 
 
     if (self == NULL) {
       return AAC_DEC_INVALID_HANDLE;
     }
 
+    pTimeData = self->pcmOutputBuffer;
+    timeDataSize = sizeof(self->pcmOutputBuffer)/sizeof(*self->pcmOutputBuffer);
+
     if (flags & AACDEC_INTR) {
       self->streamInfo.numLostAccessUnits = 0;
     }
@@ -768,6 +853,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
     /* Signal bit stream interruption to other modules if required. */
     if ( fTpInterruption || (flags & (AACDEC_INTR|AACDEC_CLRHIST)) )
     {
+      sbrDecoder_SetParam(self->hSbrDecoder, SBR_CLEAR_HISTORY, (flags&AACDEC_CLRHIST));
       aacDecoder_SignalInterruption(self);
       if ( ! (flags & AACDEC_INTR) ) {
         ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR;
@@ -783,6 +869,19 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
       self->streamInfo.numBadBytes = 0;
       self->streamInfo.numTotalBytes = 0;
     }
+    /* Reset the output delay field. The modules will add their figures one after another. */
+    self->streamInfo.outputDelay = 0;
+
+    if (self->limiterEnableUser==(UCHAR)-1) {
+      /* Enbale limiter for all non-lowdelay AOT's. */
+      self->limiterEnableCurr = ( self->flags & (AC_LD|AC_ELD) ) ? 0 : 1;
+    }
+    else {
+      /* Use limiter configuration as requested. */
+      self->limiterEnableCurr = self->limiterEnableUser;
+    }
+    /* reset limiter gain on a per frame basis */
+    self->extGain[0] = FL2FXCONST_DBL(1.0f/(float)(1<<TDL_GAIN_SCALING));
 
 
     ErrorStatus = CAacDecoder_DecodeFrame(self,
@@ -825,11 +924,16 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
     if (self->sbrEnabled)
     {
       SBR_ERROR sbrError = SBRDEC_OK;
+      int chIdx, numCoreChannel = self->streamInfo.numChannels;
+      int chOutMapIdx = ((self->chMapIndex==0) && (numCoreChannel<7)) ? numCoreChannel : self->chMapIndex;
 
       /* set params */
       sbrDecoder_SetParam ( self->hSbrDecoder,
                             SBR_SYSTEM_BITSTREAM_DELAY,
                             self->sbrParams.bsDelay);
+      sbrDecoder_SetParam ( self->hSbrDecoder,
+                            SBR_FLUSH_DATA,
+                            (flags & AACDEC_FLUSH) );
 
       if ( self->streamInfo.aot == AOT_ER_AAC_ELD ) {
         /* Configure QMF */
@@ -838,7 +942,16 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
                               (self->flags & AC_LD_MPS) ? 1 : 0 );
       }
 
+      {
+        PCMDMX_ERROR dmxErr;
+        INT  maxOutCh = 0;
 
+        dmxErr = pcmDmx_GetParam(self->hPcmUtils, MAX_NUMBER_OF_OUTPUT_CHANNELS, &maxOutCh);
+        if ( (dmxErr == PCMDMX_OK) && (maxOutCh == 1) ) {
+          /* Disable PS processing if we have to create a mono output signal. */
+          self->psPossible = 0;
+        }
+      }
 
 
       /* apply SBR processing */
@@ -846,43 +959,50 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
                                     pTimeData,
                                    &self->streamInfo.numChannels,
                                    &self->streamInfo.sampleRate,
-                                    self->channelOutputMapping[self->streamInfo.numChannels-1],
+                                    self->channelOutputMapping[chOutMapIdx],
                                     interleaved,
                                     self->frameOK,
                                    &self->psPossible);
 
 
      if (sbrError == SBRDEC_OK) {
+       #define UPS_SCALE  2  /* Maximum upsampling factor is 4 (CELP+SBR) */
+       FIXP_DBL  upsampleFactor = FL2FXCONST_DBL(1.0f/(1<<UPS_SCALE));
 
        /* Update data in streaminfo structure. Assume that the SBR upsampling factor is either 1 or 2 */
        self->flags |= AC_SBR_PRESENT;
        if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) {
          if (self->streamInfo.frameSize == 768) {
-           self->streamInfo.frameSize =  (self->streamInfo.aacSamplesPerFrame * 8) / 3;
+           upsampleFactor = FL2FXCONST_DBL(8.0f/(3<<UPS_SCALE));
          } else {
-           self->streamInfo.frameSize =  self->streamInfo.aacSamplesPerFrame << 1;
+           upsampleFactor = FL2FXCONST_DBL(2.0f/(1<<UPS_SCALE));
          }
        }
+       /* Apply upsampling factor to both the core frame length and the core delay */
+       self->streamInfo.frameSize    =       (INT)fMult((FIXP_DBL)self->streamInfo.aacSamplesPerFrame<<UPS_SCALE, upsampleFactor);
+       self->streamInfo.outputDelay  = (UINT)(INT)fMult((FIXP_DBL)self->streamInfo.outputDelay<<UPS_SCALE, upsampleFactor);
+       self->streamInfo.outputDelay += sbrDecoder_GetDelay( self->hSbrDecoder );
 
        if (self->psPossible) {
          self->flags |= AC_PS_PRESENT;
-         self->channelType[0] = ACT_FRONT;
-         self->channelType[1] = ACT_FRONT;
-         self->channelIndices[0] = 0;
-         self->channelIndices[1] = 1;
-       } else {
-         self->flags &= ~AC_PS_PRESENT;
+       }
+       for (chIdx = numCoreChannel; chIdx < self->streamInfo.numChannels; chIdx+=1) {
+         self->channelType[chIdx] = ACT_FRONT;
+         self->channelIndices[chIdx] = chIdx;
        }
      }
    }
 
 
+    {
+    INT pcmLimiterScale = 0;
+    PCMDMX_ERROR dmxErr = PCMDMX_OK;
     if ( flags & (AACDEC_INTR | AACDEC_CLRHIST) ) {
       /* delete data from the past (e.g. mixdown coeficients) */
       pcmDmx_Reset( self->hPcmUtils, PCMDMX_RESET_BS_DATA );
     }
     /* do PCM post processing */
-    pcmDmx_ApplyFrame (
+    dmxErr = pcmDmx_ApplyFrame (
             self->hPcmUtils,
             pTimeData,
             self->streamInfo.frameSize,
@@ -890,9 +1010,40 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
             interleaved,
             self->channelType,
             self->channelIndices,
-            self->channelOutputMapping
+            self->channelOutputMapping,
+            (self->limiterEnableCurr) ? &pcmLimiterScale : NULL
       );
+    if ( (ErrorStatus == AAC_DEC_OK)
+      && (dmxErr == PCMDMX_INVALID_MODE) ) {
+      /* Announce the framework that the current combination of channel configuration and downmix
+       * settings are not know to produce a predictable behavior and thus maybe produce strange output. */
+      ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
+    }
 
+    if ( flags & AACDEC_CLRHIST ) {
+      /* Delete the delayed signal. */
+      resetLimiter(self->hLimiter);
+    }
+    if (self->limiterEnableCurr)
+    {
+      /* Set actual signal parameters */
+      setLimiterNChannels(self->hLimiter, self->streamInfo.numChannels);
+      setLimiterSampleRate(self->hLimiter, self->streamInfo.sampleRate);
+
+      applyLimiter(
+              self->hLimiter,
+              pTimeData,
+              self->extGain,
+             &pcmLimiterScale,
+              1,
+              self->extGainDelay,
+              self->streamInfo.frameSize
+              );
+
+      /* Announce the additional limiter output delay */
+      self->streamInfo.outputDelay += getLimiterDelay(self->hLimiter);
+    }
+    }
 
 
     /* Signal interruption to take effect in next frame. */
@@ -908,6 +1059,19 @@ bail:
     /* Update Statistics */
     aacDecoder_UpdateBitStreamCounters(&self->streamInfo, hBs, nBits, ErrorStatus);
 
+    /* Check whether external output buffer is large enough. */
+    if (timeDataSize_extern < self->streamInfo.numChannels*self->streamInfo.frameSize) {
+      ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
+    }
+
+    /* Update external output buffer. */
+    if ( IS_OUTPUT_VALID(ErrorStatus) ) {
+      FDKmemcpy(pTimeData_extern, pTimeData, self->streamInfo.numChannels*self->streamInfo.frameSize*sizeof(*pTimeData));
+    }
+    else {
+      FDKmemclear(pTimeData_extern, timeDataSize_extern*sizeof(*pTimeData_extern));
+    }
+
     return ErrorStatus;
 }
 
@@ -917,6 +1081,9 @@ LINKSPEC_CPP void aacDecoder_Close ( HANDLE_AACDECODER self )
     return;
 
 
+  if (self->hLimiter != NULL) {
+    destroyLimiter(self->hLimiter);
+  }
 
   if (self->hPcmUtils != NULL) {
     pcmDmx_Close( &self->hPcmUtils );
@@ -974,6 +1141,7 @@ LINKSPEC_CPP INT aacDecoder_GetLibInfo ( LIB_INFO *info )
   /* Set flags */
   info->flags = 0
       | CAPF_AAC_LC
+      | CAPF_ER_AAC_SCAL
       | CAPF_AAC_VCB11
       | CAPF_AAC_HCR
       | CAPF_AAC_RVLC
@@ -984,6 +1152,7 @@ LINKSPEC_CPP INT aacDecoder_GetLibInfo ( LIB_INFO *info )
 
       | CAPF_AAC_MPEG4
 
+      | CAPF_AAC_DRM_BSFORMAT
 
       | CAPF_AAC_1024
       | CAPF_AAC_960

libAACdec/src/block.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -324,13 +324,12 @@ AAC_DECODER_ERROR CBlock_ReadSectionData(HANDLE_FDK_BITSTREAM bs,
 
       if (flags & AC_ER_HCR) {
         /* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */
-        pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band];
-        numLinesInSecIdx++;
         if (numLinesInSecIdx >= MAX_SFB_HCR) {
           return AAC_DEC_PARSE_ERROR;
         }
-        if (
-             (sect_cb == BOOKSCL) )
+        pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band];
+        numLinesInSecIdx++;
+        if (sect_cb == BOOKSCL)
         {
           return AAC_DEC_INVALID_CODE_BOOK;
         } else {

libAACdec/src/conceal.cpp

@@ -762,7 +762,6 @@ int
   CConcealment_UpdateState( hConcealmentInfo,
                             frameOk );
 
-  if ( !frameOk )
   {
     /* Create data for signal rendering according to the selected concealment method and decoder operating mode. */
 
@@ -775,11 +774,13 @@ int
       {
       default:
       case ConcealMethodMute:
-        /* Mute spectral data in case of errors */
-        FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL));
-        /* Set last window shape */
-        pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape;
-        appliedProcessing = 1;
+        if (!frameOk) {
+          /* Mute spectral data in case of errors */
+          FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL));
+          /* Set last window shape */
+          pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape;
+          appliedProcessing = 1;
+        }
         break;
 
       case ConcealMethodNoise:
@@ -801,7 +802,7 @@ int
                                    pSamplingRateInfo,
                                    samplesPerFrame,
                                    0,  /* don't use tonal improvement */
-                                   0);
+                                   frameOk);
         break;
 
       }

libAACdec/src/rvlcconceal.cpp

@@ -459,7 +459,7 @@ void BidirectionalEstimation_UseScfOfPrevFrameAsReference (
           break;
 
         case NOISE_HCB:
-          if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB) ) {
+          if ( pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB ) {
             commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]);
             pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]);
           } else {
@@ -669,7 +669,7 @@ void PredictiveInterpolation (
           break;
 
         case NOISE_HCB:
-          if ( (pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB) ) {
+          if ( pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousCodebook[bnds]==NOISE_HCB ) {
             commonMin = FDKmin(pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfFwd[bnds],pAacDecoderChannelInfo->pComData->overlay.aac.aRvlcScfBwd[bnds]);
             pAacDecoderChannelInfo->pDynData->aScaleFactor[bnds] = FDKmin(commonMin, pAacDecoderStaticChannelInfo->concealmentInfo.aRvlcPreviousScaleFactor[bnds]);
           }

libAACenc/include/aacenc_lib.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -707,7 +707,7 @@ the encoder deactivates PNS calculation internally.
 
 #define AACENCODER_LIB_VL0 3
 #define AACENCODER_LIB_VL1 4
-#define AACENCODER_LIB_VL2 12
+#define AACENCODER_LIB_VL2 22
 
 /**
  *  AAC encoder error codes.
@@ -900,11 +900,7 @@ typedef enum
                                                         This configuration can be used only with stereo input audio data.
                                                   - 23: MPEG-4 AAC Low-Delay.
                                                   - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in
-                                                        combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter.
-                                                  - 129: MPEG-2 AAC Low Complexity.
-                                                  - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC).
-                                                  - 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
-                                                         This configuration can be used only with stereo input audio data. */
+                                                        combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. */
 
   AACENC_BITRATE                  = 0x0101,  /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
                                                   - CBR: Bitrate in bits/second.
@@ -961,6 +957,16 @@ typedef enum
                                                   - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
                                                                touch this value to avoid degraded audio quality) */
 
+  AACENC_PEAK_BITRATE             = 0x0207,  /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second. 
+                                                  The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit
+                                                  and the number_of_effective_channels*6144 bit as upper limit.
+
+                                                  Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames
+                                                  will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by
+                                                  one byte in one or the other direction over various frames. However, it is not recommended to reduce
+                                                  the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the
+                                                  audio quality by a large amount. */
+
   AACENC_TRANSMUX                 = 0x0300,  /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
                                                   types can be configured in encoder library:
                                                   - 0: raw access units
@@ -1026,6 +1032,11 @@ typedef enum
                                                   - ADTS: Maximum number of sub frames restricted to 4.
                                                   - LOAS/LATM: Maximum number of sub frames restricted to 2.*/
 
+  AACENC_AUDIOMUXVER              = 0x0304,  /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented):
+                                                  - 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes.
+                                                  - 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness.
+                                                  - 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */
+
   AACENC_PROTECTION               = 0x0306,  /*!< Configure protection in tranpsort layer:
                                                   - 0: No protection. (default)
                                                   - 1: CRC active for ADTS bitstream format. */

libAACenc/src/aacEnc_rom.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -93,7 +93,7 @@ amm-info@iis.fraunhofer.de
 /*
    Huffman Tables
 */
-const INT FDKaacEnc_huff_ltab1_2[3][3][3][3]=
+const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]=
 {
   {
     { {0x000b0009,0x00090007,0x000b0009}, {0x000a0008,0x00070006,0x000a0008}, {0x000b0009,0x00090008,0x000b0009} },
@@ -113,7 +113,7 @@ const INT FDKaacEnc_huff_ltab1_2[3][3][3][3]=
 };
 
 
-const INT FDKaacEnc_huff_ltab3_4[3][3][3][3]=
+const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]=
 {
   {
     { {0x00010004,0x00040005,0x00080008}, {0x00040005,0x00050004,0x00080008}, {0x00090009,0x00090008,0x000a000b} },
@@ -132,7 +132,7 @@ const INT FDKaacEnc_huff_ltab3_4[3][3][3][3]=
   }
 };
 
-const INT FDKaacEnc_huff_ltab5_6[9][9]=
+const ULONG FDKaacEnc_huff_ltab5_6[9][9]=
 {
   {0x000d000b, 0x000c000a, 0x000b0009, 0x000b0009, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c000a, 0x000d000b},
   {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, 0x000a0008, 0x000b0009, 0x000c000a},
@@ -145,7 +145,7 @@ const INT FDKaacEnc_huff_ltab5_6[9][9]=
   {0x000d000b, 0x000c000a, 0x000c0009, 0x000b0009, 0x000a0009, 0x000a0009, 0x000b0009, 0x000c000a, 0x000d000b}
 };
 
-const INT FDKaacEnc_huff_ltab7_8[8][8]=
+const ULONG FDKaacEnc_huff_ltab7_8[8][8]=
 {
   {0x00010005, 0x00030004, 0x00060005, 0x00070006, 0x00080007, 0x00090008, 0x000a0009, 0x000b000a},
   {0x00030004, 0x00040003, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x00090008},
@@ -157,7 +157,7 @@ const INT FDKaacEnc_huff_ltab7_8[8][8]=
   {0x000b000a, 0x000a0008, 0x000a0008, 0x000a0008, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c000a}
 };
 
-const INT FDKaacEnc_huff_ltab9_10[13][13]=
+const ULONG FDKaacEnc_huff_ltab9_10[13][13]=
 {
   {0x00010006, 0x00030005, 0x00060006, 0x00080006, 0x00090007, 0x000a0008, 0x000a0009, 0x000b000a, 0x000b000a, 0x000c000a, 0x000c000b, 0x000d000b, 0x000d000c},
   {0x00030005, 0x00040004, 0x00060004, 0x00070005, 0x00080006, 0x00080007, 0x00090007, 0x000a0008, 0x000a0008, 0x000a0009, 0x000b000a, 0x000c000a, 0x000c000b},
@@ -392,7 +392,7 @@ const USHORT FDKaacEnc_huff_ctab11[21][17]=
   {0x0046, 0x00ea, 0x0034, 0x00ea, 0x0011, 0x001b, 0x00a9, 0x0094, 0x00e2, 0x0031, 0x00d0, 0x00e5, 0x0007, 0x0070, 0x0069, 0x003e, 0x0021}
 };
 
-const INT FDKaacEnc_huff_ctabscf[121]=
+const ULONG FDKaacEnc_huff_ctabscf[121]=
 {
   0x0003ffe8, 0x0003ffe6, 0x0003ffe7, 0x0003ffe5, 0x0007fff5, 0x0007fff1, 0x0007ffed, 0x0007fff6,
   0x0007ffee, 0x0007ffef, 0x0007fff0, 0x0007fffc, 0x0007fffd, 0x0007ffff, 0x0007fffe, 0x0007fff7,
@@ -624,7 +624,7 @@ const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024 = {
 };
 const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128 = {
     12,
-    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 18, 30, 32 }
+    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 16, 28, 36 }
 };
 const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024 = {
     41,
@@ -634,7 +634,7 @@ const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024 = {
 };
 const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128 = {
     12,
-    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 18, 30, 32 }
+    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 16, 28, 36 }
 };
 const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024 = {
     41,
@@ -644,7 +644,7 @@ const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024 = {
 };
 const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = {
     12,
-    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 18, 30, 32 }
+    {  4,  4,  4,  4,  4,  4,  8,  8,  8, 16, 28, 36 }
 };
 
 
@@ -657,11 +657,11 @@ const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = {
 */
 const FIXP_DBL FDKaacEnc_tnsEncCoeff3[8]=
 {
-  0x81f1d201, 0x91261481, 0xadb92301, 0xd438af00, 0x00000000, 0x37898080, 0x64130dff, 0x7cca6fff
+  (FIXP_DBL)0x81f1d201, (FIXP_DBL)0x91261481, (FIXP_DBL)0xadb92301, (FIXP_DBL)0xd438af00, (FIXP_DBL)0x00000000, (FIXP_DBL)0x37898080, (FIXP_DBL)0x64130dff, (FIXP_DBL)0x7cca6fff
 };
 const FIXP_DBL FDKaacEnc_tnsCoeff3Borders[8]={
-  0x80000001 /*-4*/,  0x87b826df /*-3*/,  0x9df24154 /*-2*/,  0xbfffffe5 /*-1*/,
-  0xe9c5e578 /* 0*/,  0x1c7b90f0 /* 1*/,  0x4fce83a9 /* 2*/,  0x7352f2c3 /* 3*/
+  (FIXP_DBL)0x80000001 /*-4*/,  (FIXP_DBL)0x87b826df /*-3*/,  (FIXP_DBL)0x9df24154 /*-2*/,  (FIXP_DBL)0xbfffffe5 /*-1*/,
+  (FIXP_DBL)0xe9c5e578 /* 0*/,  (FIXP_DBL)0x1c7b90f0 /* 1*/,  (FIXP_DBL)0x4fce83a9 /* 2*/,  (FIXP_DBL)0x7352f2c3 /* 3*/
 };
 
 /*
@@ -669,15 +669,15 @@ const FIXP_DBL FDKaacEnc_tnsCoeff3Borders[8]={
 */
 const FIXP_DBL FDKaacEnc_tnsEncCoeff4[16]=
 {
-  0x808bc881, 0x84e2e581, 0x8d6b4a01, 0x99da9201, 0xa9c45701, 0xbc9dde81, 0xd1c2d500, 0xe87ae540,
-  0x00000000, 0x1a9cd9c0, 0x340ff240, 0x4b3c8bff, 0x5f1f5e7f, 0x6ed9eb7f, 0x79bc387f, 0x7f4c7e7f
+  (FIXP_DBL)0x808bc881, (FIXP_DBL)0x84e2e581, (FIXP_DBL)0x8d6b4a01, (FIXP_DBL)0x99da9201, (FIXP_DBL)0xa9c45701, (FIXP_DBL)0xbc9dde81, (FIXP_DBL)0xd1c2d500, (FIXP_DBL)0xe87ae540,
+  (FIXP_DBL)0x00000000, (FIXP_DBL)0x1a9cd9c0, (FIXP_DBL)0x340ff240, (FIXP_DBL)0x4b3c8bff, (FIXP_DBL)0x5f1f5e7f, (FIXP_DBL)0x6ed9eb7f, (FIXP_DBL)0x79bc387f, (FIXP_DBL)0x7f4c7e7f
 };
 const FIXP_DBL FDKaacEnc_tnsCoeff4Borders[16]=
 {
-  0x80000001 /*-8*/,  0x822deff0 /*-7*/,  0x88a4bfe6 /*-6*/,  0x932c159d /*-5*/,
-  0xa16827c2 /*-4*/,  0xb2dcde27 /*-3*/,  0xc6f20b91 /*-2*/,  0xdcf89c64 /*-1*/,
-  0xf4308ce1 /* 0*/,  0x0d613054 /* 1*/,  0x278dde80 /* 2*/,  0x4000001b /* 3*/,
-  0x55a6127b /* 4*/,  0x678dde8f /* 5*/,  0x74ef0ed7 /* 6*/,  0x7d33f0da /* 7*/
+  (FIXP_DBL)0x80000001 /*-8*/,  (FIXP_DBL)0x822deff0 /*-7*/,  (FIXP_DBL)0x88a4bfe6 /*-6*/,  (FIXP_DBL)0x932c159d /*-5*/,
+  (FIXP_DBL)0xa16827c2 /*-4*/,  (FIXP_DBL)0xb2dcde27 /*-3*/,  (FIXP_DBL)0xc6f20b91 /*-2*/,  (FIXP_DBL)0xdcf89c64 /*-1*/,
+  (FIXP_DBL)0xf4308ce1 /* 0*/,  (FIXP_DBL)0x0d613054 /* 1*/,  (FIXP_DBL)0x278dde80 /* 2*/,  (FIXP_DBL)0x4000001b /* 3*/,
+  (FIXP_DBL)0x55a6127b /* 4*/,  (FIXP_DBL)0x678dde8f /* 5*/,  (FIXP_DBL)0x74ef0ed7 /* 6*/,  (FIXP_DBL)0x7d33f0da /* 7*/
 };
 const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512]={
   FL2FXCONST_DBL(0.3968502629920499),FL2FXCONST_DBL(0.3978840634868335),FL2FXCONST_DBL(0.3989185359354711),FL2FXCONST_DBL(0.3999536794661432),

libAACenc/src/aacEnc_rom.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -105,11 +105,11 @@ amm-info@iis.fraunhofer.de
 /*
   Huffman Tables
 */
-extern const INT    FDKaacEnc_huff_ltab1_2[3][3][3][3];
-extern const INT    FDKaacEnc_huff_ltab3_4[3][3][3][3];
-extern const INT    FDKaacEnc_huff_ltab5_6[9][9];
-extern const INT    FDKaacEnc_huff_ltab7_8[8][8];
-extern const INT    FDKaacEnc_huff_ltab9_10[13][13];
+extern const ULONG  FDKaacEnc_huff_ltab1_2[3][3][3][3];
+extern const ULONG  FDKaacEnc_huff_ltab3_4[3][3][3][3];
+extern const ULONG  FDKaacEnc_huff_ltab5_6[9][9];
+extern const ULONG  FDKaacEnc_huff_ltab7_8[8][8];
+extern const ULONG  FDKaacEnc_huff_ltab9_10[13][13];
 extern const UCHAR  FDKaacEnc_huff_ltab11[17][17];
 extern const UCHAR  FDKaacEnc_huff_ltabscf[121];
 extern const USHORT FDKaacEnc_huff_ctab1[3][3][3][3];
@@ -123,7 +123,7 @@ extern const USHORT FDKaacEnc_huff_ctab8[8][8];
 extern const USHORT FDKaacEnc_huff_ctab9[13][13];
 extern const USHORT FDKaacEnc_huff_ctab10[13][13];
 extern const USHORT FDKaacEnc_huff_ctab11[21][17];
-extern const INT    FDKaacEnc_huff_ctabscf[121];
+extern const ULONG  FDKaacEnc_huff_ctabscf[121];
 
 /*
   quantizer

libAACenc/src/aacenc.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -107,6 +107,39 @@ amm-info@iis.fraunhofer.de
 
 #define MIN_BUFSIZE_PER_EFF_CHAN 6144
 
+INT FDKaacEnc_CalcBitsPerFrame(
+        const INT bitRate,
+        const INT frameLength,
+        const INT samplingRate
+        )
+{
+    int shift = 0;
+    while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
+        && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
+    {
+      shift++;
+    }
+
+    return (bitRate*(frameLength>>shift)) / (samplingRate>>shift);
+}
+
+INT FDKaacEnc_CalcBitrate(
+        const INT bitsPerFrame,
+        const INT frameLength,
+        const INT samplingRate
+        )
+{
+    int shift = 0;
+    while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
+        && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
+    {
+      shift++;
+    }
+
+    return (bitsPerFrame * (samplingRate>>shift)) / ( frameLength>>shift) ;
+
+}
+
 static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate,
                                                       INT framelength,
                                                       INT ancillaryRate,
@@ -220,21 +253,19 @@ INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode)
 /**
  * \brief  Convert encoder bitreservoir value for transport library.
  *
- * \param bitrateMode           Bitratemode used in current encoder instance. Se ::AACENC_BITRATE_MODE
- * \param bitresTotal           Encoder bitreservoir level in bits.
+ * \param hAacEnc               Encoder handle
  *
  * \return  Corrected bitreservoir level used in transport library.
  */
 static INT FDKaacEnc_EncBitresToTpBitres(
-        const AACENC_BITRATE_MODE bitrateMode,
-        const INT                 bitresTotal
+        const HANDLE_AAC_ENC      hAacEnc
         )
 {
   INT transporBitreservoir = 0;
 
-  switch (bitrateMode) {
+  switch (hAacEnc->bitrateMode) {
     case AACENC_BR_MODE_CBR:
-      transporBitreservoir = bitresTotal; /* encoder bitreservoir level */
+      transporBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */
       break;
     case AACENC_BR_MODE_VBR_1:
     case AACENC_BR_MODE_VBR_2:
@@ -253,6 +284,10 @@ static INT FDKaacEnc_EncBitresToTpBitres(
       FDK_ASSERT(0);
   }
 
+  if (hAacEnc->config->audioMuxVersion==2) {
+    transporBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff;
+  }
+
   return transporBitreservoir;
 }
 
@@ -289,6 +324,7 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config)
     config->minBitsPerFrame = -1;                   /* minum number of bits in each AU */
     config->maxBitsPerFrame = -1;                   /* minum number of bits in each AU */
     config->bitreservoir    = -1;                   /* default, uninitialized value */
+    config->audioMuxVersion = -1;                   /* audio mux version not configured */
 
     /* init tabs in fixpoint_math */
     InitLdInt();
@@ -435,7 +471,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
          &averageBitsPerFrame,
           config->bitrateMode,
           config->nSubFrames
-          ) != config->bitRate )
+          ) != config->bitRate
+      && !((config->bitrateMode>=1) && (config->bitrateMode<=5))
+     )
   {
     return AAC_ENC_UNSUPPORTED_BITRATE;
   }
@@ -562,7 +600,10 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
       qcInit.averageBits     = (averageBitsPerFrame+7)&~7;
       qcInit.bitRes          = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
       qcInit.maxBits         = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
-      qcInit.minBits         = 0;
+      qcInit.maxBits         = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
+      qcInit.maxBits         = fixMax(qcInit.maxBits, (averageBitsPerFrame+7)&~7);
+      qcInit.minBits         = (config->minBitsPerFrame!=-1) ? config->minBitsPerFrame : 0;
+      qcInit.minBits         = fixMin(qcInit.minBits, averageBitsPerFrame&~7);
   }
   else
   {
@@ -573,9 +614,11 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
 
       qcInit.maxBits         = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes);
       qcInit.maxBits         = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
+      qcInit.maxBits         = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, fixMax(qcInit.maxBits, (averageBitsPerFrame+7+8)&~7));
 
       qcInit.minBits         = fixMax(0, ((averageBitsPerFrame-1)&~7)-qcInit.bitRes-transportEnc_GetStaticBits(hTpEnc, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes));
       qcInit.minBits         = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits;
+      qcInit.minBits         = fixMin(qcInit.minBits, (averageBitsPerFrame - transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits))&~7);
   }
 
   qcInit.sampleRate          = config->sampleRate;
@@ -583,11 +626,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   qcInit.nSubFrames          = config->nSubFrames;
   qcInit.padding.paddingRest = config->sampleRate;
 
-  /* Calc meanPe */
-  bw_ratio = fDivNorm((FIXP_DBL)hAacEnc->bandwidth90dB, (FIXP_DBL)(config->sampleRate>>1), &qbw);
-  qbw = DFRACT_BITS-1-qbw;
-  /* qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
-  qcInit.meanPe = fMult(bw_ratio, (FIXP_DBL)((10*config->framelength)<<16)) >> (qbw-15);
+  /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
+  bw_ratio = fDivNorm((FIXP_DBL)(10*config->framelength*hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw);
+  qcInit.meanPe = FDKmax((INT)scaleValue(bw_ratio, qbw+1-(DFRACT_BITS-1)), 1);
 
   /* Calc maxBitFac */
   mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac);
@@ -649,23 +690,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   if (ErrorStatus != AAC_ENC_OK)
     goto bail;
 
-  /* Map virtual aot's to intern aot used in bitstream writer. */
-  switch (hAacEnc->config->audioObjectType) {
-    case AOT_MP2_AAC_LC:
-    case AOT_DABPLUS_AAC_LC:
-      hAacEnc->aot = AOT_AAC_LC;
-      break;
-    case AOT_MP2_SBR:
-    case AOT_DABPLUS_SBR:
-      hAacEnc->aot = AOT_SBR;
-      break;
-    case AOT_MP2_PS:
-    case AOT_DABPLUS_PS:
-      hAacEnc->aot = AOT_PS;
-      break;
-    default:
-      hAacEnc->aot = hAacEnc->config->audioObjectType;
-  }
+  hAacEnc->aot = hAacEnc->config->audioObjectType;
 
   /* common things */
 
@@ -930,7 +955,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC       hAacEnc,
               transportEnc_WriteAccessUnit(
                     hTpEnc,
                     totalBits,
-                    FDKaacEnc_EncBitresToTpBitres(hAacEnc->bitrateMode, hAacEnc->qcKernel->bitResTot),
+                    FDKaacEnc_EncBitresToTpBitres(hAacEnc),
                     cm->nChannelsEff);
 
               /* write bitstream */

libAACenc/src/aacenc.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -98,6 +98,11 @@ amm-info@iis.fraunhofer.de
 
 #include "sbr_encoder.h"
 
+#define BITRES_MAX_LD   4000
+#define BITRES_MIN_LD    500
+#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */
+#define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */
+
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -205,6 +210,8 @@ struct AACENC_CONFIG {
   INT   maxBitsPerFrame;        /* maximum number of bits in AU */
   INT   bitreservoir;           /* size of bitreservoir */
 
+  INT   audioMuxVersion;        /* audio mux version in loas/latm transport format */
+
   UINT  sbrRatio;               /* sbr sampling rate ratio: dual- or single-rate */
 
   UCHAR useTns;                 /* flag: use temporal noise shaping */
@@ -223,6 +230,36 @@ typedef struct {
 
 typedef struct AAC_ENC *HANDLE_AAC_ENC;
 
+/**
+ * \brief Calculate framesize in bits for given bit rate, frame length and sampling rate.
+ *
+ * \param bitRate               Ttarget bitrate in bits per second.
+ * \param frameLength           Number of audio samples in one frame.
+ * \param samplingRate          Sampling rate in Hz.
+ *
+ * \return                      Framesize in bits per frame.
+*/
+INT FDKaacEnc_CalcBitsPerFrame(
+        const INT bitRate,
+        const INT frameLength,
+        const INT samplingRate
+        );
+
+/**
+ * \brief Calculate bitrate in bits per second for given framesize, frame length and sampling rate.
+ *
+ * \param bitsPerFrame          Framesize in bits per frame.
+ * \param frameLength           Number of audio samples in one frame.
+ * \param samplingRate          Sampling rate in Hz.
+ *
+ * \return                      Bitrate in bits per second.
+*/
+INT FDKaacEnc_CalcBitrate(
+        const INT bitsPerFrame,
+        const INT frameLength,
+        const INT samplingRate
+        );
+
 /**
  * \brief Limit given bit rate to a valid value
  * \param hTpEnc transport encoder handle

libAACenc/src/aacenc_lib.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -98,10 +98,15 @@ amm-info@iis.fraunhofer.de
 /* Encoder library info */
 #define AACENCODER_LIB_VL0 3
 #define AACENCODER_LIB_VL1 4
-#define AACENCODER_LIB_VL2 12
+#define AACENCODER_LIB_VL2 22
 #define AACENCODER_LIB_TITLE "AAC Encoder"
+#ifdef __ANDROID__
+#define AACENCODER_LIB_BUILD_DATE ""
+#define AACENCODER_LIB_BUILD_TIME ""
+#else
 #define AACENCODER_LIB_BUILD_DATE __DATE__
 #define AACENCODER_LIB_BUILD_TIME __TIME__
+#endif
 
 
 #include "sbr_encoder.h"
@@ -148,6 +153,7 @@ typedef struct {
     UINT              userAfterburner;
     UINT              userFramelength;
     UINT              userAncDataRate;
+    UINT              userPeakBitrate;
 
     UCHAR             userTns;               /*!< Use TNS coding. */
     UCHAR             userPns;               /*!< Use PNS coding. */
@@ -298,7 +304,7 @@ static AACENC_ERROR eldSbrConfigurator(
     int i, cfgIdx = -1;
     const ULONG channelBitrate = totalBitrate / FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;
 
-    for (i=0; i<(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) {
+    for (i=0; i<(int)(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) {
       if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate)
         && (channelBitrate >= eldSbrAutoConfigTab[i].bitrateRange) )
       {
@@ -321,10 +327,7 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
 {
     INT sbrUsed = 0;
 
-    if ( (hAacConfig->audioObjectType==AOT_SBR)         || (hAacConfig->audioObjectType==AOT_PS)
-      || (hAacConfig->audioObjectType==AOT_MP2_SBR)     || (hAacConfig->audioObjectType==AOT_MP2_PS)
-      || (hAacConfig->audioObjectType==AOT_DABPLUS_SBR) || (hAacConfig->audioObjectType==AOT_DABPLUS_PS)
-      || (hAacConfig->audioObjectType==AOT_DRM_SBR)     || (hAacConfig->audioObjectType==AOT_DRM_MPEG_PS) )
+    if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) )
     {
         sbrUsed = 1;
     }
@@ -340,10 +343,7 @@ static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
 {
     INT psUsed = 0;
 
-    if ( (audioObjectType==AOT_PS)
-      || (audioObjectType==AOT_MP2_PS)
-      || (audioObjectType==AOT_DABPLUS_PS)
-      || (audioObjectType==AOT_DRM_MPEG_PS) )
+    if ( (audioObjectType==AOT_PS) )
     {
         psUsed = 1;
     }
@@ -368,8 +368,7 @@ static SBR_PS_SIGNALING getSbrSignalingMode(
     sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
   }
 
-  if ((audioObjectType==AOT_AAC_LC)     || (audioObjectType==AOT_SBR)     || (audioObjectType==AOT_PS)    ||
-      (audioObjectType==AOT_MP2_AAC_LC) || (audioObjectType==AOT_MP2_SBR) || (audioObjectType==AOT_MP2_PS) ) {
+  if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) {
     switch (transportType) {
       case TT_MP4_ADIF:
       case TT_MP4_ADTS:
@@ -425,22 +424,7 @@ static void FDKaacEnc_MapConfig(
 
   cc->flags = 0;
 
-  /* Map virtual aot to transport aot. */
-  switch (hAacConfig->audioObjectType) {
-    case AOT_MP2_AAC_LC:
-      transport_AOT = AOT_AAC_LC;
-      break;
-    case AOT_MP2_SBR:
-      transport_AOT = AOT_SBR;
-      cc->flags |= CC_SBR;
-     break;
-    case AOT_MP2_PS:
-      transport_AOT = AOT_PS;
-      cc->flags |= CC_SBR;
-      break;
-    default:
-      transport_AOT = hAacConfig->audioObjectType;
-  }
+  transport_AOT = hAacConfig->audioObjectType;
 
   if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) {
     cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0;
@@ -506,16 +490,7 @@ static void FDKaacEnc_MapConfig(
   cc->samplingRate    = hAacConfig->sampleRate;
 
   /* Mpeg-4 signaling for transport library. */
-  switch ( hAacConfig->audioObjectType ) {
-    case AOT_MP2_AAC_LC:
-    case AOT_MP2_SBR:
-    case AOT_MP2_PS:
-      cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */
-      cc->extAOT = AOT_NULL_OBJECT;
-      break;
-    default:
-      cc->flags |= CC_MPEG_ID;
-  }
+  cc->flags |= CC_MPEG_ID;
 
   /* ER-tools signaling. */
   cc->flags     |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0;
@@ -580,6 +555,7 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
     config->userChannelMode = hAacConfig->channelMode;
     config->userBitrate     = hAacConfig->bitRate;
     config->userBitrateMode = hAacConfig->bitrateMode;
+    config->userPeakBitrate = (UINT)-1;
     config->userBandwidth   = hAacConfig->bandWidth;
     config->userTns         = hAacConfig->useTns;
     config->userPns         = hAacConfig->usePns;
@@ -787,12 +763,15 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
     hAacConfig->syntaxFlags     = 0;
     hAacConfig->epConfig        = -1;
 
+    if (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) {
+      hAacConfig->audioMuxVersion = config->userTpAmxv;
+    }
+    else {
+      hAacConfig->audioMuxVersion = -1;
+    }
+
     /* Adapt internal AOT when necessary. */
     switch ( hAacConfig->audioObjectType ) {
-      case AOT_MP2_AAC_LC:
-      case AOT_MP2_SBR:
-      case AOT_MP2_PS:
-          hAacConfig->usePns = 0;
       case AOT_AAC_LC:
       case AOT_SBR:
       case AOT_PS:
@@ -834,11 +813,16 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
     switch ( hAacConfig->audioObjectType ) {
       case AOT_ER_AAC_LD:
       case AOT_ER_AAC_ELD:
-        if (config->userBitrateMode==8) {
-          hAacConfig->bitrateMode = 0;
-        }
         if (config->userBitrateMode==0) {
-          hAacConfig->bitreservoir = 100*config->nChannels; /* default, reduced bitreservoir */
+          /* bitreservoir  = (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes; */
+          if ( isLowDelay(hAacConfig->audioObjectType) ) {
+            INT bitreservoir;
+            INT brPerChannel = hAacConfig->bitRate/hAacConfig->nChannels;
+            brPerChannel     = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));
+            FIXP_DBL slope   = fDivNorm((brPerChannel-BITRATE_MIN_LD), BITRATE_MAX_LD-BITRATE_MIN_LD); /* calc slope for interpolation */
+            bitreservoir     = fMultI(slope, (INT)(BITRES_MAX_LD-BITRES_MIN_LD)) + BITRES_MIN_LD; /* interpolate */
+            hAacConfig->bitreservoir = bitreservoir & ~7; /* align to bytes */
+          }
         }
         if (hAacConfig->bitrateMode!=0) {
           return AACENC_INVALID_CONFIG;
@@ -879,6 +863,18 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         }
     }
 
+    if ((hAacConfig->bitrateMode >= 0) && (hAacConfig->bitrateMode <= 5)) {
+      if ((INT)config->userPeakBitrate != -1) {
+        hAacConfig->maxBitsPerFrame = (FDKaacEnc_CalcBitsPerFrame(fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), hAacConfig->framelength, hAacConfig->sampleRate) + 7)&~7;
+      }
+      else {
+        hAacConfig->maxBitsPerFrame = -1;
+      }
+      if (hAacConfig->audioMuxVersion==2) {
+        hAacConfig->minBitsPerFrame = fMin(32*8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, hAacConfig->framelength, hAacConfig->sampleRate))&~7;
+      }
+    }
+
     /* Initialize SBR parameters */
     if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD)
       && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) )
@@ -909,7 +905,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
     }
     else {
       /* SBR ratio has been set by the user, so use it. */
-      hAacConfig->sbrRatio = config->userSbrRatio;
+      hAacConfig->sbrRatio = isSbrActive(hAacConfig) ? config->userSbrRatio : 0;
     }
 
     {
@@ -1041,7 +1037,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
     }
 
     /* Clear input buffer */
-    if ( (InitFlags == AACENC_INIT_ALL) ) {
+    if ( InitFlags == AACENC_INIT_ALL ) {
         FDKmemclear(hAacEncoder->inputBuffer, sizeof(INT_PCM)*hAacEncoder->nMaxAacChannels*INPUTBUFFER_SIZE);
     }
 
@@ -1134,7 +1130,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
                 hAacConfig);
 
         /* create flags for transport encoder */
-        if (config->userTpAmxv == 1) {
+        if (config->userTpAmxv != 0) {
             flags |= TP_FLAG_LATM_AMV;
         }
         /* Clear output buffer */
@@ -1564,7 +1560,7 @@ AACENC_ERROR aacEncEncode(
           && ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) )
         {
           /* Set matrix mixdown coefficient. */
-          UINT pceValue = (UINT)( (1<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
+          UINT pceValue = (UINT)( (0<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
           if (hAacEncoder->extParam.userPceAdditions != pceValue) {
             hAacEncoder->extParam.userPceAdditions = pceValue;
             hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
@@ -1780,19 +1776,16 @@ AACENC_ERROR aacEncoder_SetParam(
             /* check if AOT matches the allocated modules */
             switch ( value ) {
               case AOT_PS:
-              case AOT_MP2_PS:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) {
                   err = AACENC_INVALID_CONFIG;
                   goto bail;
                 }
               case AOT_SBR:
-              case AOT_MP2_SBR:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) {
                   err = AACENC_INVALID_CONFIG;
                   goto bail;
                 }
               case AOT_AAC_LC:
-              case AOT_MP2_AAC_LC:
               case AOT_ER_AAC_LD:
               case AOT_ER_AAC_ELD:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@@ -1818,11 +1811,7 @@ AACENC_ERROR aacEncoder_SetParam(
         if (settings->userBitrateMode != value) {
             switch ( value ) {
               case 0:
-              case 1:
-              case 2:
-              case 3:
-              case 4:
-              case 5:
+              case 1: case 2: case 3: case 4: case 5:
               case 8:
                 settings->userBitrateMode = value;
                 hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
@@ -1973,6 +1962,16 @@ AACENC_ERROR aacEncoder_SetParam(
             hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
         }
         break;
+    case AACENC_AUDIOMUXVER:
+        if (settings->userTpAmxv != value) {
+            if ( !((value==0) || (value==1) || (value==2)) ) {
+                err = AACENC_INVALID_CONFIG;
+                break;
+            }
+            settings->userTpAmxv = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
+        }
+        break;
     case AACENC_TPSUBFRAMES:
         if (settings->userTpNsubFrames != value) {
             if (! ( (value>=1) && (value<=4) ) ) {
@@ -1998,7 +1997,7 @@ AACENC_ERROR aacEncoder_SetParam(
         break;
     case AACENC_METADATA_MODE:
         if ((UINT)settings->userMetaDataMode != value) {
-            if ( !((value>=0) && (value<=2)) ) {
+            if ( !(((INT)value>=0) && ((INT)value<=2)) ) {
                 err = AACENC_INVALID_CONFIG;
                 break;
             }
@@ -2006,6 +2005,12 @@ AACENC_ERROR aacEncoder_SetParam(
             hAacEncoder->InitFlags |= AACENC_INIT_CONFIG;
         }
         break;
+    case AACENC_PEAK_BITRATE:
+        if (settings->userPeakBitrate != value) {
+            settings->userPeakBitrate = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
+        }
+        break;
     default:
       err = AACENC_UNSUPPORTED_PARAMETER;
       break;
@@ -2076,6 +2081,9 @@ UINT aacEncoder_GetParam(
     case AACENC_HEADER_PERIOD:
         value = (UINT)hAacEncoder->coderConfig.headerPeriod;
         break;
+    case AACENC_AUDIOMUXVER:
+        value = (UINT)hAacEncoder->aacConfig.audioMuxVersion;
+        break;
     case AACENC_TPSUBFRAMES:
         value = (UINT)settings->userTpNsubFrames;
         break;
@@ -2088,6 +2096,12 @@ UINT aacEncoder_GetParam(
     case AACENC_METADATA_MODE:
         value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode;
         break;
+    case AACENC_PEAK_BITRATE:
+        value = (UINT)-1; /* peak bitrate parameter is meaningless */
+        if ( ((INT)hAacEncoder->extParam.userPeakBitrate!=-1) ) {
+          value = (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, hAacEncoder->aacConfig.bitRate)); /* peak bitrate parameter is in use */
+        }
+        break;
     default:
       //err = MPS_INVALID_PARAMETER;
       break;

libAACenc/src/aacenc_tns.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -95,13 +95,7 @@ amm-info@iis.fraunhofer.de
 #include "aacEnc_rom.h"
 #include "aacenc_tns.h"
 
-enum {
-    HIFILT = 0, /* index of higher filter */
-    LOFILT = 1 /* index of lower filter */
-};
-
-
-#define FILTER_DIRECTION 0
+#define FILTER_DIRECTION 0 /* 0 = up, 1 = down */
 
 static const FIXP_DBL acfWindowLong[12+3+1] = {
   0x7fffffff,0x7fb80000,0x7ee00000,0x7d780000,0x7b800000,0x78f80000,0x75e00000,0x72380000,
@@ -112,20 +106,6 @@ static const FIXP_DBL acfWindowShort[4+3+1] = {
   0x7fffffff,0x7e000000,0x78000000,0x6e000000,0x60000000,0x4e000000,0x38000000,0x1e000000
 };
 
-
-typedef struct {
-  INT      filterEnabled[MAX_NUM_OF_FILTERS];
-  INT      threshOn[MAX_NUM_OF_FILTERS];                /* min. prediction gain for using tns TABUL*/
-  INT      filterStartFreq[MAX_NUM_OF_FILTERS];         /* lowest freq for lpc TABUL*/
-  INT      tnsLimitOrder[MAX_NUM_OF_FILTERS];           /* Limit for TNS order TABUL*/
-  INT      tnsFilterDirection[MAX_NUM_OF_FILTERS];      /* Filtering direction, 0=up, 1=down TABUL */
-  INT      acfSplit[MAX_NUM_OF_FILTERS];
-  FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS];       /* TNS max. time resolution TABUL. Should be fract but MSVC won't compile then */
-  INT      seperateFiltersAllowed;
-
-} TNS_PARAMETER_TABULATED;
-
-
 typedef struct{
   INT                      bitRateFrom[2];  /* noneSbr=0, useSbr=1 */
   INT                      bitRateTo[2];    /* noneSbr=0, useSbr=1 */
@@ -373,6 +353,7 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
                                                  INT channels,
                                                  INT blockType,
                                                  INT granuleLength,
+                                                 INT isLowDelay,
                                                  INT ldSbrPresent,
                                                  TNS_CONFIG *tC,
                                                  PSY_CONFIGURATION *pC,
@@ -385,6 +366,8 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
   if (channels <= 0)
     return (AAC_ENCODER_ERROR)1;
 
+  tC->isLowDelay = isLowDelay;
+
   /* initialize TNS filter flag, order, and coefficient resolution (in bits per coeff) */
   tC->tnsActive      = (active) ? TRUE : FALSE;
   tC->maxOrder       = (blockType == SHORT_WINDOW) ? 5 : 12;  /* maximum: 7, 20 */
@@ -450,27 +433,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
         const TNS_PARAMETER_TABULATED* pCfg = FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent);
 
         if ( pCfg != NULL ) {
+
+          FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab));
+
           tC->lpcStartBand[HIFILT]         = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
           tC->lpcStartLine[HIFILT]         = pC->sfbOffset[tC->lpcStartBand[HIFILT]];
           tC->lpcStartBand[LOFILT]         = FDKaacEnc_FreqToBandWithRounding(pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, pC->sfbOffset);
           tC->lpcStartLine[LOFILT]         = pC->sfbOffset[tC->lpcStartBand[LOFILT]];
 
-          tC->confTab.threshOn[HIFILT] = pCfg->threshOn[HIFILT];
-          tC->confTab.threshOn[LOFILT] = pCfg->threshOn[LOFILT];
-
-          tC->confTab.tnsLimitOrder[HIFILT] = pCfg->tnsLimitOrder[HIFILT];
-          tC->confTab.tnsLimitOrder[LOFILT] = pCfg->tnsLimitOrder[LOFILT];
-
-          tC->confTab.tnsFilterDirection[HIFILT] = pCfg->tnsFilterDirection[HIFILT];
-          tC->confTab.tnsFilterDirection[LOFILT] = pCfg->tnsFilterDirection[LOFILT];
-
-          tC->confTab.acfSplit[HIFILT] = pCfg->acfSplit[HIFILT];
-          tC->confTab.acfSplit[LOFILT] = pCfg->acfSplit[LOFILT];
-
-          tC->confTab.filterEnabled[HIFILT] = pCfg->filterEnabled[HIFILT];
-          tC->confTab.filterEnabled[LOFILT] = pCfg->filterEnabled[LOFILT];
-          tC->confTab.seperateFiltersAllowed = pCfg->seperateFiltersAllowed;
-
           FDKaacEnc_CalcGaussWindow(tC->acfWindow[HIFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE);
           FDKaacEnc_CalcGaussWindow(tC->acfWindow[LOFILT], tC->maxOrder+1, sampleRate, granuleLength, pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE);
         }
@@ -614,6 +584,7 @@ static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac(
 
 static void FDKaacEnc_MergedAutoCorrelation(
         const FIXP_DBL           *spectrum,
+        const INT                 isLowDelay,
         const FIXP_DBL            acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER+3+1],
         const INT                 lpcStartLine[MAX_NUM_OF_FILTERS],
         const INT                 lpcStopLine,
@@ -633,6 +604,8 @@ static void FDKaacEnc_MergedAutoCorrelation(
     FDKmemclear(&_rxx1[0], sizeof(FIXP_DBL)*(maxOrder+1));
     FDKmemclear(&_rxx2[0], sizeof(FIXP_DBL)*(maxOrder+1));
 
+    idx0 = idx1 = idx2 = idx3 = idx4 = 0;
+
     /* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters */
     if ( (acfSplit[LOFILT]==-1) || (acfSplit[HIFILT]==-1) ) {
       /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the spectrum */
@@ -676,17 +649,27 @@ static void FDKaacEnc_MergedAutoCorrelation(
     /* compute energy normalization factors, i. e. 1/energy (saves some divisions) */
     if (rxx1_0 != FL2FXCONST_DBL(0.f))
     {
-        INT sc_fac1 = -1;
-        FIXP_DBL fac1 = FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2*sc1)+nsc1), &sc_fac1);
-        _rxx1[0] = scaleValue(fMult(rxx1_0,fac1),sc_fac1);
+      INT sc_fac1 = -1;
+      FIXP_DBL fac1 = FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2*sc1)+nsc1), &sc_fac1);
+      _rxx1[0] = scaleValue(fMult(rxx1_0,fac1),sc_fac1);
 
+      if (isLowDelay)
+      {
         for (lag = 1; lag <= maxOrder; lag++) {
           /* compute energy-normalized and windowed autocorrelation values at this lag */
+          FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1);
+          _rxx1[lag] = fMult(scaleValue(fMult(x1,fac1),sc_fac1), acfWindow[LOFILT][lag]);
+        }
+      }
+      else
+      {
+        for (lag = 1; lag <= maxOrder; lag++) {
           if ((3 * lag) <= maxOrder + 3) {
               FIXP_DBL x1 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1);
               _rxx1[lag] = fMult(scaleValue(fMult(x1,fac1),sc_fac1), acfWindow[LOFILT][3*lag]);
           }
         }
+      }
     }
 
     /* auto corr over upper 3/4 of spectrum */
@@ -762,8 +745,12 @@ INT FDKaacEnc_TnsDetect(
     : &tnsData->dataRaw.Long.subBlockInfo;
 
   tnsData->filtersMerged  = FALSE;
-  tsbi->tnsActive         = FALSE;
-  tsbi->predictionGain    = 1000;
+
+  tsbi->tnsActive[HIFILT]         = FALSE;
+  tsbi->predictionGain[HIFILT]    = 1000;
+  tsbi->tnsActive[LOFILT]         = FALSE;
+  tsbi->predictionGain[LOFILT]    = 1000;
+
   tnsInfo->numOfFilters[subBlockNumber] = 0;
   tnsInfo->coefRes[subBlockNumber]      = tC->coefRes;
   for (i = 0; i < tC->maxOrder; i++) {
@@ -779,6 +766,7 @@ INT FDKaacEnc_TnsDetect(
 
     FDKaacEnc_MergedAutoCorrelation(
           spectrum,
+          tC->isLowDelay,
           tC->acfWindow,
           tC->lpcStartLine,
           tC->lpcStopLine,
@@ -788,7 +776,7 @@ INT FDKaacEnc_TnsDetect(
           rxx2);
 
     /* compute higher TNS filter in lattice (ParCor) form with LeRoux-Gueguen algorithm */
-    tsbi->predictionGain = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]);
+    tsbi->predictionGain[HIFILT] = FDKaacEnc_AutoToParcor(rxx2, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT]);
 
     /* non-linear quantization of TNS lattice coefficients with given resolution */
     FDKaacEnc_Parcor2Index(
@@ -815,9 +803,9 @@ INT FDKaacEnc_TnsDetect(
     tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT];
 
     /* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small */
-    if ((tsbi->predictionGain > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2)))
+    if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT]/2 + 2)))
     {
-      tsbi->tnsActive = TRUE;
+      tsbi->tnsActive[HIFILT] = TRUE;
       tnsInfo->numOfFilters[subBlockNumber]++;
 
       /* compute second filter for lower quarter; only allowed for long windows! */
@@ -857,6 +845,7 @@ INT FDKaacEnc_TnsDetect(
           || ( (sumSqrCoef > 9)  && (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]) ) )
         {
           /* compare lower to upper filter; if they are very similar, merge them */
+          tsbi->tnsActive[LOFILT] = TRUE;
           sumSqrCoef = 0;
           for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) {
             sumSqrCoef += FDKabs(tnsInfo->coef[subBlockNumber][HIFILT][i] - tnsInfo->coef[subBlockNumber][LOFILT][i]);
@@ -884,6 +873,8 @@ INT FDKaacEnc_TnsDetect(
             tnsInfo->numOfFilters[subBlockNumber]++;
           }
         } /* filter lower part */
+        tsbi->predictionGain[LOFILT]=predGain;
+
       } /* second filter allowed  */
     } /* if predictionGain > 1437 ... */
   } /* maxOrder > 0 && tnsActive */
@@ -944,7 +935,7 @@ void FDKaacEnc_TnsSync(
       INT doSync = 1, absDiffSum = 0;
 
       /* if TNS is active in at least one channel, check if ParCor coefficients of higher filter are similar */
-      if (pSbInfoDestW->tnsActive || pSbInfoSrcW->tnsActive) {
+      if (pSbInfoDestW->tnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) {
         for (i = 0; i < tC->maxOrder; i++) {
           absDiff = FDKabs(tnsInfoDest->coef[w][HIFILT][i] - tnsInfoSrc->coef[w][HIFILT][i]);
           absDiffSum += absDiff;
@@ -957,12 +948,12 @@ void FDKaacEnc_TnsSync(
 
         if (doSync) {
             /* if no significant difference was detected, synchronize coefficient sets */
-            if (pSbInfoSrcW->tnsActive) {
+            if (pSbInfoSrcW->tnsActive[HIFILT]) {
               /* no dest filter, or more dest than source filters: use one dest filter */
-              if ((!pSbInfoDestW->tnsActive) ||
-                  ((pSbInfoDestW->tnsActive) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w])))
+              if ((!pSbInfoDestW->tnsActive[HIFILT]) ||
+                  ((pSbInfoDestW->tnsActive[HIFILT]) && (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w])))
               {
-                pSbInfoDestW->tnsActive = tnsInfoDest->numOfFilters[w] = 1;
+                pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1;
               }
               tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged;
               tnsInfoDest->order       [w][HIFILT] = tnsInfoSrc->order       [w][HIFILT];
@@ -975,7 +966,7 @@ void FDKaacEnc_TnsSync(
               }
             }
             else
-              pSbInfoDestW->tnsActive = tnsInfoDest->numOfFilters[w] = 0;
+              pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0;
             }
         }
 
@@ -1012,8 +1003,8 @@ INT FDKaacEnc_TnsEncode(
 {
     INT i, startLine, stopLine;
 
-    if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive) )
-      || ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive) ) )
+    if ( ( (blockType == SHORT_WINDOW) && (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber].tnsActive[HIFILT]) )
+      || ( (blockType != SHORT_WINDOW) && (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]) ) )
     {
       return 1;
     }
@@ -1129,8 +1120,9 @@ static INT FDKaacEnc_AutoToParcor(
   FIXP_DBL *RESTRICT workBuffer = parcorWorkBuffer;
   const FIXP_DBL  autoCorr_0 = input[0];
 
+  FDKmemclear(reflCoeff,numOfCoeff*sizeof(FIXP_DBL));
+
   if((FIXP_DBL)input[0] == FL2FXCONST_DBL(0.0)) {
-    FDKmemclear(reflCoeff,numOfCoeff*sizeof(FIXP_DBL));
     return(predictionGain);
   }
 

libAACenc/src/aacenc_tns.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -117,21 +117,25 @@ amm-info@iis.fraunhofer.de
 
 #define MAX_NUM_OF_FILTERS 2
 
+#define HIFILT 0           /* index of higher filter   */
+#define LOFILT 1           /* index of lower filter    */
 
-typedef struct{ /*stuff that is tabulated dependent on bitrate etc. */
-  INT     filterEnabled[MAX_NUM_OF_FILTERS];
-  INT     threshOn[MAX_NUM_OF_FILTERS];                /* min. prediction gain for using tns TABUL*/
-  INT     tnsLimitOrder[MAX_NUM_OF_FILTERS];           /* Limit for TNS order TABUL*/
-  INT     tnsFilterDirection[MAX_NUM_OF_FILTERS];      /* Filtering direction, 0=up, 1=down TABUL */
-  INT     acfSplit[MAX_NUM_OF_FILTERS];
-  INT     seperateFiltersAllowed;
-
-}TNS_CONFIG_TABULATED;
 
+typedef struct{ /* stuff that is tabulated dependent on bitrate etc. */
+  INT      filterEnabled[MAX_NUM_OF_FILTERS];
+  INT      threshOn[MAX_NUM_OF_FILTERS];                /* min. prediction gain for using tns TABUL*/
+  INT      filterStartFreq[MAX_NUM_OF_FILTERS];         /* lowest freq for lpc TABUL*/
+  INT      tnsLimitOrder[MAX_NUM_OF_FILTERS];           /* Limit for TNS order TABUL*/
+  INT      tnsFilterDirection[MAX_NUM_OF_FILTERS];      /* Filtering direction, 0=up, 1=down TABUL */
+  INT      acfSplit[MAX_NUM_OF_FILTERS];
+  FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS];       /* TNS max. time resolution TABUL. Should be fract but MSVC won't compile then */
+  INT      seperateFiltersAllowed;
+} TNS_PARAMETER_TABULATED;
 
 
 typedef struct {   /*assigned at InitTime*/
-  TNS_CONFIG_TABULATED confTab;
+  TNS_PARAMETER_TABULATED confTab;
+  INT isLowDelay;
   INT tnsActive;
   INT maxOrder;                /* max. order of tns filter */
   INT coefRes;
@@ -148,8 +152,8 @@ typedef struct {   /*assigned at InitTime*/
 
 
 typedef struct {
-  INT   tnsActive;
-  INT   predictionGain;
+  INT   tnsActive[MAX_NUM_OF_FILTERS];
+  INT   predictionGain[MAX_NUM_OF_FILTERS];
 } TNS_SUBBLOCK_INFO;
 
 typedef struct{   /*changed at runTime*/

libAACenc/src/adj_thr.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -130,14 +130,14 @@ the crash recovery strategy will be activated once.
 
 typedef struct {
   INT                 bitrate;
-  LONG                bits2PeFactor_mono;
-  LONG                bits2PeFactor_mono_slope;
-  LONG                bits2PeFactor_stereo;
-  LONG                bits2PeFactor_stereo_slope;
-  LONG                bits2PeFactor_mono_scfOpt;
-  LONG                bits2PeFactor_mono_scfOpt_slope;
-  LONG                bits2PeFactor_stereo_scfOpt;
-  LONG                bits2PeFactor_stereo_scfOpt_slope;
+  ULONG               bits2PeFactor_mono;
+  ULONG               bits2PeFactor_mono_slope;
+  ULONG               bits2PeFactor_stereo;
+  ULONG               bits2PeFactor_stereo_slope;
+  ULONG               bits2PeFactor_mono_scfOpt;
+  ULONG               bits2PeFactor_mono_scfOpt_slope;
+  ULONG               bits2PeFactor_stereo_scfOpt;
+  ULONG               bits2PeFactor_stereo_scfOpt_slope;
 
 } BIT_PE_SFAC;
 
@@ -153,10 +153,10 @@ static const BIT_PE_SFAC S_Bits2PeTab16000[] = {
   { 24000, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413},
   { 32000, 0x247AE148, 0x11B1D92B, 0x23851EB8, 0x01F75105, 0x247AE148, 0x110A137F, 0x23851EB8, 0x01F75105},
   { 48000, 0x2D1EB852, 0x6833C600, 0x247AE148, 0x014F8B59, 0x2CCCCCCD, 0x68DB8BAC, 0x247AE148, 0x01F75105},
-  { 64000, 0x60000000, 0x00000000, 0x251EB852, 0x154C985F, 0x60000000, 0x00000000, 0x2570A3D7, 0x154C985F},
-  { 96000, 0x60000000, 0x00000000, 0x39EB851F, 0x088509C0, 0x60000000, 0x00000000, 0x3A3D70A4, 0x088509C0},
-  {128000, 0x60000000, 0x00000000, 0x423D70A4, 0x18A43BB4, 0x60000000, 0x00000000, 0x428F5C29, 0x181E03F7},
-  {148000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000}
+  { 64000, 0x25c28f40, 0x00000000, 0x251EB852, 0x01480000, 0x25c28f40, 0x00000000, 0x2570A3D7, 0x01480000},
+  { 96000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000, 0x25c28f40, 0x00000000, 0x26000000, 0x01000000},
+  {128000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000, 0x25c28f40, 0x00000000, 0x270a3d80, 0x01000000},
+  {148000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000, 0x25c28f40, 0x00000000, 0x28000000, 0x00000000}
 };
 
 static const BIT_PE_SFAC S_Bits2PeTab22050[] = {
@@ -166,8 +166,8 @@ static const BIT_PE_SFAC S_Bits2PeTab22050[] = {
   { 48000, 0x23d70a3d, 0x014f8b59, 0x2199999a, 0x03eea20a, 0x23d70a3d, 0x14f8b59,  0x2199999a, 0x03eea20a},
   { 64000, 0x247ae148, 0x08d8ec96, 0x23851eb8, 0x00fba882, 0x247ae148, 0x88509c0,  0x23851eb8, 0x00fba882},
   { 96000, 0x2d1eb852, 0x3419e300, 0x247ae148, 0x00a7c5ac, 0x2ccccccd, 0x346dc5d6, 0x247ae148, 0x00fba882},
-  {128000, 0x60000000, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x00000000, 0x2570a3d7, 0x009f16b1},
-  {148000, 0x60000000, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x00000000, 0x270a3d71, 0x00000000}
+  {128000, 0x25c28f40, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x25c28f40, 0x2570a3d7, 0x009f16b1},
+  {148000, 0x25c28f40, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x25c28f40, 0x270a3d71, 0x00000000}
 };
 
 static const BIT_PE_SFAC S_Bits2PeTab24000[] = {
@@ -178,21 +178,21 @@ static const BIT_PE_SFAC S_Bits2PeTab24000[] = {
   { 64000, 0x24cccccd, 0x05e5f30e, 0x22e147ae, 0x01a36e2f, 0x24cccccd, 0x05e5f30e, 0x23333333, 0x014f8b59},
   { 96000, 0x2a8f5c29, 0x24b33db0, 0x247ae148, 0x00fba882, 0x2a8f5c29, 0x26fe718b, 0x247ae148, 0x00fba882},
   {128000, 0x4e666666, 0x1cd5f99c, 0x2570a3d7, 0x010c6f7a, 0x50a3d70a, 0x192a7371, 0x2570a3d7, 0x010c6f7a},
-  {148000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000}
+  {148000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000, 0x25c28f40, 0x00000000, 0x26147ae1, 0x00000000}
 };
 
 static const BIT_PE_SFAC S_Bits2PeTab32000[] = {
-  { 16000, 0x1199999a, 0x20c49ba6, 0x00000000, 0x4577d955, 0x00000000, 0x60fe4799, 0x00000000, 0x00000000},
-  { 24000, 0x1999999a, 0x0fba8827, 0x10f5c28f, 0x1b866e44, 0x17ae147b, 0x0fba8827, 0x00000000, 0x4d551d69},
+  { 16000, 0x247ae140, 0xFFFFAC1E, 0x270a3d80, 0xFFFE9B7C, 0x14ccccc0, 0x000110A1, 0x15c28f60, 0xFFFEEF5F},
+  { 24000, 0x23333340, 0x0fba8827, 0x21999980, 0x1b866e44, 0x18f5c280, 0x0fba8827, 0x119999a0, 0x4d551d69},
   { 32000, 0x1d70a3d7, 0x07357e67, 0x17ae147b, 0x09d49518, 0x1b851eb8, 0x0a7c5ac4, 0x12e147ae, 0x110a137f},
   { 48000, 0x20f5c28f, 0x049667b6, 0x1c7ae148, 0x053e2d62, 0x20a3d70a, 0x053e2d62, 0x1b333333, 0x05e5f30e},
   { 64000, 0x23333333, 0x029f16b1, 0x1f0a3d71, 0x02f2f987, 0x23333333, 0x029f16b1, 0x1e147ae1, 0x03eea20a},
   { 96000, 0x25c28f5c, 0x2c3c9eed, 0x21eb851f, 0x01f75105, 0x25c28f5c, 0x0a7c5ac4, 0x21eb851f, 0x01a36e2f},
   {128000, 0x50f5c28f, 0x18a43bb4, 0x23d70a3d, 0x010c6f7a, 0x30000000, 0x168b5cc0, 0x23851eb8, 0x0192a737},
-  {148000, 0x60000000, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476},
-  {160000, 0x60000000, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184},
-  {200000, 0x00000000, 0x00000000, 0x2b333333, 0x0836be91, 0x00000000, 0x00000000, 0x2b333333, 0x0890390f},
-  {320000, 0x00000000, 0x00000000, 0x4947ae14, 0x00000000, 0x00000000, 0x00000000, 0x4a8f5c29, 0x00000000}
+  {148000, 0x25c28f40, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476},
+  {160000, 0x25c28f40, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184},
+  {200000, 0x25c28f40, 0x00000000, 0x2b333333, 0x0836be91, 0x25c28f40, 0x00000000, 0x2b333333, 0x0890390f},
+  {320000, 0x25c28f40, 0x00000000, 0x4947ae14, 0x00000000, 0x25c28f40, 0x00000000, 0x4a8f5c29, 0x00000000}
 };
 
 static const BIT_PE_SFAC S_Bits2PeTab44100[] = {
@@ -205,8 +205,8 @@ static const BIT_PE_SFAC S_Bits2PeTab44100[] = {
   {128000, 0x2ae147ae, 0x1b435265, 0x223d70a4, 0x0192a737, 0x2a3d70a4, 0x1040bfe4, 0x21eb851f, 0x0192a737},
   {148000, 0x3b851eb8, 0x2832069c, 0x23333333, 0x00dfb23b, 0x3428f5c3, 0x2054c288, 0x22e147ae, 0x00dfb23b},
   {160000, 0x4a3d70a4, 0xc32ebe5a, 0x23851eb8, 0x01d5c316, 0x40000000, 0xcb923a2b, 0x23333333, 0x01d5c316},
-  {200000, 0x00000000, 0x00000000, 0x25c28f5c, 0x0713f078, 0x00000000, 0x00000000, 0x2570a3d7, 0x072a4f17},
-  {320000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000}
+  {200000, 0x25c28f40, 0x00000000, 0x25c28f5c, 0x0713f078, 0x25c28f40, 0x00000000, 0x2570a3d7, 0x072a4f17},
+  {320000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000, 0x25c28f40, 0x00000000, 0x3fae147b, 0x00000000}
 };
 
 static const BIT_PE_SFAC S_Bits2PeTab48000[] = {
@@ -219,8 +219,8 @@ static const BIT_PE_SFAC S_Bits2PeTab48000[] = {
   {128000, 0x28f5c28f, 0x14727dcc, 0x2147ae14, 0x0218def4, 0x2851eb85, 0x0e27e0f0, 0x20f5c28f, 0x0218def4},
   {148000, 0x3570a3d7, 0x1cd5f99c, 0x228f5c29, 0x01bf6476, 0x30f5c28f, 0x18777e75, 0x223d70a4, 0x01bf6476},
   {160000, 0x40000000, 0xcb923a2b, 0x23333333, 0x0192a737, 0x39eb851f, 0xd08d4bae, 0x22e147ae, 0x0192a737},
-  {200000, 0x00000000, 0x00000000, 0x251eb852, 0x06775a1b, 0x00000000, 0x00000000, 0x24cccccd, 0x06a4175a},
-  {320000, 0x00000000, 0x00000000, 0x3ccccccd, 0x00000000, 0x00000000, 0x00000000, 0x3d1eb852, 0x00000000}
+  {200000, 0x25c28f40, 0x00000000, 0x251eb852, 0x06775a1b, 0x25c28f40, 0x00000000, 0x24cccccd, 0x06a4175a},
+  {320000, 0x25c28f40, 0x00000000, 0x3ccccccd, 0x00000000, 0x25c28f40, 0x00000000, 0x3d1eb852, 0x00000000}
 };
 
 static const BITS2PE_CFG_TAB bits2PeConfigTab[] = {
@@ -258,6 +258,7 @@ static void FDKaacEnc_InitBits2PeFactor(
         const INT nChannels,
         const INT sampleRate,
         const INT advancedBitsToPe,
+        const INT dZoneQuantEnable,
         const INT invQuant
         )
 {
@@ -329,7 +330,32 @@ static void FDKaacEnc_InitBits2PeFactor(
   } /* advancedBitsToPe */
 
 
-  /* return bits2pe factor */
+  if (dZoneQuantEnable)
+  {
+    if(bit2PE_m >= (FL2FXCONST_DBL(0.6f))>>bit2PE_e)
+    {
+      /* Additional headroom for addition */
+      bit2PE_m >>= 1;
+      bit2PE_e  += 1;
+    }
+
+    /* the quantTendencyCompensator compensates a lower bit consumption due to increasing the tendency to quantize low spectral values to the lower quantizer border for bitrates below a certain bitrate threshold --> see also function calcSfbDistLD in quantize.c */
+    if ((bitRate/nChannels > 32000) && (bitRate/nChannels <= 40000)) {
+      bit2PE_m += (FL2FXCONST_DBL(0.4f))>>bit2PE_e;
+    }
+    else if (bitRate/nChannels > 20000) {
+      bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e;
+    }
+    else if (bitRate/nChannels >= 16000) {
+      bit2PE_m += (FL2FXCONST_DBL(0.3f))>>bit2PE_e;
+    }
+    else {
+      bit2PE_m += (FL2FXCONST_DBL(0.0f))>>bit2PE_e;
+    }
+  }
+
+
+  /***** 3.) Return bits2pe factor *****/
   *bits2PeFactor_m = bit2PE_m;
   *bits2PeFactor_e = bit2PE_e;
 }
@@ -1649,6 +1675,7 @@ static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING*  cm,
                                 QC_OUT_ELEMENT*   qcElement[(8)],
                                 PSY_OUT_ELEMENT*  psyOutElement[(8)],
                                 const INT         desiredPe,
+                                const INT         maxIter2ndGuess,
                                 const INT         processElements,
                                 const INT         elementOffset)
 {
@@ -1733,7 +1760,7 @@ static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING*  cm,
    /* Part III: Iterate until bit constraints are met */
    /* -------------------------------------------------- */
    iter = 0;
-   while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f),desiredPe)) && (iter < 1)) {
+   while ((fixp_abs(redPeGlobal - desiredPe) > fMultI(FL2FXCONST_DBL(0.05f),desiredPe)) && (iter < maxIter2ndGuess)) {
 
      INT desiredPeNoAHGlobal;
      INT redPeNoAHGlobal = 0;
@@ -2138,7 +2165,7 @@ static FIXP_DBL FDKaacEnc_bitresCalcBitFac(const INT       bitresBits,
         bresParam->clipSpendLow, bresParam->clipSpendHigh,
         bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope);
 
-    pe_pers = fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin);
+    pe_pers = (pex > adjThrChan->peMin) ? fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin) : 0;
     tmp_fix = fMult(((FIXP_DBL)bitSpend + (FIXP_DBL)bitSave), pe_pers);
     bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2);
 
@@ -2225,7 +2252,8 @@ void FDKaacEnc_AdjThrInit(
         INT             nChannelsEff,
         INT             sampleRate,
         INT             advancedBitsToPe,
-        FIXP_DBL        vbrQualFactor
+        FIXP_DBL        vbrQualFactor,
+        const INT       dZoneQuantEnable
         )
 {
   INT i;
@@ -2233,6 +2261,10 @@ void FDKaacEnc_AdjThrInit(
   FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f);
   FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f);
 
+  /* Max number of iterations in second guess is 3 for lowdelay aot and for configurations with
+     multiple audio elements in general, otherwise iteration value is always 1. */
+  hAdjThr->maxIter2ndGuess = (advancedBitsToPe!=0 || nElements>1) ? 3 : 1;
+
   /* common for all elements: */
   /* parameters for bitres control */
   hAdjThr->bresParamLong.clipSaveLow   = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
@@ -2313,10 +2345,11 @@ void FDKaacEnc_AdjThrInit(
     FDKaacEnc_InitBits2PeFactor(
               &atsElem->bits2PeFactor_m,
               &atsElem->bits2PeFactor_e,
-              chBitrate,       /* bitrate/channel*/
+              chBitrate*nChannelsEff, /* overall bitrate */
               nChannelsEff,    /* number of channels */
               sampleRate,
               advancedBitsToPe,
+              dZoneQuantEnable,
               invQuant
               );
 
@@ -2545,6 +2578,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT*        AdjThrStateElement[(8)],
                                 QC_OUT*             qcOut,
                                 PSY_OUT_ELEMENT*    psyOutElement[(8)],
                                 INT                 CBRbitrateMode,
+                                INT                 maxIter2ndGuess,
                                 CHANNEL_MAPPING*    cm)
 {
     int i;
@@ -2570,6 +2604,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT*        AdjThrStateElement[(8)],
                                             qcElement,
                                             psyOutElement,
                                             qcElement[i]->grantedPeCorr,
+                                            maxIter2ndGuess,
                                             1,         /* Process only 1 element */
                                             i);        /* Process exactly THIS element */
 

libAACenc/src/adj_thr.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -118,7 +118,8 @@ void FDKaacEnc_AdjThrInit(ADJ_THR_STATE *hAdjThr,
                 INT nChannelsEff,
                 INT sampleRate,
                 INT advancedBitsToPe,
-                FIXP_DBL vbrQualFactor);
+                FIXP_DBL vbrQualFactor,
+                const INT dZoneQuantEnable);
  
 
 void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
@@ -140,6 +141,7 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)],
     QC_OUT*           qcOut,
     PSY_OUT_ELEMENT*  psyOutElement[(8)],
     INT               CBRbitrateMode,
+    INT               maxIter2ndGuess,
     CHANNEL_MAPPING*  cm);
 
 void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** hAdjThr);

libAACenc/src/adj_thr_data.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -145,6 +145,7 @@ typedef struct {
 typedef struct {
   BRES_PARAM bresParamLong, bresParamShort;
   ATS_ELEMENT* adjThrStateElem[(8)];
+  INT maxIter2ndGuess;
 } ADJ_THR_STATE;
 
 #endif

libAACenc/src/bandwidth.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,7 +127,7 @@ static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = {
   { 8000,  2000,  2000},
   {12000,  2000,  2300},
   {16000,  2200,  2500},
-  {24000,  5650,  6400},
+  {24000,  5650,  7200},
   {32000,  11600, 12000},
   {40000,  12000, 16000},
   {48000,  16000, 16000},
@@ -138,10 +138,10 @@ static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = {
 static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = {
   { 8000,  2000,  2000},
   {12000,  2000,  2000},
-  {24000,  4250,  5200},
+  {24000,  4250,  7200},
   {32000,  8400,  9000},
   {40000,  9400,  11300},
-  {48000,  11900, 13700},
+  {48000,  11900, 14700},
   {64000,  14800, 16000},
   {76000,  16000, 16000},
   {360001, 16000, 16000}

libAACenc/src/intensity.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -325,7 +325,6 @@ FDKaacEnc_prepareIntensityDecision(const FIXP_DBL    *sfbEnergyLeft,
 
       channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f);
 
-      FDK_ASSERT(50 >= 49);
       /* max width of scalefactorband is 96; width's are always even */
       /* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent loops */
       inv_n = GetInvInt((sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs])>>1);

libAACenc/src/metadata_compressor.cpp

@@ -193,36 +193,36 @@ struct DRC_COMP {
  *  Profile tables.
  */
 static const FIXP_DBL tabMaxBoostThr[] = {
-    (FIXP_DBL)(-43<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-53<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-55<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-65<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-50<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-40<<METADATA_FRACT_BITS)
+    (FIXP_DBL)(int)((unsigned)-43<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-53<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-55<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-65<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-50<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-40<<METADATA_FRACT_BITS)
 };
 static const FIXP_DBL tabBoostThr[] = {
-    (FIXP_DBL)(-31<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-41<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-31<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-41<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-31<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-31<<METADATA_FRACT_BITS)
+    (FIXP_DBL)(int)((unsigned)-31<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-41<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-31<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-41<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-31<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-31<<METADATA_FRACT_BITS)
 };
 static const FIXP_DBL tabEarlyCutThr[] = {
-    (FIXP_DBL)(-26<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-21<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-26<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-21<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-26<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-20<<METADATA_FRACT_BITS)
+    (FIXP_DBL)(int)((unsigned)-26<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-21<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-26<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-21<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-26<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-20<<METADATA_FRACT_BITS)
 };
 static const FIXP_DBL tabCutThr[]      = {
-    (FIXP_DBL)(-16<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-11<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-16<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-21<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-16<<METADATA_FRACT_BITS),
-    (FIXP_DBL)(-10<<METADATA_FRACT_BITS)
+    (FIXP_DBL)(int)((unsigned)-16<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-11<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-16<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-21<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-16<<METADATA_FRACT_BITS),
+    (FIXP_DBL)(int)((unsigned)-10<<METADATA_FRACT_BITS)
 };
 static const FIXP_DBL tabMaxCutThr[]   = {
     (FIXP_DBL)(4<<METADATA_FRACT_BITS),
@@ -576,7 +576,7 @@ INT FDK_DRC_Generator_Initialize(
     drcComp->channels     = channelMapping.nChannels;
 
     /* Init states. */
-    drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = (FIXP_DBL)(-135<<METADATA_FRACT_BITS);
+    drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = (FIXP_DBL)(int)((unsigned)-135<<METADATA_FRACT_BITS);
 
     FDKmemclear(drcComp->smoothGain, sizeof(drcComp->smoothGain));
     FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt));

libAACenc/src/pnsparam.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -107,6 +107,7 @@ typedef struct {
 typedef struct {
   ULONG brFrom;
   ULONG brTo;
+  UCHAR S16000;
   UCHAR S22050;
   UCHAR S24000;
   UCHAR S32000;
@@ -115,25 +116,26 @@ typedef struct {
 } AUTO_PNS_TAB;
 
 static const AUTO_PNS_TAB levelTable_mono[]= {
-  {0,      11999, 1, 1, 1, 1, 1,},
-  {12000,  19999, 1, 1, 1, 1, 1,},
-  {20000,  28999, 2, 1, 1, 1, 1,},
-  {29000,  40999, 4, 4, 4, 2, 2,},
-  {41000,  55999, 9, 9, 7, 7, 7,},
-  {56000,  79999, 0, 0, 0, 9, 9,},
-  {80000,  99999, 0, 0, 0, 0, 0,},
-  {100000,999999, 0, 0, 0, 0, 0,},
+  {0,      11999, 0, 1, 1, 1, 1, 1,},
+  {12000,  19999, 0, 1, 1, 1, 1, 1,},
+  {20000,  28999, 0, 2, 1, 1, 1, 1,},
+  {29000,  40999, 0, 4, 4, 4, 2, 2,},
+  {41000,  55999, 0, 9, 9, 7, 7, 7,},
+  {56000,  61999, 0, 0, 0, 0, 9, 9,},
+  {62000,  75999, 0, 0, 0, 0, 0, 0,},
+  {76000,  92999, 0, 0, 0, 0, 0, 0,},
+  {93000, 999999, 0, 0, 0, 0, 0, 0,},
 };
 
 static const AUTO_PNS_TAB levelTable_stereo[]= {
-  {0,      11999, 1, 1, 1, 1, 1,},
-  {12000,  19999, 3, 1, 1, 1, 1,},
-  {20000,  28999, 3, 3, 3, 2, 2,},
-  {29000,  40999, 7, 6, 6, 5, 5,},
-  {41000,  55999, 9, 9, 7, 7, 7,},
-  {56000,  79999, 0, 0, 0, 0, 0,},
-  {80000,  99999, 0, 0, 0, 0, 0,},
-  {100000,999999, 0, 0, 0, 0, 0,},
+  {0,      11999, 0, 1, 1, 1, 1, 1,},
+  {12000,  19999, 0, 3, 1, 1, 1, 1,},
+  {20000,  28999, 0, 3, 3, 3, 2, 2,},
+  {29000,  40999, 0, 7, 6, 6, 5, 5,},
+  {41000,  55999, 0, 9, 9, 7, 7, 7,},
+  {56000,  79999, 0, 0, 0, 0, 0, 0,},
+  {80000,  99999, 0, 0, 0, 0, 0, 0,},
+  {100000,999999, 0, 0, 0, 0, 0, 0,},
 };
 
 
@@ -160,11 +162,11 @@ static const PNS_INFO_TAB pnsInfoTab[] = {
 };
 
 static const AUTO_PNS_TAB levelTable_lowComplexity[]= {
-  {0,      27999, 0, 0, 0, 0, 0,},
-  {28000,  31999, 2, 2, 2, 2, 2,},
-  {32000,  47999, 3, 3, 3, 3, 3,},
-  {48000,  48000, 4, 4, 4, 4, 4,},
-  {48001, 999999, 0, 0, 0, 0, 0,},
+  {0,      27999, 0, 0, 0, 0, 0, 0,},
+  {28000,  31999, 0, 2, 2, 2, 2, 2,},
+  {32000,  47999, 0, 3, 3, 3, 3, 3,},
+  {48000,  48000, 0, 4, 4, 4, 4, 4,},
+  {48001, 999999, 0, 0, 0, 0, 0, 0,},
 };
 
 /* conversion of old LC tuning tables to new (LD enc) structure (only entries which are actually used were converted) */
@@ -211,6 +213,7 @@ int FDKaacEnc_lookUpPnsUse (int bitRate, int sampleRate, int numChan, const int
   }
 
   switch (sampleRate) {
+  case 16000: hUsePns = levelTable[i].S16000; break;
   case 22050: hUsePns = levelTable[i].S22050; break;
   case 24000: hUsePns = levelTable[i].S24000; break;
   case 32000: hUsePns = levelTable[i].S32000; break;

libAACenc/src/psy_configuration.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -634,13 +634,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT   bitrate,
         if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine)
             break;
     }
-    psyConf->sfbActive = sfb;
+    psyConf->sfbActive = FDKmax(sfb, 1);
 
     for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){
         if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE)
             break;
     }
     psyConf->sfbActiveLFE = sfb;
+    psyConf->sfbActive = FDKmax(psyConf->sfbActive, psyConf->sfbActiveLFE);
 
     /* calculate minSnr */
     FDKaacEnc_initMinSnr(bitrate,

libAACenc/src/psy_main.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -342,6 +342,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy,
         tnsChannels,
         LONG_WINDOW,
         hPsy->granuleLength,
+        isLowDelay(audioObjectType),
         (syntaxFlags&AC_SBR_PRESENT)?1:0,
        &(hPsy->psyConf[0].tnsConf),
        &hPsy->psyConf[0],
@@ -362,6 +363,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMainInit(PSY_INTERNAL *hPsy,
             tnsChannels,
             SHORT_WINDOW,
             hPsy->granuleLength,
+            isLowDelay(audioObjectType),
             (syntaxFlags&AC_SBR_PRESENT)?1:0,
            &hPsy->psyConf[1].tnsConf,
            &hPsy->psyConf[1],
@@ -447,7 +449,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
                                     INT                  totalChannels
                                    )
 {
-    INT commonWindow = 1;
+    const INT commonWindow = 1;
     INT maxSfbPerGroup[(2)];
     INT mdctSpectrum_e;
     INT ch;   /* counts through channels          */
@@ -621,7 +623,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
           FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset],
                       (windowLength[ch]-psyData[ch]->lowpassLine)*sizeof(FIXP_DBL));
 
-          if (hPsyConfLong->filterbank != FB_LC) {
+          if ( (hPsyConfLong->filterbank != FB_LC) && (psyData[ch]->lowpassLine >= FADE_OUT_LEN) ) {
             /* Do blending to reduce gibbs artifacts */
             for (int i=0; i<FADE_OUT_LEN; i++) {
               psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]);
@@ -763,7 +765,8 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
 
     /* Advance psychoacoustics: Tonality and TNS */
     if (psyStatic[0]->isLFE) {
-        tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive = 0;
+        tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] = 0;
+        tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] = 0;
     }
     else
     {
@@ -815,15 +818,19 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
                       &hThisPsyConf[1]->tnsConf);
             }
 
-            FDK_ASSERT(commonWindow=1); /* all checks for TNS do only work for common windows (which is always set)*/
+            FDK_ASSERT(1==commonWindow); /* all checks for TNS do only work for common windows (which is always set)*/
             for(w = 0; w < nWindows[0]; w++)
             {
                 if (isShortWindow[0])
-                    tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive ||
-                    ((channels == 2) ? tnsData[1]->dataRaw.Short.subBlockInfo[w].tnsActive : 0);
+                    tnsActive[w] = tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] ||
+                    tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT] ||
+                    tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] ||
+                    tnsData[channels-1]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT];
                 else
-                    tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive ||
-                    ((channels == 2) ? tnsData[1]->dataRaw.Long.subBlockInfo.tnsActive : 0);
+                    tnsActive[w] = tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] ||
+                    tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] ||
+                    tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] ||
+                    tnsData[channels-1]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT];
             }
 
             for(ch = 0; ch < channels; ch++) {
@@ -1150,8 +1157,8 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
                        psyData[ch]->sfbMaxScaleSpec.Long,
                        sfbTonality[ch],
                        psyOutChannel[ch]->tnsInfo.order[0][0],
-                       tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain,
-                       tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive,
+                       tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain[HIFILT],
+                       tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT],
                        psyOutChannel[ch]->sfbEnergyLdData,
                        psyOutChannel[ch]->noiseNrg );
         } /* !isLFE */

libAACenc/src/qc_data.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -269,6 +269,8 @@ typedef struct
   BITCNTR_STATE *hBitCounter;
   ADJ_THR_STATE *hAdjThr;
 
+  INT dZoneQuantEnable;   /* enable dead zone quantizer */
+
 } QC_STATE;
 
 #endif /* _QC_DATA_H */

libAACenc/src/qc_main.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -107,14 +107,11 @@ typedef struct {
 } TAB_VBR_QUAL_FACTOR;
 
 static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = {
-  {QCDATA_BR_MODE_CBR,   FL2FXCONST_DBL(0.00f)},
   {QCDATA_BR_MODE_VBR_1, FL2FXCONST_DBL(0.160f)}, /* 32 kbps mono   AAC-LC + SBR + PS */
   {QCDATA_BR_MODE_VBR_2, FL2FXCONST_DBL(0.148f)}, /* 64 kbps stereo AAC-LC + SBR      */
   {QCDATA_BR_MODE_VBR_3, FL2FXCONST_DBL(0.135f)}, /* 80 - 96 kbps stereo AAC-LC       */
   {QCDATA_BR_MODE_VBR_4, FL2FXCONST_DBL(0.111f)}, /* 128 kbps stereo AAC-LC           */
-  {QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)}, /* 192 kbps stereo AAC-LC           */
-  {QCDATA_BR_MODE_SFR,   FL2FXCONST_DBL(0.00f)},
-  {QCDATA_BR_MODE_FF,    FL2FXCONST_DBL(0.00f)}
+  {QCDATA_BR_MODE_VBR_5, FL2FXCONST_DBL(0.070f)}  /* 192 kbps stereo AAC-LC           */
 };
 
 static INT isConstantBitrateMode(
@@ -369,6 +366,7 @@ QCNew_bail:
 AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
                                    struct QC_INIT *init)
 {
+  int i;
   hQC->maxBitsPerFrame = init->maxBits;
   hQC->minBitsPerFrame = init->minBits;
   hQC->nElements       = init->channelMapping->nElements;
@@ -382,7 +380,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
   if ( isConstantBitrateMode(hQC->bitrateMode) ) {
     INT bitresPerChannel = (hQC->bitResTotMax / init->channelMapping->nChannelsEff);
     /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
-    hQC->bitDistributionMode = (bitresPerChannel>100) ? 0 : (bitresPerChannel>0) ? 1 : 2;
+    hQC->bitDistributionMode = (bitresPerChannel>BITRES_MIN_LD) ? 0 : (bitresPerChannel>0) ? 1 : 2;
   }
   else {
     hQC->bitDistributionMode = 0; /* full bitreservoir */
@@ -399,25 +397,22 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
                             (init->averageBits/init->nSubFrames) - hQC->globHdrBits,
                             hQC->maxBitsPerFrame/init->channelMapping->nChannelsEff);
 
-  switch(hQC->bitrateMode){
-    case QCDATA_BR_MODE_CBR:
-    case QCDATA_BR_MODE_VBR_1:
-    case QCDATA_BR_MODE_VBR_2:
-    case QCDATA_BR_MODE_VBR_3:
-    case QCDATA_BR_MODE_VBR_4:
-    case QCDATA_BR_MODE_VBR_5:
-    case QCDATA_BR_MODE_SFR:
-    case QCDATA_BR_MODE_FF:
-      if((int)hQC->bitrateMode < (int)(sizeof(tableVbrQualFactor)/sizeof(TAB_VBR_QUAL_FACTOR))){
-        hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[hQC->bitrateMode].vbrQualFactor;
-      } else {
-        hQC->vbrQualFactor = FL2FXCONST_DBL(0.f); /* default setting */
-      }
-      break;
-    case QCDATA_BR_MODE_INVALID:
-    default:
-      hQC->vbrQualFactor = FL2FXCONST_DBL(0.f);
+  hQC->vbrQualFactor = FL2FXCONST_DBL(0.f);
+  for (i=0; i<(int)(sizeof(tableVbrQualFactor)/sizeof(TAB_VBR_QUAL_FACTOR)); i++) {
+    if (hQC->bitrateMode==tableVbrQualFactor[i].bitrateMode) {
+      hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[i].vbrQualFactor;
       break;
+    }
+  }
+
+  if (init->channelMapping->nChannelsEff == 1 &&
+     (init->bitrate / init->channelMapping->nChannelsEff) < 32000 &&
+     init->advancedBitsToPe != 0
+     )
+  {
+    hQC->dZoneQuantEnable = 1;
+  } else {
+    hQC->dZoneQuantEnable = 0;
   }
 
   FDKaacEnc_AdjThrInit(
@@ -429,7 +424,8 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
         init->channelMapping->nChannelsEff,
         init->sampleRate,                 /* output sample rate */
         init->advancedBitsToPe,           /* if set, calc bits2PE factor depending on samplerate */
-        hQC->vbrQualFactor
+        hQC->vbrQualFactor,
+        hQC->dZoneQuantEnable
         );
 
   return AAC_ENC_OK;
@@ -892,6 +888,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
                                      qcOut[c],
                                      psyOut[c]->psyOutElement,
                                      isConstantBitrateMode(hQC->bitrateMode),
+                                     hQC->hAdjThr->maxIter2ndGuess,
                                      cm);
 
       } /* -end- sub frame counter */
@@ -919,6 +916,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
                       FDKaacEnc_EstimateScaleFactors(psyOut[c]->psyOutElement[i]->psyOutChannel,
                                             qcElement[c][i]->qcOutChannel,
                                             hQC->invQuant,
+                                            hQC->dZoneQuantEnable,
                                             cm->elInfo[i].nChannelsInEl);
 
 
@@ -1013,7 +1011,8 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
                                                              qcOutCh->mdctSpectrum,
                                                              qcOutCh->globalGain,
                                                              qcOutCh->scf,
-                                                             qcOutCh->quantSpec) ;
+                                                             qcOutCh->quantSpec,
+                                                             hQC->dZoneQuantEnable);
 
                                   /*-------------------------------------------- */
                                   if (FDKaacEnc_calcMaxValueInSfb(psyOutCh->sfbCnt,
@@ -1263,6 +1262,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
     case QCDATA_BR_MODE_VBR_4:
     case QCDATA_BR_MODE_VBR_5:
       qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits)&7; /* precalculate alignment bits */
+      qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
+      qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
       break;
 
     case QCDATA_BR_MODE_CBR:
@@ -1272,6 +1273,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
       /* processing fill-bits */
       INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ;
       qcOut[0]->totFillBits = fixMax((deltaBitRes&7), (deltaBitRes - (fixMax(0,bitResSpace-7)&~7)));
+      qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
+      qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
       break;
   } /* switch (qcKernel->bitrateMode) */
 

libAACenc/src/quantize.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -104,13 +104,19 @@ amm-info@iis.fraunhofer.de
 static void FDKaacEnc_quantizeLines(INT      gain,
                           INT      noOfLines,
                           FIXP_DBL *mdctSpectrum,
-                          SHORT      *quaSpectrum)
+                          SHORT    *quaSpectrum,
+                          INT      dZoneQuantEnable)
 {
   int   line;
-  FIXP_DBL k = FL2FXCONST_DBL(-0.0946f + 0.5f)>>16;
+  FIXP_DBL k = FL2FXCONST_DBL(0.0f);
   FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain)&3];
   INT      quantizershift = ((-gain)>>2)+1;
+  const INT kShift=16;
 
+  if (dZoneQuantEnable)
+    k = FL2FXCONST_DBL(0.23f)>>kShift;
+  else
+    k = FL2FXCONST_DBL(-0.0946f + 0.5f)>>kShift;
 
   for (line = 0; line < noOfLines; line++)
   {
@@ -263,7 +269,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
                       FIXP_DBL *mdctSpectrum,
                       INT globalGain,
                       INT *scalefactors,
-                      SHORT *quantizedSpectrum)
+                      SHORT *quantizedSpectrum,
+                      INT dZoneQuantEnable)
 {
   INT sfbOffs,sfb;
 
@@ -280,7 +287,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
     FDKaacEnc_quantizeLines(globalGain - scalefactor, /* QSS */
                   sfbOffset[sfbOffs+sfb+1] - sfbOffset[sfbOffs+sfb],
                   mdctSpectrum + sfbOffset[sfbOffs+sfb],
-                  quantizedSpectrum + sfbOffset[sfbOffs+sfb]);
+                  quantizedSpectrum + sfbOffset[sfbOffs+sfb],
+                  dZoneQuantEnable);
   }
 }
 
@@ -296,7 +304,8 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
 FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
                      SHORT *quantSpectrum,
                      INT noOfLines,
-                     INT gain
+                     INT gain,
+                     INT dZoneQuantEnable
                      )
 {
   INT i,scale;
@@ -311,7 +320,8 @@ FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
     FDKaacEnc_quantizeLines(gain,
                   1,
                  &mdctSpectrum[i],
-                 &quantSpectrum[i]);
+                 &quantSpectrum[i],
+                  dZoneQuantEnable);
 
     if (fAbs(quantSpectrum[i])>MAX_QUANT) {
       return FL2FXCONST_DBL(0.0f);

libAACenc/src/quantize.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -102,12 +102,14 @@ void FDKaacEnc_QuantizeSpectrum(INT sfbCnt,
                       INT sfbPerGroup,
                       INT *sfbOffset, FIXP_DBL *mdctSpectrum,
                       INT globalGain, INT *scalefactors,
-                      SHORT *quantizedSpectrum);
+                      SHORT *quantizedSpectrum,
+                      INT dZoneQuantEnable);
 
 FIXP_DBL FDKaacEnc_calcSfbDist(FIXP_DBL *mdctSpectrum,
                      SHORT *quantSpectrum,
                      INT noOfLines,
-                     INT gain);
+                     INT gain,
+                     INT dZoneQuantEnable);
 
 void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum,
                                SHORT   *quantSpectrum,

libAACenc/src/sf_estim.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -363,7 +363,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
                       INT scf,
                       INT minScf,
                       FIXP_DBL  *distLdData,
-                      INT *minScfCalculated
+                      INT *minScfCalculated,
+                      INT dZoneQuantEnable
                       )
 {
    FIXP_DBL sfbDistLdData;
@@ -375,7 +376,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
    sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
                                          quantSpec,
                                          sfbWidth,
-                                         scf);
+                                         scf,
+                                         dZoneQuantEnable);
    *minScfCalculated = scf;
    /* nmr > 1.25 -> try to improve nmr */
    if (sfbDistLdData > (threshLdData-distFactorLdData)) {
@@ -390,7 +392,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
          sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
                                                quantSpecTmp,
                                                sfbWidth,
-                                               scf);
+                                               scf,
+                                               dZoneQuantEnable);
 
          if (sfbDistLdData < sfbDistBestLdData) {
             scfBest = scf;
@@ -408,7 +411,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
          sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
                                                quantSpecTmp,
                                                sfbWidth,
-                                               scf);
+                                               scf,
+                                               dZoneQuantEnable);
 
          if (sfbDistLdData < sfbDistBestLdData) {
             scfBest = scf;
@@ -429,7 +433,8 @@ static INT FDKaacEnc_improveScf(FIXP_DBL *spec,
          sfbDistLdData = FDKaacEnc_calcSfbDist(spec,
                                                quantSpecTmp,
                                                sfbWidth,
-                                               scf);
+                                               scf,
+                                               dZoneQuantEnable);
 
          if (sfbDistLdData < sfbDistAllowedLdData) {
            *minScfCalculated = scfBest+1;
@@ -454,6 +459,7 @@ static void FDKaacEnc_assimilateSingleScf(PSY_OUT_CHANNEL *psyOutChan,
                                 QC_OUT_CHANNEL   *qcOutChannel,
                                 SHORT *quantSpec,
                                 SHORT *quantSpecTmp,
+                                INT dZoneQuantEnable,
                                 INT *scf,
                                 INT *minScf,
                                 FIXP_DBL *sfbDist,
@@ -570,7 +576,8 @@ static void FDKaacEnc_assimilateSingleScf(PSY_OUT_CHANNEL *psyOutChan,
             sfbDistNew = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs,
                                                quantSpecTmp+sfbOffs,
                                                sfbWidth,
-                                               scfAct);
+                                               scfAct,
+                                               dZoneQuantEnable);
 
             if (sfbDistNew < sfbDist[sfbAct]) {
               /* success, replace scf by new one */
@@ -629,6 +636,7 @@ static void FDKaacEnc_assimilateMultipleScf(PSY_OUT_CHANNEL *psyOutChan,
                                   QC_OUT_CHANNEL  *qcOutChannel,
                                   SHORT *quantSpec,
                                   SHORT *quantSpecTmp,
+                                  INT dZoneQuantEnable,
                                   INT *scf,
                                   INT *minScf,
                                   FIXP_DBL *sfbDist,
@@ -724,7 +732,8 @@ static void FDKaacEnc_assimilateMultipleScf(PSY_OUT_CHANNEL *psyOutChan,
                 sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs,
                                               quantSpecTmp+sfbOffs,
                                               sfbWidth,
-                                              scfAct);
+                                              scfAct,
+                                              dZoneQuantEnable);
 
                 if (sfbDistNew[sfb] >qcOutChannel->sfbThresholdLdData[sfb]) {
                   /* no improvement, skip further dist. calculations */
@@ -768,6 +777,7 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
                                    QC_OUT_CHANNEL  *qcOutChannel,
                                    SHORT *quantSpec,
                                    SHORT *quantSpecTmp,
+                                   INT dZoneQuantEnable,
                                    INT *scf,
                                    INT *minScf,
                                    FIXP_DBL *sfbDist,
@@ -883,7 +893,8 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
                 sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs[sfb],
                                               quantSpecTmp+sfbOffs[sfb],
                                               sfbOffs[sfb+1]-sfbOffs[sfb],
-                                              scfNew);
+                                              scfNew,
+                                              dZoneQuantEnable);
 
                 if (sfbDistNew[sfb] > sfbDistMax[sfb]) {
                   /* no improvement, skip further dist. calculations */
@@ -963,7 +974,8 @@ static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(PSY_OUT_CHANNEL *psyOutCh
               sfbDistNew[sfb] = FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+sfbOffs[sfb],
                                                       quantSpecTmp+sfbOffs[sfb],
                                                       sfbOffs[sfb+1]-sfbOffs[sfb],
-                                                      scfNew);
+                                                      scfNew,
+                                                      dZoneQuantEnable);
 
               if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) {
                 /* no improvement, skip further dist. calculations */
@@ -1058,7 +1070,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL   *qcOutChannel,
                             INT *RESTRICT globalGain,
                             FIXP_DBL *RESTRICT sfbFormFactorLdData
                             ,const INT invQuant,
-                            SHORT *RESTRICT quantSpec
+                            SHORT *RESTRICT quantSpec,
+                            const INT dZoneQuantEnable
                             )
 {
   INT i, j, sfb, sfbOffs;
@@ -1160,7 +1173,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL   *qcOutChannel,
                               quantSpecTmp+psyOutChannel->sfbOffsets[sfbOffs+sfb],
                               psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb],
                               threshLdData, scfInt, minSfMaxQuant[sfbOffs+sfb],
-                              &sfbDistLdData[sfbOffs+sfb], &minScfCalculated[sfbOffs+sfb]
+                              &sfbDistLdData[sfbOffs+sfb], &minScfCalculated[sfbOffs+sfb],
+                              dZoneQuantEnable
                               );
         }
         scf[sfbOffs+sfb] = scfInt;
@@ -1187,20 +1201,32 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL   *qcOutChannel,
                           sfbNRelevantLines);
 
 
-    FDKaacEnc_assimilateSingleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
+    FDKaacEnc_assimilateSingleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
+                        dZoneQuantEnable,
+                        scf,
                         minSfMaxQuant, sfbDistLdData, sfbConstPePart,
                         sfbFormFactorLdData, sfbNRelevantLines, minScfCalculated, 1);
 
+    if(invQuant > 1) {
+      FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
+                        dZoneQuantEnable,
+                        scf,
+                        minSfMaxQuant, sfbDistLdData, sfbConstPePart,
+                        sfbFormFactorLdData, sfbNRelevantLines);
 
-    FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
-                          minSfMaxQuant, sfbDistLdData, sfbConstPePart,
-                          sfbFormFactorLdData, sfbNRelevantLines);
+      FDKaacEnc_assimilateMultipleScf(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
+                        dZoneQuantEnable,
+                        scf,
+                        minSfMaxQuant, sfbDistLdData, sfbConstPePart,
+                        sfbFormFactorLdData, sfbNRelevantLines);
 
 
-    FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, scf,
-                           minSfMaxQuant, sfbDistLdData, sfbConstPePart,
-                           sfbFormFactorLdData, sfbNRelevantLines);
-
+      FDKaacEnc_FDKaacEnc_assimilateMultipleScf2(psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp,
+                        dZoneQuantEnable,
+                        scf,
+                        minSfMaxQuant, sfbDistLdData, sfbConstPePart,
+                        sfbFormFactorLdData, sfbNRelevantLines);
+    }
   }
 
 
@@ -1223,7 +1249,8 @@ FDKaacEnc_FDKaacEnc_EstimateScaleFactorsChannel(QC_OUT_CHANNEL   *qcOutChannel,
                FDKaacEnc_calcSfbDist(qcOutChannel->mdctSpectrum+psyOutChannel->sfbOffsets[sfbOffs+sfb],
                                      quantSpec+psyOutChannel->sfbOffsets[sfbOffs+sfb],
                                      psyOutChannel->sfbOffsets[sfbOffs+sfb+1]-psyOutChannel->sfbOffsets[sfbOffs+sfb],
-                                     scf[sfbOffs+sfb]
+                                     scf[sfbOffs+sfb],
+                                     dZoneQuantEnable
                                      );
         }
       }
@@ -1281,6 +1308,7 @@ void
 FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
                      QC_OUT_CHANNEL* qcOutChannel[],
                      const int invQuant,
+                     const INT dZoneQuantEnable,
                      const int nChannels)
 {
   int ch;
@@ -1293,7 +1321,8 @@ FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
                                   &qcOutChannel[ch]->globalGain,
                                   qcOutChannel[ch]->sfbFormFactorLdData
                                   ,invQuant,
-                                  qcOutChannel[ch]->quantSpec
+                                  qcOutChannel[ch]->quantSpec,
+                                  dZoneQuantEnable
                                   );
   }
 

libAACenc/src/sf_estim.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -110,6 +110,7 @@ void
 FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[],
                      QC_OUT_CHANNEL* qcOutChannel[],
                      const int invQuant,
+                     const INT dZoneQuantEnable,
                      const int nChannels);
 
 

libAACenc/src/tns_func.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -100,6 +100,7 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitrate,
                          INT channels,
                          INT blocktype,
                          INT granuleLength,
+                         INT isLowDelay,
                          INT ldSbrPresent,
                          TNS_CONFIG *tnsConfig,
                          PSY_CONFIGURATION *psyConfig,

libFDK/include/FDK_archdef.h

@@ -99,7 +99,7 @@ amm-info@iis.fraunhofer.de
 /* Take action against VisualStudio 2005 crosscompile problems. */
 
 /* Use single macro (the GCC built in macro) for architecture identification independent of the particular toolchain */
-#if defined(__i386__) || defined(__i486__) || defined(__i586__) || defined(__i686__) || (defined(_MSC_VER) && defined(_M_IX86)) || defined (__x86_64__)
+#if defined(__i386__) || defined(__i486__) || defined(__i586__) || defined(__i686__) || (defined(_MSC_VER) && defined(_M_IX86)) || defined (__x86_64__) || (defined(_MSC_VER) && defined(_M_X64))
 #define __x86__
 #endif
 
@@ -154,7 +154,6 @@ amm-info@iis.fraunhofer.de
 #endif
 
 #ifdef _M_ARM
-#include "cmnintrin.h"
 #include "armintr.h"
 #endif
 
@@ -198,6 +197,14 @@ amm-info@iis.fraunhofer.de
 #undef POW2COEFF_16BIT
 #undef LDCOEFF_16BIT
 
+#elif defined(__aarch64__) || defined(__AARCH64EL__)
+#define ARCH_PREFER_MULT_32x32
+#define ARCH_PREFER_MULT_32x16
+#define SINETABLE_16BIT
+#define POW2COEFF_16BIT
+#define LDCOEFF_16BIT
+#define WINDOWTABLE_16BIT
+
 #elif defined(__x86__)	/* cppp replaced: elif */
 #define ARCH_PREFER_MULT_32x16
 #define SINETABLE_16BIT
@@ -215,7 +222,7 @@ amm-info@iis.fraunhofer.de
 
 #else
 
-  #error  >>>> Please set architecture characterization defines for your platform (FDK_HIGH_PERFORMANCE)! <<<<
+  #warning  >>>> Please set architecture characterization defines for your platform (FDK_HIGH_PERFORMANCE)! <<<<
 
 #endif /* Architecture switches */
 

libFDK/include/aarch64/clz_aarch64.h

@@ -81,13 +81,42 @@ www.iis.fraunhofer.de/amm
 amm-info@iis.fraunhofer.de
 ----------------------------------------------------------------------------------------------------------- */
 
-/***************************  MPEG AAC Audio Encoder  *************************
+/***************************  Fraunhofer IIS FDK Tools  **********************
 
-   Initial author:       R. Boehm
-   contents/description: huffman codeword reordering
-                         based on source from aacErrRobTrans
+   Author(s):
+   Description: fixed point intrinsics
 
 ******************************************************************************/
 
-#include "aacenc_hcr.h"
+#if defined(__aarch64__) || defined(__AARCH64EL__)
+
+#if defined(__GNUC__)
+  /* aarch64 gcc*/
+
+  #define FUNCTION_fixnormz_D
+  #define FUNCTION_fixnorm_D
+
+  inline INT fixnormz_D(LONG value)
+  {
+    INT result;
+    asm("clz %w0, %w1 ": "=r"(result) : "r"(value) );
+    return result;
+  }
+
+  inline INT fixnorm_D(LONG value)
+  {
+    INT result;
+    if (value == 0) {
+      return 0;
+    }
+    if (value < 0) {
+      value = ~value;
+    }
+    result =  fixnormz_D(value);
+    return result - 1;
+  }
+
+#endif /* aarch64 toolchain */
+
+#endif /* __aarch64__ */
 

libFDK/include/aarch64/fixmul_aarch64.h

@@ -81,16 +81,33 @@ www.iis.fraunhofer.de/amm
 amm-info@iis.fraunhofer.de
 ----------------------------------------------------------------------------------------------------------- */
 
-/***************************  MPEG AAC Audio Encoder  *************************
+/***************************  Fraunhofer IIS FDK Tools  **********************
 
-   Initial author:       R. Boehm
-   contents/description: huffman codeword reordering
-                         based on source from aacErrRobTrans
+   Author(s):
+   Description: fixed point intrinsics
 
 ******************************************************************************/
 
-#ifndef _AACENC_HCR
-#define _AACENC_HCR_H
+#if defined(__aarch64__) || defined(__AARCH64EL__)
 
+#if defined(__GNUC__)	/* cppp replaced: elif */
+/* ARM with GNU compiler */
+
+#define FUNCTION_fixmuldiv2_DD
+
+#define FUNCTION_fixmuldiv2BitExact_DD
+#define fixmuldiv2BitExact_DD(a,b) fixmuldiv2_DD(a,b)
+#define FUNCTION_fixmulBitExact_DD
+#define fixmulBitExact_DD(a,b) fixmul_DD(a,b)
+
+inline INT fixmuldiv2_DD (const INT a, const INT b)
+{
+  INT result ;
+  result = ((long long)a * b)>>32;
+  return result ;
+}
+
+#endif /* defined(__GNUC__) */
+
+#endif /* __aarch64__ */
 
-#endif  /* ifndef _AACENC_HCR */

libFDK/include/clz.h

@@ -97,6 +97,9 @@ amm-info@iis.fraunhofer.de
 #if defined(__arm__)
 #include "arm/clz_arm.h"
 
+#elif defined(__aarch64__) || defined(__AARCH64EL__)
+#include "aarch64/clz_aarch64.h"
+
 #elif defined(__mips__)	/* cppp replaced: elif */
 #include "mips/clz_mips.h"
 

libFDK/include/cplx_mul.h

@@ -96,7 +96,7 @@ amm-info@iis.fraunhofer.de
 #if defined(__CC_ARM) || defined(__arm__) || defined(_M_ARM)	/* cppp replaced: elif */
 #include "arm/cplx_mul.h"
 
-#elif defined(__GNUC__) && defined(__mips__)	/* cppp replaced: elif */
+#elif defined(__GNUC__) && defined(__mips__) && __mips_isa_rev < 6
 #include "mips/cplx_mul.h"
 
 #endif /* #if defined all cores: bfin, arm, etc. */

libFDK/include/fft.h

@@ -121,7 +121,7 @@ void ifft(int length, FIXP_DBL *pInput, INT *scalefactor);
  */
 
 LNK_SECTION_CODE_L1
-static void FORCEINLINE fft_4(FIXP_DBL *x)
+static FORCEINLINE void fft_4(FIXP_DBL *x)
 {
     FIXP_DBL a00, a10, a20, a30, tmp0, tmp1;
 
@@ -149,7 +149,7 @@ static void FORCEINLINE fft_4(FIXP_DBL *x)
 }
 
 LNK_SECTION_CODE_L1
-static void FORCEINLINE fft_8(FIXP_DBL *x)
+static FORCEINLINE void fft_8(FIXP_DBL *x)
 {
       #define W_PiFOURTH STC(0x5a82799a)
 

libFDK/include/fixmul.h

@@ -98,6 +98,9 @@ amm-info@iis.fraunhofer.de
 #if defined(__arm__)
 #include "arm/fixmul_arm.h"
 
+#elif defined(__aarch64__) || defined(__AARCH64EL__)
+#include "aarch64/fixmul_aarch64.h"
+
 #elif defined(__mips__)	/* cppp replaced: elif */
 #include "mips/fixmul_mips.h"
 

libFDK/include/fixpoint_math.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -94,6 +94,35 @@ amm-info@iis.fraunhofer.de
 
 #include "common_fix.h"
 
+#if !defined(FUNCTION_fIsLessThan)
+/**
+ * \brief Compares two fixpoint values incl. scaling.
+ * \param a_m mantissa of the first input value.
+ * \param a_e exponent of the first input value.
+ * \param b_m mantissa of the second input value.
+ * \param b_e exponent of the second input value.
+ * \return non-zero if (a_m*2^a_e) < (b_m*2^b_e), 0 otherwise
+ */
+FDK_INLINE INT fIsLessThan(FIXP_DBL a_m, INT a_e, FIXP_DBL b_m, INT b_e)
+{
+  if (a_e > b_e) {
+    return (b_m >> fMin(a_e-b_e, DFRACT_BITS-1) > a_m);
+  } else {
+    return (a_m >> fMin(b_e-a_e, DFRACT_BITS-1) < b_m);
+  }
+}
+
+FDK_INLINE INT fIsLessThan(FIXP_SGL a_m, INT a_e, FIXP_SGL b_m, INT b_e)
+{
+  if (a_e > b_e) {
+    return (b_m >> fMin(a_e-b_e, FRACT_BITS-1) > a_m);
+  } else {
+    return (a_m >> fMin(b_e-a_e, FRACT_BITS-1) < b_m);
+  }
+}
+#endif
+
+
 
 #define LD_DATA_SCALING (64.0f)
 #define LD_DATA_SHIFT   6   /* pow(2, LD_DATA_SHIFT) = LD_DATA_SCALING */
@@ -438,11 +467,11 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
 
 /*****************************************************************************
 
- array for 1/n, n=1..50
+ array for 1/n, n=1..80
 
 ****************************************************************************/
 
-  extern const FIXP_DBL invCount[50];
+  extern const FIXP_DBL invCount[80];
 
   LNK_SECTION_INITCODE
   inline void InitInvInt(void) {}
@@ -450,14 +479,14 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
 
 /**
  * \brief Calculate the value of 1/i where i is a integer value. It supports
- *        input values from 1 upto 50.
+ *        input values from 1 upto 80.
  * \param intValue Integer input value.
  * \param FIXP_DBL representation of 1/intValue
  */
 inline FIXP_DBL GetInvInt(int intValue)
 {
-  FDK_ASSERT((intValue > 0) && (intValue < 50));
-  FDK_ASSERT(intValue<50);
+  FDK_ASSERT((intValue > 0) && (intValue < 80));
+  FDK_ASSERT(intValue<80);
 	return invCount[intValue];
 }
 

libFDK/include/mips/cplx_mul.h

@@ -107,24 +107,8 @@ inline void cplxMultDiv2( FIXP_DBL *c_Re,
                           FIXP_DBL  b_Re,
                           FIXP_DBL  b_Im)
 {
-  INT result;
-
-   __asm__ ("mult %[a_Re], %[b_Re];\n"
-            "msub %[a_Im], %[b_Im];\n"
-            "mfhi %[result];\n"
-       : [result]"=r"(result)
-       : [a_Re]"d"(a_Re), [b_Re]"d"(b_Re),  [a_Im]"d"(a_Im), [b_Im]"d"(b_Im)
-       : "lo");
-
-   *c_Re = result;
-
-   __asm__ ("mult %[a_Re], %[b_Im];\n"
-            "madd %[a_Im], %[b_Re];\n"
-            "mfhi %[result];\n"
-       : [result]"=r"(result) 
-       : [a_Re]"r"(a_Re), [b_Im]"r"(b_Im), [a_Im]"r"(a_Im), [b_Re]"r"(b_Re)
-       : "lo");
-   *c_Im = result;
+  *c_Re = (((long long)a_Re * (long long)b_Re) - ((long long)a_Im * (long long)b_Im))>>32;
+  *c_Im = (((long long)a_Re * (long long)b_Im) + ((long long)a_Im * (long long)b_Re))>>32;
 }
 #endif
 
@@ -136,25 +120,8 @@ inline void cplxMult( FIXP_DBL *c_Re,
                       FIXP_DBL  b_Re,
                       FIXP_DBL  b_Im)
 {
-  INT result;
-
-   __asm__ ("mult %[a_Re], %[b_Re];\n"
-            "msub %[a_Im], %[b_Im];\n"
-            "mfhi %[result];\n"
-            //"extr_w %[result], 31;\n"
-        : [result]"=r"(result)
-        : [a_Re]"r"(a_Re), [b_Re]"r"(b_Re), [a_Im]"r"(a_Im), [b_Im]"r"(b_Im)
-        : "lo");
-   *c_Re = result<<1;
-
-   __asm__ ("mult %[a_Re], %[b_Im];\n"
-            "madd %[a_Im], %[b_Re];\n"
-            "mfhi %[result];\n"
-            //"extr_w %[result], 31;\n"
-        : [result]"=r"(result)
-        : [a_Re]"r"(a_Re), [b_Im]"r"(b_Im), [a_Im]"r"(a_Im), [b_Re]"r"(b_Re)
-        : "lo");
-   *c_Im = result<<1;
+  *c_Re = ((((long long)a_Re * (long long)b_Re) - ((long long)a_Im * (long long)b_Im))>>32)<<1;
+  *c_Im = ((((long long)a_Re * (long long)b_Im) + ((long long)a_Im * (long long)b_Re))>>32)<<1;
 }
 #endif
 

libFDK/src/FDK_core.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -93,10 +93,15 @@ amm-info@iis.fraunhofer.de
 /* FDK tools library info */
 #define FDK_TOOLS_LIB_VL0 2
 #define FDK_TOOLS_LIB_VL1 3
-#define FDK_TOOLS_LIB_VL2 2
+#define FDK_TOOLS_LIB_VL2 6
 #define FDK_TOOLS_LIB_TITLE "FDK Tools"
+#ifdef __ANDROID__
+#define FDK_TOOLS_LIB_BUILD_DATE ""
+#define FDK_TOOLS_LIB_BUILD_TIME ""
+#else
 #define FDK_TOOLS_LIB_BUILD_DATE __DATE__
 #define FDK_TOOLS_LIB_BUILD_TIME __TIME__
+#endif
 
 int FDK_toolsGetLibInfo(LIB_INFO *info)
 {

libFDK/src/FDK_tools_rom.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1260,9 +1260,9 @@ const FIXP_WTP * FDKgetWindowSlope(int length, int shape)
 #define QTCFL(x)  FL2FXCONST_SGL(x)
 #define QTC(x)    FX_DBL2FXCONST_SGL(x)
 #else
-#define QFC(x) (x)
+#define QFC(x) ((FIXP_DBL)(x))
 #define QTCFL(x)  FL2FXCONST_DBL(x)
-#define QTC(x)  (x)
+#define QTC(x)  ((FIXP_DBL)(x))
 #endif /* ARCH_PREFER_MULT_32x16 */
 
 #ifndef LOW_POWER_SBR_ONLY
@@ -1902,7 +1902,7 @@ const USHORT sqrt_tab[49]={
 0xb504};
 
 LNK_SECTION_CONSTDATA_L1
-const FIXP_DBL invCount[50]=  /* This could be 16-bit wide */
+const FIXP_DBL invCount[80]=  /* This could be 16-bit wide */
 {
     0x00000000, 0x7fffffff, 0x40000000, 0x2aaaaaab, 0x20000000,
     0x1999999a, 0x15555555, 0x12492492, 0x10000000, 0x0e38e38e,
@@ -1913,7 +1913,13 @@ const FIXP_DBL invCount[50]=  /* This could be 16-bit wide */
     0x04444444, 0x04210842, 0x04000000, 0x03e0f83e, 0x03c3c3c4,
     0x03a83a84, 0x038e38e4, 0x03759f23, 0x035e50d8, 0x03483483,
     0x03333333, 0x031f3832, 0x030c30c3, 0x02fa0be8, 0x02e8ba2f,
-    0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15
+    0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15,
+    0x028f5c29, 0x02828283, 0x02762762, 0x026a439f, 0x025ed098,
+    0x0253c825, 0x02492492, 0x023ee090, 0x0234f72c, 0x022b63cc,
+    0x02222222, 0x02192e2a, 0x02108421, 0x02082082, 0x02000000,
+    0x01f81f82, 0x01f07c1f, 0x01e9131b, 0x01e1e1e2, 0x01dae607,
+    0x01d41d42, 0x01cd8569, 0x01c71c72, 0x01c0e070, 0x01bacf91,
+    0x01b4e81b, 0x01af286c, 0x01a98ef6, 0x01a41a42, 0x019ec8e9
 };
 
 
@@ -2033,19 +2039,6 @@ static const element_list_t node_aac_cpe = {
   { &node_aac_cpe0, &node_aac_cpe1 }
 };
 
-#define el_mpegsres_sce &el_aac_sce[2]
-
-static const element_list_t node_mpegsres_sce = {
-  el_mpegsres_sce,
-  { NULL, NULL }
-};
-
-static const element_list_t node_mpegsres_cpe = {
-  el_aac_cpe1,
-  { NULL, NULL }
-};
-
-
 /*
  * AOT C- {17,23}
  * epConfig = 0,1
@@ -2243,7 +2236,7 @@ static const rbd_id_t el_aac_cpe1_epc1[] = {
   ics_info,
   ms,
   ltp_data_present,
-  ltp_data,
+  /* ltp_data, */
   global_gain,
   section_data,
   scale_factor_data,
@@ -2254,7 +2247,7 @@ static const rbd_id_t el_aac_cpe1_epc1[] = {
   next_channel,
 
   ltp_data_present,
-  ltp_data,
+  /* ltp_data, */
   global_gain,
   section_data,
   scale_factor_data,
@@ -2297,7 +2290,178 @@ static const element_list_t node_aac_cpe_epc1 = {
   { &node_aac_cpe0_epc1, &node_aac_cpe1_epc1 }
 };
 
+/*
+ * AOT = 20
+ * epConfig = 0
+ */
+static const rbd_id_t el_scal_sce_epc0[] = {
+  ics_info,            /* ESC 1 */
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  esc2_rvlc,           /* ESC 2 */
+  tns_data,            /* ESC 3 */
+  spectral_data,       /* ESC 4 */
+  end_of_sequence
+};
 
+static const struct element_list node_scal_sce_epc0 = {
+  el_scal_sce_epc0,
+  { NULL, NULL }
+};
+
+static const rbd_id_t el_scal_cpe_epc0[] = {
+  ics_info,            /* ESC 0 */
+  ms,
+  tns_data_present,    /* ESC 1 (ch 0) */
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  esc2_rvlc,           /* ESC 2 (ch 0) */
+  tns_data,            /* ESC 3 (ch 0) */
+  spectral_data,       /* ESC 4 (ch 0) */
+  next_channel,
+  tns_data_present,    /* ESC 1 (ch 1) */
+  ltp_data_present,
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  esc2_rvlc,           /* ESC 2 (ch 1) */
+  tns_data,            /* ESC 3 (ch 1) */
+  spectral_data,       /* ESC 4 (ch 1) */
+  end_of_sequence
+};
+
+static const struct element_list node_scal_cpe_epc0 = {
+  el_scal_cpe_epc0,
+  { NULL, NULL }
+};
+
+/*
+ * AOT = 20
+ * epConfig = 1
+ */
+static const rbd_id_t el_scal_sce_epc1[] = {
+  ics_info,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  tns_data,
+  spectral_data,
+  end_of_sequence
+};
+
+static const struct element_list node_scal_sce_epc1 = {
+  el_scal_sce_epc1,
+  { NULL, NULL }
+};
+
+static const rbd_id_t el_scal_cpe_epc1[] = {
+  ics_info,
+  ms,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  next_channel,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  next_channel,
+  tns_data,
+  next_channel,
+  tns_data,
+  next_channel,
+  spectral_data,
+  next_channel,
+  spectral_data,
+  end_of_sequence
+};
+
+static const struct element_list node_scal_cpe_epc1 = {
+  el_scal_cpe_epc1,
+  { NULL, NULL }
+};
+
+/*
+ * Pseudo AOT for DRM/DRM+ (similar to AOT 20)
+ * Derived from epConfig = 1
+ */
+static const rbd_id_t el_drm_sce[] = {
+  drmcrc_start_reg,
+  ics_info,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  tns_data,
+  drmcrc_end_reg,
+  spectral_data,
+  end_of_sequence
+};
+
+static const struct element_list node_drm_sce = {
+  el_drm_sce,
+  { NULL, NULL }
+};
+
+static const rbd_id_t el_drm_cpe[] = {
+  drmcrc_start_reg,
+  ics_info,
+  ms,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  next_channel,
+  tns_data_present,
+  ltp_data_present,
+  /* ltp_data, */
+  global_gain,
+  section_data,
+  scale_factor_data,
+  esc1_hcr,
+  next_channel,
+  tns_data,
+  next_channel,
+  tns_data,
+  drmcrc_end_reg,
+  next_channel,
+  spectral_data,
+  next_channel,
+  spectral_data,
+  end_of_sequence
+};
+
+static const struct element_list node_drm_cpe = {
+  el_drm_cpe,
+  { NULL, NULL }
+};
 
 /*
  * AOT = 39
@@ -2412,6 +2576,19 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
           return &node_aac_cpe_epc1;
       }
       break;
+    case AOT_ER_AAC_SCAL:
+      if (nChannels == 1) {
+        if (epConfig <= 0)
+          return &node_scal_sce_epc0;
+        else
+          return &node_scal_sce_epc1;
+      } else {
+        if (epConfig <= 0)
+          return &node_scal_cpe_epc0;
+        else
+          return &node_scal_cpe_epc1;
+      }
+      break;
     case AOT_ER_AAC_ELD:
       if (nChannels == 1) {
         if (epConfig <= 0)
@@ -2424,11 +2601,14 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
         else
           return &node_eld_cpe_epc1;
       }
-    case AOT_MPEGS_RESIDUALS:
+    case AOT_DRM_AAC:
+    case AOT_DRM_SBR:
+    case AOT_DRM_MPEG_PS:
+      FDK_ASSERT(epConfig == 1);
       if (nChannels == 1) {
-        return &node_mpegsres_sce;
+        return &node_drm_sce;
       } else {
-        return &node_mpegsres_cpe;
+        return &node_drm_cpe;
       }
       break;
     default:

libFDK/src/arm/qmf_arm.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -482,42 +482,42 @@ static void qmfSynPrototypeFirSlot1_filter(FIXP_QMF *RESTRICT realSlot,
      B = p_flt[4];                        /* Bottom=[8] Top=[9]     */
      A = p_fltm[3];                       /* Bottom=[316] Top=[317] */
      sta0 = sta[0];                       /* save state[0]          */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=9...........319  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=316...........6  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=8,18,    ...318  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=9...........319  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=316...........6  */
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=8,18,    ...318  */
      B = p_flt[3];                        /* Bottom=[6] Top=[7]     */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=317...........7  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=317...........7  */
      A = p_fltm[4];                       /* Bottom=[318] Top=[319] */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=7...........317  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=318...........8  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=6...........316  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=7...........317  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=318...........8  */
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=6...........316  */
      B = p_flt[2];                        /* Bottom=[X] Top=[5]     */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=9...........319  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=9...........319  */
      A = p_fltm[2];                       /* Bottom=[X] Top=[315]   */
-     *sta++ =         SMULWT( imag, B );  /* index=5,15, ...   315  */
+     sta[0] =         SMULWT( imag, B ); sta++; /* index=5,15, ...   315  */
      result = SMLAWT( sta0,   real, A );  /* index=315...........5  */
 
-     *pMyTimeOut++ = result;
+     pMyTimeOut[0] = result; pMyTimeOut++;
 
      real = *--realSlot;
      imag = *--imagSlot;
      A = p_fltm[0];                       /* Bottom=[310] Top=[311] */
      B = p_flt[7];                        /* Bottom=[14]  Top=[15]  */
      result = SMLAWB( sta[0], real, A );  /* index=310...........0  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=14..........324  */
-     *pMyTimeOut++ = result;
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=14..........324  */
+     pMyTimeOut[0] = result; pMyTimeOut++;
      B = p_flt[6];                        /* Bottom=[12]  Top=[13]  */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=311...........1  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=311...........1  */
      A = p_fltm[1];                       /* Bottom=[312] Top=[313] */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=13..........323  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=312...........2  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=12..........322  */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=313...........3  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=13..........323  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=312...........2  */
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=12..........322  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=313...........3  */
      A = p_fltm[2];                       /* Bottom=[314] Top=[315] */
      B = p_flt[5];                        /* Bottom=[10]  Top=[11]  */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=11..........321  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=314...........4  */
-     *sta++ =         SMULWB( imag, B );  /* index=10..........320  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=11..........321  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=314...........4  */
+     sta[0] =         SMULWB( imag, B ); sta++; /* index=10..........320  */
 
 
      p_flt    += 5;
@@ -566,21 +566,21 @@ INT qmfSynPrototypeFirSlot2(
      A = p_fltm[0];                       /* Bottom=[310] Top=[311] */
      B = p_flt[7];                        /* Bottom=[14]  Top=[15]  */
      result = SMLAWB( sta[0], real, A );  /* index=310...........0  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=14..........324  */
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=14..........324  */
      B = p_flt[6];                        /* Bottom=[12]  Top=[13]  */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=311...........1  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=311...........1  */
      A = p_fltm[1];                       /* Bottom=[312] Top=[313] */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=13..........323  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=312...........2  */
-     *sta++ = SMLAWB( sta[1], imag, B );  /* index=12..........322  */
-     *sta++ = SMLAWT( sta[1], real, A );  /* index=313...........3  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=13..........323  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=312...........2  */
+     sta[0] = SMLAWB( sta[1], imag, B ); sta++; /* index=12..........322  */
+     sta[0] = SMLAWT( sta[1], real, A ); sta++; /* index=313...........3  */
      A = p_fltm[2];                       /* Bottom=[314] Top=[315] */
      B = p_flt[5];                        /* Bottom=[10]  Top=[11]  */
-     *sta++ = SMLAWT( sta[1], imag, B );  /* index=11..........321  */
-     *sta++ = SMLAWB( sta[1], real, A );  /* index=314...........4  */
-     *sta++ =         SMULWB( imag, B );  /* index=10..........320  */
+     sta[0] = SMLAWT( sta[1], imag, B ); sta++; /* index=11..........321  */
+     sta[0] = SMLAWB( sta[1], real, A ); sta++; /* index=314...........4  */
+     sta[0] =         SMULWB( imag, B ); sta++; /* index=10..........320  */
 
-     *pMyTimeOut++ = result;
+     pMyTimeOut[0] = result; pMyTimeOut++;
 
      p_fltm   -= 5;
      p_flt    += 5;
@@ -610,8 +610,8 @@ INT qmfSynPrototypeFirSlot2(
     {
       FIXP_DBL result1, result2;
 
-      result1 = *pMyTimeOut++;
-      result2 = *pMyTimeOut++;
+      result1 = pMyTimeOut[0]; pMyTimeOut++;
+      result2 = pMyTimeOut[0]; pMyTimeOut++;
 
       result1 = fMult(result1,gain);
       timeOut -= stride;
@@ -635,8 +635,8 @@ INT qmfSynPrototypeFirSlot2(
       timeOut[0] = result2 << scale;
 #endif
 
-      result1 = *pMyTimeOut++;
-      result2 = *pMyTimeOut++;
+      result1 = pMyTimeOut[0]; pMyTimeOut++;
+      result2 = pMyTimeOut[0]; pMyTimeOut++;
 
       result1 = fMult(result1,gain);
       timeOut -= stride;
@@ -666,8 +666,8 @@ INT qmfSynPrototypeFirSlot2(
     for (no_channels>>=2; no_channels--;)
     {
       FIXP_DBL result1, result2;
-      result1 = *pMyTimeOut++;
-      result2 = *pMyTimeOut++;
+      result1 = pMyTimeOut[0]; pMyTimeOut++;
+      result2 = pMyTimeOut[0]; pMyTimeOut++;
       timeOut -= stride;
       if (result1 < 0)        result1 += add_neg;
       if (result1 < max_neg)  result1 = max_neg;
@@ -688,8 +688,8 @@ INT qmfSynPrototypeFirSlot2(
       timeOut[0] = result2 << scale;
 #endif
       
-      result1 = *pMyTimeOut++;
-      result2 = *pMyTimeOut++;
+      result1 = pMyTimeOut[0]; pMyTimeOut++;
+      result2 = pMyTimeOut[0]; pMyTimeOut++;
       timeOut -= stride;
       if (result1 < 0)        result1 += add_neg;
       if (result1 < max_neg)  result1 = max_neg;

libFDK/src/dct.cpp

@@ -324,12 +324,12 @@ void dct_IV(FIXP_DBL *pDat,
   {
     FIXP_DBL *RESTRICT pDat_0 = &pDat[0];
     FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2];
-    register int i;
+    int i;
 
     /* 29 cycles on ARM926 */
     for (i = 0; i < M-1; i+=2,pDat_0+=2,pDat_1-=2)
     {
-      register FIXP_DBL accu1,accu2,accu3,accu4;
+      FIXP_DBL accu1,accu2,accu3,accu4;
 
       accu1 = pDat_1[1]; accu2 = pDat_0[0];
       accu3 = pDat_0[1]; accu4 = pDat_1[0];
@@ -342,7 +342,7 @@ void dct_IV(FIXP_DBL *pDat,
     }
     if (M&1)
     {
-      register FIXP_DBL accu1,accu2;
+      FIXP_DBL accu1,accu2;
 
       accu1 = pDat_1[1]; accu2 = pDat_0[0];
 
@@ -363,7 +363,7 @@ void dct_IV(FIXP_DBL *pDat,
   {
     FIXP_DBL *RESTRICT pDat_0 = &pDat[0];
     FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2];
-    register FIXP_DBL accu1,accu2,accu3,accu4;
+    FIXP_DBL accu1,accu2,accu3,accu4;
     int idx, i;
 
     /* Sin and Cos values are 0.0f and 1.0f */
@@ -450,12 +450,12 @@ void dst_IV(FIXP_DBL *pDat,
     FIXP_DBL *RESTRICT pDat_0 = &pDat[0];
     FIXP_DBL *RESTRICT pDat_1 = &pDat[L - 2];
 
-    register int i;
+    int i;
 
     /* 34 cycles on ARM926 */
     for (i = 0; i < M-1; i+=2,pDat_0+=2,pDat_1-=2)
     {
-      register FIXP_DBL accu1,accu2,accu3,accu4;
+      FIXP_DBL accu1,accu2,accu3,accu4;
 
       accu1 =  pDat_1[1]; accu2 = -pDat_0[0];
       accu3 =  pDat_0[1]; accu4 = -pDat_1[0];
@@ -468,7 +468,7 @@ void dst_IV(FIXP_DBL *pDat,
     }
     if (M&1)
     {
-      register FIXP_DBL accu1,accu2;
+      FIXP_DBL accu1,accu2;
 
       accu1 =  pDat_1[1]; accu2 = -pDat_0[0];
 
@@ -488,7 +488,7 @@ void dst_IV(FIXP_DBL *pDat,
   {
     FIXP_DBL *RESTRICT pDat_0;
     FIXP_DBL *RESTRICT pDat_1;
-    register FIXP_DBL accu1,accu2,accu3,accu4;
+    FIXP_DBL accu1,accu2,accu3,accu4;
     int idx, i;
 
     pDat_0 = &pDat[0];

libFDK/src/fixpoint_math.cpp

@@ -322,31 +322,31 @@ LNK_SECTION_CODE_L1 FIXP_DBL CalcInvLdData(FIXP_DBL x)
 
     LNK_SECTION_CONSTDATA_L1
     static const FIXP_DBL ldIntCoeff[] = {
-      0x80000001, 0x00000000, 0x02000000, 0x032b8034, 0x04000000, 0x04a4d3c2, 0x052b8034, 0x059d5da0,
-      0x06000000, 0x06570069, 0x06a4d3c2, 0x06eb3a9f, 0x072b8034, 0x0766a009, 0x079d5da0, 0x07d053f7,
-      0x08000000, 0x082cc7ee, 0x08570069, 0x087ef05b, 0x08a4d3c2, 0x08c8ddd4, 0x08eb3a9f, 0x090c1050,
-      0x092b8034, 0x0949a785, 0x0966a009, 0x0982809d, 0x099d5da0, 0x09b74949, 0x09d053f7, 0x09e88c6b,
-      0x0a000000, 0x0a16bad3, 0x0a2cc7ee, 0x0a423162, 0x0a570069, 0x0a6b3d79, 0x0a7ef05b, 0x0a92203d,
-      0x0aa4d3c2, 0x0ab7110e, 0x0ac8ddd4, 0x0ada3f60, 0x0aeb3a9f, 0x0afbd42b, 0x0b0c1050, 0x0b1bf312,
-      0x0b2b8034, 0x0b3abb40, 0x0b49a785, 0x0b584822, 0x0b66a009, 0x0b74b1fd, 0x0b82809d, 0x0b900e61,
-      0x0b9d5da0, 0x0baa708f, 0x0bb74949, 0x0bc3e9ca, 0x0bd053f7, 0x0bdc899b, 0x0be88c6b, 0x0bf45e09,
-      0x0c000000, 0x0c0b73cb, 0x0c16bad3, 0x0c21d671, 0x0c2cc7ee, 0x0c379085, 0x0c423162, 0x0c4caba8,
-      0x0c570069, 0x0c6130af, 0x0c6b3d79, 0x0c7527b9, 0x0c7ef05b, 0x0c88983f, 0x0c92203d, 0x0c9b8926,
-      0x0ca4d3c2, 0x0cae00d2, 0x0cb7110e, 0x0cc0052b, 0x0cc8ddd4, 0x0cd19bb0, 0x0cda3f60, 0x0ce2c97d,
-      0x0ceb3a9f, 0x0cf39355, 0x0cfbd42b, 0x0d03fda9, 0x0d0c1050, 0x0d140ca0, 0x0d1bf312, 0x0d23c41d,
-      0x0d2b8034, 0x0d3327c7, 0x0d3abb40, 0x0d423b08, 0x0d49a785, 0x0d510118, 0x0d584822, 0x0d5f7cff,
-      0x0d66a009, 0x0d6db197, 0x0d74b1fd, 0x0d7ba190, 0x0d82809d, 0x0d894f75, 0x0d900e61, 0x0d96bdad,
-      0x0d9d5da0, 0x0da3ee7f, 0x0daa708f, 0x0db0e412, 0x0db74949, 0x0dbda072, 0x0dc3e9ca, 0x0dca258e,
-      0x0dd053f7, 0x0dd6753e, 0x0ddc899b, 0x0de29143, 0x0de88c6b, 0x0dee7b47, 0x0df45e09, 0x0dfa34e1,
-      0x0e000000, 0x0e05bf94, 0x0e0b73cb, 0x0e111cd2, 0x0e16bad3, 0x0e1c4dfb, 0x0e21d671, 0x0e275460,
-      0x0e2cc7ee, 0x0e323143, 0x0e379085, 0x0e3ce5d8, 0x0e423162, 0x0e477346, 0x0e4caba8, 0x0e51daa8,
-      0x0e570069, 0x0e5c1d0b, 0x0e6130af, 0x0e663b74, 0x0e6b3d79, 0x0e7036db, 0x0e7527b9, 0x0e7a1030,
-      0x0e7ef05b, 0x0e83c857, 0x0e88983f, 0x0e8d602e, 0x0e92203d, 0x0e96d888, 0x0e9b8926, 0x0ea03232,
-      0x0ea4d3c2, 0x0ea96df0, 0x0eae00d2, 0x0eb28c7f, 0x0eb7110e, 0x0ebb8e96, 0x0ec0052b, 0x0ec474e4,
-      0x0ec8ddd4, 0x0ecd4012, 0x0ed19bb0, 0x0ed5f0c4, 0x0eda3f60, 0x0ede8797, 0x0ee2c97d, 0x0ee70525,
-      0x0eeb3a9f, 0x0eef69ff, 0x0ef39355, 0x0ef7b6b4, 0x0efbd42b, 0x0effebcd, 0x0f03fda9, 0x0f0809cf,
-      0x0f0c1050, 0x0f10113b, 0x0f140ca0, 0x0f18028d, 0x0f1bf312, 0x0f1fde3d, 0x0f23c41d, 0x0f27a4c0,
-      0x0f2b8034
+      (FIXP_DBL)0x80000001, (FIXP_DBL)0x00000000, (FIXP_DBL)0x02000000, (FIXP_DBL)0x032b8034, (FIXP_DBL)0x04000000, (FIXP_DBL)0x04a4d3c2, (FIXP_DBL)0x052b8034, (FIXP_DBL)0x059d5da0,
+      (FIXP_DBL)0x06000000, (FIXP_DBL)0x06570069, (FIXP_DBL)0x06a4d3c2, (FIXP_DBL)0x06eb3a9f, (FIXP_DBL)0x072b8034, (FIXP_DBL)0x0766a009, (FIXP_DBL)0x079d5da0, (FIXP_DBL)0x07d053f7,
+      (FIXP_DBL)0x08000000, (FIXP_DBL)0x082cc7ee, (FIXP_DBL)0x08570069, (FIXP_DBL)0x087ef05b, (FIXP_DBL)0x08a4d3c2, (FIXP_DBL)0x08c8ddd4, (FIXP_DBL)0x08eb3a9f, (FIXP_DBL)0x090c1050,
+      (FIXP_DBL)0x092b8034, (FIXP_DBL)0x0949a785, (FIXP_DBL)0x0966a009, (FIXP_DBL)0x0982809d, (FIXP_DBL)0x099d5da0, (FIXP_DBL)0x09b74949, (FIXP_DBL)0x09d053f7, (FIXP_DBL)0x09e88c6b,
+      (FIXP_DBL)0x0a000000, (FIXP_DBL)0x0a16bad3, (FIXP_DBL)0x0a2cc7ee, (FIXP_DBL)0x0a423162, (FIXP_DBL)0x0a570069, (FIXP_DBL)0x0a6b3d79, (FIXP_DBL)0x0a7ef05b, (FIXP_DBL)0x0a92203d,
+      (FIXP_DBL)0x0aa4d3c2, (FIXP_DBL)0x0ab7110e, (FIXP_DBL)0x0ac8ddd4, (FIXP_DBL)0x0ada3f60, (FIXP_DBL)0x0aeb3a9f, (FIXP_DBL)0x0afbd42b, (FIXP_DBL)0x0b0c1050, (FIXP_DBL)0x0b1bf312,
+      (FIXP_DBL)0x0b2b8034, (FIXP_DBL)0x0b3abb40, (FIXP_DBL)0x0b49a785, (FIXP_DBL)0x0b584822, (FIXP_DBL)0x0b66a009, (FIXP_DBL)0x0b74b1fd, (FIXP_DBL)0x0b82809d, (FIXP_DBL)0x0b900e61,
+      (FIXP_DBL)0x0b9d5da0, (FIXP_DBL)0x0baa708f, (FIXP_DBL)0x0bb74949, (FIXP_DBL)0x0bc3e9ca, (FIXP_DBL)0x0bd053f7, (FIXP_DBL)0x0bdc899b, (FIXP_DBL)0x0be88c6b, (FIXP_DBL)0x0bf45e09,
+      (FIXP_DBL)0x0c000000, (FIXP_DBL)0x0c0b73cb, (FIXP_DBL)0x0c16bad3, (FIXP_DBL)0x0c21d671, (FIXP_DBL)0x0c2cc7ee, (FIXP_DBL)0x0c379085, (FIXP_DBL)0x0c423162, (FIXP_DBL)0x0c4caba8,
+      (FIXP_DBL)0x0c570069, (FIXP_DBL)0x0c6130af, (FIXP_DBL)0x0c6b3d79, (FIXP_DBL)0x0c7527b9, (FIXP_DBL)0x0c7ef05b, (FIXP_DBL)0x0c88983f, (FIXP_DBL)0x0c92203d, (FIXP_DBL)0x0c9b8926,
+      (FIXP_DBL)0x0ca4d3c2, (FIXP_DBL)0x0cae00d2, (FIXP_DBL)0x0cb7110e, (FIXP_DBL)0x0cc0052b, (FIXP_DBL)0x0cc8ddd4, (FIXP_DBL)0x0cd19bb0, (FIXP_DBL)0x0cda3f60, (FIXP_DBL)0x0ce2c97d,
+      (FIXP_DBL)0x0ceb3a9f, (FIXP_DBL)0x0cf39355, (FIXP_DBL)0x0cfbd42b, (FIXP_DBL)0x0d03fda9, (FIXP_DBL)0x0d0c1050, (FIXP_DBL)0x0d140ca0, (FIXP_DBL)0x0d1bf312, (FIXP_DBL)0x0d23c41d,
+      (FIXP_DBL)0x0d2b8034, (FIXP_DBL)0x0d3327c7, (FIXP_DBL)0x0d3abb40, (FIXP_DBL)0x0d423b08, (FIXP_DBL)0x0d49a785, (FIXP_DBL)0x0d510118, (FIXP_DBL)0x0d584822, (FIXP_DBL)0x0d5f7cff,
+      (FIXP_DBL)0x0d66a009, (FIXP_DBL)0x0d6db197, (FIXP_DBL)0x0d74b1fd, (FIXP_DBL)0x0d7ba190, (FIXP_DBL)0x0d82809d, (FIXP_DBL)0x0d894f75, (FIXP_DBL)0x0d900e61, (FIXP_DBL)0x0d96bdad,
+      (FIXP_DBL)0x0d9d5da0, (FIXP_DBL)0x0da3ee7f, (FIXP_DBL)0x0daa708f, (FIXP_DBL)0x0db0e412, (FIXP_DBL)0x0db74949, (FIXP_DBL)0x0dbda072, (FIXP_DBL)0x0dc3e9ca, (FIXP_DBL)0x0dca258e,
+      (FIXP_DBL)0x0dd053f7, (FIXP_DBL)0x0dd6753e, (FIXP_DBL)0x0ddc899b, (FIXP_DBL)0x0de29143, (FIXP_DBL)0x0de88c6b, (FIXP_DBL)0x0dee7b47, (FIXP_DBL)0x0df45e09, (FIXP_DBL)0x0dfa34e1,
+      (FIXP_DBL)0x0e000000, (FIXP_DBL)0x0e05bf94, (FIXP_DBL)0x0e0b73cb, (FIXP_DBL)0x0e111cd2, (FIXP_DBL)0x0e16bad3, (FIXP_DBL)0x0e1c4dfb, (FIXP_DBL)0x0e21d671, (FIXP_DBL)0x0e275460,
+      (FIXP_DBL)0x0e2cc7ee, (FIXP_DBL)0x0e323143, (FIXP_DBL)0x0e379085, (FIXP_DBL)0x0e3ce5d8, (FIXP_DBL)0x0e423162, (FIXP_DBL)0x0e477346, (FIXP_DBL)0x0e4caba8, (FIXP_DBL)0x0e51daa8,
+      (FIXP_DBL)0x0e570069, (FIXP_DBL)0x0e5c1d0b, (FIXP_DBL)0x0e6130af, (FIXP_DBL)0x0e663b74, (FIXP_DBL)0x0e6b3d79, (FIXP_DBL)0x0e7036db, (FIXP_DBL)0x0e7527b9, (FIXP_DBL)0x0e7a1030,
+      (FIXP_DBL)0x0e7ef05b, (FIXP_DBL)0x0e83c857, (FIXP_DBL)0x0e88983f, (FIXP_DBL)0x0e8d602e, (FIXP_DBL)0x0e92203d, (FIXP_DBL)0x0e96d888, (FIXP_DBL)0x0e9b8926, (FIXP_DBL)0x0ea03232,
+      (FIXP_DBL)0x0ea4d3c2, (FIXP_DBL)0x0ea96df0, (FIXP_DBL)0x0eae00d2, (FIXP_DBL)0x0eb28c7f, (FIXP_DBL)0x0eb7110e, (FIXP_DBL)0x0ebb8e96, (FIXP_DBL)0x0ec0052b, (FIXP_DBL)0x0ec474e4,
+      (FIXP_DBL)0x0ec8ddd4, (FIXP_DBL)0x0ecd4012, (FIXP_DBL)0x0ed19bb0, (FIXP_DBL)0x0ed5f0c4, (FIXP_DBL)0x0eda3f60, (FIXP_DBL)0x0ede8797, (FIXP_DBL)0x0ee2c97d, (FIXP_DBL)0x0ee70525,
+      (FIXP_DBL)0x0eeb3a9f, (FIXP_DBL)0x0eef69ff, (FIXP_DBL)0x0ef39355, (FIXP_DBL)0x0ef7b6b4, (FIXP_DBL)0x0efbd42b, (FIXP_DBL)0x0effebcd, (FIXP_DBL)0x0f03fda9, (FIXP_DBL)0x0f0809cf,
+      (FIXP_DBL)0x0f0c1050, (FIXP_DBL)0x0f10113b, (FIXP_DBL)0x0f140ca0, (FIXP_DBL)0x0f18028d, (FIXP_DBL)0x0f1bf312, (FIXP_DBL)0x0f1fde3d, (FIXP_DBL)0x0f23c41d, (FIXP_DBL)0x0f27a4c0,
+      (FIXP_DBL)0x0f2b8034
     };
 
 

libMpegTPDec/include/tp_data.h

@@ -146,12 +146,15 @@ typedef struct
 
   UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
 
@@ -324,16 +327,23 @@ int getSamplingRateIndex( UINT samplingRate )
  */
 static inline int getNumberOfTotalChannels(int channelConfig)
 {
-  if (channelConfig > 0 && channelConfig < 8)
-    return (channelConfig == 7)?8:channelConfig;
-  else
+  switch (channelConfig) {
+  case 1: case 2: case 3:
+  case 4: case 5: case 6:
+    return channelConfig;
+  case 7: case 12: case 14:
+    return 8;
+  case 11:
+    return 7;
+  default:
     return 0;
+  }
 }
 
 static inline
 int getNumberOfEffectiveChannels(const int channelConfig)
-{
-  const int n[] = {0,1,2,3,4,5,5,7};
+{        /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0};
   return n[channelConfig];
 }
 

libMpegTPDec/include/tpdec_lib.h

@@ -151,6 +151,7 @@ typedef enum {
 #define PC_ASSOCDATA_MAX    8
 #define PC_CCEL_MAX         16 /* CC elements */
 #define PC_COMMENTLENGTH    256
+#define PC_NUM_HEIGHT_LAYER 3
 
 
 /*!
@@ -239,14 +240,20 @@ int CProgramConfig_LookupElement(
         );
 
 /**
- * \brief        Get table of elements in canonical order.
- * \param pPce   A valid program config structure.
- * \param table  An array where the element IDs are stored.
- * \return       Total element count including all SCE, CPE and LFE.
+ * \brief             Get table of elements in canonical order from a
+ *                    give program config field.
+ * \param pPce        A valid program config structure.
+ * \param table       An array where the element IDs are stored.
+ * \param elListSize  The length of the table array.
+ * \param pChMapIdx   Pointer to a field receiving the corresponding
+ *                    implicit channel configuration index of the given
+ *                    PCE. If none can be found it receives the value 0.
+ * \return            Total element count including all SCE, CPE and LFE.
  */
 int CProgramConfig_GetElementTable( const CProgramConfig *pPce,
                                     MP4_ELEMENT_ID  table[],
-                                    const INT elListSize );
+                                    const INT elListSize,
+                                    UCHAR *pChMapIdx );
 
 /**
  * \brief       Initialize a given AudioSpecificConfig structure.

libMpegTPDec/src/tpdec_asc.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -90,6 +90,9 @@ amm-info@iis.fraunhofer.de
 
 #include "tpdec_lib.h"
 #include "tp_data.h"
+#ifdef TP_PCE_ENABLE
+#include "FDK_crc.h"
+#endif
 
 
 void CProgramConfig_Reset(CProgramConfig *pPce)
@@ -111,13 +114,75 @@ int  CProgramConfig_IsValid ( const CProgramConfig *pPce )
 }
 
 #ifdef TP_PCE_ENABLE
+#define PCE_HEIGHT_EXT_SYNC  ( 0xAC )
+
+/*
+ * Read the extension for height info.
+ * return 0 if successfull or -1 if the CRC failed.
+ */
+static
+int CProgramConfig_ReadHeightExt(
+                                  CProgramConfig *pPce,
+                                  HANDLE_FDK_BITSTREAM bs,
+                                  int * const bytesAvailable,
+                                  const UINT alignmentAnchor
+                                )
+{
+  int err = 0;
+  FDK_CRCINFO crcInfo;    /* CRC state info */
+  INT crcReg;
+  FDKcrcInit(&crcInfo, 0x07, 0xFF, 8);
+  crcReg = FDKcrcStartReg(&crcInfo, bs, 0);
+  UINT startAnchor = FDKgetValidBits(bs);
+
+  FDK_ASSERT(pPce != NULL);
+  FDK_ASSERT(bs != NULL);
+  FDK_ASSERT(bytesAvailable != NULL);
+
+  if ( (startAnchor >= 24) && (*bytesAvailable >= 3)
+    && (FDKreadBits(bs,8) == PCE_HEIGHT_EXT_SYNC) )
+  {
+    int i;
+
+    for (i=0; i < pPce->NumFrontChannelElements; i++)
+    {
+      pPce->FrontElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    for (i=0; i < pPce->NumSideChannelElements; i++)
+    {
+      pPce->SideElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    for (i=0; i < pPce->NumBackChannelElements; i++)
+    {
+      pPce->BackElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    FDKbyteAlign(bs, alignmentAnchor);
+
+    FDKcrcEndReg(&crcInfo, bs, crcReg);
+    if ((USHORT)FDKreadBits(bs,8) != FDKcrcGetCRC(&crcInfo)) {
+      /* CRC failed */
+      err = -1;
+    }
+  }
+  else {
+    /* No valid extension data found -> restore the initial bitbuffer state */
+    FDKpushBack(bs, startAnchor - FDKgetValidBits(bs));
+  }
+
+  /* Always report the bytes read. */
+  *bytesAvailable -= (startAnchor - FDKgetValidBits(bs)) >> 3;
+
+  return (err);
+}
+
 void CProgramConfig_Read(
                           CProgramConfig *pPce,
                           HANDLE_FDK_BITSTREAM bs,
                           UINT alignmentAnchor
                         )
 {
-  int i;
+  int i, err = 0;
+  int commentBytes;
 
   pPce->NumEffectiveChannels = 0;
   pPce->NumChannels = 0;
@@ -190,8 +255,12 @@ void CProgramConfig_Read(
   FDKbyteAlign(bs, alignmentAnchor);
 
   pPce->CommentFieldBytes = (UCHAR) FDKreadBits(bs,8);
+  commentBytes = pPce->CommentFieldBytes;
 
-  for (i=0; i < pPce->CommentFieldBytes; i++)
+  /* Search for height info extension and read it if available */
+  err = CProgramConfig_ReadHeightExt( pPce, bs, &commentBytes, alignmentAnchor );
+
+  for (i=0; i < commentBytes; i++)
   {
     UCHAR text;
 
@@ -203,7 +272,7 @@ void CProgramConfig_Read(
     }
   }
 
-  pPce->isValid = 1;
+  pPce->isValid = (err) ? 0 : 1;
 }
 
 /*
@@ -235,6 +304,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumFrontChannelElements; el += 1) {
+          if (pPce1->FrontElementHeightInfo[el] != pPce2->FrontElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->FrontElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->FrontElementIsCpe[el] ? 2 : 1;
         }
@@ -248,6 +321,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumSideChannelElements; el += 1) {
+          if (pPce1->SideElementHeightInfo[el] != pPce2->SideElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->SideElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->SideElementIsCpe[el] ? 2 : 1;
         }
@@ -261,6 +338,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumBackChannelElements; el += 1) {
+          if (pPce1->BackElementHeightInfo[el] != pPce2->BackElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->BackElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->BackElementIsCpe[el] ? 2 : 1;
         }
@@ -290,6 +371,44 @@ void CProgramConfig_GetDefault( CProgramConfig *pPce,
 
   switch (channelConfig) {
   /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 32: /* 7.1 side channel configuration as defined in FDK_audio.h */
+    pPce->NumFrontChannelElements  = 2;
+    pPce->FrontElementIsCpe[0]     = 0;
+    pPce->FrontElementIsCpe[1]     = 1;
+    pPce->NumSideChannelElements   = 1;
+    pPce->SideElementIsCpe[0]      = 1;
+    pPce->NumBackChannelElements   = 1;
+    pPce->BackElementIsCpe[0]      = 1;
+    pPce->NumLfeChannelElements    = 1;
+    pPce->NumChannels              = 8;
+    pPce->NumEffectiveChannels     = 7;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 12:  /* 3/0/4.1ch surround back */
+    pPce->BackElementIsCpe[1]      = 1;
+    pPce->NumChannels             += 1;
+    pPce->NumEffectiveChannels    += 1;
+  case 11:  /* 3/0/3.1ch */
+    pPce->NumFrontChannelElements += 2;
+    pPce->FrontElementIsCpe[0]     = 0;
+    pPce->FrontElementIsCpe[1]     = 1;
+    pPce->NumBackChannelElements  += 2;
+    pPce->BackElementIsCpe[0]      = 1;
+    pPce->BackElementIsCpe[1]     += 0;
+    pPce->NumLfeChannelElements   += 1;
+    pPce->NumChannels             += 7;
+    pPce->NumEffectiveChannels    += 6;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 14:  /* 2/0/0-3/0/2-0.1ch front height */
+    pPce->FrontElementHeightInfo[2] = 1;      /* Top speaker */
+  case 7:   /* 5/0/2.1ch front */
+    pPce->NumFrontChannelElements += 1;
+    pPce->FrontElementIsCpe[2]     = 1;
+    pPce->NumChannels             += 2;
+    pPce->NumEffectiveChannels    += 2;
   case 6:   /* 3/0/2.1ch */
     pPce->NumLfeChannelElements   += 1;
     pPce->NumChannels             += 1;
@@ -330,13 +449,13 @@ void CProgramConfig_GetDefault( CProgramConfig *pPce,
     int el, elTagSce = 0, elTagCpe = 0;
 
     for (el = 0; el < pPce->NumFrontChannelElements; el += 1) {
-      pPce->FrontElementTagSelect[el] = (pPce->FrontElementIsCpe) ? elTagCpe++ : elTagSce++;
+      pPce->FrontElementTagSelect[el] = (pPce->FrontElementIsCpe[el]) ? elTagCpe++ : elTagSce++;
     }
     for (el = 0; el < pPce->NumSideChannelElements; el += 1) {
-      pPce->SideElementTagSelect[el] = (pPce->SideElementIsCpe) ? elTagCpe++ : elTagSce++;
+      pPce->SideElementTagSelect[el] = (pPce->SideElementIsCpe[el]) ? elTagCpe++ : elTagSce++;
     }
     for (el = 0; el < pPce->NumBackChannelElements; el += 1) {
-      pPce->BackElementTagSelect[el] = (pPce->BackElementIsCpe) ? elTagCpe++ : elTagSce++;
+      pPce->BackElementTagSelect[el] = (pPce->BackElementIsCpe[el]) ? elTagCpe++ : elTagSce++;
     }
     elTagSce = 0;
     for (el = 0; el < pPce->NumLfeChannelElements; el += 1) {
@@ -348,10 +467,11 @@ void CProgramConfig_GetDefault( CProgramConfig *pPce,
 
 /**
  * \brief get implicit audio channel type for given channelConfig and MPEG ordered channel index
- * \param channelConfig MPEG channelConfiguration from 1 upto 7
+ * \param channelConfig MPEG channelConfiguration from 1 upto 14
  * \param index MPEG channel order index
  * \return audio channel type.
  */
+static
 void getImplicitAudioChannelTypeAndIndex(
         AUDIO_CHANNEL_TYPE *chType,
         UCHAR *chIndex,
@@ -364,9 +484,9 @@ void getImplicitAudioChannelTypeAndIndex(
     *chIndex = index;
   } else {
     switch (channelConfig) {
-      case MODE_1_2_1:
-      case MODE_1_2_2:
-      case MODE_1_2_2_1:
+      case 4:  /* SCE, CPE, SCE */
+      case 5:  /* SCE, CPE, CPE */
+      case 6:  /* SCE, CPE, CPE, LFE */
         switch (index) {
           case 3:
           case 4:
@@ -379,12 +499,12 @@ void getImplicitAudioChannelTypeAndIndex(
             break;
         }
         break;
-      case MODE_1_2_2_2_1:
+      case 7:  /* SCE,CPE,CPE,CPE,LFE */
         switch (index) {
           case 3:
           case 4:
-            *chType = ACT_SIDE;
-            *chIndex = index - 3;
+            *chType = ACT_FRONT;
+            *chIndex = index;
             break;
           case 5:
           case 6:
@@ -397,6 +517,42 @@ void getImplicitAudioChannelTypeAndIndex(
             break;
         }
         break;
+      case 11:  /* SCE,CPE,CPE,SCE,LFE */
+        if (index < 6) {
+          *chType = ACT_BACK;
+          *chIndex = index - 3;
+        } else {
+          *chType = ACT_LFE;
+          *chIndex = 0;
+        }
+        break;
+      case 12:  /* SCE,CPE,CPE,CPE,LFE */
+        if (index < 7) {
+          *chType = ACT_BACK;
+          *chIndex = index - 3;
+        } else {
+          *chType = ACT_LFE;
+          *chIndex = 0;
+        }
+        break;
+      case 14:  /* SCE,CPE,CPE,LFE,CPE */
+        switch (index) {
+          case 3:
+          case 4:
+            *chType = ACT_BACK;
+            *chIndex = index - 3;
+            break;
+          case 5:
+            *chType = ACT_LFE;
+            *chIndex = 0;
+            break;
+          case 6:
+          case 7:
+            *chType = ACT_FRONT_TOP;
+            *chIndex = index - 6;  /* handle the top layer independently */
+            break;
+        }
+        break;
       default:
         *chType = ACT_NONE;
         break;
@@ -467,7 +623,24 @@ int CProgramConfig_LookupElement(
     else {
       /* Accept the additional channel(s), only if the tag is in the lists */
       int isCpe = 0, i;
-      int cc = 0, fc = 0, sc = 0, bc = 0, lc = 0, ec = 0; /* Channel and element counters */
+      /* Element counter */
+      int ec[PC_NUM_HEIGHT_LAYER] = {0};
+      /* Channel counters */
+      int cc[PC_NUM_HEIGHT_LAYER] = {0};
+      int fc[PC_NUM_HEIGHT_LAYER] = {0};
+      int sc[PC_NUM_HEIGHT_LAYER] = {0};
+      int bc[PC_NUM_HEIGHT_LAYER] = {0};
+      int lc = 0;;
+
+      /* General MPEG (PCE) composition rules:
+         - Over all:
+             <normal height channels><top height channels><bottom height channels>
+         - Within each height layer:
+             <front channels><side channels><back channels>
+         - Exception:
+             The LFE channels have no height info and thus they are arranged at the very
+             end of the normal height layer channels.
+       */
 
       switch (elType)
       {
@@ -476,87 +649,206 @@ int CProgramConfig_LookupElement(
       case ID_SCE:
         /* search in front channels */
         for (i = 0; i < pPce->NumFrontChannelElements; i++) {
+          int heightLayer = pPce->FrontElementHeightInfo[i];
           if (isCpe == pPce->FrontElementIsCpe[i] && pPce->FrontElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_FRONT;
-            chIndex[cc] = fc;
-            if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_FRONT;
-              chIndex[cc+1] = fc+1;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_FRONT);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* normal height */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
             }
-            *elMapping = ec;
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = fc[heightLayer];
+            if (isCpe) {
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = fc[heightLayer]+1;
+            }
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->FrontElementIsCpe[i]) {
-            cc+=2; fc+=2;
+            cc[heightLayer] += 2;
+            fc[heightLayer] += 2;
           } else {
-            cc++; fc++;
+            cc[heightLayer] += 1;
+            fc[heightLayer] += 1;
           }
         }
         /* search in side channels */
         for (i = 0; i < pPce->NumSideChannelElements; i++) {
+          int heightLayer = pPce->SideElementHeightInfo[i];
           if (isCpe == pPce->SideElementIsCpe[i] && pPce->SideElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_SIDE;
-            chIndex[cc] = sc;
-            if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_SIDE;
-              chIndex[cc+1] = sc+1;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_SIDE);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* LFE channels belong to the normal height layer */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
             }
-            *elMapping = ec;
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = sc[heightLayer];
+            if (isCpe) {
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = sc[heightLayer]+1;
+            }
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->SideElementIsCpe[i]) {
-            cc+=2; sc+=2;
+            cc[heightLayer] += 2;
+            sc[heightLayer] += 2;
           } else {
-            cc++; sc++;
+            cc[heightLayer] += 1;
+            sc[heightLayer] += 1;
           }
         }
         /* search in back channels */
         for (i = 0; i < pPce->NumBackChannelElements; i++) {
+          int heightLayer = pPce->BackElementHeightInfo[i];
           if (isCpe == pPce->BackElementIsCpe[i] && pPce->BackElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_BACK;
-            chIndex[cc] = bc;
-            if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_BACK;
-              chIndex[cc+1] = bc+1;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_BACK);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* normal height */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
             }
-            *elMapping = ec;
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = bc[heightLayer];
+            if (isCpe) {
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = bc[heightLayer]+1;
+            }
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->BackElementIsCpe[i]) {
-            cc+=2; bc+=2;
+            cc[heightLayer] += 2;
+            bc[heightLayer] += 2;
           } else {
-            cc++; bc++;
+            cc[heightLayer] += 1;
+            bc[heightLayer] += 1;
           }
         }
         break;
 
       case ID_LFE:
-        /* Initialize channel counter and element counter */
-        cc = pPce->NumEffectiveChannels;
-        ec = pPce->NumFrontChannelElements+ pPce->NumSideChannelElements + pPce->NumBackChannelElements;
+      { /* Unfortunately we have to go through all normal height
+           layer elements to get the position of the LFE channels.
+           Start with counting the front channels/elements at normal height */
+        for (i = 0; i < pPce->NumFrontChannelElements; i+=1) {
+          int heightLayer = pPce->FrontElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->FrontElementIsCpe[i]) ? 2 : 1;
+        }
+        /* Count side channels/elements at normal height */
+        for (i = 0; i < pPce->NumSideChannelElements; i+=1) {
+          int heightLayer = pPce->SideElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->SideElementIsCpe[i]) ? 2 : 1;
+        }
+        /* Count back channels/elements at normal height */
+        for (i = 0; i < pPce->NumBackChannelElements; i+=1) {
+          int heightLayer = pPce->BackElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->BackElementIsCpe[i]) ? 2 : 1;
+        }
+
         /* search in lfe channels */
         for (i = 0; i < pPce->NumLfeChannelElements; i++) {
+          int elIdx = ec[0];  /* LFE channels belong to the normal height layer */
+          int chIdx = cc[0];
           if ( pPce->LfeElementTagSelect[i] == tag ) {
-            chMapping[cc] = channelIdx;
-            *elMapping = ec;
-            chType[cc] = ACT_LFE;
-            chIndex[cc] = lc;
+            chMapping[chIdx] = channelIdx;
+            *elMapping = elIdx;
+            chType[chIdx] = ACT_LFE;
+            chIndex[chIdx] = lc;
             return 1;
           }
-          ec++;
-          cc++;
-          lc++;
+          ec[0] += 1;
+          cc[0] += 1;
+          lc += 1;
         }
-        break;
+      } break;
 
       /* Non audio elements */
       case ID_CCE:
@@ -590,13 +882,19 @@ int CProgramConfig_LookupElement(
 int CProgramConfig_GetElementTable(
         const CProgramConfig *pPce,
         MP4_ELEMENT_ID  elList[],
-        const INT elListSize
+        const INT elListSize,
+        UCHAR *pChMapIdx
        )
 {
   int i, el = 0;
 
-  if ( elListSize 
-    < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements 
+  FDK_ASSERT(elList != NULL);
+  FDK_ASSERT(pChMapIdx != NULL);
+
+  *pChMapIdx = 0;
+
+  if ( elListSize
+    < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements
     )
   {
     return 0;
@@ -623,6 +921,47 @@ int CProgramConfig_GetElementTable(
   }
 
 
+  /* Find an corresponding channel configuration if possible */
+  switch (pPce->NumChannels) {
+  case 1: case 2: case 3: case 4: case 5: case 6:
+    /* One and two channels have no alternatives. The other ones are mapped directly to the
+       corresponding channel config. Because of legacy reasons or for lack of alternative mappings. */
+    *pChMapIdx = pPce->NumChannels;
+    break;
+  case 7:
+    {
+      C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1);
+      /* Create a PCE for the config to test ... */
+      CProgramConfig_GetDefault(tmpPce, 11);
+      /* ... and compare it with the given one. */
+      *pChMapIdx = (!(CProgramConfig_Compare(pPce, tmpPce)&0xE)) ? 11 : 0;
+      /* If compare result is 0 or 1 we can be sure that it is channel config 11. */
+      C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1);
+    }
+    break;
+  case 8:
+    { /* Try the four possible 7.1ch configurations. One after the other. */
+      UCHAR testCfg[4] = { 32, 14, 12, 7};
+      C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1);
+      for (i=0; i<4; i+=1) {
+        /* Create a PCE for the config to test ... */
+        CProgramConfig_GetDefault(tmpPce, testCfg[i]);
+        /* ... and compare it with the given one. */
+        if (!(CProgramConfig_Compare(pPce, tmpPce)&0xE)) {
+          /* If the compare result is 0 or 1 than the two channel configurations match. */
+          /* Explicit mapping of 7.1 side channel configuration to 7.1 rear channel mapping. */
+          *pChMapIdx = (testCfg[i]==32) ? 12 : testCfg[i];
+        }
+      }
+      C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1);
+    }
+    break;
+  default:
+    /* The PCE does not match any predefined channel configuration. */
+    *pChMapIdx = 0;
+    break;
+  }
+
   return el;
 }
 #endif
@@ -714,7 +1053,7 @@ TRANSPORTDEC_ERROR GaSpecificConfig_Parse( CSGaSpecificConfig    *self,
 #ifdef TP_ELD_ENABLE
 
 static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
-                           HANDLE_FDK_BITSTREAM hBs, 
+                           HANDLE_FDK_BITSTREAM hBs,
                            CSTpCallBacks *cb )
 {
   const int channelConfiguration = asc->m_channelConfiguration;
@@ -728,6 +1067,8 @@ static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
   }
 
   switch ( channelConfiguration ) {
+    case 14:
+    case 12:
     case 7:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
     case 6:
@@ -737,6 +1078,8 @@ static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
       break;
 
+    case 11:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
     case 4:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++);
@@ -783,6 +1126,8 @@ TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
       if ( 0 != ld_sbr_header(asc, hBs, cb) ) {
         return TRANSPORTDEC_PARSE_ERROR;
       }
+    } else {
+      return TRANSPORTDEC_UNSUPPORTED_FORMAT;
     }
   }
   esc->m_useLdQmfTimeAlign = 0;
@@ -802,26 +1147,8 @@ TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
     }
 
     switch (eldExtType) {
-      case ELDEXT_LDSAC:
-        esc->m_useLdQmfTimeAlign = 1;
-        if (cb->cbSsc != NULL) {
-          ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc(
-                  cb->cbSscData,
-                  hBs,
-                  asc->m_aot,
-                  asc->m_samplingFrequency,
-                  1,  /* muxMode */
-                  len
-                  );
-        } else {
-          ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
-        }
-        if (ErrorStatus != TRANSPORTDEC_OK) {
-          goto bail;
-        }
-        break;
       default:
-        for(cnt=0; cnt<len; cnt++) {
+        for(cnt=0; cnt<eldExtLen; cnt++) {
           FDKreadBits(hBs, 8 );
         }
         break;
@@ -1047,4 +1374,133 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse(
   return (ErrorStatus);
 }
 
+TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse(
+        CSAudioSpecificConfig *self,
+        HANDLE_FDK_BITSTREAM   bs
+        )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  AudioSpecificConfig_Init(self);
+
+  if ((INT)FDKgetValidBits(bs) < 20) {
+    ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+    goto bail;
+  }
+  else {
+    /* DRM - Audio information data entity - type 9
+       - Short Id            2 bits
+       - Stream Id           2 bits
+       - audio coding        2 bits
+       - SBR flag            1 bit
+       - audio mode          2 bits
+       - audio sampling rate 3 bits
+       - text flag           1 bit
+       - enhancement flag    1 bit
+       - coder field         5 bits
+       - rfa                 1 bit  */
+
+    int audioCoding, audioMode, cSamplingFreq, coderField, sfIdx, sbrFlag;
+
+    /* Read the SDC field */
+    FDKreadBits(bs,4);   /* Short and Stream Id */
+
+    audioCoding   = FDKreadBits(bs, 2);
+    sbrFlag       = FDKreadBits(bs, 1);
+    audioMode     = FDKreadBits(bs, 2);
+    cSamplingFreq = FDKreadBits(bs, 3);    /* audio sampling rate */
+
+    FDKreadBits(bs, 2);  /* Text and enhancement flag */
+    coderField   = FDKreadBits(bs, 5);
+    FDKreadBits(bs, 1);  /* rfa */
+
+    /* Evaluate configuration and fill the ASC */
+    switch (cSamplingFreq) {
+    case 0: /*  8 kHz */
+      sfIdx = 11;
+      break;
+    case 1: /* 12 kHz */
+      sfIdx = 9;
+      break;
+    case 2: /* 16 kHz */
+      sfIdx = 8;
+      break;
+    case 3: /* 24 kHz */
+      sfIdx = 6;
+      break;
+    case 5: /* 48 kHz */
+      sfIdx = 3;
+      break;
+    case 4: /* reserved */
+    case 6: /* reserved */
+    case 7: /* reserved */
+    default:
+      ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+      goto bail;
+    }
+
+    self->m_samplingFrequencyIndex = sfIdx;
+    self->m_samplingFrequency = SamplingRateTable[sfIdx];
+
+    if ( sbrFlag ) {
+      UINT i;
+      int tmp = -1;
+      self->m_sbrPresentFlag = 1;
+      self->m_extensionAudioObjectType = AOT_SBR;
+      self->m_extensionSamplingFrequency = self->m_samplingFrequency << 1;
+      for (i=0; i<(sizeof(SamplingRateTable)/sizeof(SamplingRateTable[0])); i++){
+        if (SamplingRateTable[i] == self->m_extensionSamplingFrequency){
+          tmp = i;
+          break;
+        }
+      }
+      self->m_extensionSamplingFrequencyIndex = tmp;
+    }
+
+    switch (audioCoding) {
+      case 0: /* AAC */
+          self->m_aot = AOT_DRM_AAC     ;  /* Set pseudo AOT for Drm AAC */
+
+        switch (audioMode) {
+        case 1: /* parametric stereo */
+          self->m_psPresentFlag = 1;
+        case 0: /* mono */
+          self->m_channelConfiguration = 1;
+          break;
+        case 2: /* stereo */
+          self->m_channelConfiguration = 2;
+          break;
+        default:
+          ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+          goto bail;
+        }
+        self->m_vcb11Flag = 1;
+        self->m_hcrFlag = 1;
+        self->m_samplesPerFrame = 960;
+        self->m_epConfig = 1;
+        break;
+      case 1: /* CELP */
+        self->m_aot = AOT_ER_CELP;
+        self->m_channelConfiguration = 1;
+        break;
+      case 2: /* HVXC */
+        self->m_aot = AOT_ER_HVXC;
+        self->m_channelConfiguration = 1;
+        break;
+      case 3: /* reserved */
+      default:
+        ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+        self->m_aot = AOT_NONE;
+        break;
+    }
+
+    if (self->m_psPresentFlag && !self->m_sbrPresentFlag) {
+      ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+      goto bail;
+    }
+  }
+
+bail:
+  return (ErrorStatus);
+}
 

libMpegTPDec/src/tpdec_drm.cpp

@@ -0,0 +1,146 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Christian Griebel
+   Description: DRM transport stuff
+
+******************************************************************************/
+
+#include "tpdec_drm.h"
+
+
+#include "FDK_bitstream.h"
+
+
+
+void drmRead_CrcInit(HANDLE_DRM pDrm)      /*!< pointer to drm crc info stucture */
+{
+  FDK_ASSERT(pDrm != NULL);
+
+  FDKcrcInit(&pDrm->crcInfo, 0x001d, 0xFFFF, 8);
+}
+
+int drmRead_CrcStartReg(
+                     HANDLE_DRM pDrm,            /*!< pointer to drm stucture */
+                     HANDLE_FDK_BITSTREAM hBs,   /*!< handle to current bit buffer structure */
+                     int mBits                   /*!< number of bits in crc region */
+                   )
+{
+  FDK_ASSERT(pDrm != NULL);
+
+  FDKcrcReset(&pDrm->crcInfo);
+
+  pDrm->crcReadValue = FDKreadBits(hBs, 8);
+
+  return ( FDKcrcStartReg(&pDrm->crcInfo, hBs, mBits) );
+
+}
+
+void drmRead_CrcEndReg(
+                    HANDLE_DRM pDrm,             /*!< pointer to drm crc info stucture */
+                    HANDLE_FDK_BITSTREAM hBs,    /*!< handle to current bit buffer structure */
+                    int reg                      /*!< crc region */
+                  )
+{
+  FDK_ASSERT(pDrm != NULL);
+
+  FDKcrcEndReg(&pDrm->crcInfo, hBs, reg);
+}
+
+TRANSPORTDEC_ERROR drmRead_CrcCheck( HANDLE_DRM pDrm )
+{
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  USHORT crc;
+
+  crc = FDKcrcGetCRC(&pDrm->crcInfo) ^ 0xFF;
+  if (crc != pDrm->crcReadValue)
+  {
+    return (TRANSPORTDEC_CRC_ERROR);
+  }
+
+  return (ErrorStatus);
+}
+
+

libMpegTPDec/src/tpdec_drm.h

@@ -0,0 +1,194 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Josef Hoepfl
+   Description: DRM interface
+
+******************************************************************************/
+
+#ifndef TPDEC_DRM_H
+#define TPDEC_DRM_H
+
+#include "tpdec_lib.h"
+
+
+#include "FDK_crc.h"
+
+typedef struct {
+
+  FDK_CRCINFO crcInfo;             /* CRC state info */
+  USHORT      crcReadValue;        /* CRC value read from bitstream data */
+
+} STRUCT_DRM;
+
+typedef STRUCT_DRM *HANDLE_DRM;
+
+/*!
+  \brief Initialize DRM CRC
+
+  The function initialzes the crc buffer and the crc lookup table.
+
+  \return  none
+*/
+void drmRead_CrcInit( HANDLE_DRM pDrm );
+
+/**
+ * \brief Starts CRC region with a maximum number of bits
+ *        If mBits is positive zero padding will be used for CRC calculation, if there
+ *        are less than mBits bits available.
+ *        If mBits is negative no zero padding is done.
+ *        If mBits is zero the memory for the buffer is allocated dynamically, the
+ *        number of bits is not limited.
+ *
+ * \param pDrm DRM data handle
+ * \param hBs bitstream handle, on which the CRC region referes to
+ * \param mBits max number of bits in crc region to be considered
+ *
+ * \return  ID for the created region, -1 in case of an error
+ */
+int drmRead_CrcStartReg(
+        HANDLE_DRM pDrm,
+        HANDLE_FDK_BITSTREAM hBs,
+        int mBits
+        );
+
+/**
+ * \brief Ends CRC region identified by reg
+ *
+ * \param pDrm DRM data handle
+ * \param hBs bitstream handle, on which the CRC region referes to
+ * \param reg CRC regions ID returned by drmRead_CrcStartReg()
+ *
+ * \return  none
+ */
+void drmRead_CrcEndReg(
+        HANDLE_DRM pDrm,
+        HANDLE_FDK_BITSTREAM hBs,
+        int reg
+        );
+
+/**
+ * \brief Check CRC
+ *
+ * Checks if the currently calculated CRC matches the CRC field read from the bitstream
+ * Deletes all CRC regions.
+ *
+ * \param pDrm DRM data handle
+ *
+ * \return Returns 0 if they are identical otherwise 1
+ */
+TRANSPORTDEC_ERROR drmRead_CrcCheck( HANDLE_DRM pDrm );
+
+/**
+ * \brief Check if we have a valid DRM frame at the current bitbuffer position
+ *
+ * This function assumes enough bits in buffer for the current frame.
+ * It reads out the header bits to prepare the bitbuffer for the decode loop.
+ * In case the header bits show an invalid bitstream/frame, the whole frame is skipped.
+ *
+ * \param pDrm DRM data handle which is filled with parsed DRM header data
+ * \param bs handle of bitstream from whom the DRM header is read
+ *
+ * \return  error status
+ */
+TRANSPORTDEC_ERROR drmRead_DecodeHeader(
+        HANDLE_DRM            pDrm,
+        HANDLE_FDK_BITSTREAM  bs
+        );
+
+/**
+ * \brief   Parse a Drm specific SDC audio config from a given bitstream handle.
+ *
+ * \param pAsc                         A pointer to an allocated CSAudioSpecificConfig struct.
+ * \param hBs                          Bitstream handle.
+ *
+ * \return  Total element count including all SCE, CPE and LFE.
+ */
+TRANSPORTDEC_ERROR DrmRawSdcAudioConfig_Parse( CSAudioSpecificConfig *pAsc,
+                                               HANDLE_FDK_BITSTREAM hBs );
+
+
+
+#endif /* TPDEC_DRM_H */

libMpegTPDec/src/tpdec_lib.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -102,6 +102,7 @@ amm-info@iis.fraunhofer.de
 
 #include "tpdec_latm.h"
 
+#include "tpdec_drm.h"
 
 
 #define MODULE_NAME "transportDec"
@@ -113,6 +114,7 @@ typedef union {
 
   CLatmDemux latm;
 
+  STRUCT_DRM drm;
 
 } transportdec_parser_t;
 
@@ -182,6 +184,9 @@ HANDLE_TRANSPORTDEC transportDec_Open( const TRANSPORT_TYPE transportFmt, const
     hInput->numberOfRawDataBlocks = 0;
     break;
 
+  case TT_DRM:
+    drmRead_CrcInit(&hInput->parser.drm);
+    break;
 
   case TT_MP4_LATM_MCP0:
   case TT_MP4_LATM_MCP1:
@@ -253,6 +258,18 @@ TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *
         }
       }
       break;
+    case TT_DRM:
+      fConfigFound = 1;
+      err = DrmRawSdcAudioConfig_Parse(&hTp->asc[layer], hBs);
+      if (err == TRANSPORTDEC_OK) {
+        int errC;
+
+        errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer]);
+        if (errC != 0) {
+          err = TRANSPORTDEC_PARSE_ERROR;
+        }
+      }
+      break;
   }
 
   if (err == TRANSPORTDEC_OK && fConfigFound) {
@@ -1083,6 +1100,7 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
       break;
 
     case TT_MP4_RAW:
+    case TT_DRM:
       /* One Access Unit was filled into buffer.
          So get the length out of the buffer. */
       hTp->auLength[layer] = FDKgetValidBits(hBs);
@@ -1100,7 +1118,6 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
       }
       break;
 
-    case TT_RSVD50:
     case TT_MP4_ADTS:
     case TT_MP4_LOAS:
       err = transportDec_readStream(hTp, layer);
@@ -1268,8 +1285,13 @@ TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info )
   info += i;
 
   info->module_id  = FDK_TPDEC;
+#ifdef __ANDROID__
+  info->build_date = "";
+  info->build_time = "";
+#else
   info->build_date = __DATE__;
   info->build_time = __TIME__;
+#endif
   info->title      = TP_LIB_TITLE;
   info->version    = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
   LIB_VERSION_STRING(info);
@@ -1279,6 +1301,7 @@ TRANSPORTDEC_ERROR transportDec_GetLibInfo( LIB_INFO *info )
     | CAPF_LATM
     | CAPF_LOAS
     | CAPF_RAWPACKETS
+    | CAPF_DRM
     ;
 
   return TRANSPORTDEC_OK; /* FDKERR_NOERROR; */
@@ -1290,6 +1313,8 @@ int  transportDec_CrcStartReg(HANDLE_TRANSPORTDEC pTp, INT mBits)
   switch (pTp->transportFmt) {
   case TT_MP4_ADTS:
     return adtsRead_CrcStartReg(&pTp->parser.adts, &pTp->bitStream[0], mBits);
+  case TT_DRM:
+    return drmRead_CrcStartReg(&pTp->parser.drm, &pTp->bitStream[0], mBits);
   default:
     return 0;
   }
@@ -1301,6 +1326,9 @@ void transportDec_CrcEndReg(HANDLE_TRANSPORTDEC pTp, INT reg)
   case TT_MP4_ADTS:
     adtsRead_CrcEndReg(&pTp->parser.adts, &pTp->bitStream[0], reg);
     break;
+  case TT_DRM:
+    drmRead_CrcEndReg(&pTp->parser.drm, &pTp->bitStream[0], reg);
+    break;
   default:
     break;
   }
@@ -1317,6 +1345,9 @@ TRANSPORTDEC_ERROR transportDec_CrcCheck(HANDLE_TRANSPORTDEC pTp)
       transportDec_AdjustEndOfAccessUnit(pTp);
     }
     return adtsRead_CrcCheck(&pTp->parser.adts);
+  case TT_DRM:
+    return drmRead_CrcCheck(&pTp->parser.drm);
+    break;
   default:
     return TRANSPORTDEC_OK;
   }

libMpegTPDec/src/version

@@ -2,7 +2,12 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
+#define TP_LIB_VL2 7
 #define TP_LIB_TITLE "MPEG Transport"
+#ifdef __ANDROID__
+#define TP_LIB_BUILD_DATE ""
+#define TP_LIB_BUILD_TIME ""
+#else
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
+#endif

libMpegTPEnc/include/tp_data.h

@@ -146,12 +146,15 @@ typedef struct
 
   UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
 
@@ -324,16 +327,23 @@ int getSamplingRateIndex( UINT samplingRate )
  */
 static inline int getNumberOfTotalChannels(int channelConfig)
 {
-  if (channelConfig > 0 && channelConfig < 8)
-    return (channelConfig == 7)?8:channelConfig;
-  else
+  switch (channelConfig) {
+  case 1: case 2: case 3:
+  case 4: case 5: case 6:
+    return channelConfig;
+  case 7: case 12: case 14:
+    return 8;
+  case 11:
+    return 7;
+  default:
     return 0;
+  }
 }
 
 static inline
 int getNumberOfEffectiveChannels(const int channelConfig)
-{
-  const int n[] = {0,1,2,3,4,5,5,7};
+{        /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0};
   return n[channelConfig];
 }
 

libMpegTPEnc/src/tpenc_latm.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -296,6 +296,7 @@ CreateStreamMuxConfig(
 
   USHORT coreFrameOffset=0;
 
+  hAss->taraBufferFullness  = 0xFF;
   hAss->audioMuxVersionA    = 0; /* for future extensions */
   hAss->streamMuxConfigBits = 0;
 
@@ -339,13 +340,7 @@ CreateStreamMuxConfig(
             hAss->streamMuxConfigBits+=1;
           }
           if( (useSameConfig == 0) || (transLayer==0) ) {
-            UINT bits;
-
-            if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
-            }
-
-            bits = FDKgetValidBits( hBs );
+            const UINT alignAnchor = FDKgetValidBits(hBs);
 
             transportEnc_writeASC(
                     hBs,
@@ -353,19 +348,24 @@ CreateStreamMuxConfig(
                     cb
                     );
 
-            bits = FDKgetValidBits( hBs ) - bits;
-
             if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushBack(hBs, bits+2);
-              hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
+              UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
+              FDKbyteAlign(hBs, alignAnchor);
+              ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
+              FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
+
+              transportEnc_LatmWriteValue(hBs, ascLen);
+
               transportEnc_writeASC(
                       hBs,
                       hAss->config[prog][layer],
                       cb
                       );
+
+              FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */
             }
 
-            hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */
+            hAss->streamMuxConfigBits += FDKgetValidBits(hBs) - alignAnchor; /* add asc length to smc summary */
           }
           transLayer++;
 
@@ -384,7 +384,6 @@ CreateStreamMuxConfig(
           case AOT_ER_AAC_LD     :
           case AOT_ER_AAC_ELD    :
           case AOT_USAC:
-          case AOT_RSVD50:
             p_linfo->frameLengthType = 0;
 
             FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );                        /* frameLengthType */

libMpegTPEnc/src/tpenc_lib.cpp

@@ -619,8 +619,13 @@ TRANSPORTENC_ERROR transportEnc_GetLibInfo( LIB_INFO *info )
   info->module_id = FDK_TPENC;
   info->version = LIB_VERSION(TP_LIB_VL0, TP_LIB_VL1, TP_LIB_VL2);
   LIB_VERSION_STRING(info);
+#ifdef __ANDROID__
+  info->build_date = "";
+  info->build_time = "";
+#else
   info->build_date = __DATE__;
   info->build_time = __TIME__;
+#endif
   info->title = TP_LIB_TITLE;
 
   /* Set flags */

libMpegTPEnc/src/version

@@ -2,7 +2,12 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
+#define TP_LIB_VL2 6
 #define TP_LIB_TITLE "MPEG Transport"
+#ifdef __ANDROID__
+#define TP_LIB_BUILD_DATE ""
+#define TP_LIB_BUILD_TIME ""
+#else
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
+#endif

libPCMutils/include/limiter.h

@@ -0,0 +1,233 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2013 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
+----------------------------------------------------------------------------------------------------------- */
+
+/************************  FDK PCM postprocessor module  *********************
+
+   Author(s):   Matthias Neusinger
+   Description: Hard limiter for clipping prevention
+
+*******************************************************************************/
+
+#ifndef _LIMITER_H_
+#define _LIMITER_H_
+
+
+#include "common_fix.h"
+
+#define TDL_ATTACK_DEFAULT_MS      (15)              /* default attack  time in ms */
+#define TDL_RELEASE_DEFAULT_MS     (50)              /* default release time in ms */
+
+#define TDL_GAIN_SCALING           (15)              /* scaling of gain value. */
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+typedef enum {
+  TDLIMIT_OK = 0,
+
+  __error_codes_start = -100,
+
+  TDLIMIT_INVALID_HANDLE,
+  TDLIMIT_INVALID_PARAMETER,
+
+  __error_codes_end
+} TDLIMITER_ERROR;
+
+struct TDLimiter;
+typedef struct TDLimiter* TDLimiterPtr;
+
+/******************************************************************************
+* createLimiter                                                               *
+* maxAttackMs:   maximum and initial attack/lookahead time in milliseconds    *
+* releaseMs:     release time in milliseconds (90% time constant)             *
+* threshold:     limiting threshold                                           *
+* maxChannels:   maximum and initial number of channels                       *
+* maxSampleRate: maximum and initial sampling rate in Hz                      *
+* returns:       limiter handle                                               *
+******************************************************************************/
+TDLimiterPtr createLimiter(unsigned int  maxAttackMs,
+                           unsigned int  releaseMs,
+                           INT_PCM       threshold,
+                           unsigned int  maxChannels,
+                           unsigned int  maxSampleRate);
+
+
+/******************************************************************************
+* resetLimiter                                                                *
+* limiter: limiter handle                                                     *
+* returns: error code                                                         *
+******************************************************************************/
+TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter);
+
+
+/******************************************************************************
+* destroyLimiter                                                              *
+* limiter: limiter handle                                                     *
+* returns: error code                                                         *
+******************************************************************************/
+TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter);
+
+/******************************************************************************
+* applyLimiter                                                                *
+* limiter:    limiter handle                                                  *
+* pGain :     pointer to gains to be applied to the signal before limiting,   *
+*             which are downscaled by TDL_GAIN_SCALING bit.                   *
+*             These gains are delayed by gain_delay, and smoothed.            *
+*             Smoothing is done by a butterworth lowpass filter with a cutoff *
+*             frequency which is fixed with respect to the sampling rate.     *
+*             It is a substitute for the smoothing due to windowing and       *
+*             overlap/add, if a gain is applied in frequency domain.          *
+* gain_scale: pointer to scaling exponents to be applied to the signal before *
+*             limiting, without delay and without smoothing                   *
+* gain_size:  number of elements in pGain, currently restricted to 1          *
+* gain_delay: delay [samples] with which the gains in pGain shall be applied  *
+*             gain_delay <= nSamples                                          *
+* samples:    input/output buffer containing interleaved samples              *
+*             precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits   *
+* nSamples:   number of samples per channel                                   *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
+                 INT_PCM*    samples,
+                 FIXP_DBL*    pGain,
+                 const INT*   gain_scale,
+                 const UINT   gain_size,
+                 const UINT   gain_delay,
+                 const UINT   nSamples);
+
+/******************************************************************************
+* getLimiterDelay                                                             *
+* limiter: limiter handle                                                     *
+* returns: exact delay caused by the limiter in samples                       *
+******************************************************************************/
+unsigned int getLimiterDelay(TDLimiterPtr limiter);
+
+/******************************************************************************
+* setLimiterNChannels                                                         *
+* limiter:   limiter handle                                                   *
+* nChannels: number of channels ( <= maxChannels specified on create)         *
+* returns:   error code                                                       *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels);
+
+/******************************************************************************
+* setLimiterSampleRate                                                        *
+* limiter:    limiter handle                                                  *
+* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create)     *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate);
+
+/******************************************************************************
+* setLimiterAttack                                                            *
+* limiter:    limiter handle                                                  *
+* attackMs:   attack time in ms ( <= maxAttackMs specified on create)         *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs);
+
+/******************************************************************************
+* setLimiterRelease                                                           *
+* limiter:    limiter handle                                                  *
+* releaseMs:  release time in ms                                              *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs);
+
+/******************************************************************************
+* setLimiterThreshold                                                         *
+* limiter:    limiter handle                                                  *
+* threshold:  limiter threshold                                               *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold);
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif  //#ifndef _LIMITER_H_

libPCMutils/include/pcmutils_lib.h

@@ -95,69 +95,88 @@ amm-info@iis.fraunhofer.de
 #include "machine_type.h"
 #include "common_fix.h"
 #include "FDK_audio.h"
+#include "FDK_bitstream.h"
 
-/* ------------------------ *
- *     MODULE SETTINGS:     *
- * ------------------------ */
-/* #define PCM_UPMIX_ENABLE */       /*!< Generally enable up mixing.                           */
-#define PCM_DOWNMIX_ENABLE           /*!< Generally enable down mixing.                         */
-#define DVB_MIXDOWN_ENABLE           /*!< Enable this to support DVB ancillary data for encoder
-                                          assisted downmixing of MPEG-4 AAC and
-                                          MPEG-1/2 layer 2 streams. PCM_DOWNMIX_ENABLE has to
-                                          be enabled, too!                                      */
-#define MPEG_PCE_MIXDOWN_ENABLE      /*!< Enable this to support MPEG matrix mixdown with a
-                                          coefficient carried in the PCE. PCM_DOWNMIX_ENABLE
-                                          has to be enabled, too!                               */
-/* #define ARIB_MIXDOWN_ENABLE */    /*!< Enable modifications to the MPEG PCE mixdown method
-                                          to fulfill ARIB standard. MPEG_PCE_MIXDOWN_ENABLE has
-                                          to be set.                                            */
 
 /* ------------------------ *
  *     ERROR CODES:         *
  * ------------------------ */
 typedef enum
 {
-  PCMDMX_OK              = 0x0,    /*!< No error happened.                                     */
-  PCMDMX_OUT_OF_MEMORY   = 0x2,    /*!< Not enough memory to set up an instance of the module. */
-  PCMDMX_UNKNOWN         = 0x5,    /*!< Error condition is of unknown reason, or from a third
-                                          party module.                                        */
-  PCMDMX_INVALID_HANDLE,           /*!< The given instance handle is not valid.                */
-  PCMDMX_INVALID_ARGUMENT,         /*!< One of the parameters handed over is invalid.          */
-  PCMDMX_INVALID_CH_CONFIG,        /*!< The given channel configuration is not supported and
-                                          thus no processing was performed.                    */
-  PCMDMX_INVALID_MODE,             /*!< The set configuration/mode is not applicable.          */
-  PCMDMX_UNKNOWN_PARAM,            /*!< The handed parameter is not known/supported.           */
-  PCMDMX_UNABLE_TO_SET_PARAM,      /*!< Unable to set the specific parameter. Most probably
-                                          the value ist out of range.                          */
-  PCMDMX_CORRUPT_ANC_DATA          /*!< The read ancillary data was corrupt.                   */
+  PCMDMX_OK              = 0x0,   /*!< No error happened.                                        */
+
+  pcm_dmx_fatal_error_start,
+  PCMDMX_OUT_OF_MEMORY   = 0x2,   /*!< Not enough memory to set up an instance of the module.    */
+  PCMDMX_UNKNOWN         = 0x5,   /*!< Error condition is of unknown reason, or from a third
+                                       party module.                                             */
+  pcm_dmx_fatal_error_end,
+
+  PCMDMX_INVALID_HANDLE,          /*!< The given instance handle is not valid.                   */
+  PCMDMX_INVALID_ARGUMENT,        /*!< One of the parameters handed over is invalid.             */
+  PCMDMX_INVALID_CH_CONFIG,       /*!< The given channel configuration is not supported and thus
+                                       no processing was performed.                              */
+  PCMDMX_INVALID_MODE,            /*!< The set configuration/mode is not applicable.             */
+  PCMDMX_UNKNOWN_PARAM,           /*!< The handed parameter is not known/supported.              */
+  PCMDMX_UNABLE_TO_SET_PARAM,     /*!< Unable to set the specific parameter. Most probably the
+                                       value ist out of range.                                   */
+  PCMDMX_CORRUPT_ANC_DATA         /*!< The read ancillary data was corrupt.                      */
 
 } PCMDMX_ERROR;
 
+/** Macro to identify fatal errors. */
+#define PCMDMX_IS_FATAL_ERROR(err)   ( (((err)>=pcm_dmx_fatal_error_start)   && ((err)<=pcm_dmx_fatal_error_end))   ? 1 : 0)
 
 /* ------------------------ *
  *     RUNTIME PARAMS:      *
  * ------------------------ */
 typedef enum
 {
-  DMX_BS_DATA_EXPIRY_FRAME,          /*!< The number of frames without new metadata that have to
-                                            go by before the bitstream data expires. The value 0
-                                            disables expiry.                                     */
-  DMX_BS_DATA_DELAY,                 /*!< The number of delay frames of the output samples
-                                            compared to the bitstream data.                      */
-  NUMBER_OF_OUTPUT_CHANNELS ,        /*!< The number of output channels (equals the number of
-                                            channels processed by the audio output setup).       */
-  DUAL_CHANNEL_DOWNMIX_MODE          /*!< Downmix mode for two channel audio data.               */
-
+  DMX_BS_DATA_EXPIRY_FRAME,       /*!< The number of frames without new metadata that have to go
+                                       by before the bitstream data expires. The value 0 disables
+                                       expiry.                                                   */
+  DMX_BS_DATA_DELAY,              /*!< The number of delay frames of the output samples compared
+                                       to the bitstream data.                                    */
+  MIN_NUMBER_OF_OUTPUT_CHANNELS,  /*!< The minimum number of output channels. For all input
+                                       configurations that have less than the given channels the
+                                       module will modify the output automatically to obtain the
+                                       given number of output channels. Mono signals will be
+                                       duplicated. If more than two output channels are desired
+                                       the module just adds empty channels. The parameter value
+                                       must be either -1, 0, 1, 2, 6 or 8. If the value is
+                                       greater than zero and exceeds the value of parameter
+                                       MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to
+                                       the same value. Both values -1 and 0 disable the feature. */
+  MAX_NUMBER_OF_OUTPUT_CHANNELS,  /*!< The maximum number of output channels. For all input
+                                       configurations that have more than the given channels the
+                                       module will apply a mixdown automatically to obtain the
+                                       given number of output channels. The value must be either
+                                       -1, 0, 1, 2, 6 or 8. If it is greater than zero and lower
+                                       or equal than the value of MIN_NUMBER_OF_OUTPUT_CHANNELS
+                                       parameter the latter will be set to the same value.
+                                       The values -1 and 0 disable the feature.                  */
+  DMX_DUAL_CHANNEL_MODE,          /*!< Downmix mode for two channel audio data.                  */
+  DMX_PSEUDO_SURROUND_MODE        /*!< Defines how module handles pseudo surround compatible
+                                       signals. See PSEUDO_SURROUND_MODE type for details.       */
 } PCMDMX_PARAM;
 
+/* Parameter value types */
+typedef enum
+{
+  NEVER_DO_PS_DMX = -1,           /*!< Never create a pseudo surround compatible downmix.        */
+  AUTO_PS_DMX     =  0,           /*!< Create a pseudo surround compatible downmix only if
+                                       signalled in bitstreams meta data. (Default)              */
+  FORCE_PS_DMX    =  1            /*!< Always create a pseudo surround compatible downmix.
+                                       CAUTION: This can lead to excessive signal cancellations
+                                       and signal level differences for non-compatible signals.  */
+} PSEUDO_SURROUND_MODE;
 
 typedef enum
 {
-  STEREO_MODE = 0x0,           /*!< Leave stereo signals as they are.                            */
-  CH1_MODE    = 0x1,           /*!< Create a dual mono output signal from channel 1.             */
-  CH2_MODE    = 0x2,           /*!< Create a dual mono output signal from channel 2.             */
-  MIXED_MODE  = 0x3            /*!< Create a dual mono output signal by mixing the two channels. */
-
+  STEREO_MODE = 0x0,              /*!< Leave stereo signals as they are.                         */
+  CH1_MODE    = 0x1,              /*!< Create a dual mono output signal from channel 1.          */
+  CH2_MODE    = 0x2,              /*!< Create a dual mono output signal from channel 2.          */
+  MIXED_MODE  = 0x3               /*!< Create a dual mono output signal by mixing the two
+                                       channels.                                                 */
 } DUAL_CHANNEL_MODE;
 
 
@@ -178,7 +197,7 @@ extern "C"
 
 /** Open and initialize an instance of the PCM downmix module
  * @param [out] Pointer to a buffer receiving the handle of the new instance.
- * @returns Returns an error code.
+ * @returns     Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_Open (
     HANDLE_PCM_DOWNMIX *pSelf
@@ -188,20 +207,46 @@ PCMDMX_ERROR pcmDmx_Open (
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Parameter to be set.
  * @param [in] Parameter value.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_SetParam (
     HANDLE_PCM_DOWNMIX  self,
-    PCMDMX_PARAM        param,
-    UINT                value
+    const PCMDMX_PARAM  param,
+    const INT           value
   );
 
-/** Read the ancillary data transported in DSEs of DVB streams with MPEG-4 content
+/** Get one parameter value of one PCM downmix module instance.
+ * @param [in] Handle of PCM downmix module instance.
+ * @param [in] Parameter to be set.
+ * @param [out] Pointer to buffer receiving the parameter value.
+ * @returns Returns an error code.
+ **/
+PCMDMX_ERROR pcmDmx_GetParam (
+    HANDLE_PCM_DOWNMIX  self,
+    const PCMDMX_PARAM  param,
+    INT * const         pValue
+  );
+
+/** Read downmix meta-data directly from a given bitstream.
+ * @param [in] Handle of PCM downmix instance.
+ * @param [in] Handle of FDK bitstream buffer.
+ * @param [in] Length of ancillary data in bits.
+ * @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
+ * @returns    Returns an error code.
+ **/
+PCMDMX_ERROR pcmDmx_Parse (
+    HANDLE_PCM_DOWNMIX  self,
+    HANDLE_FDK_BITSTREAM  hBitStream,
+    UINT   ancDataBits,
+    int    isMpeg2
+  );
+
+/** Read downmix meta-data from a given data buffer.
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Pointer to ancillary data buffer.
- * @param [in] Size of ancillary data.
+ * @param [in] Size of ancillary data in bytes.
  * @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_ReadDvbAncData (
     HANDLE_PCM_DOWNMIX  self,
@@ -211,12 +256,11 @@ PCMDMX_ERROR pcmDmx_ReadDvbAncData (
   );
 
 /** Set the matrix mixdown information extracted from the PCE of an AAC bitstream.
- *  Note: Call only if matrix_mixdown_idx_present is true.
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Matrix mixdown index present flag extracted from PCE.
  * @param [in] The 2 bit matrix mixdown index extracted from PCE.
  * @param [in] The pseudo surround enable flag extracted from PCE.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce (
     HANDLE_PCM_DOWNMIX  self,
@@ -235,34 +279,42 @@ PCMDMX_ERROR pcmDmx_Reset (
     UINT                flags
   );
 
-/** Apply down or up mixing.
+/** Create a mixdown, bypass or extend the output signal depending on the modules settings and the
+ *  respective given input configuration.
  *
  * \param [in]    Handle of PCM downmix module instance.
  * \param [inout] Pointer to time buffer with decoded PCM samples.
- * \param [in]    Pointer where the amount of output samples is returned into.
- * \param [inout] Pointer where the amount of output channels is returned into.
- * \param [in]    Flag which indicates if output time data are writtern interleaved or as subsequent blocks.
- * \param [inout] Array were the corresponding channel type for each output audio channel is stored into.
- * \param [inout] Array were the corresponding channel type index for each output audio channel is stored into.
- * \param [in]    Array containing the output channel mapping to be used (From MPEG PCE ordering to whatever is required).
- *
- * @returns Returns an error code.
+ * \param [in]    The I/O block size which is the number of samples per channel.
+ * \param [inout] Pointer to buffer that holds the number of input channels and where the
+ *                amount of output channels is written to.
+ * \param [in]    Flag which indicates if output time data is writtern interleaved or as
+ *                subsequent blocks.
+ * \param [inout] Array were the corresponding channel type for each output audio channel is
+ *                stored into.
+ * \param [inout] Array were the corresponding channel type index for each output audio channel
+ *                is stored into.
+ * \param [in]    Array containing the output channel mapping to be used (from MPEG PCE ordering
+ *                to whatever is required).
+ * \param [out]   Pointer on a field receiving the scale factor that has to be applied on all
+ *                samples afterwards. If the handed pointer is NULL the final scaling is done
+ *                internally.
+ * @returns       Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_ApplyFrame (
     HANDLE_PCM_DOWNMIX      self,
     INT_PCM                *pPcmBuf,
     UINT                    frameSize,
     INT                    *nChannels,
-
     int                     fInterleaved,
     AUDIO_CHANNEL_TYPE      channelType[],
     UCHAR                   channelIndices[],
-    const UCHAR             channelMapping[][8]
+    const UCHAR             channelMapping[][8],
+    INT                    *pDmxOutScale
   );
 
 /** Close an instance of the PCM downmix module.
  * @param [inout] Pointer to a buffer containing the handle of the instance.
- * @returns Returns an error code.
+ * @returns       Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_Close (
     HANDLE_PCM_DOWNMIX *pSelf
@@ -270,7 +322,7 @@ PCMDMX_ERROR pcmDmx_Close (
 
 /** Get library info for this module.
  * @param [out] Pointer to an allocated LIB_INFO structure.
- * @returns Returns an error code.
+ * @returns     Returns an error code.
  */
 PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info );
 

libPCMutils/src/limiter.cpp

@@ -0,0 +1,498 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2013 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
+----------------------------------------------------------------------------------------------------------- */
+
+/************************  FDK PCM postprocessor module  *********************
+
+   Author(s):   Matthias Neusinger
+   Description: Hard limiter for clipping prevention
+
+*******************************************************************************/
+
+#include "limiter.h"
+
+
+struct TDLimiter {
+  unsigned int  attack;
+  FIXP_DBL      attackConst, releaseConst;
+  unsigned int  attackMs, releaseMs, maxAttackMs;
+  FIXP_PCM      threshold;
+  unsigned int  channels, maxChannels;
+  unsigned int  sampleRate, maxSampleRate;
+  FIXP_DBL      cor, max;
+  FIXP_DBL*     maxBuf;
+  FIXP_DBL*     delayBuf;
+  unsigned int  maxBufIdx, delayBufIdx;
+  FIXP_DBL      smoothState0;
+  FIXP_DBL      minGain;
+
+  FIXP_DBL      additionalGainPrev;
+  FIXP_DBL      additionalGainFilterState;
+  FIXP_DBL      additionalGainFilterState1;
+};
+
+/* create limiter */
+TDLimiterPtr createLimiter(
+                           unsigned int  maxAttackMs,
+                           unsigned int  releaseMs,
+                           INT_PCM       threshold,
+                           unsigned int  maxChannels,
+                           unsigned int  maxSampleRate
+                           )
+{
+  TDLimiterPtr limiter = NULL;
+  unsigned int attack, release;
+  FIXP_DBL attackConst, releaseConst, exponent;
+  INT e_ans;
+
+  /* calc attack and release time in samples */
+  attack = (unsigned int)(maxAttackMs * maxSampleRate / 1000);
+  release = (unsigned int)(releaseMs * maxSampleRate / 1000);
+
+  /* alloc limiter struct */
+  limiter = (TDLimiterPtr)FDKcalloc(1, sizeof(struct TDLimiter));
+  if (!limiter) return NULL;
+
+  /* alloc max and delay buffers */
+  limiter->maxBuf   = (FIXP_DBL*)FDKcalloc(attack + 1, sizeof(FIXP_DBL));
+  limiter->delayBuf = (FIXP_DBL*)FDKcalloc(attack * maxChannels, sizeof(FIXP_DBL));
+
+  if (!limiter->maxBuf || !limiter->delayBuf) {
+    destroyLimiter(limiter);
+    return NULL;
+  }
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  /* init parameters */
+  limiter->attackMs      = maxAttackMs;
+  limiter->maxAttackMs   = maxAttackMs;
+  limiter->releaseMs     = releaseMs;
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->releaseConst  = releaseConst;
+  limiter->threshold     = (FIXP_PCM)threshold;
+  limiter->channels      = maxChannels;
+  limiter->maxChannels   = maxChannels;
+  limiter->sampleRate    = maxSampleRate;
+  limiter->maxSampleRate = maxSampleRate;
+
+  resetLimiter(limiter);
+
+  return limiter;
+}
+
+
+/* reset limiter */
+TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter)
+{
+  if (limiter != NULL) {
+
+    limiter->maxBufIdx = 0;
+    limiter->delayBufIdx = 0;
+    limiter->max = (FIXP_DBL)0;
+    limiter->cor = FL2FXCONST_DBL(1.0f/(1<<1));
+    limiter->smoothState0 = FL2FXCONST_DBL(1.0f/(1<<1));
+    limiter->minGain = FL2FXCONST_DBL(1.0f/(1<<1));
+
+    limiter->additionalGainPrev = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+    limiter->additionalGainFilterState = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+    limiter->additionalGainFilterState1 = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+
+    FDKmemset(limiter->maxBuf,   0, (limiter->attack + 1) * sizeof(FIXP_DBL) );
+    FDKmemset(limiter->delayBuf, 0, limiter->attack * limiter->channels * sizeof(FIXP_DBL) );
+  }
+  else {
+    return TDLIMIT_INVALID_HANDLE;
+  }
+
+  return TDLIMIT_OK;
+}
+
+
+/* destroy limiter */
+TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter)
+{
+  if (limiter != NULL) {
+    FDKfree(limiter->maxBuf);
+    FDKfree(limiter->delayBuf);
+
+    FDKfree(limiter);
+  }
+  else {
+    return TDLIMIT_INVALID_HANDLE;
+  }
+  return TDLIMIT_OK;
+}
+
+/* apply limiter */
+TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
+                 INT_PCM*    samples,
+                 FIXP_DBL*    pGain,
+                 const INT*   gain_scale,
+                 const UINT   gain_size,
+                 const UINT   gain_delay,
+                 const UINT   nSamples)
+{
+  unsigned int i, j;
+  FIXP_PCM tmp1, tmp2;
+  FIXP_DBL tmp, old, gain, additionalGain, additionalGainUnfiltered;
+  FIXP_DBL minGain = FL2FXCONST_DBL(1.0f/(1<<1));
+
+  FDK_ASSERT(gain_size == 1);
+  FDK_ASSERT(gain_delay <= nSamples);
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  {
+    unsigned int channels       = limiter->channels;
+    unsigned int attack         = limiter->attack;
+    FIXP_DBL     attackConst    = limiter->attackConst;
+    FIXP_DBL     releaseConst   = limiter->releaseConst;
+    FIXP_DBL     threshold      = FX_PCM2FX_DBL(limiter->threshold)>>TDL_GAIN_SCALING;
+
+    FIXP_DBL     max            = limiter->max;
+    FIXP_DBL*    maxBuf         = limiter->maxBuf;
+    unsigned int maxBufIdx      = limiter->maxBufIdx;
+    FIXP_DBL     cor            = limiter->cor;
+    FIXP_DBL*    delayBuf       = limiter->delayBuf;
+    unsigned int delayBufIdx    = limiter->delayBufIdx;
+
+    FIXP_DBL     smoothState0   = limiter->smoothState0;
+    FIXP_DBL     additionalGainSmoothState = limiter->additionalGainFilterState;
+    FIXP_DBL     additionalGainSmoothState1 = limiter->additionalGainFilterState1;
+
+    for (i = 0; i < nSamples; i++) {
+
+      if (i < gain_delay) {
+        additionalGainUnfiltered = limiter->additionalGainPrev;
+      } else {
+        additionalGainUnfiltered = pGain[0];
+      }
+
+      /* Smooth additionalGain */
+      /* [b,a] = butter(1, 0.01) */
+      static const FIXP_SGL b[] = { FL2FXCONST_SGL(0.015466*2.0), FL2FXCONST_SGL( 0.015466*2.0) };
+      static const FIXP_SGL a[] = { FL2FXCONST_SGL(1.000000), FL2FXCONST_SGL(-0.96907) };
+      /* [b,a] = butter(1, 0.001) */
+      //static const FIXP_SGL b[] = { FL2FXCONST_SGL(0.0015683*2.0), FL2FXCONST_SGL( 0.0015683*2.0) };
+      //static const FIXP_SGL a[] = { FL2FXCONST_SGL(1.0000000), FL2FXCONST_SGL(-0.99686) };
+      additionalGain = - fMult(additionalGainSmoothState, a[1]) + fMultDiv2( additionalGainUnfiltered, b[0]) + fMultDiv2(additionalGainSmoothState1, b[1]);
+      additionalGainSmoothState1 = additionalGainUnfiltered;
+      additionalGainSmoothState = additionalGain;
+
+      /* Apply the additional scaling that has no delay and no smoothing */
+      if (gain_scale[0] > 0) {
+        additionalGain <<= gain_scale[0];
+      } else {
+        additionalGain >>= gain_scale[0];
+      }
+
+      /* get maximum absolute sample value of all channels, including the additional gain. */
+      tmp1 = (FIXP_PCM)0;
+      for (j = 0; j < channels; j++) {
+          tmp2 = (FIXP_PCM)samples[i * channels + j];
+          if (tmp2 == (FIXP_PCM)SAMPLE_MIN) /* protect fAbs from -1.0 value */
+              tmp2 = (FIXP_PCM)(SAMPLE_MIN+1);
+        tmp1 = fMax(tmp1, fAbs(tmp2));
+      }
+      tmp = SATURATE_LEFT_SHIFT(fMultDiv2(tmp1, additionalGain), 1, DFRACT_BITS);
+
+      /* set threshold as lower border to save calculations in running maximum algorithm */
+      tmp = fMax(tmp, threshold);
+
+      /* running maximum */
+      old = maxBuf[maxBufIdx];
+      maxBuf[maxBufIdx] = tmp;
+
+      if (tmp >= max) {
+        /* new sample is greater than old maximum, so it is the new maximum */
+        max = tmp;
+      }
+      else if (old < max) {
+        /* maximum does not change, as the sample, which has left the window was
+           not the maximum */
+      }
+      else {
+        /* the old maximum has left the window, we have to search the complete
+           buffer for the new max */
+        max = maxBuf[0];
+        for (j = 1; j <= attack; j++) {
+          if (maxBuf[j] > max) max = maxBuf[j];
+        }
+      }
+      maxBufIdx++;
+      if (maxBufIdx >= attack+1) maxBufIdx = 0;
+
+      /* calc gain */
+      /* gain is downscaled by one, so that gain = 1.0 can be represented */
+      if (max > threshold) {
+        gain = fDivNorm(threshold, max)>>1;
+      }
+      else {
+        gain = FL2FXCONST_DBL(1.0f/(1<<1));
+      }
+
+      /* gain smoothing, method: TDL_EXPONENTIAL */
+      /* first order IIR filter with attack correction to avoid overshoots */
+
+      /* correct the 'aiming' value of the exponential attack to avoid the remaining overshoot */
+      if (gain < smoothState0) {
+        cor = fMin(cor, fMultDiv2((gain - fMultDiv2(FL2FXCONST_SGL(0.1f*(1<<1)),smoothState0)), FL2FXCONST_SGL(1.11111111f/(1<<1)))<<2);
+      }
+      else {
+        cor = gain;
+      }
+
+      /* smoothing filter */
+      if (cor < smoothState0) {
+        smoothState0 = fMult(attackConst,(smoothState0 - cor)) + cor;  /* attack */
+        smoothState0 = fMax(smoothState0, gain); /* avoid overshooting target */
+      }
+      else {
+        /* sign inversion twice to round towards +infinity,
+           so that gain can converge to 1.0 again,
+           for bit-identical output when limiter is not active */
+        smoothState0 = -fMult(releaseConst,-(smoothState0 - cor)) + cor; /* release */
+      }
+
+      gain = smoothState0;
+
+      /* lookahead delay, apply gain */
+      for (j = 0; j < channels; j++) {
+
+        tmp = delayBuf[delayBufIdx * channels + j];
+        delayBuf[delayBufIdx * channels + j] = fMult((FIXP_PCM)samples[i * channels + j], additionalGain);
+
+        /* Apply gain to delayed signal */
+        if (gain < FL2FXCONST_DBL(1.0f/(1<<1)))
+          tmp = fMult(tmp,gain<<1);
+
+        samples[i * channels + j] = FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(tmp,TDL_GAIN_SCALING,DFRACT_BITS));
+      }
+      delayBufIdx++;
+      if (delayBufIdx >= attack) delayBufIdx = 0;
+
+      /* save minimum gain factor */
+      if (gain < minGain) minGain = gain;
+    }
+
+
+    limiter->max = max;
+    limiter->maxBufIdx = maxBufIdx;
+    limiter->cor = cor;
+    limiter->delayBufIdx = delayBufIdx;
+
+    limiter->smoothState0 = smoothState0;
+    limiter->additionalGainFilterState = additionalGainSmoothState;
+    limiter->additionalGainFilterState1 = additionalGainSmoothState1;
+
+    limiter->minGain = minGain;
+
+    limiter->additionalGainPrev = pGain[0];
+
+    return TDLIMIT_OK;
+  }
+}
+
+/* get delay in samples */
+unsigned int getLimiterDelay(TDLimiterPtr limiter)
+{
+  FDK_ASSERT(limiter != NULL);
+  return limiter->attack;
+}
+
+/* set number of channels */
+TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels)
+{
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (nChannels > limiter->maxChannels) return TDLIMIT_INVALID_PARAMETER;
+
+  limiter->channels = nChannels;
+  //resetLimiter(limiter);
+
+  return TDLIMIT_OK;
+}
+
+/* set sampling rate */
+TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate)
+{
+  unsigned int attack, release;
+  FIXP_DBL attackConst, releaseConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER;
+
+  /* update attack and release time in samples */
+  attack = (unsigned int)(limiter->attackMs * sampleRate / 1000);
+  release = (unsigned int)(limiter->releaseMs * sampleRate / 1000);
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->releaseConst  = releaseConst;
+  limiter->sampleRate    = sampleRate;
+
+  /* reset */
+  //resetLimiter(limiter);
+
+  return TDLIMIT_OK;
+}
+
+/* set attack time */
+TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs)
+{
+  unsigned int attack;
+  FIXP_DBL attackConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (attackMs > limiter->maxAttackMs) return TDLIMIT_INVALID_PARAMETER;
+
+  /* calculate attack time in samples */
+  attack = (unsigned int)(attackMs * limiter->sampleRate / 1000);
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->attackMs      = attackMs;
+
+  return TDLIMIT_OK;
+}
+
+/* set release time */
+TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs)
+{
+  unsigned int release;
+  FIXP_DBL releaseConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  /* calculate  release time in samples */
+  release = (unsigned int)(releaseMs * limiter->sampleRate / 1000);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  limiter->releaseConst  = releaseConst;
+  limiter->releaseMs     = releaseMs;
+
+  return TDLIMIT_OK;
+}
+
+/* set limiter threshold */
+TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold)
+{
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  limiter->threshold = (FIXP_PCM)threshold;
+
+  return TDLIMIT_OK;
+}

libSBRdec/include/sbrdecoder.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -145,6 +145,8 @@ typedef enum
   SBR_SYSTEM_BITSTREAM_DELAY,          /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */
   SBR_QMF_MODE,                        /*!< Set QMF mode, either complex or low power. */
   SBR_LD_QMF_TIME_ALIGN,               /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */
+  SBR_FLUSH_DATA,                      /*!< Set internal state to flush the decoder with the next process call. */
+  SBR_CLEAR_HISTORY,                   /*!< Clear all internal states (delay lines, QMF states, ...). */
   SBR_BS_INTERRUPTION                  /*!< Signal bit stream interruption. Value is ignored. */
 } SBRDEC_PARAM;
 
@@ -260,6 +262,7 @@ void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
  *         into *count if a payload length is given (byPayLen > 0). If no SBR payload length is
  *         given (bsPayLen < 0) then the bit stream position on return will be random after this
  *         function call in case of errors, and any further decoding will be completely pointless.
+ *         This function accepts either normal ordered SBR data or reverse ordered DRM SBR data.
  *
  * \param self           SBR decoder handle.
  * \param hBs            Bit stream handle as data source.
@@ -308,7 +311,7 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER    self,
                              INT_PCM             *timeData,
                              int                 *numChannels,
                              int                 *sampleRate,
-                             const UCHAR          channelMapping[(6)],
+                             const UCHAR          channelMapping[(8)],
                              const int            interleaved,
                              const int            coreDecodedOk,
                              UCHAR               *psDecoded );
@@ -329,6 +332,13 @@ SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *self );
  */
 INT sbrDecoder_GetLibInfo( LIB_INFO *info );    
 
+/**
+ * \brief       Determine the modules output signal delay in samples.
+ * \param self  SBR decoder handle.
+ * \return      The number of samples signal delay added by the module.
+ */
+UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self );
+
 
 #ifdef __cplusplus
 }

libSBRdec/src/arm/lpp_tran_arm.cpp

@@ -97,7 +97,10 @@ amm-info@iis.fraunhofer.de
 #ifdef FUNCTION_LPPTRANSPOSER_func1
 
 /* Note: This code requires only 43 cycles per iteration instead of 61 on ARM926EJ-S */
-__attribute__ ((noinline)) static void lppTransposer_func1(
+#ifdef __GNUC__
+__attribute__ ((noinline))
+#endif
+static void lppTransposer_func1(
   FIXP_DBL *lowBandReal,
   FIXP_DBL *lowBandImag,
   FIXP_DBL **qmfBufferReal,

libSBRdec/src/env_calc.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -151,13 +151,13 @@ typedef struct
 }
 ENV_CALC_NRGS;
 
-/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
+static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
                                   SCHAR   *filtBuffer_e,
                                   FIXP_DBL *NrgGain,
                                   SCHAR   *NrgGain_e,
                                   int    subbands);
 
-/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
+static void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
                               FIXP_DBL  **analysBufferImag,
                               int       lowSubband, int highSubband,
                               int       start_pos,  int next_pos,
@@ -165,7 +165,7 @@ ENV_CALC_NRGS;
                               FIXP_DBL *nrgEst,
                               SCHAR    *nrgEst_e );
 
-/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
+static void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
                           FIXP_DBL  **analysBufferImag,
                           int       nSfb,
                           UCHAR    *freqBandTable,
@@ -174,13 +174,13 @@ ENV_CALC_NRGS;
                           FIXP_DBL *nrg_est,
                           SCHAR    *nrg_est_e );
 
-/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
+static void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
                             FIXP_DBL  tmpNoise, SCHAR tmpNoise_e,
                             UCHAR     sinePresentFlag,
                             UCHAR     sineMapped,
                             int       noNoiseFlag);
 
-/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+static void calcAvgGain(ENV_CALC_NRGS* nrgs,
                         int        lowSubband,
                         int        highSubband,
                         FIXP_DBL  *sumRef_m,
@@ -188,7 +188,7 @@ ENV_CALC_NRGS;
                         FIXP_DBL  *ptrAvgGain_m,
                         SCHAR     *ptrAvgGain_e);
 
-/*static*/ void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
+static void adjustTimeSlot_EldGrid(FIXP_DBL  *ptrReal,
                            ENV_CALC_NRGS* nrgs,
                            UCHAR *ptrHarmIndex,
                            int    lowSubbands,
@@ -196,8 +196,17 @@ ENV_CALC_NRGS;
                            int    scale_change,
                            int    noNoiseFlag,
                            int   *ptrPhaseIndex,
-                           int    fCldfb);
-/*static*/ void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
+                           int    scale_diff_low);
+
+static void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
+                           ENV_CALC_NRGS* nrgs,
+                           UCHAR *ptrHarmIndex,
+                           int    lowSubbands,
+                           int    noSubbands,
+                           int    scale_change,
+                           int    noNoiseFlag,
+                           int   *ptrPhaseIndex);
+static void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
                            FIXP_DBL  *ptrImag,
                            HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                            ENV_CALC_NRGS* nrgs,
@@ -224,7 +233,7 @@ ENV_CALC_NRGS;
   Additionally, the flags in harmFlagsPrev are being updated by this function
   for the next frame.
 */
-/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
+static void mapSineFlags(UCHAR *freqBandTable,         /*!< Band borders (there's only 1 flag per band) */
                          int nSfb,                     /*!< Number of bands in the table */
                          UCHAR *addHarmonics,           /*!< vector with 1 flag per sfb */
                          int *harmFlagsPrev,           /*!< Packed 'addHarmonics' */
@@ -990,7 +999,6 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
           /* Prevent the smoothing filter from running on constant levels */
           if (j-start_pos < smooth_length)
             smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];
-
           else
             smooth_ratio = FL2FXCONST_SGL(0.0f);
 
@@ -1007,7 +1015,8 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
         }
         else
         {
-          adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+          if (flags & SBRDEC_ELD_GRID) {
+            adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband],
                            pNrgs,
                           &h_sbr_cal_env->harmIndex,
                            lowSubband,
@@ -1015,7 +1024,18 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
                            scale_change,
                            noNoiseFlag,
                           &h_sbr_cal_env->phaseIndex,
-                           (flags & SBRDEC_ELD_GRID));
+                           EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale);
+          } else
+          {
+            adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+                           pNrgs,
+                          &h_sbr_cal_env->harmIndex,
+                           lowSubband,
+                           noSubbands,
+                           scale_change,
+                           noNoiseFlag,
+                          &h_sbr_cal_env->phaseIndex);
+          }
         }
       } // for
 
@@ -1176,7 +1196,7 @@ resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to env
   can be performed.
   This function is called once for each envelope before adjusting.
 */
-/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
+static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
                                   SCHAR    *filtBuffer_e,   /*!< exponents of bufferd gains */
                                   FIXP_DBL *nrgGain,        /*!< gains for current envelope */
                                   SCHAR    *nrgGain_e,      /*!< exponents of gains for current envelope */
@@ -1331,7 +1351,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   This function is used when interpolFreq is true.
 */
-/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
+static void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
                               FIXP_DBL  **analysBufferImag, /*!< Imaginary part of subband samples */
                               int       lowSubband,           /*!< Begin of the SBR frequency range */
                               int       highSubband,          /*!< High end of the SBR frequency range */
@@ -1452,7 +1472,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   This function is used when interpolFreq is false.
 */
-/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
+static void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
                           FIXP_DBL  **analysBufferImag,  /*!< Imaginary part of subband samples */
                           int       nSfb,                /*!< Number of scale factor bands */
                           UCHAR    *freqBandTable,       /*!< First Subband for each Sfb */
@@ -1585,7 +1605,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   The resulting energy gain is given by mantissa and exponent.
 */
-/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
+static void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
                             SCHAR     nrgRef_e,          /*!< Reference Energy according to envelope data (exponent) */
                             ENV_CALC_NRGS* nrgs,
                             int       i,
@@ -1689,7 +1709,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   The result is used as a relative limit for all gains within the
   current "limiter band" (a certain frequency range).
 */
-/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+static void calcAvgGain(ENV_CALC_NRGS* nrgs,
                         int        lowSubband,    /*!< Begin of the limiter band */
                         int        highSubband,   /*!< High end of the limiter band */
                         FIXP_DBL  *ptrSumRef,
@@ -1728,21 +1748,101 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   *ptrSumRef_e = sumRef_e;
 }
 
+static void adjustTimeSlot_EldGrid(
+                              FIXP_DBL *ptrReal,        /*!< Subband samples to be adjusted, real part */
+                              ENV_CALC_NRGS* nrgs,
+                              UCHAR    *ptrHarmIndex,   /*!< Harmonic index */
+                              int       lowSubband,     /*!< Lowest QMF-channel in the currently used SBR range. */
+                              int       noSubbands,     /*!< Number of QMF subbands */
+                              int       scale_change,   /*!< Number of bits to shift adjusted samples */
+                              int       noNoiseFlag,    /*!< Flag to suppress noise addition */
+                              int      *ptrPhaseIndex,  /*!< Start index to random number array */
+                              int       scale_diff_low) /*!<  */
+{
+  int k;
+  FIXP_DBL  signalReal, sbNoise;
+  int tone_count = 0;
+
+  FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
+  FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
+  FIXP_DBL *pSineLevel  = nrgs->nrgSine;     /*!< Sine levels */
+
+  int    phaseIndex = *ptrPhaseIndex;
+  UCHAR  harmIndex  = *ptrHarmIndex;
+
+  static const INT harmonicPhase [2][4] = {
+    { 1, 0, -1,  0},
+    { 0, 1,  0, -1}
+  };
+
+  static const FIXP_DBL harmonicPhaseX [2][4] = {
+    { FL2FXCONST_DBL(2.0*1.245183154539139e-001),  FL2FXCONST_DBL(2.0*-1.123767859325028e-001),  FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001) },
+    { FL2FXCONST_DBL(2.0*1.245183154539139e-001),  FL2FXCONST_DBL(2.0* 1.123767859325028e-001),  FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001) }
+  };
+
+  for (k=0; k < noSubbands; k++) {
+
+    phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
+
+    if( (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) || (noNoiseFlag == 1) ){
+      sbNoise = FL2FXCONST_DBL(0.0f);
+    } else {
+      sbNoise = pNoiseLevel[0];
+    }
+
+    signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
+
+    signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise)<<4);
+
+    signalReal += pSineLevel[0] * harmonicPhase[0][harmIndex];
+
+    *ptrReal = signalReal;
+
+    if (k == 0) {
+      *(ptrReal-1) += scaleValue(fMultDiv2(harmonicPhaseX[lowSubband&1][harmIndex], pSineLevel[0]), -scale_diff_low)  ;
+      if (k < noSubbands - 1) {
+        *(ptrReal)   += fMultDiv2(pSineLevel[1], harmonicPhaseX[(lowSubband+1)&1][harmIndex]);
+      }
+    }
+    if (k > 0 && k < noSubbands - 1 && tone_count < 16) {
+      *(ptrReal)   += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1]  [harmIndex]);
+      *(ptrReal)   += fMultDiv2(pSineLevel[+ 1], harmonicPhaseX [(lowSubband+k+1)&1][harmIndex]);
+    }
+    if (k == noSubbands - 1 && tone_count < 16) {
+      if (k > 0) {
+        *(ptrReal)   += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1][harmIndex]);
+      }
+      if (k + lowSubband + 1< 63) {
+        *(ptrReal+1) += fMultDiv2(pSineLevel[0], harmonicPhaseX[(lowSubband+k+1)&1][harmIndex]);
+      }
+    }
+
+    if(pSineLevel[0] != FL2FXCONST_DBL(0.0f)){
+      tone_count++;
+    }
+    ptrReal++;
+    pNoiseLevel++;
+    pGain++;
+    pSineLevel++;
+  }
+
+  *ptrHarmIndex = (harmIndex + 1) & 3;
+  *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1);
+}
 
 /*!
   \brief   Amplify one timeslot of the signal with the calculated gains
            and add the noisefloor.
 */
 
-/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
+static void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
                              ENV_CALC_NRGS* nrgs,
                              UCHAR    *ptrHarmIndex,  /*!< Harmonic index */
                              int       lowSubband,    /*!< Lowest QMF-channel in the currently used SBR range. */
                              int       noSubbands,    /*!< Number of QMF subbands */
                              int       scale_change,  /*!< Number of bits to shift adjusted samples */
                              int       noNoiseFlag,   /*!< Flag to suppress noise addition */
-                             int      *ptrPhaseIndex, /*!< Start index to random number array */
-                             int       fCldfb)        /*!< CLDFB 80 flag */
+                             int      *ptrPhaseIndex) /*!< Start index to random number array */
 {
   FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
   FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
@@ -1775,41 +1875,10 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
 
   if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;
-
   else if (!noNoiseFlag)
         /* Add noisefloor to the amplified signal */
         signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
 
-  if (fCldfb) {
-
-    if (!(harmIndex&0x1)) {
-      /* harmIndex 0,2 */
-      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
-      *ptrReal++ = signalReal;
-    }
-    else {
-      /* harmIndex 1,3 in combination with freqInvFlag */
-      int shift = (int) (scale_change+1);
-      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
-
-      FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );
-
-      FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);
-
-
-      /* save switch and compare operations and reduce to XOR statement */
-      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
-          *(ptrReal-1) += tmp1;
-          signalReal   -= tmp2;
-      } else {
-          *(ptrReal-1) -= tmp1;
-          signalReal   += tmp2;
-      }
-      *ptrReal++ = signalReal;
-      freqInvFlag = !freqInvFlag;
-    }
-
-  } else
   {
     if (!(harmIndex&0x1)) {
       /* harmIndex 0,2 */
@@ -1933,8 +2002,9 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   *ptrHarmIndex = (harmIndex + 1) & 3;
   *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
 }
-void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal,           /*!< Subband samples to be adjusted, real part */
-                      FIXP_DBL *RESTRICT ptrImag,               /*!< Subband samples to be adjusted, imag part */
+static void adjustTimeSlotHQ(
+                      FIXP_DBL *RESTRICT ptrReal,      /*!< Subband samples to be adjusted, real part */
+                      FIXP_DBL *RESTRICT ptrImag,      /*!< Subband samples to be adjusted, imag part */
                       HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                       ENV_CALC_NRGS* nrgs,
                       int       lowSubband,            /*!< Lowest QMF-channel in the currently used SBR range. */
@@ -1961,7 +2031,7 @@ void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal,           /*!< Subband samples
   FIXP_SGL direct_ratio = /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio;
   int    index = *ptrPhaseIndex;
   UCHAR   harmIndex = *ptrHarmIndex;
-  register int freqInvFlag = (lowSubband & 1);
+  int freqInvFlag = (lowSubband & 1);
   FIXP_DBL sineLevel;
   int shift;
 
@@ -2137,7 +2207,6 @@ ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QM
   UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
   int patchBorders[MAX_NUM_PATCHES + 1];
   int kx, k2;
-  FIXP_DBL temp;
 
   int lowSubband = freqBandTable[0];
   int highSubband = freqBandTable[noFreqBands];
@@ -2169,13 +2238,32 @@ ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QM
 
 
     while (hiLimIndex <= tempNoLim) {
+      FIXP_DBL div_m, oct_m, temp;
+      INT div_e  = 0, oct_e  = 0, temp_e = 0;
+
       k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
       kx = workLimiterBandTable[loLimIndex] + lowSubband;
 
-      temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
-      temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);
+      div_m  = fDivNorm(k2, kx, &div_e);
+
+      /* calculate number of octaves */
+      oct_m  = fLog2(div_m, div_e, &oct_e);
+
+      /* multiply with limiterbands per octave    */
+      /* values 1, 1.2, 2, 3 -> scale factor of 2 */
+      temp = fMultNorm(oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], &temp_e);
+
+      /* overall scale factor of temp ist addition of scalefactors from log2 calculation,
+         limiter bands scalefactor (2) and limiter bands multiplication */
+      temp_e += oct_e + 2;
+
+      /*    div can be a maximum of 64 (k2 = 64 and kx = 1)
+         -> oct can be a maximum of 6
+         -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum factor of 3)
+         -> we need a scale factor of 5 for comparisson
+      */
+      if (temp >> (5 - temp_e) < FL2FXCONST_DBL (0.49f) >> 5) {
 
-      if (temp < FL2FXCONST_DBL (0.49f)>>5) {
         if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
           workLimiterBandTable[hiLimIndex] = highSubband;
           nBands--;

libSBRdec/src/env_dec.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -369,7 +369,7 @@ leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control d
   FIXP_SGL step;    /* speed of fade */
   int i;
 
-  int currentStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+  int currentStartPos = FDKmax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots);
   int currentStopPos = hHeaderData->numberTimeSlots;
 
 

libSBRdec/src/env_extr.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -327,7 +327,7 @@ sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData,
   }
 
   /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */
-  if(hHeaderData->syncState != SBR_ACTIVE ||
+  if(hHeaderData->syncState < SBR_HEADER ||
      lastHeader.startFreq   != pBsData->startFreq   ||
      lastHeader.stopFreq    != pBsData->stopFreq    ||
      lastHeader.freqScale   != pBsData->freqScale   ||
@@ -904,6 +904,9 @@ static const FRAME_INFO v_frame_info4_8 = { 0, 4, {0, 2, 4, 6, 8}, {1, 1, 1, 1},
             break;
         default:
             FDK_ASSERT(0);
+            /* in case assertion checks are disabled, force a definite memory fault at first access */
+            pTable = NULL;
+            break;
     }
 
     /* look number of envelopes in table */

libSBRdec/src/env_extr.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -125,6 +125,7 @@ amm-info@iis.fraunhofer.de
 typedef enum
 {
   HEADER_NOT_PRESENT,
+  HEADER_ERROR,
   HEADER_OK,
   HEADER_RESET
 }
@@ -132,10 +133,10 @@ SBR_HEADER_STATUS;
 
 typedef enum
 {
-  SBR_NOT_INITIALIZED,
-  UPSAMPLING,
-  SBR_HEADER,
-  SBR_ACTIVE
+  SBR_NOT_INITIALIZED = 0,
+  UPSAMPLING = 1,
+  SBR_HEADER = 2,
+  SBR_ACTIVE = 3
 }
 SBR_SYNC_STATE;
 
@@ -179,6 +180,10 @@ typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
 #define SBRDEC_LOW_POWER      16  /* Flag indicating that Low Power QMF mode shall be used. */
 #define SBRDEC_PS_DECODED     32  /* Flag indicating that PS was decoded and rendered. */
 #define SBRDEC_LD_MPS_QMF    512  /* Flag indicating that the LD-MPS QMF shall be used. */
+#define SBRDEC_SYNTAX_DRM   2048  /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */
+#define SBRDEC_DOWNSAMPLE   8192  /* Flag indicating that the downsampling mode is used. */
+#define SBRDEC_FLUSH       16384  /* Flag is used to flush all elements in use. */
+#define SBRDEC_FORCE_RESET 32768  /* Flag is used to force a reset of all elements in use. */
 
 #define SBRDEC_HDR_STAT_RESET  1
 #define SBRDEC_HDR_STAT_UPDATE 2

libSBRdec/src/sbr_dec.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -225,7 +225,14 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
   }
 
   if (resetAnaQmf) {
-    int qmfErr = qmfInitAnalysisFilterBank (
+    QMF_FILTER_BANK prvAnaQmf;
+    int  qmfErr;
+
+    /* Store current configuration */
+    FDKmemcpy(&prvAnaQmf, &hSbrDec->AnalysiscQMF, sizeof(QMF_FILTER_BANK));
+
+    /* Reset analysis QMF */
+    qmfErr = qmfInitAnalysisFilterBank (
            &hSbrDec->AnalysiscQMF,
             hSbrDec->anaQmfStates,
             hSbrDec->AnalysiscQMF.no_col,
@@ -234,13 +241,22 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
             hSbrDec->AnalysiscQMF.no_channels,
             anaQmfFlags | QMF_FLAG_KEEP_STATES
             );
+
     if (qmfErr != 0) {
-      FDK_ASSERT(0);
+      /* Restore old configuration of analysis QMF */
+      FDKmemcpy(&hSbrDec->AnalysiscQMF, &prvAnaQmf, sizeof(QMF_FILTER_BANK));
     }
   }
 
   if (resetSynQmf) {
-    int qmfErr = qmfInitSynthesisFilterBank (
+    QMF_FILTER_BANK prvSynQmf;
+    int  qmfErr;
+
+    /* Store current configuration */
+    FDKmemcpy(&prvSynQmf, &hSbrDec->SynthesisQMF, sizeof(QMF_FILTER_BANK));
+
+    /* Reset synthesis QMF */
+    qmfErr = qmfInitSynthesisFilterBank (
            &hSbrDec->SynthesisQMF,
             hSbrDec->pSynQmfStates,
             hSbrDec->SynthesisQMF.no_col,
@@ -251,7 +267,8 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
             );
 
     if (qmfErr != 0) {
-      FDK_ASSERT(0);
+      /* Restore old configuration of synthesis QMF */
+      FDKmemcpy(&hSbrDec->SynthesisQMF, &prvSynQmf, sizeof(QMF_FILTER_BANK));
     }
   }
 }
@@ -321,7 +338,8 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
           HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
           const int applyProcessing,         /*!< Flag for SBR operation */
           HANDLE_PS_DEC h_ps_d,
-          const UINT flags
+          const UINT flags,
+          const int codecFrameSize
          )
 {
   int i, slot, reserve;
@@ -348,6 +366,33 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
   if (flags & SBRDEC_ELD_GRID) {
     /* Choose the right low delay filter bank */
     changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 );
+
+    /* If the LD-MPS QMF is not available delay the signal by (96-48*ldSbrSamplingRate)
+     * samples according to ISO/IEC 14496-3:2009/FDAM 2:2010(E) chapter 4.5.2.13. */
+    if ( (flags & SBRDEC_LD_MPS_QMF)
+      && (hSbrDec->AnalysiscQMF.flags & QMF_FLAG_CLDFB) )
+    {
+      INT_PCM *pDlyBuf = hSbrDec->coreDelayBuf;  /* DLYBUF */
+      int smpl, delay = 96 >> (!(flags & SBRDEC_DOWNSAMPLE) ? 1 : 0);
+      /* Create TMPBUF */
+      C_AALLOC_SCRATCH_START(pcmTemp, INT_PCM, (96));
+      /* Copy delay samples from INBUF to TMPBUF */
+      for (smpl = 0; smpl < delay; smpl += 1) {
+        pcmTemp[smpl] = timeIn[(codecFrameSize-delay+smpl)*strideIn];
+      }
+      /* Move input signal remainder to the very end of INBUF */
+      for (smpl = (codecFrameSize-delay-1)*strideIn; smpl >= 0; smpl -= strideIn) {
+        timeIn[smpl+delay] = timeIn[smpl];
+      }
+      /* Copy delayed samples from last frame from DLYBUF to the very beginning of INBUF */
+      for (smpl = 0; smpl < delay; smpl += 1) {
+        timeIn[smpl*strideIn] = pDlyBuf[smpl];
+      }
+      /* Copy TMPBUF to DLYBUF */
+      FDKmemcpy(pDlyBuf, pcmTemp, delay*sizeof(INT_PCM));
+      /* Destory TMPBUF */
+      C_AALLOC_SCRATCH_END(pcmTemp, INT_PCM, (96));
+    }
   }
 
   /*
@@ -761,7 +806,7 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
   {
     int qmfErr;
     /* Adapted QMF analysis post-twiddles for down-sampled HQ SBR */
-    const UINT downSampledFlag = (downsampleFac==2) ? QMF_FLAG_DOWNSAMPLED : 0;
+    const UINT downSampledFlag = (flags & SBRDEC_DOWNSAMPLE) ? QMF_FLAG_DOWNSAMPLED : 0;
 
     qmfErr = qmfInitAnalysisFilterBank (
                     &hs->AnalysiscQMF,
@@ -836,6 +881,9 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
     }
   }
 
+  /* Clear input delay line */
+  FDKmemclear(hs->coreDelayBuf, (96)*sizeof(INT_PCM));
+
   /* assign qmf time slots */
   assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP);
 

libSBRdec/src/sbr_dec.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -118,6 +118,9 @@ typedef struct
   FIXP_DBL * WorkBuffer1;
   FIXP_DBL * WorkBuffer2;
 
+  /* Delayed time input signal needed to align CLDFD with LD-MPS QMF. */
+  INT_PCM    coreDelayBuf[(96)];
+
   /* QMF filter states */
   FIXP_QAS   anaQmfStates[(320)];
   FIXP_QSS * pSynQmfStates;
@@ -182,7 +185,8 @@ sbr_dec (HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
          HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
          const int applyProcessing,         /*!< Flag for SBR operation */
          HANDLE_PS_DEC h_ps_d,
-         const UINT flags
+         const UINT flags,
+         const int codecFrameSize
         );
 
 

libSBRdec/src/sbr_ram.cpp

@@ -107,19 +107,19 @@ amm-info@iis.fraunhofer.de
 /*! SBR Decoder main structure */
 C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1)
 /*! SBR Decoder element data  <br>
-  Dimension: (4) */
-C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (4))
+  Dimension: (8) */
+C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8))
 /*! SBR Decoder individual channel data  <br>
-  Dimension: (6) */
-C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (6)+1)
+  Dimension: (8) */
+C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8)+1)
 
 /*! Filter states for QMF-synthesis.  <br>
-  Dimension: #(6) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
-C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (6)+1, SECT_DATA_L1)
+  Dimension: #(8) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
+C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (8)+1, SECT_DATA_L1)
 
 /*! Delayed spectral data needed for the dynamic framing of SBR.
   For mp3PRO, 1/3 of a frame is buffered (#(6) 6) */
-C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (6)+1)
+C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (8)+1)
 
 /*! Static Data of PS */
 

libSBRdec/src/sbr_ram.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -118,8 +118,8 @@ typedef struct
 
 struct SBR_DECODER_INSTANCE
 {
-  SBR_DECODER_ELEMENT  *pSbrElement[(4)];
-  SBR_HEADER_DATA       sbrHeader[(4)][(1)+1];      /* Sbr header for each individual channel of an element */
+  SBR_DECODER_ELEMENT  *pSbrElement[(8)];
+  SBR_HEADER_DATA       sbrHeader[(8)][(1)+1];      /* Sbr header for each individual channel of an element */
 
   FIXP_DBL *workBuffer1;
   FIXP_DBL *workBuffer2;
@@ -135,6 +135,7 @@ struct SBR_DECODER_INSTANCE
   USHORT codecFrameSize;
   UCHAR synDownsampleFac;
   UCHAR numDelayFrames;        /* The current number of additional delay frames used for processing. */
+  UCHAR numFlushedFrames;      /* The variable counts the number of frames which are flushed consecutively. */
 
   UINT flags;
 

libSBRdec/src/sbr_rom.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -189,6 +189,15 @@ const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] =
   FL2FXCONST_SGL(3.0f / 4.0f)
 };
 
+/*! Constants for calculating the number of limiter bands */
+const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] =
+{
+  FL2FXCONST_DBL(1.0f / 4.0f),
+  FL2FXCONST_DBL(1.2f / 4.0f),
+  FL2FXCONST_DBL(2.0f / 4.0f),
+  FL2FXCONST_DBL(3.0f / 4.0f)
+};
+
 /*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */
 const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = {
   FL2FXCONST_SGL(0.66666666666666f),
@@ -1176,7 +1185,7 @@ const FIXP_DBL Alphas[NO_ICC_LEVELS] = {
 #define  FL2FXCONST_PS FL2FXCONST_SGL
 #else
 #define  FIXP_PS FIXP_DBL
-#define  FXP_CAST
+#define  FXP_CAST(x) ((FIXP_DBL)(x))
 #define  FL2FXCONST_PS FL2FXCONST_DBL
 #endif
 

libSBRdec/src/sbr_rom.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -124,6 +124,7 @@ extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRI
 extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4];
 extern const UCHAR    FDK_sbrDecoder_sbr_limGains_e[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4];
+extern const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2];
 extern const FIXP_SGL harmonicPhaseX [2][4];

libSBRdec/src/sbrdec_drc.h

@@ -95,7 +95,7 @@ amm-info@iis.fraunhofer.de
 
 
 
-#define SBRDEC_MAX_DRC_CHANNELS  (6)
+#define SBRDEC_MAX_DRC_CHANNELS  (8)
 #define SBRDEC_MAX_DRC_BANDS     ( 16 )
 
 typedef struct

libSBRdec/src/sbrdecoder.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -128,6 +128,7 @@ amm-info@iis.fraunhofer.de
 #include "lpp_tran.h"
 #include "transcendent.h"
 
+#include "FDK_crc.h"
 
 #include "sbrdec_drc.h"
 
@@ -137,10 +138,15 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 2
 #define SBRDECODER_LIB_VL1 2
-#define SBRDECODER_LIB_VL2 3
+#define SBRDECODER_LIB_VL2 12
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
+#ifdef __ANDROID__
+#define SBRDECODER_LIB_BUILD_DATE ""
+#define SBRDECODER_LIB_BUILD_TIME ""
+#else
 #define SBRDECODER_LIB_BUILD_DATE __DATE__
 #define SBRDECODER_LIB_BUILD_TIME __TIME__
+#endif
 
 
 
@@ -189,6 +195,33 @@ static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_
   hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi;
 }
 
+static int compareSbrHeader( const HANDLE_SBR_HEADER_DATA hHdr1, const HANDLE_SBR_HEADER_DATA hHdr2 )
+{
+  int result = 0;
+
+  /* compare basic data */
+  result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0;
+  result |= (hHdr1->status != hHdr2->status) ? 1 : 0;
+  result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0;
+  result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0;
+  result |= (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0;
+  result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0;
+  result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0;
+
+  /* compare bitstream data */
+  result |= FDKmemcmp( &hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS) );
+  result |= FDKmemcmp( &hHdr1->bs_info, &hHdr2->bs_info, sizeof(SBR_HEADER_DATA_BS_INFO) );
+
+  /* compare frequency band data */
+  result |= FDKmemcmp( &hHdr1->freqBandData, &hHdr2->freqBandData, (8+MAX_NUM_LIMITERS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableLo, hHdr2->freqBandData.freqBandTableLo, (MAX_FREQ_COEFFS/2+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableHi, hHdr2->freqBandData.freqBandTableHi, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableNoise, hHdr2->freqBandData.freqBandTableNoise, (MAX_NOISE_COEFFS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
+
+  return result;
+}
+
 
 /*!
   \brief Reset SBR decoder.
@@ -251,8 +284,10 @@ SBR_ERROR sbrDecoder_ResetElement (
 
   if ( sampleRateIn == sampleRateOut ) {
     synDownsampleFac = 2;
+    self->flags |=  SBRDEC_DOWNSAMPLE;
   } else {
     synDownsampleFac = 1;
+    self->flags &= ~SBRDEC_DOWNSAMPLE;
   }
 
   self->synDownsampleFac = synDownsampleFac;
@@ -311,7 +346,6 @@ SBR_ERROR sbrDecoder_ResetElement (
     case AOT_PS:
     case AOT_ER_AAC_SCAL:
     case AOT_DRM_AAC:
-    case AOT_DRM_SURROUND:
       if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
         sbrError = SBRDEC_CREATE_ERROR;
         goto bail;
@@ -385,6 +419,7 @@ int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec)
     case AOT_PS:
     case AOT_ER_AAC_SCAL:
     case AOT_ER_AAC_ELD:
+    case AOT_DRM_AAC:
       return 1;
     default:
       return 0;
@@ -428,7 +463,7 @@ SBR_ERROR sbrDecoder_InitElement (
   int nSbrElementsStart = self->numSbrElements;
 
   /* Check core codec AOT */
-  if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (4)) {
+  if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (8)) {
     sbrError = SBRDEC_UNSUPPORTED_CONFIG;
     goto bail;
   }
@@ -444,6 +479,7 @@ SBR_ERROR sbrDecoder_InitElement (
      && self->coreCodec == coreCodec
      && self->pSbrElement[elementIndex] != NULL
      && self->pSbrElement[elementIndex]->elementID == elementID
+     && !(self->flags & SBRDEC_FORCE_RESET)
      )
   {
      /* Nothing to do */
@@ -456,6 +492,8 @@ SBR_ERROR sbrDecoder_InitElement (
 
   self->flags = 0;
   self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0;
+  self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0;
+  self->flags |= (coreCodec == AOT_DRM_AAC)     ? SBRDEC_SYNTAX_SCAL|SBRDEC_SYNTAX_DRM : 0;
 
   /* Init SBR elements */
   {
@@ -495,7 +533,6 @@ SBR_ERROR sbrDecoder_InitElement (
         case AOT_PS:
         case AOT_ER_AAC_SCAL:
         case AOT_DRM_AAC:
-        case AOT_DRM_SURROUND:
           elChannels = 2;
           break;
         default:
@@ -550,8 +587,9 @@ bail:
     if (nSbrElementsStart < self->numSbrElements) {
       /* Free the memory allocated for this element */
       sbrDecoder_DestroyElement( self, elementIndex );
-    } else if (self->pSbrElement[elementIndex] != NULL) {
-      /* Set error flag to trigger concealment */
+    } else if ( (self->pSbrElement[elementIndex] != NULL)
+             && (elementIndex < (8)))
+    { /* Set error flag to trigger concealment */
       self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;
     }
   }
@@ -615,7 +653,7 @@ INT sbrDecoder_Header (
   SBR_ERROR sbrError = SBRDEC_OK;
   int headerIndex;
 
-  if ( self == NULL || elementIndex > (4) )
+  if ( self == NULL || elementIndex > (8) )
   {
     return SBRDEC_UNSUPPORTED_CONFIG;
   }
@@ -728,6 +766,24 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
       }
     }
     break;
+  case SBR_FLUSH_DATA:
+    if (value != 0) {
+      if (self == NULL) {
+        errorStatus = SBRDEC_NOT_INITIALIZED;
+      } else {
+        self->flags |= SBRDEC_FLUSH;
+      }
+    }
+    break;
+  case SBR_CLEAR_HISTORY:
+    if (value != 0) {
+      if (self == NULL) {
+        errorStatus = SBRDEC_NOT_INITIALIZED;
+      } else {
+        self->flags |= SBRDEC_FORCE_RESET;
+      }
+    }
+    break;
   case SBR_BS_INTERRUPTION:
     {
       int elementIndex;
@@ -738,7 +794,8 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
       }
 
       /* Loop over SBR elements */
-      for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++)
+      for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++) {
+      if (self->pSbrElement[elementIndex] != NULL)
       {
         HANDLE_SBR_HEADER_DATA hSbrHeader;
         int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
@@ -750,7 +807,7 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
            This switches off bitstream parsing until a new header arrives. */
         hSbrHeader->syncState = UPSAMPLING;
         hSbrHeader->status   |= SBRDEC_HDR_STAT_UPDATE;
-      }
+      } }
     }
     break;
   default:
@@ -767,7 +824,7 @@ SBRDEC_DRC_CHANNEL * sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, con
   SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
   int elementIndex, elChanIdx=0, numCh=0;
 
-  for (elementIndex = 0; (elementIndex < (4)) && (numCh <= channel); elementIndex++)
+  for (elementIndex = 0; (elementIndex < (8)) && (numCh <= channel); elementIndex++)
   {
     SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex];
     int c, elChannels;
@@ -829,7 +886,7 @@ SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER  self,
   if (self == NULL) {
     return SBRDEC_NOT_INITIALIZED;
   }
-  if (ch > (6) || pNextFact_mag == NULL) {
+  if (ch > (8) || pNextFact_mag == NULL) {
     return SBRDEC_SET_PARAM_FAIL;
   }
 
@@ -874,7 +931,7 @@ void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
   SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
 
   if ( (self == NULL)
-    || (ch > (6))
+    || (ch > (8))
     || (self->numSbrElements == 0)
     || (self->numSbrChannels == 0) ) {
     return;
@@ -902,24 +959,73 @@ SBR_ERROR sbrDecoder_Parse(
         )
 {
   SBR_DECODER_ELEMENT   *hSbrElement;
-  HANDLE_SBR_HEADER_DATA hSbrHeader;
+  HANDLE_SBR_HEADER_DATA hSbrHeader = NULL;
   HANDLE_SBR_CHANNEL    *pSbrChannel;
 
   SBR_FRAME_DATA *hFrameDataLeft;
   SBR_FRAME_DATA *hFrameDataRight;
 
   SBR_ERROR errorStatus = SBRDEC_OK;
-  SBR_SYNC_STATE initialSyncState;
   SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
 
   INT  startPos;
   INT  CRCLen = 0;
+  HANDLE_FDK_BITSTREAM hBsOriginal = hBs;
+  FDK_CRCINFO  crcInfo;         /* shall be used for all other CRCs in the future (TBD) */
+  INT          crcReg = 0;
+  USHORT       drmSbrCrc = 0;
 
   int  stereo;
   int  fDoDecodeSbrData = 1;
 
   int lastSlot, lastHdrSlot = 0, thisHdrSlot;
 
+  /* Reverse bits of DRM SBR payload */
+  if ( (self->flags & SBRDEC_SYNTAX_DRM) && *count > 0 )
+  {
+    UCHAR *bsBufferDrm = (UCHAR*)self->workBuffer1;
+    HANDLE_FDK_BITSTREAM hBsBwd = (HANDLE_FDK_BITSTREAM) (bsBufferDrm + (512));
+    int dataBytes, dataBits;
+
+    dataBits = *count;
+
+    if (dataBits > ((512)*8)) {
+      /* do not flip more data than needed */
+      dataBits = (512)*8;
+    }
+
+    dataBytes = (dataBits+7)>>3;
+
+    int j;
+
+    if ((j = (int)FDKgetValidBits(hBs)) != 8) {
+      FDKpushBiDirectional(hBs, (j-8));
+    }
+
+    j = 0;
+    for ( ; dataBytes > 0; dataBytes--)
+    {
+      int i;
+      UCHAR tmpByte;
+      UCHAR buffer = 0x00;
+
+      tmpByte = (UCHAR) FDKreadBits(hBs, 8);
+      for (i = 0; i < 4; i++) {
+        int shift = 2 * i + 1;
+        buffer |= (tmpByte & (0x08>>i)) << shift;
+        buffer |= (tmpByte & (0x10<<i)) >> shift;
+      }
+      bsBufferDrm[j++] = buffer;
+      FDKpushBack(hBs, 16);
+    }
+
+    FDKinitBitStream(hBsBwd, bsBufferDrm, (512), dataBits, BS_READER);
+
+    /* Use reversed data */
+    hBs = hBsBwd;
+    bsPayLen = *count;
+  }
+
   /* Remember start position of  SBR element */
   startPos = FDKgetValidBits(hBs);
 
@@ -944,7 +1050,6 @@ SBR_ERROR sbrDecoder_Parse(
   hFrameDataLeft  = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
   hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
 
-  initialSyncState = hSbrHeader->syncState;
 
   /* reset PS flag; will be set after PS was found */
   self->flags &= ~SBRDEC_PS_DECODED;
@@ -980,12 +1085,19 @@ SBR_ERROR sbrDecoder_Parse(
   */
   if (fDoDecodeSbrData)
   {
-    if (crcFlag == 1) {
+    if (crcFlag) {
       switch (self->coreCodec) {
       case AOT_ER_AAC_ELD:
         FDKpushFor (hBs, 10);
         /* check sbrcrc later: we don't know the payload length now */
         break;
+      case AOT_DRM_AAC:
+        drmSbrCrc = (USHORT)FDKreadBits(hBs, 8);
+        /* Setup CRC decoder */
+        FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8);
+        /* Start CRC region */
+        crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
+        break;
       default:
         CRCLen = bsPayLen - 10;                     /* change: 0 => i */
         if (CRCLen < 0) {
@@ -1030,6 +1142,7 @@ SBR_ERROR sbrDecoder_Parse(
         hSbrHeader->syncState = SBR_HEADER;
       } else {
         hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+        headerStatus = HEADER_ERROR;
       }
     }
 
@@ -1079,7 +1192,7 @@ SBR_ERROR sbrDecoder_Parse(
         valBits = (INT)FDKgetValidBits(hBs);
       }
 
-      if ( crcFlag == 1 ) {
+      if ( crcFlag ) {
         switch (self->coreCodec) {
         case AOT_ER_AAC_ELD:
           {
@@ -1091,6 +1204,14 @@ SBR_ERROR sbrDecoder_Parse(
             FDKpushFor(hBs, crcLen);
           }
           break;
+        case AOT_DRM_AAC:
+          /* End CRC region */
+          FDKcrcEndReg(&crcInfo, hBs, crcReg);
+          /* Check CRC */
+          if ((FDKcrcGetCRC(&crcInfo)^0xFF) != drmSbrCrc) {
+            fDoDecodeSbrData = 0;
+          }
+          break;
         default:
           break;
         }
@@ -1119,6 +1240,10 @@ SBR_ERROR sbrDecoder_Parse(
         }
       }
     }
+  } else {
+    /* The returned bit count will not be the actual payload size since we did not
+       parse the frame data. Return an error so that the caller can react respectively. */
+    errorStatus = SBRDEC_PARSE_ERROR;
   }
 
   if (!fDoDecodeSbrData) {
@@ -1137,8 +1262,25 @@ SBR_ERROR sbrDecoder_Parse(
   }
 
 bail:
-  if (errorStatus == SBRDEC_OK) {
-    if (headerStatus == HEADER_NOT_PRESENT) {
+
+  if ( self->flags & SBRDEC_SYNTAX_DRM )
+  {
+    hBs = hBsOriginal;
+  }
+
+  if ( (errorStatus == SBRDEC_OK)
+    || ( (errorStatus == SBRDEC_PARSE_ERROR)
+      && (headerStatus != HEADER_ERROR) ) )
+  {
+    int useOldHdr = ( (headerStatus == HEADER_NOT_PRESENT)
+                   || (headerStatus == HEADER_ERROR) ) ? 1 : 0;
+
+    if (!useOldHdr && (thisHdrSlot != lastHdrSlot)) {
+      useOldHdr |= ( compareSbrHeader( hSbrHeader,
+                                      &self->sbrHeader[elementIndex][lastHdrSlot] ) == 0 ) ? 1 : 0;
+    }
+
+    if (useOldHdr != 0) {
       /* Use the old header for this frame */
       hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot;
     } else {
@@ -1198,6 +1340,24 @@ sbrDecoder_DecodeElement (
   int  stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
   int  numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
 
+  if (self->flags & SBRDEC_FLUSH) {
+    if ( self->numFlushedFrames > self->numDelayFrames ) {
+      int hdrIdx;
+      /* No valid SBR payload available, hence switch to upsampling (in all headers) */
+      for (hdrIdx = 0; hdrIdx < ((1)+1); hdrIdx += 1) {
+        self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+      }
+    }
+    else {
+      /* Move frame pointer to the next slot which is up to be decoded/applied next */
+      hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
+      /* Update header and frame data pointer because they have already been set */
+      hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
+      hFrameDataLeft  = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+      hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+    }
+  }
+
   /* Update the header error flag */
   hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot];
 
@@ -1315,7 +1475,8 @@ sbrDecoder_DecodeElement (
            &pSbrChannel[0]->prevFrameData,
             (hSbrHeader->syncState == SBR_ACTIVE),
             h_ps_d,
-            self->flags
+            self->flags,
+            codecFrameSize
           );
 
   if (stereo) {
@@ -1332,7 +1493,8 @@ sbrDecoder_DecodeElement (
              &pSbrChannel[1]->prevFrameData,
               (hSbrHeader->syncState == SBR_ACTIVE),
               NULL,
-              self->flags
+              self->flags,
+              codecFrameSize
             );
   }
 
@@ -1348,20 +1510,21 @@ sbrDecoder_DecodeElement (
     if ( !(self->flags & SBRDEC_PS_DECODED) ) {
       /* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */
       /* So copy left channel to right channel.                                                              */
+      int copyFrameSize = codecFrameSize * 2 / self->synDownsampleFac;
       if (interleaved) {
         INT_PCM *ptr;
         INT i;
         FDK_ASSERT(strideOut == 2);
 
         ptr = timeData;
-        for (i = codecFrameSize; i--; ) 
+        for (i = copyFrameSize>>1; i--; )
         {
           INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */
           tmp = *ptr++; *ptr++ = tmp;
           tmp = *ptr++; *ptr++ = tmp;
         }
       } else {
-        FDKmemcpy( timeData+2*codecFrameSize, timeData, 2*codecFrameSize*sizeof(INT_PCM) );
+        FDKmemcpy( timeData+copyFrameSize, timeData, copyFrameSize*sizeof(INT_PCM) );
       }
     }
     *numOutChannels = 2;  /* Output minimum two channels when PS is enabled. */
@@ -1375,7 +1538,7 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
                              INT_PCM            *timeData,
                              int                *numChannels,
                              int                *sampleRate,
-                             const UCHAR         channelMapping[(6)],
+                             const UCHAR         channelMapping[(8)],
                              const int           interleaved,
                              const int           coreDecodedOk,
                              UCHAR              *psDecoded )
@@ -1425,14 +1588,23 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
     self->flags &= ~SBRDEC_PS_DECODED;
   }
 
+  if ( self->flags & SBRDEC_FLUSH ) {
+    /* flushing is signalized, hence increment the flush frame counter */
+    self->numFlushedFrames++;
+  }
+  else {
+    /* no flushing is signalized, hence reset the flush frame counter */
+    self->numFlushedFrames = 0;
+  }
+
   /* Loop over SBR elements */
   for (sbrElementNum = 0; sbrElementNum<self->numSbrElements; sbrElementNum++)
   {
     int numElementChan;
 
     if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) {
-      errorStatus = SBRDEC_UNSUPPORTED_CONFIG;
-      goto bail;
+      /* Disable PS and try decoding SBR mono. */
+      psPossible = 0;
     }
 
     numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1;
@@ -1472,6 +1644,10 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
 
 
 
+  /* Clear reset and flush flag because everything seems to be done successfully. */
+  self->flags &= ~SBRDEC_FORCE_RESET;
+  self->flags &= ~SBRDEC_FLUSH;
+
 bail:
 
   return errorStatus;
@@ -1496,7 +1672,7 @@ SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *pSelf )
       FreeRam_SbrDecWorkBuffer2(&self->workBuffer2);
     }
 
-    for (i = 0; i < (4); i++) {
+    for (i = 0; i < (8); i++) {
       sbrDecoder_DestroyElement( self, i );
     }
 
@@ -1536,6 +1712,7 @@ INT sbrDecoder_GetLibInfo( LIB_INFO *info )
     | CAPF_SBR_HQ
     | CAPF_SBR_LP
     | CAPF_SBR_PS_MPEG
+    | CAPF_SBR_DRM_BS
     | CAPF_SBR_CONCEALMENT
     | CAPF_SBR_DRC
       ;
@@ -1544,3 +1721,37 @@ INT sbrDecoder_GetLibInfo( LIB_INFO *info )
   return 0;
 }
 
+
+UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self )
+{
+  UINT outputDelay = 0;
+
+  if ( self != NULL) {
+    UINT flags = self->flags;
+
+    /* See chapter 1.6.7.2 of ISO/IEC 14496-3 for the GA-SBR figures below. */
+
+    /* Are we initialized? */
+    if ( (self->numSbrChannels > 0)
+      && (self->numSbrElements > 0) )
+    {
+      /* Add QMF synthesis delay */
+      if ( (flags & SBRDEC_ELD_GRID)
+        && IS_LOWDELAY(self->coreCodec) ) {
+        /* Low delay SBR: */
+        {
+          outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64;   /* QMF synthesis */
+          if (flags & SBRDEC_LD_MPS_QMF) {
+            outputDelay += 32;
+          }
+        }
+      }
+      else if (!IS_USAC(self->coreCodec)) {
+        /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) branch: */
+        outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 481 : 962;
+      }
+    }
+  }
+
+  return (outputDelay);
+}

libSBRenc/include/sbr_encoder.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -135,6 +135,12 @@ enum
   SBR_SYNTAX_DRM_CRC   = 0x0008
 };
 
+typedef enum
+{
+  FREQ_RES_LOW = 0,
+  FREQ_RES_HIGH
+} FREQ_RES;
+
 typedef struct
 {
   CODEC_TYPE       coreCoder;        /*!< LC or ELD */
@@ -168,8 +174,9 @@ typedef struct sbrConfiguration
   INT dynBwSupported;         /*!< Flag: support for dynamic bandwidth in this combination. */
   INT parametricCoding;       /*!< Flag: usage of parametric coding tool. */
   INT downSampleFactor;       /*!< Sampling rate relation between the SBR and the core encoder. */
-  int freq_res_fixfix[3];     /*!< Frequency resolution of envelopes in frame class FIXFIX
-                                 0=1 Env; 1=2 Env; 2=4 Env; */
+  FREQ_RES freq_res_fixfix[2];/*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */
+  UCHAR fResTransIsLow;       /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */
+
   /*
      core coder dependent tuning parameters
   */
@@ -221,6 +228,8 @@ typedef struct sbrConfiguration
   INT sbr_interpol_freq;      /*!< Flag: use interpolation in freq. direction. */
   INT sbr_smoothing_length;   /*!< Flag: choose length 4 or 0 (=on, off). */
   UCHAR init_amp_res_FF;
+  FIXP_DBL threshold_AmpRes_FF_m;
+  SCHAR threshold_AmpRes_FF_e;
 } sbrConfiguration, *sbrConfigurationPtr ;
 
 typedef struct SBR_CONFIG_DATA
@@ -237,7 +246,7 @@ typedef struct SBR_CONFIG_DATA
   INT noQmfBands;                       /**< Number of QMF frequency bands. */
   INT noQmfSlots;                       /**< Number of QMF slots. */
 
-  UCHAR *freqBandTable[2];              /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeefs actually needed for lowres. */
+  UCHAR *freqBandTable[2];              /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */
   UCHAR *v_k_master;                    /**< Master BandTable where freqBandTable is derived from. */
 
 
@@ -249,6 +258,8 @@ typedef struct SBR_CONFIG_DATA
   INT xposCtrlSwitch;                   /**< Flag indicates whether to switch xpos ctrl on the fly. */
   INT switchTransposers;                /**< Flag indicates whether to switch xpos on the fly .     */
   UCHAR initAmpResFF;
+  FIXP_DBL thresholdAmpResFF_m;
+  SCHAR thresholdAmpResFF_e;
 } SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
 
 typedef struct {

libSBRenc/src/bit_sbr.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -283,9 +283,7 @@ void sbrEncoder_GetHeader(SBR_ENCODER   *sbrEncoder,
                           INT            element_index,
                           int            fSendHeaders)
 {
-  int bits;
-
-  bits = encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs);
+  encodeSbrHeaderData (&sbrEncoder->sbrElement[element_index]->sbrHeaderData, hBs);
     
   if (fSendHeaders == 0) {
     /* Prevent header being embedded into the SBR payload. */

libSBRenc/src/bit_sbr.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -141,8 +141,8 @@ struct SBR_ENV_DATA
 {
 
   INT sbr_xpos_ctrl;
-  INT freq_res_fixfix;
-
+  FREQ_RES freq_res_fixfix[2];
+  UCHAR fResTransIsLow;
 
   INVF_MODE sbr_invf_mode;
   INVF_MODE sbr_invf_mode_vec[MAX_NUM_NOISE_VALUES];
@@ -205,6 +205,8 @@ struct SBR_ENV_DATA
   INT balance;
   AMP_RES init_sbr_amp_res;
   AMP_RES currentAmpResFF;
+  FIXP_DBL ton_HF[SBR_GLOBAL_TONALITY_VALUES]; /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */
+  FIXP_DBL global_tonality;
 
   /* extended data */
   INT extended_data;

libSBRenc/src/env_est.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -103,6 +103,114 @@ static const UCHAR panTable[2][10] = { { 0, 2, 4, 6, 8,12,16,20,24},
 static const UCHAR maxIndex[2] = {9, 5};
 
 
+/******************************************************************************
+ Functionname:  FDKsbrEnc_GetTonality
+******************************************************************************/
+/***************************************************************************/
+/*!
+
+  \brief      Calculates complete energy per band from the energy values
+              of the QMF subsamples.
+
+  \brief      quotaMatrix - calculated in FDKsbrEnc_CalculateTonalityQuotas()
+  \brief      noEstPerFrame - number of estimations per frame
+  \brief      startIndex - start index for the quota matrix
+  \brief      Energies - energy matrix
+  \brief      startBand - start band
+  \brief      stopBand - number of QMF bands
+  \brief      numberCols - number of QMF subsamples
+
+  \return     mean tonality of the 5 bands with the highest energy
+              scaled by 2^(RELAXATION_SHIFT+2)*RELAXATION_FRACT
+
+****************************************************************************/
+static FIXP_DBL FDKsbrEnc_GetTonality(
+        const FIXP_DBL *const *quotaMatrix,
+        const INT              noEstPerFrame,
+        const INT              startIndex,
+        const FIXP_DBL *const *Energies,
+        const UCHAR            startBand,
+        const INT              stopBand,
+        const INT              numberCols
+        )
+{
+  UCHAR b, e, k;
+  INT      no_enMaxBand[SBR_MAX_ENERGY_VALUES] = { -1, -1, -1, -1, -1 };
+  FIXP_DBL energyMax[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) };
+  FIXP_DBL energyMaxMin = MAXVAL_DBL; /* min. energy in energyMax array */
+  UCHAR    posEnergyMaxMin = 0;       /* min. energy in energyMax array position */
+  FIXP_DBL tonalityBand[SBR_MAX_ENERGY_VALUES] = { FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f), FL2FXCONST_DBL(0.0f) };
+  FIXP_DBL globalTonality = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL energyBand[QMF_CHANNELS];
+  INT      maxNEnergyValues; /* max. number of max. energy values */
+
+  /*** Sum up energies for each band ***/
+  FDK_ASSERT(numberCols==15||numberCols==16);
+  /* numberCols is always 15 or 16 for ELD. In case of 16 bands, the
+      energyBands are initialized with the [15]th column.
+      The rest of the column energies are added in the next step.   */
+  if (numberCols==15) {
+    for (b=startBand; b<stopBand; b++) {
+      energyBand[b]=FL2FXCONST_DBL(0.0f);
+    }
+  } else {
+    for (b=startBand; b<stopBand; b++) {
+      energyBand[b]=Energies[15][b]>>4;
+    }
+  }
+
+  for (k=0; k<15; k++) {
+    for (b=startBand; b<stopBand; b++) {
+      energyBand[b] += Energies[k][b]>>4;
+    }
+  }
+
+  /*** Determine 5 highest band-energies ***/
+  maxNEnergyValues = fMin(SBR_MAX_ENERGY_VALUES, stopBand-startBand);
+
+  /* Get min. value in energyMax array */
+  energyMaxMin = energyMax[0] = energyBand[startBand];
+  no_enMaxBand[0] = startBand;
+  posEnergyMaxMin = 0;
+  for (k=1; k<maxNEnergyValues; k++) {
+    energyMax[k] = energyBand[startBand+k];
+    no_enMaxBand[k] = startBand+k;
+    if (energyMaxMin > energyMax[k]) {
+      energyMaxMin = energyMax[k];
+      posEnergyMaxMin = k;
+    }
+  }
+
+  for (b=startBand+maxNEnergyValues; b<stopBand; b++) {
+    if (energyBand[b] > energyMaxMin) {
+      energyMax[posEnergyMaxMin] = energyBand[b];
+      no_enMaxBand[posEnergyMaxMin] = b;
+
+      /* Again, get min. value in energyMax array */
+      energyMaxMin = energyMax[0];
+      posEnergyMaxMin = 0;
+      for (k=1; k<maxNEnergyValues; k++) {
+        if (energyMaxMin > energyMax[k]) {
+          energyMaxMin = energyMax[k];
+          posEnergyMaxMin = k;
+        }
+      }
+    }
+  }
+  /*** End determine 5 highest band-energies ***/
+
+  /* Get tonality values for 5 highest energies */
+  for (e=0; e<maxNEnergyValues; e++) {
+    tonalityBand[e]=FL2FXCONST_DBL(0.0f);
+    for (k=0; k<noEstPerFrame; k++) {
+      tonalityBand[e] += quotaMatrix[startIndex + k][no_enMaxBand[e]] >> 1;
+    }
+    globalTonality += tonalityBand[e] >> 2; /* headroom of 2+1 (max. 5 additions) */
+  }
+
+  return globalTonality;
+}
+
 /***************************************************************************/
 /*!
 
@@ -919,10 +1027,42 @@ FDKsbrEnc_extractSbrEnvelope1 (
                                      hEnvChan->qmfScale);
 
 
+  if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
+    FIXP_DBL tonality = FDKsbrEnc_GetTonality (
+          hEnvChan->TonCorr.quotaMatrix,
+          hEnvChan->TonCorr.numberOfEstimatesPerFrame,
+          hEnvChan->TonCorr.startIndexMatrix,
+          sbrExtrEnv->YBuffer + sbrExtrEnv->YBufferWriteOffset,
+          h_con->freqBandTable[HI][0]+1,
+          h_con->noQmfBands,
+          sbrExtrEnv->no_cols
+        );
+
+    hEnvChan->encEnvData.ton_HF[1] = hEnvChan->encEnvData.ton_HF[0];
+    hEnvChan->encEnvData.ton_HF[0] = tonality;
+
+    /* tonality is scaled by 2^19/0.524288f (fract part of RELAXATION) */
+    hEnvChan->encEnvData.global_tonality = (hEnvChan->encEnvData.ton_HF[0]>>1) + (hEnvChan->encEnvData.ton_HF[1]>>1);
+  }
+
+
 
   /*
     Transient detection COEFF Transform OK
   */
+  if(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+  {
+    FDKsbrEnc_fastTransientDetect(
+            &hEnvChan->sbrFastTransientDetector,
+             sbrExtrEnv->YBuffer,
+             sbrExtrEnv->YBufferScale,
+             sbrExtrEnv->YBufferWriteOffset,
+             eData->transient_info
+             );
+
+  }
+  else
+  {
   FDKsbrEnc_transientDetect(&hEnvChan->sbrTransientDetector,
                              sbrExtrEnv->YBuffer,
                              sbrExtrEnv->YBufferScale,
@@ -931,6 +1071,7 @@ FDKsbrEnc_extractSbrEnvelope1 (
                              sbrExtrEnv->YBufferSzShift,
                              sbrExtrEnv->time_step,
                              hEnvChan->SbrEnvFrame.frameMiddleSlot);
+  }
 
 
 
@@ -951,7 +1092,8 @@ FDKsbrEnc_extractSbrEnvelope1 (
                           sbrExtrEnv->YBufferSzShift,
                           h_con->nSfb[1],
                           sbrExtrEnv->time_step,
-                          sbrExtrEnv->no_cols);
+                          sbrExtrEnv->no_cols,
+                         &hEnvChan->encEnvData.global_tonality);
 
 
 }
@@ -1128,12 +1270,26 @@ FDKsbrEnc_extractSbrEnvelope2 (
         && ( ed->nEnvelopes == 1 ) )
    {
 
-     if (hEnvChan->encEnvData.ldGrid)
-       hEnvChan->encEnvData.currentAmpResFF = (AMP_RES)h_con->initAmpResFF;
-     else
+     if (h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+     {
+       /* Note: global_tonaliy_float_value == ((float)hEnvChan->encEnvData.global_tonality/((INT64)(1)<<(31-(19+2)))/0.524288*(2.0/3.0)));
+                threshold_float_value == ((float)h_con->thresholdAmpResFF_m/((INT64)(1)<<(31-(h_con->thresholdAmpResFF_e)))/0.524288*(2.0/3.0))); */
+       /* decision of SBR_AMP_RES */
+       if (fIsLessThan( /* global_tonality > threshold ? */
+             h_con->thresholdAmpResFF_m, h_con->thresholdAmpResFF_e,
+             hEnvChan->encEnvData.global_tonality, RELAXATION_SHIFT+2 )
+          )
+       {
+         hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+       }
+       else {
+         hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_3_0;
+       }
+     } else {
        hEnvChan->encEnvData.currentAmpResFF = SBR_AMP_RES_1_5;
+     }
 
-     if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
+      if ( hEnvChan->encEnvData.currentAmpResFF != hEnvChan->encEnvData.init_sbr_amp_res) {
 
         FDKsbrEnc_InitSbrHuffmanTables(&hEnvChan->encEnvData,
                                        &hEnvChan->sbrCodeEnvelope,
@@ -1172,7 +1328,12 @@ FDKsbrEnc_extractSbrEnvelope2 (
     }
 
     /* Low energy in low band fix */
-    if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03))
+    if ( hEnvChan->sbrTransientDetector.prevLowBandEnergy < hEnvChan->sbrTransientDetector.prevHighBandEnergy
+      && hEnvChan->sbrTransientDetector.prevHighBandEnergy > FL2FX_DBL(0.03)
+      /* The fix needs the non-fast transient detector running.
+         It sets prevLowBandEnergy and prevHighBandEnergy.      */
+      && !(h_con->sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+      )
     {
       int i;
 

libSBRenc/src/env_est.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,6 +127,7 @@ typedef SBR_EXTRACT_ENVELOPE *HANDLE_SBR_EXTRACT_ENVELOPE;
 
 struct ENV_CHANNEL
 {
+  FAST_TRAN_DETECTOR sbrFastTransientDetector;
   SBR_TRANSIENT_DETECTOR sbrTransientDetector;
   SBR_CODE_ENVELOPE sbrCodeEnvelope;
   SBR_CODE_ENVELOPE sbrCodeNoiseFloor;

libSBRenc/src/fram_gen.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -266,7 +266,7 @@ static void calcCtrlSignal (HANDLE_SBR_GRID hSbrGrid, FRAME_CLASS frameClass,
 
 static void ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
                                   HANDLE_SBR_FRAME_INFO hFrameInfo,
-                                  INT freq_res_fixfix);
+                                  FREQ_RES *freq_res_fixfix);
 
 
 /* table for 8 time slot index */
@@ -341,8 +341,9 @@ static const FREQ_RES freqRes_table_16[16] = {
 static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
                                  HANDLE_SBR_GRID hSbrGrid,
                                  int tranPosInternal,
-                                 int numberTimeSlots
-                               );
+                                 int numberTimeSlots,
+                                 UCHAR fResTransIsLow
+                                 );
 
 
 /*!
@@ -402,11 +403,10 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
   const int *v_tuningFreq = v_tuning + 3;
 
   hSbrEnvFrame->v_tuningSegm = v_tuningSegm;
-  INT freq_res_fixfix = hSbrEnvFrame->freq_res_fixfix;
 
   if (ldGrid) {
     /* in case there was a transient at the very end of the previous frame, start with a transient envelope */
-    if(v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance)){
+    if ( !tranFlag && v_transient_info_pre[1] && (numberTimeSlots - v_transient_info_pre[0] < minFrameTranDistance) ){
       tranFlag = 1;
       tranPos  = 0;
     }
@@ -529,7 +529,8 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
       generateFixFixOnly ( &(hSbrEnvFrame->SbrFrameInfo),
                            &(hSbrEnvFrame->SbrGrid),
                            tranPosInternal,
-                           numberTimeSlots
+                           numberTimeSlots,
+                           hSbrEnvFrame->fResTransIsLow
                            );
 
       return &(hSbrEnvFrame->SbrFrameInfo);
@@ -677,7 +678,7 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
   ---------------------------------------------------------------------------*/
   ctrlSignal2FrameInfo (&hSbrEnvFrame->SbrGrid,
                         &hSbrEnvFrame->SbrFrameInfo,
-                        freq_res_fixfix);
+                         hSbrEnvFrame->freq_res_fixfix);
 
   return &hSbrEnvFrame->SbrFrameInfo;
 }
@@ -692,7 +693,8 @@ FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
 static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
                                  HANDLE_SBR_GRID hSbrGrid,
                                  int tranPosInternal,
-                                 int numberTimeSlots
+                                 int numberTimeSlots,
+                                 UCHAR fResTransIsLow
                                )
 {
   int nEnv, i, k=0, tranIdx;
@@ -727,8 +729,12 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
   /* adjust segment-frequency-resolution according to the segment-length */
   for (i=0; i<nEnv; i++){
     k = hSbrFrameInfo->borders[i+1] - hSbrFrameInfo->borders[i];
-    hSbrFrameInfo->freqRes[i] = freqResTable[k];
-    hSbrGrid->v_f[i] = freqResTable[k];
+    if (!fResTransIsLow)
+      hSbrFrameInfo->freqRes[i] = freqResTable[k];
+    else
+      hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
+
+    hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i];
   }
 
   hSbrFrameInfo->nEnvelopes = nEnv;
@@ -765,15 +771,16 @@ static void generateFixFixOnly ( HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
 
 *******************************************************************************/
 void
-FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME  hSbrEnvFrame,
-                          INT allowSpread,
-                          INT numEnvStatic,
-                          INT staticFraming,
-                          INT timeSlots,
-                          INT freq_res_fixfix
-                          ,int ldGrid
-                          )
-
+FDKsbrEnc_initFrameInfoGenerator (
+              HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+              INT       allowSpread,
+              INT       numEnvStatic,
+              INT       staticFraming,
+              INT       timeSlots,
+        const FREQ_RES* freq_res_fixfix
+             ,UCHAR     fResTransIsLow,
+              INT       ldGrid
+        )
 {                               /* FH 00-06-26 */
 
   FDKmemclear(hSbrEnvFrame,sizeof(SBR_ENVELOPE_FRAME ));
@@ -786,7 +793,9 @@ FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME  hSbrEnvFrame,
   hSbrEnvFrame->allowSpread = allowSpread;
   hSbrEnvFrame->numEnvStatic = numEnvStatic;
   hSbrEnvFrame->staticFraming = staticFraming;
-  hSbrEnvFrame->freq_res_fixfix = freq_res_fixfix;
+  hSbrEnvFrame->freq_res_fixfix[0] = freq_res_fixfix[0];
+  hSbrEnvFrame->freq_res_fixfix[1] = freq_res_fixfix[1];
+  hSbrEnvFrame->fResTransIsLow     = fResTransIsLow;
 
   hSbrEnvFrame->length_v_bord = 0;
   hSbrEnvFrame->length_v_bordFollow = 0;
@@ -804,6 +813,7 @@ FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME  hSbrEnvFrame,
       hSbrEnvFrame->dmin = 2;
       hSbrEnvFrame->dmax = 16;
       hSbrEnvFrame->frameMiddleSlot = FRAME_MIDDLE_SLOT_512LD;
+      hSbrEnvFrame->SbrGrid.bufferFrameStart = 0;
   } else
   switch(timeSlots){
     case NUMBER_TIME_SLOTS_1920:
@@ -1862,19 +1872,28 @@ createDefFrameInfo(HANDLE_SBR_FRAME_INFO hSbrFrameInfo, INT nEnv, INT nTimeSlots
  Functionname:  ctrlSignal2FrameInfo
  *******************************************************************************
 
- Description: Calculates frame_info struct from control signal.
+ Description: Convert "clear-text" sbr_grid() to "frame info" used by the
+              envelope and noise floor estimators.
+              This is basically (except for "low level" calculations) the
+              bitstream decoder defined in the MPEG-4 standard, sub clause
+              4.6.18.3.3, Time / Frequency Grid.  See inline comments for
+              explanation of the shorten and noise border algorithms.
 
  Arguments:   hSbrGrid - source
               hSbrFrameInfo - destination
+              freq_res_fixfix - frequency resolution for FIXFIX frames
 
  Return:      void; hSbrFrameInfo contains the updated FRAME_INFO struct
 
 *******************************************************************************/
 static void
-ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
-                      HANDLE_SBR_FRAME_INFO hSbrFrameInfo,
-                      INT freq_res_fixfix)
+ctrlSignal2FrameInfo (
+        HANDLE_SBR_GRID        hSbrGrid,       /* input : the grid handle       */
+        HANDLE_SBR_FRAME_INFO  hSbrFrameInfo,  /* output: the frame info handle */
+        FREQ_RES              *freq_res_fixfix /* in/out: frequency resolution for FIXFIX frames */
+        )
 {
+  INT frameSplit = 0;
   INT nEnv = 0, border = 0, i, k, p /*?*/;
   INT *v_r = hSbrGrid->bs_rel_bord;
   INT *v_f = hSbrGrid->v_f;
@@ -1887,17 +1906,10 @@ ctrlSignal2FrameInfo (HANDLE_SBR_GRID hSbrGrid,
   case FIXFIX:
     createDefFrameInfo(hSbrFrameInfo, hSbrGrid->bs_num_env, numberTimeSlots);
 
-    /* At this point all frequency resolutions are set to FREQ_RES_HIGH, so
-     * only if freq_res_fixfix is set to FREQ_RES_LOW, they all have to be
-     * changed.
-     * snd */
-    if (freq_res_fixfix == FREQ_RES_LOW) {
-      for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
-        hSbrFrameInfo->freqRes[i] = FREQ_RES_LOW;
-      }
+    frameSplit = (hSbrFrameInfo->nEnvelopes > 1);
+    for (i = 0; i < hSbrFrameInfo->nEnvelopes; i++) {
+      hSbrGrid->v_f[i] = hSbrFrameInfo->freqRes[i] = freq_res_fixfix[frameSplit];
     }
-    /* ELD: store current frequency resolution */
-    hSbrGrid->v_f[0] = hSbrFrameInfo->freqRes[0];
     break;
 
   case FIXVAR:

libSBRenc/src/fram_gen.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -89,6 +89,7 @@ amm-info@iis.fraunhofer.de
 #define _FRAM_GEN_H
 
 #include "sbr_def.h" /* for MAX_ENVELOPES and MAX_NOISE_ENVELOPES in struct FRAME_INFO and CODEC_TYPE */
+#include "sbr_encoder.h" /* for FREQ_RES */
 
 #define MAX_ENVELOPES_VARVAR MAX_ENVELOPES /*!< worst case number of envelopes in a VARVAR frame */
 #define MAX_ENVELOPES_FIXVAR_VARFIX 4 /*!< worst case number of envelopes in VARFIX and FIXVAR frames */
@@ -114,7 +115,7 @@ typedef enum {
 #define NUMBER_TIME_SLOTS_1920  15
 
 #define LD_PRETRAN_OFF           3
-#define FRAME_MIDDLE_SLOT_512LD  0
+#define FRAME_MIDDLE_SLOT_512LD  4
 #define NUMBER_TIME_SLOTS_512LD  8
 #define TRANSIENT_OFFSET_LD      0
 
@@ -248,9 +249,10 @@ typedef struct
   INT frameMiddleSlot;      /*!< transient detector offset in SBR timeslots */
 
   /* basic tuning parameters */
-  INT staticFraming;        /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */
-  INT numEnvStatic;         /*!< number of envelopes per frame for static framing */
-  INT freq_res_fixfix;      /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX */
+  INT staticFraming;           /*!< 1: run static framing in time, i.e. exclusive use of bs_frame_class = FIXFIX */
+  INT numEnvStatic;            /*!< number of envelopes per frame for static framing */
+  FREQ_RES freq_res_fixfix[2]; /*!< envelope frequency resolution to use for bs_frame_class = FIXFIX; single env and split */
+  UCHAR fResTransIsLow;        /*!< frequency resolution for transient frames - always low (0) or according to table (1) */
 
   /* expert tuning parameters */
   const int *v_tuningSegm;  /*!< segment lengths to use around transient */
@@ -286,14 +288,16 @@ typedef SBR_ENVELOPE_FRAME *HANDLE_SBR_ENVELOPE_FRAME;
 
 
 void
-FDKsbrEnc_initFrameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME  hSbrEnvFrame,
-                          INT allowSpread,
-                          INT numEnvStatic,
-                          INT staticFraming,
-                          INT timeSlots,
-                          INT freq_res_fixfix
-                          ,int ldGrid
-                          );
+FDKsbrEnc_initFrameInfoGenerator (
+              HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,
+              INT       allowSpread,
+              INT       numEnvStatic,
+              INT       staticFraming,
+              INT       timeSlots,
+        const FREQ_RES* freq_res_fixfix
+             ,UCHAR     fResTransIsLow,
+              INT       ldGrid
+        );
 
 HANDLE_SBR_FRAME_INFO
 FDKsbrEnc_frameInfoGenerator (HANDLE_SBR_ENVELOPE_FRAME hSbrEnvFrame,

libSBRenc/src/mh_det.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -663,10 +663,27 @@ static void transientCleanUp(FIXP_DBL **quotaBuffer,
 }
 
 
-/**************************************************************************/
+/*****************************************************************************/
 /*!
-  \brief     Do detection for one tonality estimate.
+  \brief     Detection for one tonality estimate.
 
+  This is the actual missing harmonics detection, using information from the
+  previous detection.
+
+  If a missing harmonic was detected (in a previous frame) due to too high
+  tonality differences, but there was not enough tonality difference in the
+  current frame, the detection algorithm still continues to trace the strongest
+  tone in the scalefactor band (assuming that this is the tone that is going to
+  be replaced in the decoder). This is done to avoid abrupt endings of sines
+  fading out (e.g. in the glockenspiel).
+
+  The function also tries to estimate where one sine is going to be replaced
+  with multiple sines (due to the patching). This is done by comparing the
+  tonality flatness measure of the original and the SBR signal.
+
+  The function also tries to estimate (for the scalefactor bands only
+  containing one qmf subband) when a strong tone in the original will be
+  replaced by a strong tone in the adjacent QMF subband.
 
   \return    none.
 
@@ -694,10 +711,10 @@ static void detection(FIXP_DBL *quotaBuffer,
   for(i=0;i<nSfb;i++){
 
     thresTemp = (guideVectors.guideVectorDiff[i] != FL2FXCONST_DBL(0.0f))
-                  ? fixMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide)
+                  ? fMax(fMult(mhThresh.decayGuideDiff,guideVectors.guideVectorDiff[i]), mhThresh.thresHoldDiffGuide)
                   : mhThresh.thresHoldDiff;
 
-    thresTemp = fixMin(thresTemp, mhThresh.thresHoldDiff);
+    thresTemp = fMin(thresTemp, mhThresh.thresHoldDiff);
 
     if(pDiffVecScfb[i] > thresTemp){
       pHarmVec[i] = 1;
@@ -813,8 +830,11 @@ static void detectionWithPrediction(FIXP_DBL **quotaBuffer,
 
   if(newDetectionAllowed){
 
+    /* Since we don't want to use the transient region for detection (since the tonality values
+       tend to be a bit unreliable for this region) the guide-values are copied to the current
+       starting point. */
     if(totNoEst > 1){
-      start = detectionStart;
+      start = detectionStart+1;
 
       if (start != 0) {
         FDKmemcpy(guideVectors[start].guideVectorDiff,guideVectors[0].guideVectorDiff,nSfb*sizeof(FIXP_DBL));

libSBRenc/src/nf_est.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -268,8 +268,9 @@ qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel,              /*!< Pointer to v
   /*
    * Add a noise floor offset to compensate for bias in the detector
    *****************************************************************/
-  if(!missingHarmonicFlag)
-    *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING);
+  if(!missingHarmonicFlag) {
+    *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING;
+  }
 
   /*
    * check to see that we don't exceed the maximum allowed level
@@ -297,7 +298,7 @@ FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFlo
                          SCHAR    *indexVector,              /*!< Index vector to obtain the patched data. */
                          INT missingHarmonicsFlag,           /*!< Flag indicating if a strong tonal component will be missing. */
                          INT startIndex,                     /*!< Start index. */
-                         int numberOfEstimatesPerFrame,      /*!< The number of tonality estimates per frame. */
+                         UINT numberOfEstimatesPerFrame,     /*!< The number of tonality estimates per frame. */
                          int transientFrame,                 /*!< A flag indicating if a transient is present. */
                          INVF_MODE* pInvFiltLevels,          /*!< Pointer to the vector holding the inverse filtering levels. */
                          UINT sbrSyntaxFlags

libSBRenc/src/nf_est.h

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -119,7 +119,7 @@ FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFlo
                          SCHAR* indexVector,                 /*!< Index vector to obtain the patched data. */
                          INT missingHarmonicsFlag,           /*!< Flag indicating if a strong tonal component will be missing. */
                          INT startIndex,                     /*!< Start index. */
-                         int numberOfEstimatesPerFrame,      /*!< The number of tonality estimates per frame. */
+                         UINT numberOfEstimatesPerFrame,     /*!< The number of tonality estimates per frame. */
                          INT transientFrame,                 /*!< A flag indicating if a transient is present. */
                          INVF_MODE* pInvFiltLevels,          /*!< Pointer to the vector holding the inverse filtering levels. */
                          UINT sbrSyntaxFlags

libSBRenc/src/ps_bitenc.cpp

@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -261,21 +261,23 @@ static const UINT opdDeltaTime_Code[] =
   0x00000001,  0x00000002,  0x00000001,  0x00000007,  0x00000006,  0000000000,  0x00000002,  0x00000003
 };
 
-static const INT psBands[] =
+static INT getNoBands(const INT mode)
 {
-  PS_BANDS_COARSE,
-  PS_BANDS_MID
-};
+  INT noBands = 0;
 
-static INT getNoBands(UINT mode)
-{
-  if(mode>=6)
-    return 0;
+  switch (mode) {
+    case 0: case 3: /* coarse */
+      noBands = PS_BANDS_COARSE;
+      break;
+    case 1: case 4: /* mid */
+      noBands = PS_BANDS_MID;
+      break;
+    case 2: case 5: /* fine not supported */
+    default:        /* coarse as default */
+      noBands = PS_BANDS_COARSE;
+  }
 
-  if(mode>=3)
-    mode = mode-3;
-
-  return psBands[mode];
+  return noBands;
 }
 
 static INT getIIDRes(INT iidMode)
@@ -524,7 +526,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT        psOut,
       bitCnt += FDKsbrEnc_EncodeIpd( hBitBuf,
                            psOut->ipd[env],
                            ipdLast,
-                           getNoBands((UINT)psOut->iidMode),
+                           getNoBands(psOut->iidMode),
                            psOut->deltaIPD[env],
                            &error);
 
@@ -532,7 +534,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT        psOut,
       bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf,
                            psOut->opd[env],
                            opdLast,
-                           getNoBands((UINT)psOut->iidMode),
+                           getNoBands(psOut->iidMode),
                            psOut->deltaOPD[env],
                            &error );
     }
@@ -661,7 +663,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT   psOut,
         bitCnt += FDKsbrEnc_EncodeIid( hBitBuf,
                              psOut->iid[env],
                              iidLast,
-                             getNoBands((UINT)psOut->iidMode),
+                             getNoBands(psOut->iidMode),
                              (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode),
                              psOut->deltaIID[env],
                              &error );
@@ -677,7 +679,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT   psOut,
         bitCnt += FDKsbrEnc_EncodeIcc( hBitBuf,
                              psOut->icc[env],
                              iccLast,
-                             getNoBands((UINT)psOut->iccMode),
+                             getNoBands(psOut->iccMode),
                              psOut->deltaICC[env],
                              &error);