diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 2aa0ffc3eb..27345d9af6 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -159,6 +159,8 @@ OBJS-$(CONFIG_FLIC_DECODER) += flicvideo.o OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o OBJS-$(CONFIG_FRWU_DECODER) += frwu.o +OBJS-$(CONFIG_G723_1_DECODER) += g723_1.o acelp_vectors.o \ + celp_filters.o celp_math.o OBJS-$(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/allcodecs.c b/libavcodec/allcodecs.c index 72a32c444c..03cc80769e 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -258,6 +258,7 @@ void avcodec_register_all(void) REGISTER_DECODER (DSICINAUDIO, dsicinaudio); REGISTER_ENCDEC (EAC3, eac3); REGISTER_ENCDEC (FLAC, flac); + REGISTER_DECODER (G723_1, g723_1); REGISTER_DECODER (G729, g729); REGISTER_DECODER (GSM, gsm); REGISTER_DECODER (GSM_MS, gsm_ms); diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index e04513a29a..cb82ec7648 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -345,6 +345,7 @@ enum CodecID { CODEC_ID_QDMC, CODEC_ID_CELT, CODEC_ID_G729 = 0x15800, + CODEC_ID_G723_1= 0x15801, /* subtitle codecs */ CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs. diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c new file mode 100644 index 0000000000..da4f0d2a04 --- /dev/null +++ b/libavcodec/g723_1.c @@ -0,0 +1,1081 @@ +/* + * G.723.1 compatible decoder + * Copyright (c) 2006 Benjamin Larsson + * Copyright (c) 2010 Mohamed Naufal Basheer + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * G.723.1 compatible decoder + */ + +#include "avcodec.h" +#define ALT_BITSTREAM_READER_LE +#include "get_bits.h" +#include "acelp_vectors.h" +#include "celp_filters.h" +#include "celp_math.h" +#include "lsp.h" +#include "libavutil/lzo.h" +#include "g723_1_data.h" + +typedef struct g723_1_context { + G723_1_Subframe subframe[4]; + FrameType cur_frame_type; + FrameType past_frame_type; + 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]; + 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; ///< formant postfilter + ///< gain scaling unit memory +} G723_1_Context; + +static av_cold int g723_1_decode_init(AVCodecContext *avctx) +{ + G723_1_Context *p = avctx->priv_data; + + avctx->sample_fmt = SAMPLE_FMT_S16; + p->pf_gain = 1 << 12; + memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t)); + + return 0; +} + +/** + * Unpack the frame into parameters. + * + * @param p the context + * @param buf pointer to the input buffer + * @param buf_size size of the input buffer + */ +static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf, + int buf_size) +{ + 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 = UntransmittedFrame; + 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 = SIDFrame; + p->subframe[0].amp_index = get_bits(&gb, 6); + return 0; + } + + /* Extract the info common to both rates */ + p->cur_rate = info_bits ? Rate5k3 : Rate6k3; + p->cur_frame_type = ActiveFrame; + + 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 == Rate6k3 && 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 == Rate6k3) { + 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; + + 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); + + 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 { /* Rate5k3 */ + 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); + + 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); + } + + return 0; +} + +/** + * Bitexact implementation of sqrt(val/2). + */ +static int16_t square_root(int val) +{ + 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; +} + +/** + * 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 i = 0; + int bits = (width) ? 31 : 15; + int limit = 1 << (bits - 1); + + if (num) { + if (num == -1) + return bits; + if (num < 0) + num = ~num; + for (i = 0; num < limit; i++) + num <<= 1; + } + return i; +} + +/** + * Scale vector contents based on the largest of their absolutes. + */ +static int scale_vector(int16_t *vector, int length) +{ + int bits, scale, max = 0; + int i; + + const int16_t shift_table[16] = { + 0x0001, 0x0002, 0x0004, 0x0008, 0x0010, 0x0020, 0x0040, 0x0080, + 0x0100, 0x0200, 0x0400, 0x0800, 0x1000, 0x2000, 0x4000, 0x7fff + }; + + for (i = 0; i < length; i++) + max = FFMAX(max, FFABS(vector[i])); + + bits = normalize_bits(max, 0); + scale = shift_table[bits]; + + for (i = 0; i < length; i++) + vector[i] = (int16_t)(av_clipl_int32(vector[i] * scale << 1) >> 4); + + return bits - 3; +} + +/** + * 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) +{ + int min_dist, pred; + int i, j, temp, stable; + + /* 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; + } + + /* 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]; + + /* 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; + } + + 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); + + /* 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; + } + if (!stable) + memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(int16_t)); +} + +/** + * 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. + * + * @param lpc buffer for LPC coefficients + */ +static void lsp2lpc(int16_t *lpc) +{ + int f1[LPC_ORDER / 2 + 1]; + int f2[LPC_ORDER / 2 + 1]; + int i, j; + + /* 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; + + lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16); + } + + /* + * Compute sum and difference polynomial coefficients + * (bitexact alternative to lsp2poly() in lsp.c) + */ + /* Initialize with values in Q28 */ + f1[0] = 1 << 28; + f1[1] = (lpc[0] << 14) + (lpc[2] << 14); + f1[2] = lpc[0] * lpc[2] + (2 << 28); + + f2[0] = 1 << 28; + f2[1] = (lpc[1] << 14) + (lpc[3] << 14); + f2[2] = lpc[1] * lpc[3] + (2 << 28); + + /* + * Calculate and scale the coefficients by 1/2 in + * each iteration for a final scaling factor of Q25 + */ + for (i = 2; i < LPC_ORDER / 2; i++) { + f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); + f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); + + for (j = i; j >= 2; j--) { + f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + + (f1[j] >> 1) + (f1[j - 2] >> 1); + f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + + (f2[j] >> 1) + (f2[j - 2] >> 1); + } + + 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; + } + + /* Convert polynomial coefficients to LPC coefficients */ + for (i = 0; i < LPC_ORDER / 2; i++) { + 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[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) +{ + int i; + int16_t *lpc_ptr = lpc; + + /* cur_lsp * 0.25 + prev_lsp * 0.75 */ + ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, + 4096, 12288, 1 << 13, 14, LPC_ORDER); + ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, + 8192, 8192, 1 << 13, 14, LPC_ORDER); + ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, + 12288, 4096, 1 << 13, 14, LPC_ORDER); + memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(int16_t)); + + for (i = 0; i < SUBFRAMES; i++) { + lsp2lpc(lpc_ptr); + lpc_ptr += LPC_ORDER; + } +} + +/** + * 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(int16_t)); + 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, + Rate cur_rate, int pitch_lag, int index) +{ + int temp, i, j; + + memset(vector, 0, SUBFRAME_LEN * sizeof(int16_t)); + + if (cur_rate == Rate6k3) { + 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 { /* Rate5k3 */ + 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]; +} + +/** + * Generate adaptive codebook excitation. + */ +static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, + int pitch_lag, G723_1_Subframe subfrm, + 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 == Rate6k3 && pitch_lag < SUBFRAME_LEN - 2) { + cb_ptr = adaptive_cb_gain85; + } else + cb_ptr = adaptive_cb_gain170; + + /* Calculate adaptive vector */ + cb_ptr += subfrm.ad_cb_gain * 20; + for (i = 0; i < SUBFRAME_LEN; i++) { + sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER, 1); + 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); + limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3); + + for (i = pitch_lag - 3; i <= limit; i++) { + ccr = ff_dot_product(buf, buf + dir * i, length, 1); + + 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, 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, 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] = ff_dot_product(buf, buf, SUBFRAME_LEN, 1); + + /* Compute forward residual energy */ + if (fwd_lag) + energy[2] = ff_dot_product(buf + fwd_lag, buf + fwd_lag, + SUBFRAME_LEN, 1); + + /* Compute backward residual energy */ + if (back_lag) + energy[4] = ff_dot_product(buf - back_lag, buf - back_lag, + SUBFRAME_LEN, 1); + + /* Normalize and shorten */ + temp1 = 0; + for (i = 0; i < 5; i++) + temp1 = FFMAX(energy[i], temp1); + + scale = normalize_bits(temp1, 1); + for (i = 0; i < 5; i++) + energy[i] = av_clipl_int32(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 = ff_dot_product(buf, buf, SUBFRAME_LEN * 2, 1); + *exc_eng = av_clipl_int32(tgt_eng + (1 << 15)) >> 16; + + if (ccr <= 0) + return 0; + + /* Compute best energy */ + best_eng = ff_dot_product(buf - index, buf - index, + SUBFRAME_LEN * 2, 1); + 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(int16_t), + FRAME_LEN * sizeof(int16_t)); + memcpy(out, vector_ptr, FRAME_LEN * sizeof(int16_t)); + } 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(int16_t)); + } +} + +/** + * Perform IIR filtering. + * + * @param fir_coef FIR coefficients + * @param iir_coef IIR coefficients + * @param src source vector + * @param dest destination vector + * @param width width of the output, 16 bits(0) / 32 bits(1) + */ +#define iir_filter(fir_coef, iir_coef, src, dest, width)\ +{\ + int m, n;\ + int res_shift = 16 & ~-(width);\ + int in_shift = 16 - res_shift;\ +\ + for (m = 0; m < SUBFRAME_LEN; m++) {\ + int64_t filter = 0;\ + for (n = 1; n <= LPC_ORDER; n++) {\ + filter -= (fir_coef)[n - 1] * (src)[m - n] -\ + (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\ + }\ +\ + (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\ + (1 << 15)) >> res_shift;\ + }\ +} + +/** + * Adjust gain of postfiltered signal. + * + * @param p the context + * @param buf postfiltered output vector + * @param energy input energy coefficient + */ +static void gain_scale(G723_1_Context *p, int16_t * buf, int energy) +{ + int num, denom, gain, bits1, bits2; + int i; + + num = energy; + denom = 0; + for (i = 0; i < SUBFRAME_LEN; i++) { + int64_t temp = buf[i] >> 2; + temp = av_clipl_int32(MUL64(temp, temp) << 1); + denom = av_clipl_int32(denom + temp); + } + + if (num && denom) { + bits1 = normalize_bits(num, 1); + bits2 = normalize_bits(denom, 1); + num = num << bits1 >> 1; + denom <<= bits2; + + bits2 = 5 + bits1 - bits2; + bits2 = FFMAX(0, bits2); + + gain = (num >> 1) / (denom >> 16); + gain = square_root(gain << 16 >> bits2); + } else { + gain = 1 << 12; + } + + for (i = 0; i < SUBFRAME_LEN; i++) { + p->pf_gain = ((p->pf_gain << 4) - 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(int16_t)); + memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(int)); + + for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { + for (k = 0; k < LPC_ORDER; k++) { + filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] + + (1 << 14)) >> 15; + filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] + + (1 << 14)) >> 15; + } + iir_filter(filter_coef[0], filter_coef[1], buf + i, + filter_signal + i, 1); + } + + memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t)); + memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int)); + + buf_ptr = buf + LPC_ORDER; + signal_ptr = filter_signal + LPC_ORDER; + for (i = 0; i < SUBFRAMES; i++) { + int16_t temp_vector[SUBFRAME_LEN]; + int16_t temp; + int auto_corr[2]; + int scale, energy; + + /* Normalize */ + memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(int16_t)); + scale = scale_vector(temp_vector, SUBFRAME_LEN); + + /* Compute auto correlation coefficients */ + auto_corr[0] = ff_dot_product(temp_vector, temp_vector + 1, + SUBFRAME_LEN - 1, 1); + auto_corr[1] = ff_dot_product(temp_vector, temp_vector, + SUBFRAME_LEN, 1); + + /* Compute reflection coefficient */ + temp = auto_corr[1] >> 16; + if (temp) { + temp = (auto_corr[0] >> 2) / temp; + } + p->reflection_coef = ((p->reflection_coef << 2) - p->reflection_coef + + temp + 2) >> 2; + temp = (p->reflection_coef * 0xffffc >> 3) & 0xfffc; + + /* Compensation filter */ + for (j = 0; j < SUBFRAME_LEN; j++) { + buf_ptr[j] = av_clipl_int32(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 *data_size, AVPacket *avpkt) +{ + G723_1_Context *p = avctx->priv_data; + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + int16_t *out = data; + 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; + int bad_frame = 0, i, j; + + if (!buf_size || buf_size < frame_size[dec_mode]) { + *data_size = 0; + return buf_size; + } + + if (unpack_bitstream(p, buf, buf_size) < 0) { + bad_frame = 1; + p->cur_frame_type = p->past_frame_type == ActiveFrame ? + ActiveFrame : UntransmittedFrame; + } + + *data_size = FRAME_LEN * sizeof(int16_t); + if(p->cur_frame_type == ActiveFrame) { + 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(int16_t)); + + /* Generate the excitation for the frame */ + memcpy(p->excitation, p->prev_excitation, PITCH_MAX * sizeof(int16_t)); + 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(out, vector_ptr, FRAME_LEN * sizeof(int16_t)); + + p->interp_index = comp_interp_index(p, p->pitch_lag[1], + &p->sid_gain, &p->cur_gain); + + for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) + comp_ppf_coeff(p, i, p->pitch_lag[j >> 1], + ppf + j, p->cur_rate); + + /* Restore the original excitation */ + memcpy(p->excitation, p->prev_excitation, + PITCH_MAX * sizeof(int16_t)); + memcpy(vector_ptr, out, FRAME_LEN * sizeof(int16_t)); + + /* Peform pitch postfiltering */ + for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) + ff_acelp_weighted_vector_sum(out + 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(int16_t)); + memset(out, 0, (FRAME_LEN + LPC_ORDER) * sizeof(int16_t)); + } else { + /* Regenerate frame */ + residual_interp(p->excitation, out + LPC_ORDER, p->interp_index, + p->interp_gain, &p->random_seed); + } + } + /* Save the excitation for the next frame */ + memcpy(p->prev_excitation, p->excitation + FRAME_LEN, + PITCH_MAX * sizeof(int16_t)); + } else { + memset(out, 0, *data_size); + av_log(avctx, AV_LOG_WARNING, + "G.723.1: Comfort noise generation not supported yet\n"); + return frame_size[dec_mode]; + } + + p->past_frame_type = p->cur_frame_type; + + memcpy(out, p->synth_mem, LPC_ORDER * sizeof(int16_t)); + for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) + ff_celp_lp_synthesis_filter(out + i, &lpc[j * LPC_ORDER], + out + i, SUBFRAME_LEN, LPC_ORDER, + 0, 1, 1 << 12); + memcpy(p->synth_mem, out + FRAME_LEN, LPC_ORDER * sizeof(int16_t)); + + formant_postfilter(p, lpc, out); + + memmove(out, out + LPC_ORDER, *data_size); + + return frame_size[dec_mode]; +} + +AVCodec ff_g723_1_decoder = { + .name = "g723_1", + .type = AVMEDIA_TYPE_AUDIO, + .id = 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, +}; diff --git a/libavcodec/version.h b/libavcodec/version.h index 872857d845..1b5ee01c95 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -21,7 +21,7 @@ #define AVCODEC_VERSION_H #define LIBAVCODEC_VERSION_MAJOR 53 -#define LIBAVCODEC_VERSION_MINOR 18 +#define LIBAVCODEC_VERSION_MINOR 19 #define LIBAVCODEC_VERSION_MICRO 0 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ diff --git a/libavformat/Makefile b/libavformat/Makefile index 04e995fc07..5f3bf4dde1 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -98,6 +98,7 @@ OBJS-$(CONFIG_GXF_DEMUXER) += gxf.o OBJS-$(CONFIG_GXF_MUXER) += gxfenc.o audiointerleave.o OBJS-$(CONFIG_G722_DEMUXER) += rawdec.o OBJS-$(CONFIG_G722_MUXER) += rawenc.o +OBJS-$(CONFIG_G723_1_DEMUXER) += g723_1.o OBJS-$(CONFIG_H261_DEMUXER) += h261dec.o rawdec.o OBJS-$(CONFIG_H261_MUXER) += rawenc.o OBJS-$(CONFIG_H263_DEMUXER) += h263dec.o rawdec.o diff --git a/libavformat/allformats.c b/libavformat/allformats.c index 429ccc51fa..94421aba07 100644 --- a/libavformat/allformats.c +++ b/libavformat/allformats.c @@ -100,6 +100,7 @@ void av_register_all(void) REGISTER_MUXER (FRAMECRC, framecrc); REGISTER_MUXER (FRAMEMD5, framemd5); REGISTER_MUXDEMUX (G722, g722); + REGISTER_DEMUXER (G723_1, g723_1); REGISTER_MUXER (GIF, gif); REGISTER_DEMUXER (GSM, gsm); REGISTER_MUXDEMUX (GXF, gxf); diff --git a/libavformat/g723_1.c b/libavformat/g723_1.c new file mode 100644 index 0000000000..5ffd7c7619 --- /dev/null +++ b/libavformat/g723_1.c @@ -0,0 +1,83 @@ +/* + * G.723.1 demuxer + * Copyright (c) 2010 Mohamed Naufal Basheer + * + * This file is part of FFmpeg. + * + * FFmpeg 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. + * + * FFmpeg 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 FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * G.723.1 demuxer + */ + +#include "avformat.h" + +static const uint8_t frame_size[4] = {24, 20, 4, 1}; + +static int g723_1_init(AVFormatContext *s, AVFormatParameters *ap) +{ + AVStream *st; + + st = av_new_stream(s, 0); + if (!st) + return AVERROR(ENOMEM); + + st->codec->codec_type = AVMEDIA_TYPE_AUDIO; + st->codec->codec_id = CODEC_ID_G723_1; + st->codec->channels = 1; + st->codec->sample_rate = 8000; + + av_set_pts_info(st, 64, 1, st->codec->sample_rate); + + return 0; +} + +static int g723_1_read_packet(AVFormatContext *s, AVPacket *pkt) +{ + int size, byte, ret; + + pkt->pos = url_ftell(s->pb); + byte = get_byte(s->pb); + size = frame_size[byte & 3]; + + ret = av_new_packet(pkt, size); + if (ret < 0) + return ret; + + pkt->data[0] = byte; + pkt->duration = 240; + pkt->stream_index = 0; + + ret = get_buffer(s->pb, pkt->data + 1, size - 1); + if (ret < size - 1) { + av_free_packet(pkt); + return ret < 0 ? ret : AVERROR_EOF; + } + + return pkt->size; +} + +AVInputFormat ff_g723_1_demuxer = { + "g723_1", + NULL_IF_CONFIG_SMALL("G.723.1 format"), + 0, + NULL, + g723_1_init, + g723_1_read_packet, + .extensions = "tco", + .flags = AVFMT_GENERIC_INDEX +}; diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 6028fe0b82..127b540748 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -44,7 +44,7 @@ static const struct { {0, "PCMU", AVMEDIA_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, {3, "GSM", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, - {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {4, "G723", AVMEDIA_TYPE_AUDIO, CODEC_ID_G723_1, 8000, 1}, {5, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {6, "DVI4", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, {7, "LPC", AVMEDIA_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},