* 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.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/aac.h
+ * @file
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
#include "avcodec.h"
#include "dsputil.h"
+#include "fft.h"
#include "mpeg4audio.h"
+#include "sbr.h"
+#include "fmtconvert.h"
#include <stdint.h>
-#define AAC_INIT_VLC_STATIC(num, size) \
- INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
- ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
- ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
- size);
-
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
+#define MAX_LTP_LONG_SFB 40
enum RawDataBlockType {
TYPE_SCE,
#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,
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 {
+ 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
*/
#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
+#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
+
+/**
+ * Long Term Prediction
+ */
+typedef struct {
+ int8_t present;
+ int16_t lag;
+ float coef;
+ int8_t used[MAX_LTP_LONG_SFB];
+} LongTermPrediction;
/**
* Individual Channel Stream
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
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_16(float, coeffs[1024]); ///< coefficients for IMDCT
- DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap
- DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
+ 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)[2048]; ///< PCM output
+ DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
+ SpectralBandReplication sbr;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
- AVCodecContext * avccontext;
-
- MPEG4AudioConfig m4ac;
+ AVCodecContext *avctx;
+ AVFrame frame;
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
- * @defgroup elements Channel element related data.
+ * @name 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];
+ 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.)
+ * @name temporary aligned temporary buffers
+ * (We do not want to have these on the stack.)
* @{
*/
- DECLARE_ALIGNED_16(float, buf_mdct[1024]);
+ DECLARE_ALIGNED(32, float, buf_mdct)[1024];
/** @} */
/**
- * @defgroup tables Computed / set up during initialization.
+ * @name Computed / set up during initialization
* @{
*/
- MDCTContext mdct;
- MDCTContext mdct_small;
+ FFTContext mdct;
+ FFTContext mdct_small;
+ FFTContext mdct_ltp;
DSPContext dsp;
+ FmtConvertContext fmt_conv;
int random_state;
/** @} */
/**
- * @defgroup output Members used for output interleaving.
+ * @name Members used for output interleaving
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
- float add_bias; ///< offset for dsp.float_to_int16
- float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
- int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
- DECLARE_ALIGNED(16, float, temp[128]);
+ DECLARE_ALIGNED(32, float, temp)[128];
+
+ OutputConfiguration oc[2];
} AACContext;
#endif /* AVCODEC_AAC_H */
projects / ffmpeg / blobdiff