1 /* -----------------------------------------------------------------------------
2 Software License for The Fraunhofer FDK AAC Codec Library for Android
3 
4 © Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
5 Forschung e.V. All rights reserved.
6 
7  1.    INTRODUCTION
8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
10 scheme for digital audio. This FDK AAC Codec software is intended to be used on
11 a wide variety of Android devices.
12 
13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
14 general perceptual audio codecs. AAC-ELD is considered the best-performing
15 full-bandwidth communications codec by independent studies and is widely
16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG
17 specifications.
18 
19 Patent licenses for necessary patent claims for the FDK AAC Codec (including
20 those of Fraunhofer) may be obtained through Via Licensing
21 (www.vialicensing.com) or through the respective patent owners individually for
22 the purpose of encoding or decoding bit streams in products that are compliant
23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
24 Android devices already license these patent claims through Via Licensing or
25 directly from the patent owners, and therefore FDK AAC Codec software may
26 already be covered under those patent licenses when it is used for those
27 licensed purposes only.
28 
29 Commercially-licensed AAC software libraries, including floating-point versions
30 with enhanced sound quality, are also available from Fraunhofer. Users are
31 encouraged to check the Fraunhofer website for additional applications
32 information and documentation.
33 
34 2.    COPYRIGHT LICENSE
35 
36 Redistribution and use in source and binary forms, with or without modification,
37 are permitted without payment of copyright license fees provided that you
38 satisfy the following conditions:
39 
40 You must retain the complete text of this software license in redistributions of
41 the FDK AAC Codec or your modifications thereto in source code form.
42 
43 You must retain the complete text of this software license in the documentation
44 and/or other materials provided with redistributions of the FDK AAC Codec or
45 your modifications thereto in binary form. You must make available free of
46 charge copies of the complete source code of the FDK AAC Codec and your
47 modifications thereto to recipients of copies in binary form.
48 
49 The name of Fraunhofer may not be used to endorse or promote products derived
50 from this library without prior written permission.
51 
52 You may not charge copyright license fees for anyone to use, copy or distribute
53 the FDK AAC Codec software or your modifications thereto.
54 
55 Your modified versions of the FDK AAC Codec must carry prominent notices stating
56 that you changed the software and the date of any change. For modified versions
57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
59 AAC Codec Library for Android."
60 
61 3.    NO PATENT LICENSE
62 
63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
65 Fraunhofer provides no warranty of patent non-infringement with respect to this
66 software.
67 
68 You may use this FDK AAC Codec software or modifications thereto only for
69 purposes that are authorized by appropriate patent licenses.
70 
71 4.    DISCLAIMER
72 
73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
75 including but not limited to the implied warranties of merchantability and
76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
78 or consequential damages, including but not limited to procurement of substitute
79 goods or services; loss of use, data, or profits, or business interruption,
80 however caused and on any theory of liability, whether in contract, strict
81 liability, or tort (including negligence), arising in any way out of the use of
82 this software, even if advised of the possibility of such damage.
83 
84 5.    CONTACT INFORMATION
85 
86 Fraunhofer Institute for Integrated Circuits IIS
87 Attention: Audio and Multimedia Departments - FDK AAC LL
88 Am Wolfsmantel 33
89 91058 Erlangen, Germany
90 
91 www.iis.fraunhofer.de/amm
92 amm-info@iis.fraunhofer.de
93 ----------------------------------------------------------------------------- */
94 
95 /******************* MPEG transport format encoder library *********************
96 
97    Author(s):   Manuel Jander
98 
99    Description: MPEG Transport data tables
100 
101 *******************************************************************************/
102 
103 #ifndef TP_DATA_H
104 #define TP_DATA_H
105 
106 #include "machine_type.h"
107 #include "FDK_audio.h"
108 #include "FDK_bitstream.h"
109 
110 /*
111  * Configuration
112  */
113 
114 #define TP_USAC_MAX_SPEAKERS (24)
115 
116 #define TP_USAC_MAX_EXT_ELEMENTS ((24))
117 
118 #define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS)
119 
120 #define TP_USAC_MAX_CONFIG_LEN                                         \
121   512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \
122          AudioPreRoll() (285) */
123 
124 #define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \
125   (1) /* Number of frames for config change in USAC */
126 
127 enum {
128   TPDEC_FLUSH_OFF = 0,
129   TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
130   TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
131   TPDEC_USAC_DASH_IPF_FLUSH_ON = 3
132 };
133 
134 enum {
135   TPDEC_BUILD_UP_OFF = 0,
136   TPDEC_RSV60_BUILD_UP_ON = 1,
137   TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
138   TPDEC_USAC_BUILD_UP_ON = 3,
139   TPDEC_RSV60_BUILD_UP_IDLE = 4,
140   TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
141 };
142 
143 /**
144  * ProgramConfig struct.
145  */
146 /* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
147 #define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
148 #define PC_LFE_CHANNELS_MAX 4
149 #define PC_ASSOCDATA_MAX 8
150 #define PC_CCEL_MAX 16 /* CC elements */
151 #define PC_COMMENTLENGTH 256
152 #define PC_NUM_HEIGHT_LAYER 3
153 
154 typedef struct {
155   /* PCE bitstream elements: */
156   UCHAR ElementInstanceTag;
157   UCHAR Profile;
158   UCHAR SamplingFrequencyIndex;
159   UCHAR NumFrontChannelElements;
160   UCHAR NumSideChannelElements;
161   UCHAR NumBackChannelElements;
162   UCHAR NumLfeChannelElements;
163   UCHAR NumAssocDataElements;
164   UCHAR NumValidCcElements;
165 
166   UCHAR MonoMixdownPresent;
167   UCHAR MonoMixdownElementNumber;
168 
169   UCHAR StereoMixdownPresent;
170   UCHAR StereoMixdownElementNumber;
171 
172   UCHAR MatrixMixdownIndexPresent;
173   UCHAR MatrixMixdownIndex;
174   UCHAR PseudoSurroundEnable;
175 
176   UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
177   UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
178   UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
179 
180   UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
181   UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
182   UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
183 
184   UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
185   UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
186   UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
187 
188   UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
189 
190   UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
191 
192   UCHAR CcElementIsIndSw[PC_CCEL_MAX];
193   UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
194 
195   UCHAR CommentFieldBytes;
196   UCHAR Comment[PC_COMMENTLENGTH];
197 
198   /* Helper variables for administration: */
199   UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */
200   UCHAR
201   NumChannels; /*!< Amount of audio channels summing all channel elements
202                   including LFEs */
203   UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs
204                                  and CPEs */
205   UCHAR elCounter;
206 
207 } CProgramConfig;
208 
209 typedef enum {
210   ASCEXT_UNKOWN = -1,
211   ASCEXT_SBR = 0x2b7,
212   ASCEXT_PS = 0x548,
213   ASCEXT_MPS = 0x76a,
214   ASCEXT_SAOC = 0x7cb,
215   ASCEXT_LDMPS = 0x7cc
216 
217 } TP_ASC_EXTENSION_ID;
218 
219 /**
220  * GaSpecificConfig struct
221  */
222 typedef struct {
223   UINT m_frameLengthFlag;
224   UINT m_dependsOnCoreCoder;
225   UINT m_coreCoderDelay;
226 
227   UINT m_extensionFlag;
228   UINT m_extensionFlag3;
229 
230   UINT m_layer;
231   UINT m_numOfSubFrame;
232   UINT m_layerLength;
233 
234 } CSGaSpecificConfig;
235 
236 typedef enum {
237   ELDEXT_TERM = 0x0,         /* Termination tag */
238   ELDEXT_SAOC = 0x1,         /* SAOC config */
239   ELDEXT_LDSAC = 0x2,        /* LD MPEG Surround config */
240   ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */
241   /* reserved */
242 } ASC_ELD_EXT_TYPE;
243 
244 typedef struct {
245   UCHAR m_frameLengthFlag;
246 
247   UCHAR m_sbrPresentFlag;
248   UCHAR
249   m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
250   UCHAR m_sbrSamplingRate;
251   UCHAR m_sbrCrcFlag;
252   UINT m_downscaledSamplingFrequency;
253 
254 } CSEldSpecificConfig;
255 
256 typedef struct {
257   USAC_EXT_ELEMENT_TYPE usacExtElementType;
258   USHORT usacExtElementConfigLength;
259   USHORT usacExtElementDefaultLength;
260   UCHAR usacExtElementPayloadFrag;
261   UCHAR usacExtElementHasAudioPreRoll;
262 } CSUsacExtElementConfig;
263 
264 typedef struct {
265   MP4_ELEMENT_ID usacElementType;
266   UCHAR m_noiseFilling;
267   UCHAR m_harmonicSBR;
268   UCHAR m_interTes;
269   UCHAR m_pvc;
270   UCHAR m_stereoConfigIndex;
271   CSUsacExtElementConfig extElement;
272 } CSUsacElementConfig;
273 
274 typedef struct {
275   UCHAR m_frameLengthFlag;
276   UCHAR m_coreSbrFrameLengthIndex;
277   UCHAR m_sbrRatioIndex;
278   UCHAR m_nUsacChannels; /* number of audio channels signaled in
279                             UsacDecoderConfig() / rsv603daDecoderConfig() via
280                             numElements and usacElementType */
281   UCHAR m_channelConfigurationIndex;
282   UINT m_usacNumElements;
283   CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS];
284 
285   UCHAR numAudioChannels;
286   UCHAR m_usacConfigExtensionPresent;
287   UCHAR elementLengthPresent;
288   UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN];
289   USHORT UsacConfigBits;
290 } CSUsacConfig;
291 
292 /**
293  * Audio configuration struct, suitable for encoder and decoder configuration.
294  */
295 typedef struct {
296   /* XYZ Specific Data */
297   union {
298     CSGaSpecificConfig
299         m_gaSpecificConfig; /**< General audio specific configuration. */
300     CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
301     CSUsacConfig m_usacConfig; /**< USAC specific configuration               */
302   } m_sc;
303 
304   /* Common ASC parameters */
305   CProgramConfig m_progrConfigElement; /**< Program configuration. */
306 
307   AUDIO_OBJECT_TYPE m_aot;  /**< Audio Object Type.  */
308   UINT m_samplingFrequency; /**< Samplerate. */
309   UINT m_samplesPerFrame;   /**< Amount of samples per frame.   */
310   UINT m_directMapping; /**< Document this please !!                         */
311 
312   AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */
313   UINT m_extensionSamplingFrequency;            /**< Samplerate            */
314 
315   SCHAR m_channelConfiguration; /**< Channel configuration index */
316 
317   SCHAR m_epConfig;  /**< Error protection index                           */
318   SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag                     */
319   SCHAR m_rvlcFlag;  /**< aacScalefactorDataResilienceFlag                 */
320   SCHAR m_hcrFlag;   /**< aacSpectralDataResilienceFlag                    */
321 
322   SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the
323                              bitstream               */
324   SCHAR
325   m_psPresentFlag; /**< Flag indicating the presence of parametric stereo
326                       data in the bitstream */
327   UCHAR m_samplingFrequencyIndex;          /**< Samplerate index          */
328   UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
329   SCHAR m_extensionChannelConfiguration;   /**< Channel configuration index   */
330 
331   UCHAR
332   configMode; /**< The flag indicates if the callback shall work in memory
333                  allocation mode or in config change detection mode */
334   UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
335                              parameter has changed that requires a memory
336                              reconfiguration, otherwise it will be cleared */
337   UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
338                              parameter has changed that requires a memory
339                              reconfiguration, otherwise it will be cleared */
340   UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
341                              parameter has changed that requires a memory
342                              reconfiguration, otherwise it will be cleared */
343 
344   UCHAR
345   config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
346   UINT configBits;                /**< Configuration length in bits */
347 
348 } CSAudioSpecificConfig;
349 
350 typedef struct {
351   SCHAR flushCnt;      /**< Flush frame counter */
352   UCHAR flushStatus;   /**< Flag indicates flush mode: on|off */
353   SCHAR buildUpCnt;    /**< Build up frame counter */
354   UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
355   UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
356                        needs to be initialized again via callback. Make sure
357                        that memory is freed before initialization. */
358   UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
359                            right truncation occured before */
360   UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
361                            even if new config is the same */
362 } CCtrlCFGChange;
363 
364 typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
365                                 const UCHAR configMode, UCHAR *configChanged);
366 typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
367 typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
368 typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
369                        const AUDIO_OBJECT_TYPE coreCodec,
370                        const INT samplingRate, const INT frameSize,
371                        const INT stereoConfigIndex,
372                        const INT coreSbrFrameLengthIndex, const INT configBytes,
373                        const UCHAR configMode, UCHAR *configChanged);
374 
375 typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
376                        const INT sampleRateIn, const INT sampleRateOut,
377                        const INT samplesPerFrame,
378                        const AUDIO_OBJECT_TYPE coreCodec,
379                        const MP4_ELEMENT_ID elementID, const INT elementIndex,
380                        const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
381                        const UCHAR configMode, UCHAR *configChanged,
382                        const INT downscaleFactor);
383 
384 typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
385 
386 typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
387                           const INT fullPayloadLength, const INT payloadType,
388                           const INT subStreamIndex, const INT payloadStart,
389                           const AUDIO_OBJECT_TYPE);
390 
391 typedef struct {
392   cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
393                                       notify callback.  */
394   void *cbUpdateConfigData; /*!< User data pointer for Config change notify
395                                callback. */
396   cbFreeMem_t cbFreeMem;    /*!< Function pointer for free memory callback.  */
397   void *cbFreeMemData;      /*!< User data pointer for free memory callback. */
398   cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
399                                         control callback. */
400   void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
401                                 callback. */
402   cbSsc_t cbSsc;             /*!< Function pointer for SSC parser callback. */
403   void *cbSscData;           /*!< User data pointer for SSC parser callback. */
404   cbSbr_t cbSbr;   /*!< Function pointer for SBR header parser callback. */
405   void *cbSbrData; /*!< User data pointer for SBR header parser callback. */
406   cbUsac_t cbUsac;
407   void *cbUsacData;
408   cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
409                           loudnessInfoSet parser callback. */
410   void *cbUniDrcData;  /*!< User data pointer for uniDrcConfig and
411                           loudnessInfoSet parser callback. */
412 } CSTpCallBacks;
413 
414 static const UINT SamplingRateTable[] = {
415     96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
416     8000,  7350,  0,     0,     57600, 51200, 40000, 38400, 34150, 28800, 25600,
417     20000, 19200, 17075, 14400, 12800, 9600,  0,     0,     0,     0};
418 
getSamplingRateIndex(UINT samplingRate,UINT nBits)419 static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
420   UINT sf_index;
421   UINT tableSize = (1 << nBits) - 1;
422 
423   for (sf_index = 0; sf_index < tableSize; sf_index++) {
424     if (SamplingRateTable[sf_index] == samplingRate) break;
425   }
426 
427   if (sf_index > tableSize) {
428     return tableSize - 1;
429   }
430 
431   return sf_index;
432 }
433 
434 /*
435  * Get Channel count from channel configuration
436  */
getNumberOfTotalChannels(int channelConfig)437 static inline int getNumberOfTotalChannels(int channelConfig) {
438   switch (channelConfig) {
439     case 1:
440     case 2:
441     case 3:
442     case 4:
443     case 5:
444     case 6:
445       return channelConfig;
446     case 7:
447     case 12:
448     case 14:
449       return 8;
450     case 11:
451       return 7;
452     case 13:
453       return 24;
454     default:
455       return 0;
456   }
457 }
458 
getNumberOfEffectiveChannels(const int channelConfig)459 static inline int getNumberOfEffectiveChannels(
460     const int
461         channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
462   const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0};
463   return n[channelConfig];
464 }
465 
466 #endif /* TP_DATA_H */
467