adding ffmpeg specific version

1 files changed, 333 insertions, 0 deletions

diff --git a/ffmpeg1/libavcodec/aac.h b/ffmpeg1/libavcodec/aac.h
new file mode 100644
index 0000000..d586e27
--- /dev/null
+++ b/ffmpeg1/libavcodec/aac.h
@@ -0,0 +1,333 @@
+/*
+ * AAC definitions and structures
+ * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
+ * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * AAC definitions and structures
+ * @author Oded Shimon  ( ods15 ods15 dyndns org )
+ * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
+ */
+
+#ifndef AVCODEC_AAC_H
+#define AVCODEC_AAC_H
+
+#include "libavutil/float_dsp.h"
+#include "avcodec.h"
+#include "fft.h"
+#include "mpeg4audio.h"
+#include "sbr.h"
+#include "fmtconvert.h"
+
+#include <stdint.h>
+
+#define MAX_CHANNELS 64
+#define MAX_ELEM_ID 16
+
+#define TNS_MAX_ORDER 20
+#define MAX_LTP_LONG_SFB 40
+
+enum RawDataBlockType {
+    TYPE_SCE,
+    TYPE_CPE,
+    TYPE_CCE,
+    TYPE_LFE,
+    TYPE_DSE,
+    TYPE_PCE,
+    TYPE_FIL,
+    TYPE_END,
+};
+
+enum ExtensionPayloadID {
+    EXT_FILL,
+    EXT_FILL_DATA,
+    EXT_DATA_ELEMENT,
+    EXT_DYNAMIC_RANGE = 0xb,
+    EXT_SBR_DATA      = 0xd,
+    EXT_SBR_DATA_CRC  = 0xe,
+};
+
+enum WindowSequence {
+    ONLY_LONG_SEQUENCE,
+    LONG_START_SEQUENCE,
+    EIGHT_SHORT_SEQUENCE,
+    LONG_STOP_SEQUENCE,
+};
+
+enum BandType {
+    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
+    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
+    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
+    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
+};
+
+#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
+
+enum ChannelPosition {
+    AAC_CHANNEL_OFF   = 0,
+    AAC_CHANNEL_FRONT = 1,
+    AAC_CHANNEL_SIDE  = 2,
+    AAC_CHANNEL_BACK  = 3,
+    AAC_CHANNEL_LFE   = 4,
+    AAC_CHANNEL_CC    = 5,
+};
+
+/**
+ * The point during decoding at which channel coupling is applied.
+ */
+enum CouplingPoint {
+    BEFORE_TNS,
+    BETWEEN_TNS_AND_IMDCT,
+    AFTER_IMDCT = 3,
+};
+
+/**
+ * Output configuration status
+ */
+enum OCStatus {
+    OC_NONE,        ///< Output unconfigured
+    OC_TRIAL_PCE,   ///< Output configuration under trial specified by an inband PCE
+    OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
+    OC_GLOBAL_HDR,  ///< Output configuration set in a global header but not yet locked
+    OC_LOCKED,      ///< Output configuration locked in place
+};
+
+typedef struct OutputConfiguration {
+    MPEG4AudioConfig m4ac;
+    uint8_t layout_map[MAX_ELEM_ID*4][3];
+    int layout_map_tags;
+    int channels;
+    uint64_t channel_layout;
+    enum OCStatus status;
+} OutputConfiguration;
+
+/**
+ * Predictor State
+ */
+typedef struct PredictorState {
+    float cor0;
+    float cor1;
+    float var0;
+    float var1;
+    float r0;
+    float r1;
+} PredictorState;
+
+#define MAX_PREDICTORS 672
+
+#define SCALE_DIV_512    36    ///< scalefactor difference that corresponds to scale difference in 512 times
+#define SCALE_ONE_POS   140    ///< scalefactor index that corresponds to scale=1.0
+#define SCALE_MAX_POS   255    ///< scalefactor index maximum value
+#define SCALE_MAX_DIFF   60    ///< maximum scalefactor difference allowed by standard
+#define SCALE_DIFF_ZERO  60    ///< codebook index corresponding to zero scalefactor indices difference
+
+/**
+ * Long Term Prediction
+ */
+typedef struct LongTermPrediction {
+    int8_t present;
+    int16_t lag;
+    float coef;
+    int8_t used[MAX_LTP_LONG_SFB];
+} LongTermPrediction;
+
+/**
+ * Individual Channel Stream
+ */
+typedef struct IndividualChannelStream {
+    uint8_t max_sfb;            ///< number of scalefactor bands per group
+    enum WindowSequence window_sequence[2];
+    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
+    int num_window_groups;
+    uint8_t group_len[8];
+    LongTermPrediction ltp;
+    const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
+    const uint8_t *swb_sizes;   ///< table of scalefactor band sizes for a particular window
+    int num_swb;                ///< number of scalefactor window bands
+    int num_windows;
+    int tns_max_bands;
+    int predictor_present;
+    int predictor_initialized;
+    int predictor_reset_group;
+    uint8_t prediction_used[41];
+} IndividualChannelStream;
+
+/**
+ * Temporal Noise Shaping
+ */
+typedef struct TemporalNoiseShaping {
+    int present;
+    int n_filt[8];
+    int length[8][4];
+    int direction[8][4];
+    int order[8][4];
+    float coef[8][4][TNS_MAX_ORDER];
+} TemporalNoiseShaping;
+
+/**
+ * Dynamic Range Control - decoded from the bitstream but not processed further.
+ */
+typedef struct DynamicRangeControl {
+    int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
+    int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
+    int dyn_rng_ctl[17];                            ///< DRC magnitude information
+    int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
+    int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
+    int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
+    int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
+    int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
+                                                     *   channels combined.
+                                                     */
+} DynamicRangeControl;
+
+typedef struct Pulse {
+    int num_pulse;
+    int start;
+    int pos[4];
+    int amp[4];
+} Pulse;
+
+/**
+ * coupling parameters
+ */
+typedef struct ChannelCoupling {
+    enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
+    int num_coupled;       ///< number of target elements
+    enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
+    int id_select[8];      ///< element id
+    int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
+                            *   [2] list of gains for left channel; [3] lists of gains for both channels
+                            */
+    float gain[16][120];
+} ChannelCoupling;
+
+/**
+ * Single Channel Element - used for both SCE and LFE elements.
+ */
+typedef struct SingleChannelElement {
+    IndividualChannelStream ics;
+    TemporalNoiseShaping tns;
+    Pulse pulse;
+    enum BandType band_type[128];                   ///< band types
+    int band_type_run_end[120];                     ///< band type run end points
+    float sf[120];                                  ///< scalefactors
+    int sf_idx[128];                                ///< scalefactor indices (used by encoder)
+    uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
+    DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT
+    DECLARE_ALIGNED(32, float,   saved)[1024];      ///< overlap
+    DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
+    DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
+    PredictorState predictor_state[MAX_PREDICTORS];
+    float *ret;                                     ///< PCM output
+} SingleChannelElement;
+
+/**
+ * channel element - generic struct for SCE/CPE/CCE/LFE
+ */
+typedef struct ChannelElement {
+    // CPE specific
+    int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
+    int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
+    uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
+    // shared
+    SingleChannelElement ch[2];
+    // CCE specific
+    ChannelCoupling coup;
+    SpectralBandReplication sbr;
+} ChannelElement;
+
+/**
+ * main AAC context
+ */
+struct AACContext {
+    AVClass        *class;
+    AVCodecContext *avctx;
+    AVFrame *frame;
+
+    int is_saved;                 ///< Set if elements have stored overlap from previous frame.
+    DynamicRangeControl che_drc;
+
+    /**
+     * @name Channel element related data
+     * @{
+     */
+    ChannelElement          *che[4][MAX_ELEM_ID];
+    ChannelElement  *tag_che_map[4][MAX_ELEM_ID];
+    int tags_mapped;
+    /** @} */
+
+    /**
+     * @name temporary aligned temporary buffers
+     * (We do not want to have these on the stack.)
+     * @{
+     */
+    DECLARE_ALIGNED(32, float, buf_mdct)[1024];
+    /** @} */
+
+    /**
+     * @name Computed / set up during initialization
+     * @{
+     */
+    FFTContext mdct;
+    FFTContext mdct_small;
+    FFTContext mdct_ltp;
+    FmtConvertContext fmt_conv;
+    AVFloatDSPContext fdsp;
+    int random_state;
+    /** @} */
+
+    /**
+     * @name Members used for output
+     * @{
+     */
+    SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
+    /** @} */
+
+
+    /**
+     * @name Japanese DTV specific extension
+     * @{
+     */
+    int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
+    int dmono_mode;      ///< 0->not dmono, 1->use first channel, 2->use second channel
+    /** @} */
+
+    DECLARE_ALIGNED(32, float, temp)[128];
+
+    OutputConfiguration oc[2];
+    int warned_num_aac_frames;
+
+    /* aacdec functions pointers */
+    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
+    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
+                      IndividualChannelStream *ics, int decode);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
+                                   float *in, IndividualChannelStream *ics);
+    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
+
+};
+
+void ff_aacdec_init_mips(AACContext *c);
+
+#endif /* AVCODEC_AAC_H */