* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/**
- * @file
- * G.723.1 compatible decoder
- */
+#include <stdint.h>
+
+#include "libavutil/common.h"
-#define BITSTREAM_READER_LE
-#include "libavutil/channel_layout.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "avcodec.h"
-#include "get_bits.h"
#include "acelp_vectors.h"
-#include "celp_filters.h"
+#include "avcodec.h"
#include "celp_math.h"
-#include "g723_1_data.h"
-#include "internal.h"
-
-#define CNG_RANDOM_SEED 12345
-
-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; ///< formant postfilter
- ///< gain scaling unit memory
- int postfilter;
-
- int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
- int16_t prev_data[HALF_FRAME_LEN];
- int16_t prev_weight_sig[PITCH_MAX];
-
-
- int16_t hpf_fir_mem; ///< highpass filter fir
- int hpf_iir_mem; ///< and iir memories
- int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
- int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
-
- int16_t harmonic_mem[PITCH_MAX];
-} G723_1_Context;
-
-static av_cold int g723_1_decode_init(AVCodecContext *avctx)
-{
- G723_1_Context *p = avctx->priv_data;
-
- avctx->channel_layout = AV_CH_LAYOUT_MONO;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- avctx->channels = 1;
- p->pf_gain = 1 << 12;
-
- memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
-
- p->cng_random_seed = CNG_RANDOM_SEED;
- p->past_frame_type = SID_FRAME;
+#include "g723_1.h"
- return 0;
-}
-
-/**
- * Unpack the frame into parameters.
- *
- * @param p the context
- * @param buf pointer to the input buffer
- * @param buf_size size of the input buffer
- */
-static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
- int buf_size)
+int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length)
{
- GetBitContext gb;
- int ad_cb_len;
- int temp, info_bits, i;
-
- init_get_bits(&gb, buf, buf_size * 8);
-
- /* Extract frame type and rate info */
- info_bits = get_bits(&gb, 2);
-
- if (info_bits == 3) {
- p->cur_frame_type = UNTRANSMITTED_FRAME;
- return 0;
- }
-
- /* Extract 24 bit lsp indices, 8 bit for each band */
- p->lsp_index[2] = get_bits(&gb, 8);
- p->lsp_index[1] = get_bits(&gb, 8);
- p->lsp_index[0] = get_bits(&gb, 8);
-
- if (info_bits == 2) {
- p->cur_frame_type = SID_FRAME;
- p->subframe[0].amp_index = get_bits(&gb, 6);
- return 0;
- }
-
- /* Extract the info common to both rates */
- p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
- p->cur_frame_type = ACTIVE_FRAME;
-
- p->pitch_lag[0] = get_bits(&gb, 7);
- if (p->pitch_lag[0] > 123) /* test if forbidden code */
- return -1;
- p->pitch_lag[0] += PITCH_MIN;
- p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
-
- p->pitch_lag[1] = get_bits(&gb, 7);
- if (p->pitch_lag[1] > 123)
- return -1;
- p->pitch_lag[1] += PITCH_MIN;
- p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
- p->subframe[0].ad_cb_lag = 1;
- p->subframe[2].ad_cb_lag = 1;
-
- for (i = 0; i < SUBFRAMES; i++) {
- /* Extract combined gain */
- temp = get_bits(&gb, 12);
- ad_cb_len = 170;
- p->subframe[i].dirac_train = 0;
- if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
- p->subframe[i].dirac_train = temp >> 11;
- temp &= 0x7FF;
- ad_cb_len = 85;
- }
- p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
- if (p->subframe[i].ad_cb_gain < ad_cb_len) {
- p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
- GAIN_LEVELS;
- } else {
- return -1;
- }
- }
-
- p->subframe[0].grid_index = get_bits1(&gb);
- p->subframe[1].grid_index = get_bits1(&gb);
- p->subframe[2].grid_index = get_bits1(&gb);
- p->subframe[3].grid_index = get_bits1(&gb);
-
- if (p->cur_rate == RATE_6300) {
- skip_bits1(&gb); /* skip reserved bit */
-
- /* Compute pulse_pos index using the 13-bit combined position index */
- temp = get_bits(&gb, 13);
- p->subframe[0].pulse_pos = temp / 810;
-
- temp -= p->subframe[0].pulse_pos * 810;
- p->subframe[1].pulse_pos = FASTDIV(temp, 90);
-
- temp -= p->subframe[1].pulse_pos * 90;
- p->subframe[2].pulse_pos = FASTDIV(temp, 9);
- p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
+ int bits, max = 0;
+ int i;
- p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
- get_bits(&gb, 16);
- p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
- get_bits(&gb, 14);
- p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
- get_bits(&gb, 16);
- p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
- get_bits(&gb, 14);
+ for (i = 0; i < length; i++)
+ max |= FFABS(vector[i]);
- p->subframe[0].pulse_sign = get_bits(&gb, 6);
- p->subframe[1].pulse_sign = get_bits(&gb, 5);
- p->subframe[2].pulse_sign = get_bits(&gb, 6);
- p->subframe[3].pulse_sign = get_bits(&gb, 5);
- } else { /* 5300 bps */
- p->subframe[0].pulse_pos = get_bits(&gb, 12);
- p->subframe[1].pulse_pos = get_bits(&gb, 12);
- p->subframe[2].pulse_pos = get_bits(&gb, 12);
- p->subframe[3].pulse_pos = get_bits(&gb, 12);
+ bits= 14 - av_log2_16bit(max);
+ bits= FFMAX(bits, 0);
- p->subframe[0].pulse_sign = get_bits(&gb, 4);
- p->subframe[1].pulse_sign = get_bits(&gb, 4);
- p->subframe[2].pulse_sign = get_bits(&gb, 4);
- p->subframe[3].pulse_sign = get_bits(&gb, 4);
- }
+ for (i = 0; i < length; i++)
+ dst[i] = vector[i] << bits >> 3;
- return 0;
+ return bits - 3;
}
-/**
- * Bitexact implementation of sqrt(val/2).
- */
-static int16_t square_root(unsigned val)
+int ff_g723_1_normalize_bits(int num, int width)
{
- av_assert2(!(val & 0x80000000));
-
- return (ff_sqrt(val << 1) >> 1) & (~1);
+ return width - av_log2(num) - 1;
}
-/**
- * Calculate the number of left-shifts required for normalizing the input.
- *
- * @param num input number
- * @param width width of the input, 15 or 31 bits
- */
-static int normalize_bits(int num, int width)
+int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length)
{
- return width - av_log2(num) - 1;
+ int sum = ff_dot_product(a, b, length);
+ return av_sat_add32(sum, sum);
}
-#define normalize_bits_int16(num) normalize_bits(num, 15)
-#define normalize_bits_int32(num) normalize_bits(num, 31)
-
-/**
- * Scale vector contents based on the largest of their absolutes.
- */
-static int scale_vector(int16_t *dst, const int16_t *vector, int length)
+void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
+ int lag)
{
- int bits, max = 0;
+ int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
int i;
- for (i = 0; i < length; i++)
- max |= FFABS(vector[i]);
-
- bits= 14 - av_log2_16bit(max);
- bits= FFMAX(bits, 0);
-
- for (i = 0; i < length; i++)
- dst[i] = vector[i] << bits >> 3;
+ residual[0] = prev_excitation[offset];
+ residual[1] = prev_excitation[offset + 1];
- return bits - 3;
+ offset += 2;
+ for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
+ residual[i] = prev_excitation[offset + (i - 2) % lag];
}
-/**
- * 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
- */
-static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
- uint8_t *lsp_index, int bad_frame)
+void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag)
{
- int min_dist, pred;
- int i, j, temp, stable;
+ int16_t vector[SUBFRAME_LEN];
+ int i, j;
- /* Check for frame erasure */
- if (!bad_frame) {
- min_dist = 0x100;
- pred = 12288;
- } else {
- min_dist = 0x200;
- pred = 23552;
- lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
+ memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
+ for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
+ for (j = 0; j < SUBFRAME_LEN - i; j++)
+ buf[i + j] += vector[j];
}
+}
- /* Get the VQ table entry corresponding to the transmitted index */
- cur_lsp[0] = lsp_band0[lsp_index[0]][0];
- cur_lsp[1] = lsp_band0[lsp_index[0]][1];
- cur_lsp[2] = lsp_band0[lsp_index[0]][2];
- cur_lsp[3] = lsp_band1[lsp_index[1]][0];
- cur_lsp[4] = lsp_band1[lsp_index[1]][1];
- cur_lsp[5] = lsp_band1[lsp_index[1]][2];
- cur_lsp[6] = lsp_band2[lsp_index[2]][0];
- cur_lsp[7] = lsp_band2[lsp_index[2]][1];
- cur_lsp[8] = lsp_band2[lsp_index[2]][2];
- cur_lsp[9] = lsp_band2[lsp_index[2]][3];
+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)
+{
+ int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
+ const int16_t *cb_ptr;
+ int lag = pitch_lag + subfrm->ad_cb_lag - 1;
- /* Add predicted vector & DC component to the previously quantized vector */
- for (i = 0; i < LPC_ORDER; i++) {
- temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
- cur_lsp[i] += dc_lsp[i] + temp;
- }
+ int i;
+ int sum;
- for (i = 0; i < LPC_ORDER; i++) {
- cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
- cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
+ ff_g723_1_get_residual(residual, prev_excitation, lag);
- /* Stability check */
- for (j = 1; j < LPC_ORDER; j++) {
- temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
- if (temp > 0) {
- temp >>= 1;
- cur_lsp[j - 1] -= temp;
- cur_lsp[j] += temp;
- }
- }
- stable = 1;
- for (j = 1; j < LPC_ORDER; j++) {
- temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
- if (temp > 0) {
- stable = 0;
- break;
- }
- }
- if (stable)
- break;
+ /* Select quantization table */
+ if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
+ cb_ptr = adaptive_cb_gain85;
+ } else
+ cb_ptr = adaptive_cb_gain170;
+
+ /* Calculate adaptive vector */
+ cb_ptr += subfrm->ad_cb_gain * 20;
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
+ vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
}
- if (!stable)
- memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
}
-/**
- * Bitexact implementation of 2ab scaled by 1/2^16.
- *
- * @param a 32 bit multiplicand
- * @param b 16 bit multiplier
- */
-#define MULL2(a, b) \
- MULL(a,b,15)
-
/**
* Convert LSP frequencies to LPC coefficients.
*
}
}
-/**
- * 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
- */
-static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
+void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp,
+ int16_t *prev_lsp)
{
int i;
int16_t *lpc_ptr = lpc;
}
}
-/**
- * Generate a train of dirac functions with period as pitch lag.
- */
-static void gen_dirac_train(int16_t *buf, int pitch_lag)
+void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
+ uint8_t *lsp_index, int bad_frame)
{
- int16_t vector[SUBFRAME_LEN];
- int i, j;
+ int min_dist, pred;
+ int i, j, temp, stable;
- memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
- for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
- for (j = 0; j < SUBFRAME_LEN - i; j++)
- buf[i + j] += vector[j];
+ /* Check for frame erasure */
+ if (!bad_frame) {
+ min_dist = 0x100;
+ pred = 12288;
+ } else {
+ min_dist = 0x200;
+ pred = 23552;
+ lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
}
-}
-/**
- * Generate fixed codebook excitation vector.
- *
- * @param vector decoded excitation vector
- * @param subfrm current subframe
- * @param cur_rate current bitrate
- * @param pitch_lag closed loop pitch lag
- * @param index current subframe index
- */
-static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
- enum Rate cur_rate, int pitch_lag, int index)
-{
- int temp, i, j;
+ /* Get the VQ table entry corresponding to the transmitted index */
+ cur_lsp[0] = lsp_band0[lsp_index[0]][0];
+ cur_lsp[1] = lsp_band0[lsp_index[0]][1];
+ cur_lsp[2] = lsp_band0[lsp_index[0]][2];
+ cur_lsp[3] = lsp_band1[lsp_index[1]][0];
+ cur_lsp[4] = lsp_band1[lsp_index[1]][1];
+ cur_lsp[5] = lsp_band1[lsp_index[1]][2];
+ cur_lsp[6] = lsp_band2[lsp_index[2]][0];
+ cur_lsp[7] = lsp_band2[lsp_index[2]][1];
+ cur_lsp[8] = lsp_band2[lsp_index[2]][2];
+ cur_lsp[9] = lsp_band2[lsp_index[2]][3];
- memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
+ /* Add predicted vector & DC component to the previously quantized vector */
+ for (i = 0; i < LPC_ORDER; i++) {
+ temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
+ cur_lsp[i] += dc_lsp[i] + temp;
+ }
- if (cur_rate == RATE_6300) {
- if (subfrm->pulse_pos >= max_pos[index])
- return;
+ for (i = 0; i < LPC_ORDER; i++) {
+ cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
+ cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
- /* Decode amplitudes and positions */
- j = PULSE_MAX - pulses[index];
- temp = subfrm->pulse_pos;
- for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
- temp -= combinatorial_table[j][i];
- if (temp >= 0)
- continue;
- temp += combinatorial_table[j++][i];
- if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
- vector[subfrm->grid_index + GRID_SIZE * i] =
- -fixed_cb_gain[subfrm->amp_index];
- } else {
- vector[subfrm->grid_index + GRID_SIZE * i] =
- fixed_cb_gain[subfrm->amp_index];
+ /* Stability check */
+ for (j = 1; j < LPC_ORDER; j++) {
+ temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
+ if (temp > 0) {
+ temp >>= 1;
+ cur_lsp[j - 1] -= temp;
+ cur_lsp[j] += temp;
}
- if (j == PULSE_MAX)
- break;
- }
- if (subfrm->dirac_train == 1)
- gen_dirac_train(vector, pitch_lag);
- } else { /* 5300 bps */
- int cb_gain = fixed_cb_gain[subfrm->amp_index];
- int cb_shift = subfrm->grid_index;
- int cb_sign = subfrm->pulse_sign;
- int cb_pos = subfrm->pulse_pos;
- int offset, beta, lag;
-
- for (i = 0; i < 8; i += 2) {
- offset = ((cb_pos & 7) << 3) + cb_shift + i;
- vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
- cb_pos >>= 3;
- cb_sign >>= 1;
}
-
- /* Enhance harmonic components */
- lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
- subfrm->ad_cb_lag - 1;
- beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
-
- if (lag < SUBFRAME_LEN - 2) {
- for (i = lag; i < SUBFRAME_LEN; i++)
- vector[i] += beta * vector[i - lag] >> 15;
+ stable = 1;
+ for (j = 1; j < LPC_ORDER; j++) {
+ temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
+ if (temp > 0) {
+ stable = 0;
+ break;
+ }
}
+ if (stable)
+ break;
}
+ if (!stable)
+ memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
}
-
-/**
- * Get delayed contribution from the previous excitation vector.
- */
-static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
-{
- int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
- int i;
-
- residual[0] = prev_excitation[offset];
- residual[1] = prev_excitation[offset + 1];
-
- offset += 2;
- for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
- residual[i] = prev_excitation[offset + (i - 2) % lag];
-}
-
-static int dot_product(const int16_t *a, const int16_t *b, int length)
-{
- int sum = ff_dot_product(a,b,length);
- return av_sat_add32(sum, sum);
-}
-
-/**
- * Generate adaptive codebook excitation.
- */
-static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
- int pitch_lag, G723_1_Subframe *subfrm,
- enum Rate cur_rate)
-{
- int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
- const int16_t *cb_ptr;
- int lag = pitch_lag + subfrm->ad_cb_lag - 1;
-
- int i;
- int sum;
-
- get_residual(residual, prev_excitation, lag);
-
- /* Select quantization table */
- if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
- cb_ptr = adaptive_cb_gain85;
- } else
- cb_ptr = adaptive_cb_gain170;
-
- /* Calculate adaptive vector */
- cb_ptr += subfrm->ad_cb_gain * 20;
- for (i = 0; i < SUBFRAME_LEN; i++) {
- sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
- vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
- }
-}
-
-/**
- * Estimate maximum auto-correlation around pitch lag.
- *
- * @param buf buffer with offset applied
- * @param offset offset of the excitation vector
- * @param ccr_max pointer to the maximum auto-correlation
- * @param pitch_lag decoded pitch lag
- * @param length length of autocorrelation
- * @param dir forward lag(1) / backward lag(-1)
- */
-static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
- int pitch_lag, int length, int dir)
-{
- int limit, ccr, lag = 0;
- int i;
-
- pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
- if (dir > 0)
- limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
- else
- limit = pitch_lag + 3;
-
- for (i = pitch_lag - 3; i <= limit; i++) {
- ccr = dot_product(buf, buf + dir * i, length);
-
- if (ccr > *ccr_max) {
- *ccr_max = ccr;
- lag = i;
- }
- }
- return lag;
-}
-
-/**
- * Calculate pitch postfilter optimal and scaling gains.
- *
- * @param lag pitch postfilter forward/backward lag
- * @param ppf pitch postfilter parameters
- * @param cur_rate current bitrate
- * @param tgt_eng target energy
- * @param ccr cross-correlation
- * @param res_eng residual energy
- */
-static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
- int tgt_eng, int ccr, int res_eng)
-{
- int pf_residual; /* square of postfiltered residual */
- int temp1, temp2;
-
- ppf->index = lag;
-
- temp1 = tgt_eng * res_eng >> 1;
- temp2 = ccr * ccr << 1;
-
- if (temp2 > temp1) {
- if (ccr >= res_eng) {
- ppf->opt_gain = ppf_gain_weight[cur_rate];
- } else {
- ppf->opt_gain = (ccr << 15) / res_eng *
- ppf_gain_weight[cur_rate] >> 15;
- }
- /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
- temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
- temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
- pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
-
- if (tgt_eng >= pf_residual << 1) {
- temp1 = 0x7fff;
- } else {
- temp1 = (tgt_eng << 14) / pf_residual;
- }
-
- /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
- ppf->sc_gain = square_root(temp1 << 16);
- } else {
- ppf->opt_gain = 0;
- ppf->sc_gain = 0x7fff;
- }
-
- ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
-}
-
-/**
- * Calculate pitch postfilter parameters.
- *
- * @param p the context
- * @param offset offset of the excitation vector
- * @param pitch_lag decoded pitch lag
- * @param ppf pitch postfilter parameters
- * @param cur_rate current bitrate
- */
-static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
- PPFParam *ppf, enum Rate cur_rate)
-{
-
- int16_t scale;
- int i;
- int temp1, temp2;
-
- /*
- * 0 - target energy
- * 1 - forward cross-correlation
- * 2 - forward residual energy
- * 3 - backward cross-correlation
- * 4 - backward residual energy
- */
- int energy[5] = {0, 0, 0, 0, 0};
- int16_t *buf = p->audio + LPC_ORDER + offset;
- int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
- SUBFRAME_LEN, 1);
- int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
- SUBFRAME_LEN, -1);
-
- ppf->index = 0;
- ppf->opt_gain = 0;
- ppf->sc_gain = 0x7fff;
-
- /* Case 0, Section 3.6 */
- if (!back_lag && !fwd_lag)
- return;
-
- /* Compute target energy */
- energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
-
- /* Compute forward residual energy */
- if (fwd_lag)
- energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
-
- /* Compute backward residual energy */
- if (back_lag)
- energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
-
- /* Normalize and shorten */
- temp1 = 0;
- for (i = 0; i < 5; i++)
- temp1 = FFMAX(energy[i], temp1);
-
- scale = normalize_bits(temp1, 31);
- for (i = 0; i < 5; i++)
- energy[i] = (energy[i] << scale) >> 16;
-
- if (fwd_lag && !back_lag) { /* Case 1 */
- comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
- energy[2]);
- } else if (!fwd_lag) { /* Case 2 */
- comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
- energy[4]);
- } else { /* Case 3 */
-
- /*
- * Select the largest of energy[1]^2/energy[2]
- * and energy[3]^2/energy[4]
- */
- temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
- temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
- if (temp1 >= temp2) {
- comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
- energy[2]);
- } else {
- comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
- energy[4]);
- }
- }
-}
-
-/**
- * Classify frames as voiced/unvoiced.
- *
- * @param p the context
- * @param pitch_lag decoded pitch_lag
- * @param exc_eng excitation energy estimation
- * @param scale scaling factor of exc_eng
- *
- * @return residual interpolation index if voiced, 0 otherwise
- */
-static int comp_interp_index(G723_1_Context *p, int pitch_lag,
- int *exc_eng, int *scale)
-{
- int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
- int16_t *buf = p->audio + LPC_ORDER;
-
- int index, ccr, tgt_eng, best_eng, temp;
-
- *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
- buf += offset;
-
- /* Compute maximum backward cross-correlation */
- ccr = 0;
- index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
- ccr = av_sat_add32(ccr, 1 << 15) >> 16;
-
- /* Compute target energy */
- tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);
- *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
-
- if (ccr <= 0)
- return 0;
-
- /* Compute best energy */
- best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
- best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
-
- temp = best_eng * *exc_eng >> 3;
-
- if (temp < ccr * ccr) {
- return index;
- } else
- return 0;
-}
-
-/**
- * Peform residual interpolation based on frame classification.
- *
- * @param buf decoded excitation vector
- * @param out output vector
- * @param lag decoded pitch lag
- * @param gain interpolated gain
- * @param rseed seed for random number generator
- */
-static void residual_interp(int16_t *buf, int16_t *out, int lag,
- int gain, int *rseed)
-{
- int i;
- if (lag) { /* Voiced */
- int16_t *vector_ptr = buf + PITCH_MAX;
- /* Attenuate */
- for (i = 0; i < lag; i++)
- out[i] = vector_ptr[i - lag] * 3 >> 2;
- av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
- (FRAME_LEN - lag) * sizeof(*out));
- } else { /* Unvoiced */
- for (i = 0; i < FRAME_LEN; i++) {
- *rseed = *rseed * 521 + 259;
- out[i] = gain * *rseed >> 15;
- }
- memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
- }
-}
-
-/**
- * Perform IIR filtering.
- *
- * @param fir_coef FIR coefficients
- * @param iir_coef IIR coefficients
- * @param src source vector
- * @param dest destination vector
- * @param width width of the output, 16 bits(0) / 32 bits(1)
- */
-#define iir_filter(fir_coef, iir_coef, src, dest, width)\
-{\
- int m, n;\
- int res_shift = 16 & ~-(width);\
- int in_shift = 16 - res_shift;\
-\
- for (m = 0; m < SUBFRAME_LEN; m++) {\
- int64_t filter = 0;\
- for (n = 1; n <= LPC_ORDER; n++) {\
- filter -= (fir_coef)[n - 1] * (src)[m - n] -\
- (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
- }\
-\
- (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
- (1 << 15)) >> res_shift;\
- }\
-}
-
-/**
- * Adjust gain of postfiltered signal.
- *
- * @param p the context
- * @param buf postfiltered output vector
- * @param energy input energy coefficient
- */
-static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
-{
- int num, denom, gain, bits1, bits2;
- int i;
-
- num = energy;
- denom = 0;
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int temp = buf[i] >> 2;
- temp *= temp;
- denom = av_sat_dadd32(denom, temp);
- }
-
- if (num && denom) {
- bits1 = normalize_bits(num, 31);
- bits2 = normalize_bits(denom, 31);
- num = num << bits1 >> 1;
- denom <<= bits2;
-
- bits2 = 5 + bits1 - bits2;
- bits2 = FFMAX(0, bits2);
-
- gain = (num >> 1) / (denom >> 16);
- gain = square_root(gain << 16 >> bits2);
- } else {
- gain = 1 << 12;
- }
-
- for (i = 0; i < SUBFRAME_LEN; i++) {
- p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
- buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
- (1 << 10)) >> 11);
- }
-}
-
-/**
- * Perform formant filtering.
- *
- * @param p the context
- * @param lpc quantized lpc coefficients
- * @param buf input buffer
- * @param dst output buffer
- */
-static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
- int16_t *buf, int16_t *dst)
-{
- int16_t filter_coef[2][LPC_ORDER];
- int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
- int i, j, k;
-
- memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
- memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
-
- for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
- for (k = 0; k < LPC_ORDER; k++) {
- filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
- (1 << 14)) >> 15;
- filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
- (1 << 14)) >> 15;
- }
- iir_filter(filter_coef[0], filter_coef[1], buf + i,
- filter_signal + i, 1);
- lpc += LPC_ORDER;
- }
-
- memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
- memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
-
- buf += LPC_ORDER;
- signal_ptr = filter_signal + LPC_ORDER;
- for (i = 0; i < SUBFRAMES; i++) {
- int temp;
- int auto_corr[2];
- int scale, energy;
-
- /* Normalize */
- scale = scale_vector(dst, buf, SUBFRAME_LEN);
-
- /* Compute auto correlation coefficients */
- auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
- auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN);
-
- /* Compute reflection coefficient */
- temp = auto_corr[1] >> 16;
- if (temp) {
- temp = (auto_corr[0] >> 2) / temp;
- }
- p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
- temp = -p->reflection_coef >> 1 & ~3;
-
- /* Compensation filter */
- for (j = 0; j < SUBFRAME_LEN; j++) {
- dst[j] = av_sat_dadd32(signal_ptr[j],
- (signal_ptr[j - 1] >> 16) * temp) >> 16;
- }
-
- /* Compute normalized signal energy */
- temp = 2 * scale + 4;
- if (temp < 0) {
- energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
- } else
- energy = auto_corr[1] >> temp;
-
- gain_scale(p, dst, energy);
-
- buf += SUBFRAME_LEN;
- signal_ptr += SUBFRAME_LEN;
- dst += SUBFRAME_LEN;
- }
-}
-
-static int sid_gain_to_lsp_index(int gain)
-{
- if (gain < 0x10)
- return gain << 6;
- else if (gain < 0x20)
- return gain - 8 << 7;
- else
- return gain - 20 << 8;
-}
-
-static inline int cng_rand(int *state, int base)
-{
- *state = (*state * 521 + 259) & 0xFFFF;
- return (*state & 0x7FFF) * base >> 15;
-}
-
-static int estimate_sid_gain(G723_1_Context *p)
-{
- int i, shift, seg, seg2, t, val, val_add, x, y;
-
- shift = 16 - p->cur_gain * 2;
- if (shift > 0)
- t = p->sid_gain << shift;
- else
- t = p->sid_gain >> -shift;
- x = t * cng_filt[0] >> 16;
-
- if (x >= cng_bseg[2])
- return 0x3F;
-
- if (x >= cng_bseg[1]) {
- shift = 4;
- seg = 3;
- } else {
- shift = 3;
- seg = (x >= cng_bseg[0]);
- }
- seg2 = FFMIN(seg, 3);
-
- val = 1 << shift;
- val_add = val >> 1;
- for (i = 0; i < shift; i++) {
- t = seg * 32 + (val << seg2);
- t *= t;
- if (x >= t)
- val += val_add;
- else
- val -= val_add;
- val_add >>= 1;
- }
-
- t = seg * 32 + (val << seg2);
- y = t * t - x;
- if (y <= 0) {
- t = seg * 32 + (val + 1 << seg2);
- t = t * t - x;
- val = (seg2 - 1 << 4) + val;
- if (t >= y)
- val++;
- } else {
- t = seg * 32 + (val - 1 << seg2);
- t = t * t - x;
- val = (seg2 - 1 << 4) + val;
- if (t >= y)
- val--;
- }
-
- return val;
-}
-
-static void generate_noise(G723_1_Context *p)
-{
- int i, j, idx, t;
- int off[SUBFRAMES];
- int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
- int tmp[SUBFRAME_LEN * 2];
- int16_t *vector_ptr;
- int64_t sum;
- int b0, c, delta, x, shift;
-
- p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
- p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
-
- for (i = 0; i < SUBFRAMES; i++) {
- p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
- p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];
- }
-
- for (i = 0; i < SUBFRAMES / 2; i++) {
- t = cng_rand(&p->cng_random_seed, 1 << 13);
- off[i * 2] = t & 1;
- off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
- t >>= 2;
- for (j = 0; j < 11; j++) {
- signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
- t >>= 1;
- }
- }
-
- idx = 0;
- for (i = 0; i < SUBFRAMES; i++) {
- for (j = 0; j < SUBFRAME_LEN / 2; j++)
- tmp[j] = j;
- t = SUBFRAME_LEN / 2;
- for (j = 0; j < pulses[i]; j++, idx++) {
- int idx2 = cng_rand(&p->cng_random_seed, t);
-
- pos[idx] = tmp[idx2] * 2 + off[i];
- tmp[idx2] = tmp[--t];
- }
- }
-
- vector_ptr = p->audio + LPC_ORDER;
- memcpy(vector_ptr, p->prev_excitation,
- PITCH_MAX * sizeof(*p->excitation));
- for (i = 0; i < SUBFRAMES; i += 2) {
- gen_acb_excitation(vector_ptr, vector_ptr,
- p->pitch_lag[i >> 1], &p->subframe[i],
- p->cur_rate);
- gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
- vector_ptr + SUBFRAME_LEN,
- p->pitch_lag[i >> 1], &p->subframe[i + 1],
- p->cur_rate);
-
- t = 0;
- for (j = 0; j < SUBFRAME_LEN * 2; j++)
- t |= FFABS(vector_ptr[j]);
- t = FFMIN(t, 0x7FFF);
- if (!t) {
- shift = 0;
- } else {
- shift = -10 + av_log2(t);
- if (shift < -2)
- shift = -2;
- }
- sum = 0;
- if (shift < 0) {
- for (j = 0; j < SUBFRAME_LEN * 2; j++) {
- t = vector_ptr[j] << -shift;
- sum += t * t;
- tmp[j] = t;
- }
- } else {
- for (j = 0; j < SUBFRAME_LEN * 2; j++) {
- t = vector_ptr[j] >> shift;
- sum += t * t;
- tmp[j] = t;
- }
- }
-
- b0 = 0;
- for (j = 0; j < 11; j++)
- b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
- b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
-
- c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
- if (shift * 2 + 3 >= 0)
- c >>= shift * 2 + 3;
- else
- c <<= -(shift * 2 + 3);
- c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
-
- delta = b0 * b0 * 2 - c;
- if (delta <= 0) {
- x = -b0;
- } else {
- delta = square_root(delta);
- x = delta - b0;
- t = delta + b0;
- if (FFABS(t) < FFABS(x))
- x = -t;
- }
- shift++;
- if (shift < 0)
- x >>= -shift;
- else
- x <<= shift;
- x = av_clip(x, -10000, 10000);
-
- for (j = 0; j < 11; j++) {
- idx = (i / 2) * 11 + j;
- vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
- (x * signs[idx] >> 15));
- }
-
- /* copy decoded data to serve as a history for the next decoded subframes */
- memcpy(vector_ptr + PITCH_MAX, vector_ptr,
- sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
- vector_ptr += SUBFRAME_LEN * 2;
- }
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
- PITCH_MAX * sizeof(*p->excitation));
-}
-
-static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
-{
- G723_1_Context *p = avctx->priv_data;
- AVFrame *frame = data;
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
- int dec_mode = buf[0] & 3;
-
- PPFParam ppf[SUBFRAMES];
- int16_t cur_lsp[LPC_ORDER];
- int16_t lpc[SUBFRAMES * LPC_ORDER];
- int16_t acb_vector[SUBFRAME_LEN];
- int16_t *out;
- int bad_frame = 0, i, j, ret;
- int16_t *audio = p->audio;
-
- if (buf_size < frame_size[dec_mode]) {
- if (buf_size)
- av_log(avctx, AV_LOG_WARNING,
- "Expected %d bytes, got %d - skipping packet\n",
- frame_size[dec_mode], buf_size);
- *got_frame_ptr = 0;
- return buf_size;
- }
-
- if (unpack_bitstream(p, buf, buf_size) < 0) {
- bad_frame = 1;
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_frame_type = ACTIVE_FRAME;
- else
- p->cur_frame_type = UNTRANSMITTED_FRAME;
- }
-
- frame->nb_samples = FRAME_LEN;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
- return ret;
-
- out = (int16_t *)frame->data[0];
-
- if (p->cur_frame_type == ACTIVE_FRAME) {
- if (!bad_frame)
- p->erased_frames = 0;
- else if (p->erased_frames != 3)
- p->erased_frames++;
-
- inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
- lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
-
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
-
- /* Generate the excitation for the frame */
- memcpy(p->excitation, p->prev_excitation,
- PITCH_MAX * sizeof(*p->excitation));
- if (!p->erased_frames) {
- int16_t *vector_ptr = p->excitation + PITCH_MAX;
-
- /* Update interpolation gain memory */
- p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
- p->subframe[3].amp_index) >> 1];
- for (i = 0; i < SUBFRAMES; i++) {
- gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
- p->pitch_lag[i >> 1], i);
- gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
- p->pitch_lag[i >> 1], &p->subframe[i],
- p->cur_rate);
- /* Get the total excitation */
- for (j = 0; j < SUBFRAME_LEN; j++) {
- int v = av_clip_int16(vector_ptr[j] << 1);
- vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
- }
- vector_ptr += SUBFRAME_LEN;
- }
-
- vector_ptr = p->excitation + PITCH_MAX;
-
- p->interp_index = comp_interp_index(p, p->pitch_lag[1],
- &p->sid_gain, &p->cur_gain);
-
- /* Peform pitch postfiltering */
- if (p->postfilter) {
- i = PITCH_MAX;
- for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
- ppf + j, p->cur_rate);
-
- for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
- vector_ptr + i,
- vector_ptr + i + ppf[j].index,
- ppf[j].sc_gain,
- ppf[j].opt_gain,
- 1 << 14, 15, SUBFRAME_LEN);
- } else {
- audio = vector_ptr - LPC_ORDER;
- }
-
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
- PITCH_MAX * sizeof(*p->excitation));
- } else {
- p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
- if (p->erased_frames == 3) {
- /* Mute output */
- memset(p->excitation, 0,
- (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
- memset(p->prev_excitation, 0,
- PITCH_MAX * sizeof(*p->excitation));
- memset(frame->data[0], 0,
- (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
- } else {
- int16_t *buf = p->audio + LPC_ORDER;
-
- /* Regenerate frame */
- residual_interp(p->excitation, buf, p->interp_index,
- p->interp_gain, &p->random_seed);
-
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
- PITCH_MAX * sizeof(*p->excitation));
- }
- }
- p->cng_random_seed = CNG_RANDOM_SEED;
- } else {
- if (p->cur_frame_type == SID_FRAME) {
- p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
- inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
- } else if (p->past_frame_type == ACTIVE_FRAME) {
- p->sid_gain = estimate_sid_gain(p);
- }
-
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_gain = p->sid_gain;
- else
- p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
- generate_noise(p);
- lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- }
-
- p->past_frame_type = p->cur_frame_type;
-
- memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
- for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
- audio + i, SUBFRAME_LEN, LPC_ORDER,
- 0, 1, 1 << 12);
- memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
-
- if (p->postfilter) {
- formant_postfilter(p, lpc, p->audio, out);
- } else { // if output is not postfiltered it should be scaled by 2
- for (i = 0; i < FRAME_LEN; i++)
- out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
- }
-
- *got_frame_ptr = 1;
-
- return frame_size[dec_mode];
-}
-
-#define OFFSET(x) offsetof(G723_1_Context, x)
-#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
-
-static const AVOption options[] = {
- { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
- { .i64 = 1 }, 0, 1, AD },
- { NULL }
-};
-
-
-static const AVClass g723_1dec_class = {
- .class_name = "G.723.1 decoder",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-AVCodec ff_g723_1_decoder = {
- .name = "g723_1",
- .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_G723_1,
- .priv_data_size = sizeof(G723_1_Context),
- .init = g723_1_decode_init,
- .decode = g723_1_decode_frame,
- .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
- .priv_class = &g723_1dec_class,
-};
-
-#if CONFIG_G723_1_ENCODER
-#define BITSTREAM_WRITER_LE
-#include "put_bits.h"
-
-static av_cold int g723_1_encode_init(AVCodecContext *avctx)
-{
- G723_1_Context *p = avctx->priv_data;
-
- if (avctx->sample_rate != 8000) {
- av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
- return -1;
- }
-
- if (avctx->channels != 1) {
- av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
- return AVERROR(EINVAL);
- }
-
- if (avctx->bit_rate == 6300) {
- p->cur_rate = RATE_6300;
- } else if (avctx->bit_rate == 5300) {
- av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
- return AVERROR_PATCHWELCOME;
- } else {
- av_log(avctx, AV_LOG_ERROR,
- "Bitrate not supported, use 6.3k\n");
- return AVERROR(EINVAL);
- }
- avctx->frame_size = 240;
- memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
-
- return 0;
-}
-
-/**
- * Remove DC component from the input signal.
- *
- * @param buf input signal
- * @param fir zero memory
- * @param iir pole memory
- */
-static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
-{
- int i;
- for (i = 0; i < FRAME_LEN; i++) {
- *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
- *fir = buf[i];
- buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
- }
-}
-
-/**
- * Estimate autocorrelation of the input vector.
- *
- * @param buf input buffer
- * @param autocorr autocorrelation coefficients vector
- */
-static void comp_autocorr(int16_t *buf, int16_t *autocorr)
-{
- int i, scale, temp;
- int16_t vector[LPC_FRAME];
-
- scale_vector(vector, buf, LPC_FRAME);
-
- /* Apply the Hamming window */
- for (i = 0; i < LPC_FRAME; i++)
- vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
-
- /* Compute the first autocorrelation coefficient */
- temp = ff_dot_product(vector, vector, LPC_FRAME);
-
- /* Apply a white noise correlation factor of (1025/1024) */
- temp += temp >> 10;
-
- /* Normalize */
- scale = normalize_bits_int32(temp);
- autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
- (1 << 15)) >> 16;
-
- /* Compute the remaining coefficients */
- if (!autocorr[0]) {
- memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
- } else {
- for (i = 1; i <= LPC_ORDER; i++) {
- temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
- temp = MULL2((temp << scale), binomial_window[i - 1]);
- autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
- }
- }
-}
-
-/**
- * Use Levinson-Durbin recursion to compute LPC coefficients from
- * autocorrelation values.
- *
- * @param lpc LPC coefficients vector
- * @param autocorr autocorrelation coefficients vector
- * @param error prediction error
- */
-static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
-{
- int16_t vector[LPC_ORDER];
- int16_t partial_corr;
- int i, j, temp;
-
- memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
-
- for (i = 0; i < LPC_ORDER; i++) {
- /* Compute the partial correlation coefficient */
- temp = 0;
- for (j = 0; j < i; j++)
- temp -= lpc[j] * autocorr[i - j - 1];
- temp = ((autocorr[i] << 13) + temp) << 3;
-
- if (FFABS(temp) >= (error << 16))
- break;
-
- partial_corr = temp / (error << 1);
-
- lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
- (1 << 15)) >> 16;
-
- /* Update the prediction error */
- temp = MULL2(temp, partial_corr);
- error = av_clipl_int32((int64_t)(error << 16) - temp +
- (1 << 15)) >> 16;
-
- memcpy(vector, lpc, i * sizeof(int16_t));
- for (j = 0; j < i; j++) {
- temp = partial_corr * vector[i - j - 1] << 1;
- lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
- (1 << 15)) >> 16;
- }
- }
-}
-
-/**
- * Calculate LPC coefficients for the current frame.
- *
- * @param buf current frame
- * @param prev_data 2 trailing subframes of the previous frame
- * @param lpc LPC coefficients vector
- */
-static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
-{
- int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
- int16_t *autocorr_ptr = autocorr;
- int16_t *lpc_ptr = lpc;
- int i, j;
-
- for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
- comp_autocorr(buf + i, autocorr_ptr);
- levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
-
- lpc_ptr += LPC_ORDER;
- autocorr_ptr += LPC_ORDER + 1;
- }
-}
-
-static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
-{
- int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
- ///< polynomials (F1, F2) ordered as
- ///< f1[0], f2[0], ...., f1[5], f2[5]
-
- int max, shift, cur_val, prev_val, count, p;
- int i, j;
- int64_t temp;
-
- /* Initialize f1[0] and f2[0] to 1 in Q25 */
- for (i = 0; i < LPC_ORDER; i++)
- lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
-
- /* Apply bandwidth expansion on the LPC coefficients */
- f[0] = f[1] = 1 << 25;
-
- /* Compute the remaining coefficients */
- for (i = 0; i < LPC_ORDER / 2; i++) {
- /* f1 */
- f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
- /* f2 */
- f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
- }
-
- /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
- f[LPC_ORDER] >>= 1;
- f[LPC_ORDER + 1] >>= 1;
-
- /* Normalize and shorten */
- max = FFABS(f[0]);
- for (i = 1; i < LPC_ORDER + 2; i++)
- max = FFMAX(max, FFABS(f[i]));
-
- shift = normalize_bits_int32(max);
-
- for (i = 0; i < LPC_ORDER + 2; i++)
- f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
-
- /**
- * Evaluate F1 and F2 at uniform intervals of pi/256 along the
- * unit circle and check for zero crossings.
- */
- p = 0;
- temp = 0;
- for (i = 0; i <= LPC_ORDER / 2; i++)
- temp += f[2 * i] * cos_tab[0];
- prev_val = av_clipl_int32(temp << 1);
- count = 0;
- for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
- /* Evaluate */
- temp = 0;
- for (j = 0; j <= LPC_ORDER / 2; j++)
- temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
- cur_val = av_clipl_int32(temp << 1);
-
- /* Check for sign change, indicating a zero crossing */
- if ((cur_val ^ prev_val) < 0) {
- int abs_cur = FFABS(cur_val);
- int abs_prev = FFABS(prev_val);
- int sum = abs_cur + abs_prev;
-
- shift = normalize_bits_int32(sum);
- sum <<= shift;
- abs_prev = abs_prev << shift >> 8;
- lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
-
- if (count == LPC_ORDER)
- break;
-
- /* Switch between sum and difference polynomials */
- p ^= 1;
-
- /* Evaluate */
- temp = 0;
- for (j = 0; j <= LPC_ORDER / 2; j++){
- temp += f[LPC_ORDER - 2 * j + p] *
- cos_tab[i * j % COS_TBL_SIZE];
- }
- cur_val = av_clipl_int32(temp<<1);
- }
- prev_val = cur_val;
- }
-
- if (count != LPC_ORDER)
- memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
-}
-
-/**
- * Quantize the current LSP subvector.
- *
- * @param num band number
- * @param offset offset of the current subvector in an LPC_ORDER vector
- * @param size size of the current subvector
- */
-#define get_index(num, offset, size) \
-{\
- int error, max = -1;\
- int16_t temp[4];\
- int i, j;\
- for (i = 0; i < LSP_CB_SIZE; i++) {\
- for (j = 0; j < size; j++){\
- temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
- (1 << 14)) >> 15;\
- }\
- error = dot_product(lsp + (offset), temp, size) << 1;\
- error -= dot_product(lsp_band##num[i], temp, size);\
- if (error > max) {\
- max = error;\
- lsp_index[num] = i;\
- }\
- }\
-}
-
-/**
- * Vector quantize the LSP frequencies.
- *
- * @param lsp the current lsp vector
- * @param prev_lsp the previous lsp vector
- */
-static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
-{
- int16_t weight[LPC_ORDER];
- int16_t min, max;
- int shift, i;
-
- /* Calculate the VQ weighting vector */
- weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
- weight[LPC_ORDER - 1] = (1 << 20) /
- (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
-
- for (i = 1; i < LPC_ORDER - 1; i++) {
- min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
- if (min > 0x20)
- weight[i] = (1 << 20) / min;
- else
- weight[i] = INT16_MAX;
- }
-
- /* Normalize */
- max = 0;
- for (i = 0; i < LPC_ORDER; i++)
- max = FFMAX(weight[i], max);
-
- shift = normalize_bits_int16(max);
- for (i = 0; i < LPC_ORDER; i++) {
- weight[i] <<= shift;
- }
-
- /* Compute the VQ target vector */
- for (i = 0; i < LPC_ORDER; i++) {
- lsp[i] -= dc_lsp[i] +
- (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
- }
-
- get_index(0, 0, 3);
- get_index(1, 3, 3);
- get_index(2, 6, 4);
-}
-
-/**
- * Apply the formant perceptual weighting filter.
- *
- * @param flt_coef filter coefficients
- * @param unq_lpc unquantized lpc vector
- */
-static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
- int16_t *unq_lpc, int16_t *buf)
-{
- int16_t vector[FRAME_LEN + LPC_ORDER];
- int i, j, k, l = 0;
-
- memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
- memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
- memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
-
- for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
- for (k = 0; k < LPC_ORDER; k++) {
- flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
- (1 << 14)) >> 15;
- flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
- percept_flt_tbl[1][k] +
- (1 << 14)) >> 15;
- }
- iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
- buf + i, 0);
- l += LPC_ORDER;
- }
- memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
- memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
-}
-
-/**
- * Estimate the open loop pitch period.
- *
- * @param buf perceptually weighted speech
- * @param start estimation is carried out from this position
- */
-static int estimate_pitch(int16_t *buf, int start)
-{
- int max_exp = 32;
- int max_ccr = 0x4000;
- int max_eng = 0x7fff;
- int index = PITCH_MIN;
- int offset = start - PITCH_MIN + 1;
-
- int ccr, eng, orig_eng, ccr_eng, exp;
- int diff, temp;
-
- int i;
-
- orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
-
- for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
- offset--;
-
- /* Update energy and compute correlation */
- orig_eng += buf[offset] * buf[offset] -
- buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
- ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
- if (ccr <= 0)
- continue;
-
- /* Split into mantissa and exponent to maintain precision */
- exp = normalize_bits_int32(ccr);
- ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
- exp <<= 1;
- ccr *= ccr;
- temp = normalize_bits_int32(ccr);
- ccr = ccr << temp >> 16;
- exp += temp;
-
- temp = normalize_bits_int32(orig_eng);
- eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
- exp -= temp;
-
- if (ccr >= eng) {
- exp--;
- ccr >>= 1;
- }
- if (exp > max_exp)
- continue;
-
- if (exp + 1 < max_exp)
- goto update;
-
- /* Equalize exponents before comparison */
- if (exp + 1 == max_exp)
- temp = max_ccr >> 1;
- else
- temp = max_ccr;
- ccr_eng = ccr * max_eng;
- diff = ccr_eng - eng * temp;
- if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
-update:
- index = i;
- max_exp = exp;
- max_ccr = ccr;
- max_eng = eng;
- }
- }
- return index;
-}
-
-/**
- * Compute harmonic noise filter parameters.
- *
- * @param buf perceptually weighted speech
- * @param pitch_lag open loop pitch period
- * @param hf harmonic filter parameters
- */
-static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
-{
- int ccr, eng, max_ccr, max_eng;
- int exp, max, diff;
- int energy[15];
- int i, j;
-
- for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
- /* Compute residual energy */
- energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
- /* Compute correlation */
- energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
- }
-
- /* Compute target energy */
- energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
-
- /* Normalize */
- max = 0;
- for (i = 0; i < 15; i++)
- max = FFMAX(max, FFABS(energy[i]));
-
- exp = normalize_bits_int32(max);
- for (i = 0; i < 15; i++) {
- energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
- (1 << 15)) >> 16;
- }
-
- hf->index = -1;
- hf->gain = 0;
- max_ccr = 1;
- max_eng = 0x7fff;
-
- for (i = 0; i <= 6; i++) {
- eng = energy[i << 1];
- ccr = energy[(i << 1) + 1];
-
- if (ccr <= 0)
- continue;
-
- ccr = (ccr * ccr + (1 << 14)) >> 15;
- diff = ccr * max_eng - eng * max_ccr;
- if (diff > 0) {
- max_ccr = ccr;
- max_eng = eng;
- hf->index = i;
- }
- }
-
- if (hf->index == -1) {
- hf->index = pitch_lag;
- return;
- }
-
- eng = energy[14] * max_eng;
- eng = (eng >> 2) + (eng >> 3);
- ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
- if (eng < ccr) {
- eng = energy[(hf->index << 1) + 1];
-
- if (eng >= max_eng)
- hf->gain = 0x2800;
- else
- hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
- }
- hf->index += pitch_lag - 3;
-}
-
-/**
- * Apply the harmonic noise shaping filter.
- *
- * @param hf filter parameters
- */
-static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
-{
- int i;
-
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int64_t temp = hf->gain * src[i - hf->index] << 1;
- dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
- }
-}
-
-static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
-{
- int i;
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int64_t temp = hf->gain * src[i - hf->index] << 1;
- dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
- (1 << 15)) >> 16;
-
- }
-}
-
-/**
- * Combined synthesis and formant perceptual weighting filer.
- *
- * @param qnt_lpc quantized lpc coefficients
- * @param perf_lpc perceptual filter coefficients
- * @param perf_fir perceptual filter fir memory
- * @param perf_iir perceptual filter iir memory
- * @param scale the filter output will be scaled by 2^scale
- */
-static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
- int16_t *perf_fir, int16_t *perf_iir,
- const int16_t *src, int16_t *dest, int scale)
-{
- int i, j;
- int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
- int64_t buf[SUBFRAME_LEN];
-
- int16_t *bptr_16 = buf_16 + LPC_ORDER;
-
- memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
- memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
-
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int64_t temp = 0;
- for (j = 1; j <= LPC_ORDER; j++)
- temp -= qnt_lpc[j - 1] * bptr_16[i - j];
-
- buf[i] = (src[i] << 15) + (temp << 3);
- bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
- }
-
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int64_t fir = 0, iir = 0;
- for (j = 1; j <= LPC_ORDER; j++) {
- fir -= perf_lpc[j - 1] * bptr_16[i - j];
- iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
- }
- dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
- (1 << 15)) >> 16;
- }
- memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
- memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
- sizeof(int16_t) * LPC_ORDER);
-}
-
-/**
- * Compute the adaptive codebook contribution.
- *
- * @param buf input signal
- * @param index the current subframe index
- */
-static void acb_search(G723_1_Context *p, int16_t *residual,
- int16_t *impulse_resp, const int16_t *buf,
- int index)
-{
-
- int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
-
- const int16_t *cb_tbl = adaptive_cb_gain85;
-
- int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
-
- int pitch_lag = p->pitch_lag[index >> 1];
- int acb_lag = 1;
- int acb_gain = 0;
- int odd_frame = index & 1;
- int iter = 3 + odd_frame;
- int count = 0;
- int tbl_size = 85;
-
- int i, j, k, l, max;
- int64_t temp;
-
- if (!odd_frame) {
- if (pitch_lag == PITCH_MIN)
- pitch_lag++;
- else
- pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
- }
-
- for (i = 0; i < iter; i++) {
- get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
-
- for (j = 0; j < SUBFRAME_LEN; j++) {
- temp = 0;
- for (k = 0; k <= j; k++)
- temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
- flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
- (1 << 15)) >> 16;
- }
-
- for (j = PITCH_ORDER - 2; j >= 0; j--) {
- flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
- for (k = 1; k < SUBFRAME_LEN; k++) {
- temp = (flt_buf[j + 1][k - 1] << 15) +
- residual[j] * impulse_resp[k];
- flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
- }
- }
-
- /* Compute crosscorrelation with the signal */
- for (j = 0; j < PITCH_ORDER; j++) {
- temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
- ccr_buf[count++] = av_clipl_int32(temp << 1);
- }
-
- /* Compute energies */
- for (j = 0; j < PITCH_ORDER; j++) {
- ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
- SUBFRAME_LEN);
- }
-
- for (j = 1; j < PITCH_ORDER; j++) {
- for (k = 0; k < j; k++) {
- temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
- ccr_buf[count++] = av_clipl_int32(temp<<2);
- }
- }
- }
-
- /* Normalize and shorten */
- max = 0;
- for (i = 0; i < 20 * iter; i++)
- max = FFMAX(max, FFABS(ccr_buf[i]));
-
- temp = normalize_bits_int32(max);
-
- for (i = 0; i < 20 * iter; i++){
- ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
- (1 << 15)) >> 16;
- }
-
- max = 0;
- for (i = 0; i < iter; i++) {
- /* Select quantization table */
- if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
- odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
- cb_tbl = adaptive_cb_gain170;
- tbl_size = 170;
- }
-
- for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
- temp = 0;
- for (l = 0; l < 20; l++)
- temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
- temp = av_clipl_int32(temp);
-
- if (temp > max) {
- max = temp;
- acb_gain = j;
- acb_lag = i;
- }
- }
- }
-
- if (!odd_frame) {
- pitch_lag += acb_lag - 1;
- acb_lag = 1;
- }
-
- p->pitch_lag[index >> 1] = pitch_lag;
- p->subframe[index].ad_cb_lag = acb_lag;
- p->subframe[index].ad_cb_gain = acb_gain;
-}
-
-/**
- * Subtract the adaptive codebook contribution from the input
- * to obtain the residual.
- *
- * @param buf target vector
- */
-static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
- int16_t *buf)
-{
- int i, j;
- /* Subtract adaptive CB contribution to obtain the residual */
- for (i = 0; i < SUBFRAME_LEN; i++) {
- int64_t temp = buf[i] << 14;
- for (j = 0; j <= i; j++)
- temp -= residual[j] * impulse_resp[i - j];
-
- buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
- }
-}
-
-/**
- * Quantize the residual signal using the fixed codebook (MP-MLQ).
- *
- * @param optim optimized fixed codebook parameters
- * @param buf excitation vector
- */
-static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
- int16_t *buf, int pulse_cnt, int pitch_lag)
-{
- FCBParam param;
- int16_t impulse_r[SUBFRAME_LEN];
- int16_t temp_corr[SUBFRAME_LEN];
- int16_t impulse_corr[SUBFRAME_LEN];
-
- int ccr1[SUBFRAME_LEN];
- int ccr2[SUBFRAME_LEN];
- int amp, err, max, max_amp_index, min, scale, i, j, k, l;
-
- int64_t temp;
-
- /* Update impulse response */
- memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
- param.dirac_train = 0;
- if (pitch_lag < SUBFRAME_LEN - 2) {
- param.dirac_train = 1;
- gen_dirac_train(impulse_r, pitch_lag);
- }
-
- for (i = 0; i < SUBFRAME_LEN; i++)
- temp_corr[i] = impulse_r[i] >> 1;
-
- /* Compute impulse response autocorrelation */
- temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
-
- scale = normalize_bits_int32(temp);
- impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
-
- for (i = 1; i < SUBFRAME_LEN; i++) {
- temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
- impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
- }
-
- /* Compute crosscorrelation of impulse response with residual signal */
- scale -= 4;
- for (i = 0; i < SUBFRAME_LEN; i++){
- temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
- if (scale < 0)
- ccr1[i] = temp >> -scale;
- else
- ccr1[i] = av_clipl_int32(temp << scale);
- }
-
- /* Search loop */
- for (i = 0; i < GRID_SIZE; i++) {
- /* Maximize the crosscorrelation */
- max = 0;
- for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
- temp = FFABS(ccr1[j]);
- if (temp >= max) {
- max = temp;
- param.pulse_pos[0] = j;
- }
- }
-
- /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
- amp = max;
- min = 1 << 30;
- max_amp_index = GAIN_LEVELS - 2;
- for (j = max_amp_index; j >= 2; j--) {
- temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
- impulse_corr[0] << 1);
- temp = FFABS(temp - amp);
- if (temp < min) {
- min = temp;
- max_amp_index = j;
- }
- }
-
- max_amp_index--;
- /* Select additional gain values */
- for (j = 1; j < 5; j++) {
- for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
- temp_corr[k] = 0;
- ccr2[k] = ccr1[k];
- }
- param.amp_index = max_amp_index + j - 2;
- amp = fixed_cb_gain[param.amp_index];
-
- param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
- temp_corr[param.pulse_pos[0]] = 1;
-
- for (k = 1; k < pulse_cnt; k++) {
- max = INT_MIN;
- for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
- if (temp_corr[l])
- continue;
- temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
- temp = av_clipl_int32((int64_t)temp *
- param.pulse_sign[k - 1] << 1);
- ccr2[l] -= temp;
- temp = FFABS(ccr2[l]);
- if (temp > max) {
- max = temp;
- param.pulse_pos[k] = l;
- }
- }
-
- param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
- -amp : amp;
- temp_corr[param.pulse_pos[k]] = 1;
- }
-
- /* Create the error vector */
- memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
-
- for (k = 0; k < pulse_cnt; k++)
- temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
-
- for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
- temp = 0;
- for (l = 0; l <= k; l++) {
- int prod = av_clipl_int32((int64_t)temp_corr[l] *
- impulse_r[k - l] << 1);
- temp = av_clipl_int32(temp + prod);
- }
- temp_corr[k] = temp << 2 >> 16;
- }
-
- /* Compute square of error */
- err = 0;
- for (k = 0; k < SUBFRAME_LEN; k++) {
- int64_t prod;
- prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
- err = av_clipl_int32(err - prod);
- prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
- err = av_clipl_int32(err + prod);
- }
-
- /* Minimize */
- if (err < optim->min_err) {
- optim->min_err = err;
- optim->grid_index = i;
- optim->amp_index = param.amp_index;
- optim->dirac_train = param.dirac_train;
-
- for (k = 0; k < pulse_cnt; k++) {
- optim->pulse_sign[k] = param.pulse_sign[k];
- optim->pulse_pos[k] = param.pulse_pos[k];
- }
- }
- }
- }
-}
-
-/**
- * Encode the pulse position and gain of the current subframe.
- *
- * @param optim optimized fixed CB parameters
- * @param buf excitation vector
- */
-static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
- int16_t *buf, int pulse_cnt)
-{
- int i, j;
-
- j = PULSE_MAX - pulse_cnt;
-
- subfrm->pulse_sign = 0;
- subfrm->pulse_pos = 0;
-
- for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
- int val = buf[optim->grid_index + (i << 1)];
- if (!val) {
- subfrm->pulse_pos += combinatorial_table[j][i];
- } else {
- subfrm->pulse_sign <<= 1;
- if (val < 0) subfrm->pulse_sign++;
- j++;
-
- if (j == PULSE_MAX) break;
- }
- }
- subfrm->amp_index = optim->amp_index;
- subfrm->grid_index = optim->grid_index;
- subfrm->dirac_train = optim->dirac_train;
-}
-
-/**
- * Compute the fixed codebook excitation.
- *
- * @param buf target vector
- * @param impulse_resp impulse response of the combined filter
- */
-static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
- int16_t *buf, int index)
-{
- FCBParam optim;
- int pulse_cnt = pulses[index];
- int i;
-
- optim.min_err = 1 << 30;
- get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
-
- if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
- get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
- p->pitch_lag[index >> 1]);
- }
-
- /* Reconstruct the excitation */
- memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
- for (i = 0; i < pulse_cnt; i++)
- buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
-
- pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
-
- if (optim.dirac_train)
- gen_dirac_train(buf, p->pitch_lag[index >> 1]);
-}
-
-/**
- * Pack the frame parameters into output bitstream.
- *
- * @param frame output buffer
- * @param size size of the buffer
- */
-static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
-{
- PutBitContext pb;
- int info_bits, i, temp;
-
- init_put_bits(&pb, frame, size);
-
- if (p->cur_rate == RATE_6300) {
- info_bits = 0;
- put_bits(&pb, 2, info_bits);
- }else
- av_assert0(0);
-
- put_bits(&pb, 8, p->lsp_index[2]);
- put_bits(&pb, 8, p->lsp_index[1]);
- put_bits(&pb, 8, p->lsp_index[0]);
-
- put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
- put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
- put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
- put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
-
- /* Write 12 bit combined gain */
- for (i = 0; i < SUBFRAMES; i++) {
- temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
- p->subframe[i].amp_index;
- if (p->cur_rate == RATE_6300)
- temp += p->subframe[i].dirac_train << 11;
- put_bits(&pb, 12, temp);
- }
-
- put_bits(&pb, 1, p->subframe[0].grid_index);
- put_bits(&pb, 1, p->subframe[1].grid_index);
- put_bits(&pb, 1, p->subframe[2].grid_index);
- put_bits(&pb, 1, p->subframe[3].grid_index);
-
- if (p->cur_rate == RATE_6300) {
- skip_put_bits(&pb, 1); /* reserved bit */
-
- /* Write 13 bit combined position index */
- temp = (p->subframe[0].pulse_pos >> 16) * 810 +
- (p->subframe[1].pulse_pos >> 14) * 90 +
- (p->subframe[2].pulse_pos >> 16) * 9 +
- (p->subframe[3].pulse_pos >> 14);
- put_bits(&pb, 13, temp);
-
- put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
- put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
- put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
- put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
-
- put_bits(&pb, 6, p->subframe[0].pulse_sign);
- put_bits(&pb, 5, p->subframe[1].pulse_sign);
- put_bits(&pb, 6, p->subframe[2].pulse_sign);
- put_bits(&pb, 5, p->subframe[3].pulse_sign);
- }
-
- flush_put_bits(&pb);
- return frame_size[info_bits];
-}
-
-static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
- const AVFrame *frame, int *got_packet_ptr)
-{
- G723_1_Context *p = avctx->priv_data;
- int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
- int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
- int16_t cur_lsp[LPC_ORDER];
- int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
- int16_t vector[FRAME_LEN + PITCH_MAX];
- int offset, ret;
- int16_t *in_orig = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t));
- int16_t *in = in_orig;
-
- HFParam hf[4];
- int i, j;
-
- if (!in)
- return AVERROR(ENOMEM);
-
- highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
-
- memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
- memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
-
- comp_lpc_coeff(vector, unq_lpc);
- lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
- lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
-
- /* Update memory */
- memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
- sizeof(int16_t) * SUBFRAME_LEN);
- memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
- sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
- memcpy(p->prev_data, in + HALF_FRAME_LEN,
- sizeof(int16_t) * HALF_FRAME_LEN);
- memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
-
- perceptual_filter(p, weighted_lpc, unq_lpc, vector);
-
- memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
- memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
- memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
-
- scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
-
- p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
- p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
-
- for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
-
- memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
- memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
- memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
-
- for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
-
- inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
- lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
-
- memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
-
- offset = 0;
- for (i = 0; i < SUBFRAMES; i++) {
- int16_t impulse_resp[SUBFRAME_LEN];
- int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
- int16_t flt_in[SUBFRAME_LEN];
- int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
-
- /**
- * Compute the combined impulse response of the synthesis filter,
- * formant perceptual weighting filter and harmonic noise shaping filter
- */
- memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
- memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
- memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
-
- flt_in[0] = 1 << 13; /* Unit impulse */
- synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
- zero, zero, flt_in, vector + PITCH_MAX, 1);
- harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
-
- /* Compute the combined zero input response */
- flt_in[0] = 0;
- memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
- memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
-
- synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
- fir, iir, flt_in, vector + PITCH_MAX, 0);
- memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
- harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
-
- acb_search(p, residual, impulse_resp, in, i);
- gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
- &p->subframe[i], p->cur_rate);
- sub_acb_contrib(residual, impulse_resp, in);
-
- fcb_search(p, impulse_resp, in, i);
-
- /* Reconstruct the excitation */
- gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
- &p->subframe[i], RATE_6300);
-
- memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
- sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
- for (j = 0; j < SUBFRAME_LEN; j++)
- in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
- memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
- sizeof(int16_t) * SUBFRAME_LEN);
-
- /* Update filter memories */
- synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
- p->perf_fir_mem, p->perf_iir_mem,
- in, vector + PITCH_MAX, 0);
- memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
- sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
- memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
- sizeof(int16_t) * SUBFRAME_LEN);
-
- in += SUBFRAME_LEN;
- offset += LPC_ORDER;
- }
-
- av_freep(&in_orig); in = NULL;
-
- if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
- return ret;
-
- *got_packet_ptr = 1;
- avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
- return 0;
-}
-
-AVCodec ff_g723_1_encoder = {
- .name = "g723_1",
- .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_G723_1,
- .priv_data_size = sizeof(G723_1_Context),
- .init = g723_1_encode_init,
- .encode2 = g723_1_encode_frame,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
- AV_SAMPLE_FMT_NONE},
-};
-#endif
projects / ffmpeg / blobdiff