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author | Vittorio Giovara <vittorio.giovara@gmail.com> | 2015-11-23 17:10:53 -0500 |
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committer | Vittorio Giovara <vittorio.giovara@gmail.com> | 2015-11-30 10:58:45 -0500 |
commit | 165cc6fb9defcd79fd71c08167f3e8df26b058ff (patch) | |
tree | 5e2cb0a1893dad8df5c1446f2122002b01eec0c8 /libavcodec/g723_1.h | |
parent | aac996cc01042194bf621d845bbe684549b5882e (diff) | |
download | ffmpeg-165cc6fb9defcd79fd71c08167f3e8df26b058ff.tar.gz |
g723_1: Move sharable functions to a separate file
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Diffstat (limited to 'libavcodec/g723_1.h')
-rw-r--r-- | libavcodec/g723_1.h | 141 |
1 files changed, 139 insertions, 2 deletions
diff --git a/libavcodec/g723_1.h b/libavcodec/g723_1.h index 71e2df4ad3..391ca464a9 100644 --- a/libavcodec/g723_1.h +++ b/libavcodec/g723_1.h @@ -1,5 +1,5 @@ /* - * G.723.1 compatible decoder data tables. + * G.723.1 common header and data tables * Copyright (c) 2006 Benjamin Larsson * Copyright (c) 2010 Mohamed Naufal Basheer * @@ -22,7 +22,7 @@ /** * @file - * G.723.1 compatible decoder data tables + * G.723.1 types, functions and data tables */ #ifndef AVCODEC_G723_1_H @@ -44,6 +44,143 @@ #define GAIN_LEVELS 24 #define COS_TBL_SIZE 512 +/** + * Bitexact implementation of 2ab scaled by 1/2^16. + * + * @param a 32 bit multiplicand + * @param b 16 bit multiplier + */ +#define MULL2(a, b) \ + ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) + +/** + * G723.1 frame types + */ +enum FrameType { + ACTIVE_FRAME, ///< Active speech + SID_FRAME, ///< Silence Insertion Descriptor frame + UNTRANSMITTED_FRAME +}; + +/** + * G723.1 rate values + */ +enum Rate { + RATE_6300, + RATE_5300 +}; + +/** + * G723.1 unpacked data subframe + */ +typedef struct G723_1_Subframe { + int ad_cb_lag; ///< adaptive codebook lag + int ad_cb_gain; + int dirac_train; + int pulse_sign; + int grid_index; + int amp_index; + int pulse_pos; +} G723_1_Subframe; + +/** + * Pitch postfilter parameters + */ +typedef struct PPFParam { + int index; ///< postfilter backward/forward lag + int16_t opt_gain; ///< optimal gain + int16_t sc_gain; ///< scaling gain +} PPFParam; + +typedef struct g723_1_context { + AVClass *class; + + G723_1_Subframe subframe[4]; + enum FrameType cur_frame_type; + enum FrameType past_frame_type; + enum Rate cur_rate; + uint8_t lsp_index[LSP_BANDS]; + int pitch_lag[2]; + int erased_frames; + + int16_t prev_lsp[LPC_ORDER]; + int16_t sid_lsp[LPC_ORDER]; + int16_t prev_excitation[PITCH_MAX]; + int16_t excitation[PITCH_MAX + FRAME_LEN + 4]; + int16_t synth_mem[LPC_ORDER]; + int16_t fir_mem[LPC_ORDER]; + int iir_mem[LPC_ORDER]; + + int random_seed; + int cng_random_seed; + int interp_index; + int interp_gain; + int sid_gain; + int cur_gain; + int reflection_coef; + int pf_gain; + int postfilter; + + int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4]; +} G723_1_Context; + + +/** + * Scale vector contents based on the largest of their absolutes. + */ +int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length); + +/** + * Calculate the number of left-shifts required for normalizing the input. + * + * @param num input number + * @param width width of the input, 16 bits(0) / 32 bits(1) + */ +int ff_g723_1_normalize_bits(int num, int width); + +int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length); + +/** + * Get delayed contribution from the previous excitation vector. + */ +void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation, + int lag); + +/** + * Generate a train of dirac functions with period as pitch lag. + */ +void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag); + + +/** + * Generate adaptive codebook excitation. + */ +void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, + int pitch_lag, G723_1_Subframe *subfrm, + enum Rate cur_rate); +/** + * Quantize LSP frequencies by interpolation and convert them to + * the corresponding LPC coefficients. + * + * @param lpc buffer for LPC coefficients + * @param cur_lsp the current LSP vector + * @param prev_lsp the previous LSP vector + */ +void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, + int16_t *prev_lsp); + +/** + * Perform inverse quantization of LSP frequencies. + * + * @param cur_lsp the current LSP vector + * @param prev_lsp the previous LSP vector + * @param lsp_index VQ indices + * @param bad_frame bad frame flag + */ +void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, + uint8_t *lsp_index, int bad_frame); + + static const uint8_t frame_size[4] = { 24, 20, 4, 1 }; /* Postfilter gain weighting factors scaled by 2^15 */ |