/* * FAAC - Freeware Advanced Audio Coder * Copyright (C) 2001 Menno Bakker * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * * $Id: frame.h,v 1.27 2003/11/24 18:09:38 knik Exp $ */ #ifndef FRAME_H #define FRAME_H #ifdef HAVE_CONFIG_H #include "config.h" #endif #ifdef HAVE_SYS_TYPES_H # include #endif #ifdef HAVE_INTTYPES_H # include #endif #ifdef HAVE_STDINT_H # include #endif #ifndef HAVE_INT32_T typedef int int32_t; #endif #ifdef __cplusplus extern "C" { #endif /* __cplusplus */ #include "coder.h" #include "channels.h" #include "psych.h" #include "aacquant.h" #include "fft.h" #ifdef WIN32 #ifndef FAACAPI #define FAACAPI __stdcall #endif #else #ifndef FAACAPI #define FAACAPI #endif #endif #pragma pack(push, 1) typedef struct { psymodel_t *model; char *name; } psymodellist_t; #include typedef struct { /* number of channels in AAC file */ unsigned int numChannels; /* samplerate of AAC file */ unsigned long sampleRate; unsigned int sampleRateIdx; unsigned int usedBytes; /* frame number */ unsigned int frameNum; unsigned int flushFrame; /* Scalefactorband data */ SR_INFO *srInfo; /* sample buffers of current next and next next frame*/ double *sampleBuff[MAX_CHANNELS]; double *nextSampleBuff[MAX_CHANNELS]; double *next2SampleBuff[MAX_CHANNELS]; double *next3SampleBuff[MAX_CHANNELS]; double *ltpTimeBuff[MAX_CHANNELS]; /* Filterbank buffers */ double *sin_window_long; double *sin_window_short; double *kbd_window_long; double *kbd_window_short; double *freqBuff[MAX_CHANNELS]; double *overlapBuff[MAX_CHANNELS]; double *msSpectrum[MAX_CHANNELS]; /* Channel and Coder data for all channels */ CoderInfo coderInfo[MAX_CHANNELS]; ChannelInfo channelInfo[MAX_CHANNELS]; /* Psychoacoustics data */ PsyInfo psyInfo[MAX_CHANNELS]; GlobalPsyInfo gpsyInfo; /* Configuration data */ faacEncConfiguration config; psymodel_t *psymodel; /* quantizer specific config */ AACQuantCfg aacquantCfg; /* FFT Tables */ FFT_Tables fft_tables; /* output bits difference in average bitrate mode */ int bitDiff; } faacEncStruct, *faacEncHandle; int FAACAPI faacEncGetVersion(char **faac_id_string, char **faac_copyright_string); int FAACAPI faacEncGetDecoderSpecificInfo(faacEncHandle hEncoder, unsigned char** ppBuffer, unsigned long* pSizeOfDecoderSpecificInfo); faacEncConfigurationPtr FAACAPI faacEncGetCurrentConfiguration(faacEncHandle hEncoder); int FAACAPI faacEncSetConfiguration (faacEncHandle hEncoder, faacEncConfigurationPtr config); faacEncHandle FAACAPI faacEncOpen(unsigned long sampleRate, unsigned int numChannels, unsigned long *inputSamples, unsigned long *maxOutputBytes); int FAACAPI faacEncEncode(faacEncHandle hEncoder, int32_t *inputBuffer, unsigned int samplesInput, unsigned char *outputBuffer, unsigned int bufferSize ); int FAACAPI faacEncClose(faacEncHandle hEncoder); #pragma pack(pop) #ifdef __cplusplus } #endif /* __cplusplus */ #endif /* FRAME_H */