summaryrefslogtreecommitdiffstats
path: root/fdk-aac/libDRCdec/src/drcDec_gainDecoder.h
diff options
context:
space:
mode:
Diffstat (limited to 'fdk-aac/libDRCdec/src/drcDec_gainDecoder.h')
-rw-r--r--fdk-aac/libDRCdec/src/drcDec_gainDecoder.h264
1 files changed, 264 insertions, 0 deletions
diff --git a/fdk-aac/libDRCdec/src/drcDec_gainDecoder.h b/fdk-aac/libDRCdec/src/drcDec_gainDecoder.h
new file mode 100644
index 0000000..2f4df4c
--- /dev/null
+++ b/fdk-aac/libDRCdec/src/drcDec_gainDecoder.h
@@ -0,0 +1,264 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1. INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2. COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3. NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4. DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5. CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/************************* MPEG-D DRC decoder library **************************
+
+ Author(s):
+
+ Description:
+
+*******************************************************************************/
+
+#ifndef DRCDEC_GAINDECODER_H
+#define DRCDEC_GAINDECODER_H
+
+#include "drcDecoder.h"
+
+/* Definitions common to gainDecoder submodule */
+
+#define NUM_LNB_FRAMES \
+ 5 /* previous frame + this frame + one frame for DM_REGULAR_DELAY + (maximum \
+ delaySamples)/frameSize */
+
+/* QMF64 */
+#define SUBBAND_NUM_BANDS_QMF64 64
+#define SUBBAND_DOWNSAMPLING_FACTOR_QMF64 64
+#define SUBBAND_ANALYSIS_DELAY_QMF64 320
+
+/* QMF71 (according to ISO/IEC 23003-1:2007) */
+#define SUBBAND_NUM_BANDS_QMF71 71
+#define SUBBAND_DOWNSAMPLING_FACTOR_QMF71 64
+#define SUBBAND_ANALYSIS_DELAY_QMF71 320 + 384
+
+/* STFT256 (according to ISO/IEC 23008-3:2015/AMD3) */
+#define SUBBAND_NUM_BANDS_STFT256 256
+#define SUBBAND_DOWNSAMPLING_FACTOR_STFT256 256
+#define SUBBAND_ANALYSIS_DELAY_STFT256 256
+
+typedef enum {
+ GAIN_DEC_DRC1,
+ GAIN_DEC_DRC1_DRC2,
+ GAIN_DEC_DRC2,
+ GAIN_DEC_DRC3,
+ GAIN_DEC_DRC2_DRC3
+} GAIN_DEC_LOCATION;
+
+typedef struct {
+ FIXP_DBL gainLin; /* e = 7 */
+ SHORT time;
+} NODE_LIN;
+
+typedef struct {
+ GAIN_INTERPOLATION_TYPE gainInterpolationType;
+ int nNodes[NUM_LNB_FRAMES]; /* number of nodes, saturated to 16 */
+ NODE_LIN linearNode[NUM_LNB_FRAMES][16];
+} LINEAR_NODE_BUFFER;
+
+typedef struct {
+ int lnbPointer;
+ LINEAR_NODE_BUFFER linearNodeBuffer[12];
+ LINEAR_NODE_BUFFER dummyLnb;
+ FIXP_DBL channelGain[8][NUM_LNB_FRAMES]; /* e = 8 */
+} DRC_GAIN_BUFFERS;
+
+typedef struct {
+ int activeDrcOffset;
+ DRC_INSTRUCTIONS_UNI_DRC* pInst;
+ DRC_COEFFICIENTS_UNI_DRC* pCoef;
+
+ DUCKING_MODIFICATION duckingModificationForChannelGroup[8];
+ SCHAR channelGroupForChannel[8];
+
+ UCHAR bandCountForChannelGroup[8];
+ UCHAR gainElementForGroup[8];
+ UCHAR channelGroupIsParametricDrc[8];
+ UCHAR gainElementCount; /* number of different DRC gains inluding all DRC
+ bands */
+ int lnbIndexForChannel[8][NUM_LNB_FRAMES];
+ int subbandGainsReady;
+} ACTIVE_DRC;
+
+typedef struct {
+ int deltaTminDefault;
+ INT frameSize;
+ FIXP_DBL loudnessNormalisationGainDb;
+ DELAY_MODE delayMode;
+
+ int nActiveDrcs;
+ ACTIVE_DRC activeDrc[MAX_ACTIVE_DRCS];
+ int multiBandActiveDrcIndex;
+ int channelGainActiveDrcIndex;
+ FIXP_DBL channelGain[8]; /* e = 8 */
+
+ DRC_GAIN_BUFFERS drcGainBuffers;
+ FIXP_DBL subbandGains[12][4 * 1024 / 256];
+ FIXP_DBL dummySubbandGains[4 * 1024 / 256];
+
+ int status;
+ int timeDomainSupported;
+ SUBBAND_DOMAIN_MODE subbandDomainSupported;
+} DRC_GAIN_DECODER, *HANDLE_DRC_GAIN_DECODER;
+
+/* init functions */
+DRC_ERROR
+drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec);
+
+DRC_ERROR
+drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec, const int frameSize,
+ const int sampleRate);
+
+DRC_ERROR
+drcDec_GainDecoder_SetCodecDependentParameters(
+ HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode,
+ const int timeDomainSupported,
+ const SUBBAND_DOMAIN_MODE subbandDomainSupported);
+
+DRC_ERROR
+drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec,
+ HANDLE_UNI_DRC_CONFIG hUniDrcConfig,
+ const UCHAR numSelectedDrcSets,
+ const SCHAR* selectedDrcSetIds,
+ const UCHAR* selectedDownmixIds);
+
+/* close functions */
+DRC_ERROR
+drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec);
+
+/* process functions */
+
+/* call drcDec_GainDecoder_Preprocess first */
+DRC_ERROR
+drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec,
+ HANDLE_UNI_DRC_GAIN hUniDrcGain,
+ const FIXP_DBL loudnessNormalizationGainDb,
+ const FIXP_SGL boost, const FIXP_SGL compress);
+
+/* Then call one of drcDec_GainDecoder_ProcessTimeDomain or
+ * drcDec_GainDecoder_ProcessSubbandDomain */
+DRC_ERROR
+drcDec_GainDecoder_ProcessTimeDomain(
+ HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,
+ const GAIN_DEC_LOCATION drcLocation, const int channelOffset,
+ const int drcChannelOffset, const int numChannelsProcessed,
+ const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer);
+
+DRC_ERROR
+drcDec_GainDecoder_ProcessSubbandDomain(
+ HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,
+ GAIN_DEC_LOCATION drcLocation, const int channelOffset,
+ const int drcChannelOffset, const int numChannelsProcessed,
+ const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[],
+ FIXP_DBL* audioIOBufferImag[]);
+
+DRC_ERROR
+drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec,
+ HANDLE_UNI_DRC_CONFIG hUniDrcConfig,
+ HANDLE_UNI_DRC_GAIN hUniDrcGain);
+
+DRC_ERROR
+drcDec_GainDecoder_SetLoudnessNormalizationGainDb(
+ HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb);
+
+int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec);
+
+int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec);
+
+void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec,
+ const int numChannels,
+ const int frameSize,
+ const FIXP_DBL* channelGainDb,
+ const int audioBufferChannelOffset,
+ FIXP_DBL* audioBuffer);
+
+#endif