*/
/**
- * @file aac.h
+ * @file libavcodec/aac.h
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
-#ifndef FFMPEG_AAC_H
-#define FFMPEG_AAC_H
+#ifndef AVCODEC_AAC_H
+#define AVCODEC_AAC_H
+#include "libavutil/internal.h"
#include "avcodec.h"
#include "dsputil.h"
#include "mpeg4audio.h"
size);
#define MAX_CHANNELS 64
+#define MAX_ELEM_ID 16
-#define IVQUANT_SIZE 1024
-
-enum AudioObjectType {
- AOT_NULL,
- // Support? Name
- AOT_AAC_MAIN, ///< Y Main
- AOT_AAC_LC, ///< Y Low Complexity
- AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
- AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
- AOT_SBR, ///< N (in progress) Spectral Band Replication
- AOT_AAC_SCALABLE, ///< N Scalable
- AOT_TWINVQ, ///< N Twin Vector Quantizer
- AOT_CELP, ///< N Code Excited Linear Prediction
- AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
- AOT_TTSI = 12, ///< N Text-To-Speech Interface
- AOT_MAINSYNTH, ///< N Main Synthesis
- AOT_WAVESYNTH, ///< N Wavetable Synthesis
- AOT_MIDI, ///< N General MIDI
- AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
- AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
- AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
- AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
- AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
- AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
- AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
- AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
- AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
- AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
- AOT_ER_PARAM, ///< N Error Resilient Parametric
- AOT_SSC, ///< N SinuSoidal Coding
+#define TNS_MAX_ORDER 20
+
+enum RawDataBlockType {
+ TYPE_SCE,
+ TYPE_CPE,
+ TYPE_CCE,
+ TYPE_LFE,
+ TYPE_DSE,
+ TYPE_PCE,
+ TYPE_FIL,
+ TYPE_END,
};
enum ExtensionPayloadID {
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,
+};
+
+/**
+ * Predictor State
+ */
+typedef struct {
+ float cor0;
+ float cor1;
+ float var0;
+ float var1;
+ float r0;
+ float r1;
+} PredictorState;
+
+#define MAX_PREDICTORS 672
+
+/**
+ * Individual Channel Stream
+ */
+typedef struct {
+ 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];
+ const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, 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 {
+ 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 {
+ 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 {
int num_pulse;
- int start;
- int offset[4];
+ int pos[4];
int amp[4];
} Pulse;
* coupling parameters
*/
typedef struct {
+ 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 {
+ IndividualChannelStream ics;
+ TemporalNoiseShaping tns;
+ enum BandType band_type[120]; ///< band types
+ int band_type_run_end[120]; ///< band type run end points
+ float sf[120]; ///< scalefactors
+ DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
+ DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
+ DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
+ PredictorState predictor_state[MAX_PREDICTORS];
+} SingleChannelElement;
+
+/**
+ * channel element - generic struct for SCE/CPE/CCE/LFE
+ */
+typedef struct {
+ // CPE specific
+ uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
+ // shared
+ SingleChannelElement ch[2];
+ // CCE specific
+ ChannelCoupling coup;
+} ChannelElement;
/**
* main AAC context
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
+ /**
+ * @defgroup elements Channel element related data.
+ * @{
+ */
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
* first index as the first 4 raw data block types
*/
+ ChannelElement * che[4][MAX_ELEM_ID];
+ ChannelElement * tag_che_map[4][MAX_ELEM_ID];
+ int tags_mapped;
+ /** @} */
+
+ /**
+ * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
+ * @{
+ */
+ DECLARE_ALIGNED_16(float, buf_mdct[1024]);
+ /** @} */
/**
* @defgroup tables Computed / set up during initialization.
MDCTContext mdct;
MDCTContext mdct_small;
DSPContext dsp;
+ int random_state;
/** @} */
/**
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
+ DECLARE_ALIGNED(16, float, temp[128]);
} AACContext;
-#endif /* FFMPEG_AAC_H */
+#endif /* AVCODEC_AAC_H */
projects / frescor / ffmpeg.git / blobdiff