diff --git a/doc/general.texi b/doc/general.texi
index a10b7567f9f381ef5e5498c2853887370737cfb5..32c180cf3be0bf7b4814aea37d0d283ef753f17c 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -984,7 +984,7 @@ following image formats are supported:
@item Enhanced AC-3 @tab X @tab X
@item EVRC (Enhanced Variable Rate Codec) @tab @tab X
@item FLAC (Free Lossless Audio Codec) @tab X @tab IX
-@item G.723.1 @tab X @tab X
+@item G.723.1 @tab X @tab X
@item G.729 @tab @tab X
@item GSM @tab E @tab X
@tab encoding supported through external library libgsm
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 4b95e16058888db8b8157d4871c9104f9bb3f8ee..1ad3787eb0399e376f77227f1710ffa2e16c152d 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -276,9 +276,10 @@ OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
OBJS-$(CONFIG_G2M_DECODER) += g2meet.o elsdec.o
-OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o g723_1.o acelp_vectors.o \
- celp_filters.o celp_math.o
-OBJS-$(CONFIG_G723_1_ENCODER) += g723_1dec.o g723_1.o acelp_vectors.o celp_math.o
+OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o g723_1.o \
+ acelp_vectors.o celp_filters.o celp_math.o
+OBJS-$(CONFIG_G723_1_ENCODER) += g723_1enc.o g723_1.o \
+ acelp_vectors.o celp_filters.o
OBJS-$(CONFIG_G729_DECODER) += g729dec.o lsp.o celp_math.o acelp_filters.o acelp_pitch_delay.o acelp_vectors.o g729postfilter.o
OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o
diff --git a/libavcodec/celp_math.h b/libavcodec/celp_math.h
index 18d3ad94d1e7e28a1169a24525faa730a511925c..18888a425d95d8d0dc835b985295f2b73a570711 100644
--- a/libavcodec/celp_math.h
+++ b/libavcodec/celp_math.h
@@ -61,6 +61,16 @@ int ff_exp2(uint16_t power);
*/
int ff_log2_q15(uint32_t value);
+/**
+ * Calculate the dot product of 2 int16_t vectors.
+ * @param a input data array
+ * @param b input data array
+ * @param length number of elements
+ *
+ * @return dot product = sum of elementwise products
+ */
+int64_t ff_dot_product(const int16_t *a, const int16_t *b, int length);
+
/**
* Shift value left or right depending on sign of offset parameter.
* @param value value to shift
@@ -74,16 +84,6 @@ static inline int bidir_sal(int value, int offset)
else return value << offset;
}
-/**
- * returns the dot product of 2 int16_t vectors.
- * @param a input data array
- * @param b input data array
- * @param length number of elements
- *
- * @return dot product = sum of elementwise products
- */
-int64_t ff_dot_product(const int16_t *a, const int16_t *b, int length);
-
/**
* Return the dot product.
* @param a input data array
diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
index 71f3514a0230a636e220cc2e69ec59017c7efa2a..a11fec8a9eeac2092247cf2d2020150c22439a81 100644
--- a/libavcodec/g723_1.c
+++ b/libavcodec/g723_1.c
@@ -53,7 +53,7 @@ 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)
{
- int sum = ff_dot_product(a,b,length);
+ int sum = ff_dot_product(a, b, length);
return av_sat_add32(sum, sum);
}
diff --git a/libavcodec/g723_1.h b/libavcodec/g723_1.h
index d8c904712c57f7dcc26ca7ebcaced7e6020b7d09..cf982a331ca8c12f8df3fc8a202af0be6d62db86 100644
--- a/libavcodec/g723_1.h
+++ b/libavcodec/g723_1.h
@@ -145,10 +145,11 @@ typedef struct g723_1_context {
int postfilter;
int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
+
+ /* encoder */
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
diff --git a/libavcodec/g723_1dec.c b/libavcodec/g723_1dec.c
index 47d22b54bc4263b61cb83e4260a166760e159c9d..3e8c4897d29f913d1df49acb03a6294ca166f320 100644
--- a/libavcodec/g723_1dec.c
+++ b/libavcodec/g723_1dec.c
@@ -186,9 +186,6 @@ static int16_t square_root(unsigned val)
return (ff_sqrt(val << 1) >> 1) & (~1);
}
-#define normalize_bits_int16(num) ff_g723_1_normalize_bits(num, 15)
-#define normalize_bits_int32(num) ff_g723_1_normalize_bits(num, 31)
-
/**
* Generate fixed codebook excitation vector.
*
@@ -1028,1145 +1025,3 @@ AVCodec ff_g723_1_decoder = {
.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];
-
- ff_g723_1_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 = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1;\
- error -= ff_g723_1_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++) {
- ff_g723_1_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++] = ff_g723_1_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;
- ff_g723_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 = ff_g723_1_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 = ff_g723_1_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 = ff_g723_1_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)
- ff_g723_1_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);
-
- ff_g723_1_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);
-
- ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
- ff_g723_1_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);
- ff_g723_1_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 */
- ff_g723_1_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
diff --git a/libavcodec/g723_1enc.c b/libavcodec/g723_1enc.c
new file mode 100644
index 0000000000000000000000000000000000000000..974caf0d9c7b7473b0132a15da2719c00a65f619
--- /dev/null
+++ b/libavcodec/g723_1enc.c
@@ -0,0 +1,1202 @@
+/*
+ * G.723.1 compatible encoder
+ * Copyright (c) Mohamed Naufal <naufal22@gmail.com>
+ *
+ * This file is part of Libav.
+ *
+ * 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.
+ *
+ * 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 Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * G.723.1 compatible encoder
+ */
+
+#include <stdint.h>
+#include <string.h>
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/mem.h"
+#include "libavutil/opt.h"
+
+#include "avcodec.h"
+#include "celp_math.h"
+#include "g723_1.h"
+#include "internal.h"
+
+#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 AVERROR(EINVAL);
+ }
+
+ 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 6300\n");
+ return AVERROR_PATCHWELCOME;
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\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];
+
+ ff_g723_1_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 = ff_g723_1_normalize_bits(temp, 31);
+ 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 = ff_g723_1_normalize_bits(max, 31);
+
+ 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 = ff_g723_1_normalize_bits(sum, 31);
+ 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 = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \
+ error -= ff_g723_1_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 = ff_g723_1_normalize_bits(max, 15);
+ 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);
+}
+
+/**
+ * Perform IIR filtering.
+ *
+ * @param fir_coef FIR coefficients
+ * @param iir_coef IIR coefficients
+ * @param src source vector
+ * @param dest destination vector
+ */
+static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
+ int16_t *src, int16_t *dest)
+{
+ int m, n;
+
+ 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];
+ }
+
+ dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
+ (1 << 15)) >> 16;
+ }
+}
+
+/**
+ * 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);
+ 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 = ff_g723_1_normalize_bits(ccr, 31);
+ ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
+ exp <<= 1;
+ ccr *= ccr;
+ temp = ff_g723_1_normalize_bits(ccr, 31);
+ ccr = ccr << temp >> 16;
+ exp += temp;
+
+ temp = ff_g723_1_normalize_bits(orig_eng, 31);
+ 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 = ff_g723_1_normalize_bits(max, 31);
+ 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++) {
+ ff_g723_1_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++] = ff_g723_1_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 = ff_g723_1_normalize_bits(max, 31);
+
+ 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;
+ ff_g723_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 = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
+
+ scale = ff_g723_1_normalize_bits(temp, 31);
+ impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
+
+ for (i = 1; i < SUBFRAME_LEN; i++) {
+ temp = ff_g723_1_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 = ff_g723_1_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)
+ ff_g723_1_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, AVPacket *avpkt)
+{
+ PutBitContext pb;
+ int info_bits = 0;
+ int i, temp;
+
+ init_put_bits(&pb, avpkt->data, avpkt->size);
+
+ put_bits(&pb, 2, info_bits);
+
+ 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, i, j;
+ int16_t *in, *start;
+ HFParam hf[4];
+
+ /* duplicate input */
+ start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
+ if (!in)
+ return AVERROR(ENOMEM);
+ memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
+
+ 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);
+
+ ff_g723_1_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);
+
+ ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
+ ff_g723_1_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);
+ ff_g723_1_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 */
+ ff_g723_1_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_free(start);
+
+ if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
+ return ret;
+
+ *got_packet_ptr = 1;
+ avpkt->size = pack_bitstream(p, avpkt);
+ 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
+ },
+};