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authorVittorio Giovara <vittorio.giovara@gmail.com>2015-11-23 17:10:53 -0500
committerVittorio Giovara <vittorio.giovara@gmail.com>2015-11-30 10:58:45 -0500
commit165cc6fb9defcd79fd71c08167f3e8df26b058ff (patch)
tree5e2cb0a1893dad8df5c1446f2122002b01eec0c8 /libavcodec/g723_1.h
parentaac996cc01042194bf621d845bbe684549b5882e (diff)
downloadffmpeg-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.h141
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 */