X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fg723_1.c;h=3d45f9d1cfdc48514050967ba2f78f55a3e5dbf6;hb=a2ca8ed903b435446031a8a0792ca535e6ee2913;hp=f91f629311cce2d0216837d38228fb04de42af2e;hpb=1eb1f6f281eb6036d363e0317c1500be4a2708f2;p=ffmpeg diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c index f91f629311c..3d45f9d1cfd 100644 --- a/libavcodec/g723_1.c +++ b/libavcodec/g723_1.c @@ -20,363 +20,96 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/** - * @file - * G.723.1 compatible decoder - */ - -#define BITSTREAM_READER_LE -#include "libavutil/audioconvert.h" -#include "libavutil/lzo.h" -#include "libavutil/opt.h" -#include "avcodec.h" -#include "get_bits.h" -#include "acelp_vectors.h" -#include "celp_filters.h" -#include "g723_1_data.h" - -/** - * G723.1 frame types - */ -enum FrameType { - ACTIVE_FRAME, ///< Active speech - SID_FRAME, ///< Silence Insertion Descriptor frame - UNTRANSMITTED_FRAME -}; +#include -enum Rate { - RATE_6300, - RATE_5300 -}; +#include "libavutil/common.h" -/** - * G723.1 unpacked data subframe - */ -typedef struct { - int ad_cb_lag; ///< adaptive codebook lag - int ad_cb_gain; - int dirac_train; - int pulse_sign; - int grid_index; - int amp_index; - int pulse_pos; -} G723_1_Subframe; - -/** - * Pitch postfilter parameters - */ -typedef struct { - int index; ///< postfilter backward/forward lag - int16_t opt_gain; ///< optimal gain - int16_t sc_gain; ///< scaling gain -} PPFParam; - -typedef struct g723_1_context { - AVClass *class; - AVFrame frame; - - 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 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 interp_index; - int interp_gain; - int sid_gain; - int cur_gain; - int reflection_coef; - int pf_gain; - int postfilter; - - int16_t audio[FRAME_LEN + LPC_ORDER]; -} 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; - avctx->sample_rate = 8000; - p->pf_gain = 1 << 12; - - avcodec_get_frame_defaults(&p->frame); - avctx->coded_frame = &p->frame; - - memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); - - return 0; -} +#include "acelp_vectors.h" +#include "avcodec.h" +#include "celp_math.h" +#include "g723_1.h" -/** - * 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_bits(&gb, 1); - p->subframe[1].grid_index = get_bits(&gb, 1); - p->subframe[2].grid_index = get_bits(&gb, 1); - p->subframe[3].grid_index = get_bits(&gb, 1); - - if (p->cur_rate == RATE_6300) { - skip_bits(&gb, 1); /* 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); + max = FFMIN(max, 0x7FFF); + bits = ff_g723_1_normalize_bits(max, 15); - 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(int val) +int ff_g723_1_normalize_bits(int num, int width) { - int16_t res = 0; - int16_t exp = 0x4000; - int i; - - for (i = 0; i < 14; i ++) { - int res_exp = res + exp; - if (val >= res_exp * res_exp << 1) - res += exp; - exp >>= 1; - } - return res; + 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, 16 bits(0) / 32 bits(1) - */ -static int normalize_bits(int num, int width) +int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length) { - if (!num) - return 0; - if (num == -1) - return width; - if (num < 0) - num = ~num; - - return width - av_log2(num) - 1; + int sum = ff_dot_product(a, b, length); + return av_sat_add32(sum, sum); } -/** - * Scale vector contents based on the largest of their absolutes. - */ -static int scale_vector(int16_t *vector, int length) +void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation, + int lag) { - int bits, max = 0; - int64_t scale; + int offset = PITCH_MAX - PITCH_ORDER / 2 - lag; int i; + residual[0] = prev_excitation[offset]; + residual[1] = prev_excitation[offset + 1]; - for (i = 0; i < length; i++) - max = FFMAX(max, FFABS(vector[i])); - - max = FFMIN(max, 0x7FFF); - bits = normalize_bits(max, 15); - scale = (bits == 15) ? 0x7FFF : (1 << bits); - - for (i = 0; i < length; i++) - vector[i] = av_clipl_int32(vector[i] * scale << 1) >> 4; - - 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_g723_1_dot_product(residual + i, cb_ptr, PITCH_ORDER); + vector[i] = av_sat_dadd32(1 << 15, 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) \ - ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) - /** * Convert LSP frequencies to LPC coefficients. * @@ -390,13 +123,13 @@ static void lsp2lpc(int16_t *lpc) /* Calculate negative cosine */ for (j = 0; j < LPC_ORDER; j++) { - int index = lpc[j] >> 7; - int offset = lpc[j] & 0x7f; - int64_t temp1 = cos_tab[index] << 16; - int temp2 = (cos_tab[index + 1] - cos_tab[index]) * - ((offset << 8) + 0x80) << 1; + int index = (lpc[j] >> 7) & 0x1FF; + int offset = lpc[j] & 0x7f; + int temp1 = cos_tab[index] << 16; + int temp2 = (cos_tab[index + 1] - cos_tab[index]) * + ((offset << 8) + 0x80) << 1; - lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16); + lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); } /* @@ -429,8 +162,8 @@ static void lsp2lpc(int16_t *lpc) f1[0] >>= 1; f2[0] >>= 1; - f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; - f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; + f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; + f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; } /* Convert polynomial coefficients to LPC coefficients */ @@ -438,21 +171,15 @@ static void lsp2lpc(int16_t *lpc) int64_t ff1 = f1[i + 1] + f1[i]; int64_t ff2 = f2[i + 1] - f2[i]; - lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16; + lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + + (1 << 15)) >> 16; lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + (1 << 15)) >> 16; } } -/** - * 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; @@ -472,709 +199,64 @@ static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp) } } -/** - * Generate a train of dirac functions with period as pitch lag. - */ -static void gen_dirac_train(int16_t *buf, int pitch_lag) -{ - int16_t vector[SUBFRAME_LEN]; - int i, j; - - 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]; - } -} - -/** - * 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; - - memset(vector, 0, SUBFRAME_LEN * sizeof(*vector)); - - if (cur_rate == RATE_6300) { - if (subfrm.pulse_pos >= max_pos[index]) - return; - - /* 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]; - } - 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; - } - } -} - -/** - * 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 i, sum = 0; - - for (i = 0; i < length; i++) { - int64_t prod = av_clipl_int32((int64_t)(a[i] * b[i]) << 1); - sum = av_clipl_int32(sum + prod); - } - return 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; - int64_t 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 = dot_product(residual + i, cb_ptr, PITCH_ORDER); - vector[i] = av_clipl_int32((sum << 1) + (1 << 15)) >> 16; - } -} - -/** - * Estimate maximum auto-correlation around pitch lag. - * - * @param p the context - * @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(G723_1_Context *p, int offset, int *ccr_max, - int pitch_lag, int length, int dir) -{ - int limit, ccr, lag = 0; - int16_t *buf = p->excitation + offset; - 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 */ - int64_t 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_clipl_int32(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; - int64_t 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->excitation + offset; - int fwd_lag = autocorr_max(p, offset, &energy[1], pitch_lag, - SUBFRAME_LEN, 1); - int back_lag = autocorr_max(p, 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->excitation + offset; - - int index, ccr, tgt_eng, best_eng, temp; - - *scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX); - - /* Compute maximum backward cross-correlation */ - ccr = 0; - index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1); - ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16; - - /* Compute target energy */ - tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2); - *exc_eng = av_clipl_int32((int64_t)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_clipl_int32((int64_t)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++) - vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2; - av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(*vector_ptr), - FRAME_LEN * sizeof(*vector_ptr)); - memcpy(out, vector_ptr, FRAME_LEN * sizeof(*vector_ptr)); - } 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 - */ -static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef, - int16_t *src, int *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] >> 16); - } - - dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15)); - } -} - -/** - * 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) +void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, + uint8_t *lsp_index, int bad_frame) { - 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_clipl_int32((int64_t)denom + (temp << 1)); - } - - 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); + int min_dist, pred; + int i, j, temp, stable; - gain = (num >> 1) / (denom >> 16); - gain = square_root(gain << 16 >> bits2); + /* Check for frame erasure */ + if (!bad_frame) { + min_dist = 0x100; + pred = 12288; } 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 output buffer - */ -static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf) -{ - int16_t filter_coef[2][LPC_ORDER], *buf_ptr; - 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); - lpc += LPC_ORDER; - } - - memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem)); - memcpy(p->iir_mem, filter_signal + FRAME_LEN, - LPC_ORDER * sizeof(*p->iir_mem)); - - buf_ptr = buf + LPC_ORDER; - signal_ptr = filter_signal + LPC_ORDER; - for (i = 0; i < SUBFRAMES; i++) { - int16_t temp_vector[SUBFRAME_LEN]; - int temp; - int auto_corr[2]; - int scale, energy; - - /* Normalize */ - memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(*temp_vector)); - scale = scale_vector(temp_vector, SUBFRAME_LEN); - - /* Compute auto correlation coefficients */ - auto_corr[0] = dot_product(temp_vector, temp_vector + 1, - SUBFRAME_LEN - 1); - auto_corr[1] = dot_product(temp_vector, temp_vector, 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++) { - buf_ptr[j] = av_clipl_int32((int64_t)signal_ptr[j] + - ((signal_ptr[j - 1] >> 16) * - temp << 1)) >> 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, buf_ptr, energy); - - buf_ptr += SUBFRAME_LEN; - signal_ptr += SUBFRAME_LEN; - } -} - -static int g723_1_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) -{ - G723_1_Context *p = avctx->priv_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 *vector_ptr; - int16_t *out; - int bad_frame = 0, i, j, ret; - - 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; + min_dist = 0x200; + pred = 23552; + lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; } - 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; - } + /* 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]; - p->frame.nb_samples = FRAME_LEN; - if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); - return ret; + /* 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; } - out = (int16_t *)p->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)); - vector_ptr = p->excitation + PITCH_MAX; - if (!p->erased_frames) { - /* 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++) { - vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1); - vector_ptr[j] = av_clip_int16(vector_ptr[j] + - acb_vector[j]); - } - vector_ptr += SUBFRAME_LEN; - } - - vector_ptr = p->excitation + PITCH_MAX; - - /* Save the excitation */ - memcpy(p->audio + LPC_ORDER, vector_ptr, FRAME_LEN * sizeof(*p->audio)); - - p->interp_index = comp_interp_index(p, p->pitch_lag[1], - &p->sid_gain, &p->cur_gain); + 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); - 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); + /* 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; } - - /* Restore the original excitation */ - memcpy(p->excitation, p->prev_excitation, - PITCH_MAX * sizeof(*p->excitation)); - memcpy(vector_ptr, p->audio + LPC_ORDER, FRAME_LEN * sizeof(*vector_ptr)); - - /* Peform pitch postfiltering */ - if (p->postfilter) - 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 { - 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->frame.data[0], 0, - (FRAME_LEN + LPC_ORDER) * sizeof(int16_t)); - } else { - /* Regenerate frame */ - residual_interp(p->excitation, p->audio + LPC_ORDER, p->interp_index, - p->interp_gain, &p->random_seed); + } + 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; } } - /* Save the excitation for the next frame */ - memcpy(p->prev_excitation, p->excitation + FRAME_LEN, - PITCH_MAX * sizeof(*p->excitation)); - } else { - memset(out, 0, FRAME_LEN * 2); - av_log(avctx, AV_LOG_WARNING, - "G.723.1: Comfort noise generation not supported yet\n"); - - *got_frame_ptr = 1; - *(AVFrame *)data = p->frame; - return frame_size[dec_mode]; - } - - 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], - p->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); - memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2); - } 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); + if (stable) + break; } - - *got_frame_ptr = 1; - *(AVFrame *)data = p->frame; - - return frame_size[dec_mode]; + if (!stable) + memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); } - -#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, - { 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", - .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, - .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), - .capabilities = CODEC_CAP_SUBFRAMES, - .priv_class = &g723_1dec_class, -};