| /* |
| * QCELP decoder |
| * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet |
| * |
| * 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 |
| * QCELP decoder |
| * @author Reynaldo H. Verdejo Pinochet |
| * @remark FFmpeg merging spearheaded by Kenan Gillet |
| * @remark Development mentored by Benjamin Larson |
| */ |
| |
| #include <stddef.h> |
| |
| #include "libavutil/avassert.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/float_dsp.h" |
| #include "avcodec.h" |
| #include "internal.h" |
| #include "get_bits.h" |
| #include "qcelpdata.h" |
| #include "celp_filters.h" |
| #include "acelp_filters.h" |
| #include "acelp_vectors.h" |
| #include "lsp.h" |
| |
| typedef enum { |
| I_F_Q = -1, /**< insufficient frame quality */ |
| SILENCE, |
| RATE_OCTAVE, |
| RATE_QUARTER, |
| RATE_HALF, |
| RATE_FULL |
| } qcelp_packet_rate; |
| |
| typedef struct QCELPContext { |
| GetBitContext gb; |
| qcelp_packet_rate bitrate; |
| QCELPFrame frame; /**< unpacked data frame */ |
| |
| uint8_t erasure_count; |
| uint8_t octave_count; /**< count the consecutive RATE_OCTAVE frames */ |
| float prev_lspf[10]; |
| float predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */ |
| float pitch_synthesis_filter_mem[303]; |
| float pitch_pre_filter_mem[303]; |
| float rnd_fir_filter_mem[180]; |
| float formant_mem[170]; |
| float last_codebook_gain; |
| int prev_g1[2]; |
| int prev_bitrate; |
| float pitch_gain[4]; |
| uint8_t pitch_lag[4]; |
| uint16_t first16bits; |
| uint8_t warned_buf_mismatch_bitrate; |
| |
| /* postfilter */ |
| float postfilter_synth_mem[10]; |
| float postfilter_agc_mem; |
| float postfilter_tilt_mem; |
| } QCELPContext; |
| |
| /** |
| * Initialize the speech codec according to the specification. |
| * |
| * TIA/EIA/IS-733 2.4.9 |
| */ |
| static av_cold int qcelp_decode_init(AVCodecContext *avctx) |
| { |
| QCELPContext *q = avctx->priv_data; |
| int i; |
| |
| avctx->channels = 1; |
| avctx->channel_layout = AV_CH_LAYOUT_MONO; |
| avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| |
| for (i = 0; i < 10; i++) |
| q->prev_lspf[i] = (i + 1) / 11.0; |
| |
| return 0; |
| } |
| |
| /** |
| * Decode the 10 quantized LSP frequencies from the LSPV/LSP |
| * transmission codes of any bitrate and check for badly received packets. |
| * |
| * @param q the context |
| * @param lspf line spectral pair frequencies |
| * |
| * @return 0 on success, -1 if the packet is badly received |
| * |
| * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3 |
| */ |
| static int decode_lspf(QCELPContext *q, float *lspf) |
| { |
| int i; |
| float tmp_lspf, smooth, erasure_coeff; |
| const float *predictors; |
| |
| if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) { |
| predictors = q->prev_bitrate != RATE_OCTAVE && |
| q->prev_bitrate != I_F_Q ? q->prev_lspf |
| : q->predictor_lspf; |
| |
| if (q->bitrate == RATE_OCTAVE) { |
| q->octave_count++; |
| |
| for (i = 0; i < 10; i++) { |
| q->predictor_lspf[i] = |
| lspf[i] = (q->frame.lspv[i] ? QCELP_LSP_SPREAD_FACTOR |
| : -QCELP_LSP_SPREAD_FACTOR) + |
| predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR + |
| (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11); |
| } |
| smooth = q->octave_count < 10 ? .875 : 0.1; |
| } else { |
| erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR; |
| |
| av_assert2(q->bitrate == I_F_Q); |
| |
| if (q->erasure_count > 1) |
| erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7; |
| |
| for (i = 0; i < 10; i++) { |
| q->predictor_lspf[i] = |
| lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 + |
| erasure_coeff * predictors[i]; |
| } |
| smooth = 0.125; |
| } |
| |
| // Check the stability of the LSP frequencies. |
| lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR); |
| for (i = 1; i < 10; i++) |
| lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR); |
| |
| lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR); |
| for (i = 9; i > 0; i--) |
| lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR); |
| |
| // Low-pass filter the LSP frequencies. |
| ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10); |
| } else { |
| q->octave_count = 0; |
| |
| tmp_lspf = 0.0; |
| for (i = 0; i < 5; i++) { |
| lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001; |
| lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001; |
| } |
| |
| // Check for badly received packets. |
| if (q->bitrate == RATE_QUARTER) { |
| if (lspf[9] <= .70 || lspf[9] >= .97) |
| return -1; |
| for (i = 3; i < 10; i++) |
| if (fabs(lspf[i] - lspf[i - 2]) < .08) |
| return -1; |
| } else { |
| if (lspf[9] <= .66 || lspf[9] >= .985) |
| return -1; |
| for (i = 4; i < 10; i++) |
| if (fabs(lspf[i] - lspf[i - 4]) < .0931) |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| /** |
| * Convert codebook transmission codes to GAIN and INDEX. |
| * |
| * @param q the context |
| * @param gain array holding the decoded gain |
| * |
| * TIA/EIA/IS-733 2.4.6.2 |
| */ |
| static void decode_gain_and_index(QCELPContext *q, float *gain) |
| { |
| int i, subframes_count, g1[16]; |
| float slope; |
| |
| if (q->bitrate >= RATE_QUARTER) { |
| switch (q->bitrate) { |
| case RATE_FULL: subframes_count = 16; break; |
| case RATE_HALF: subframes_count = 4; break; |
| default: subframes_count = 5; |
| } |
| for (i = 0; i < subframes_count; i++) { |
| g1[i] = 4 * q->frame.cbgain[i]; |
| if (q->bitrate == RATE_FULL && !((i + 1) & 3)) { |
| g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32); |
| } |
| |
| gain[i] = qcelp_g12ga[g1[i]]; |
| |
| if (q->frame.cbsign[i]) { |
| gain[i] = -gain[i]; |
| q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127; |
| } |
| } |
| |
| q->prev_g1[0] = g1[i - 2]; |
| q->prev_g1[1] = g1[i - 1]; |
| q->last_codebook_gain = qcelp_g12ga[g1[i - 1]]; |
| |
| if (q->bitrate == RATE_QUARTER) { |
| // Provide smoothing of the unvoiced excitation energy. |
| gain[7] = gain[4]; |
| gain[6] = 0.4 * gain[3] + 0.6 * gain[4]; |
| gain[5] = gain[3]; |
| gain[4] = 0.8 * gain[2] + 0.2 * gain[3]; |
| gain[3] = 0.2 * gain[1] + 0.8 * gain[2]; |
| gain[2] = gain[1]; |
| gain[1] = 0.6 * gain[0] + 0.4 * gain[1]; |
| } |
| } else if (q->bitrate != SILENCE) { |
| if (q->bitrate == RATE_OCTAVE) { |
| g1[0] = 2 * q->frame.cbgain[0] + |
| av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54); |
| subframes_count = 8; |
| } else { |
| av_assert2(q->bitrate == I_F_Q); |
| |
| g1[0] = q->prev_g1[1]; |
| switch (q->erasure_count) { |
| case 1 : break; |
| case 2 : g1[0] -= 1; break; |
| case 3 : g1[0] -= 2; break; |
| default: g1[0] -= 6; |
| } |
| if (g1[0] < 0) |
| g1[0] = 0; |
| subframes_count = 4; |
| } |
| // This interpolation is done to produce smoother background noise. |
| slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count; |
| for (i = 1; i <= subframes_count; i++) |
| gain[i - 1] = q->last_codebook_gain + slope * i; |
| |
| q->last_codebook_gain = gain[i - 2]; |
| q->prev_g1[0] = q->prev_g1[1]; |
| q->prev_g1[1] = g1[0]; |
| } |
| } |
| |
| /** |
| * If the received packet is Rate 1/4 a further sanity check is made of the |
| * codebook gain. |
| * |
| * @param cbgain the unpacked cbgain array |
| * @return -1 if the sanity check fails, 0 otherwise |
| * |
| * TIA/EIA/IS-733 2.4.8.7.3 |
| */ |
| static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain) |
| { |
| int i, diff, prev_diff = 0; |
| |
| for (i = 1; i < 5; i++) { |
| diff = cbgain[i] - cbgain[i-1]; |
| if (FFABS(diff) > 10) |
| return -1; |
| else if (FFABS(diff - prev_diff) > 12) |
| return -1; |
| prev_diff = diff; |
| } |
| return 0; |
| } |
| |
| /** |
| * Compute the scaled codebook vector Cdn From INDEX and GAIN |
| * for all rates. |
| * |
| * The specification lacks some information here. |
| * |
| * TIA/EIA/IS-733 has an omission on the codebook index determination |
| * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says |
| * you have to subtract the decoded index parameter from the given scaled |
| * codebook vector index 'n' to get the desired circular codebook index, but |
| * it does not mention that you have to clamp 'n' to [0-9] in order to get |
| * RI-compliant results. |
| * |
| * The reason for this mistake seems to be the fact they forgot to mention you |
| * have to do these calculations per codebook subframe and adjust given |
| * equation values accordingly. |
| * |
| * @param q the context |
| * @param gain array holding the 4 pitch subframe gain values |
| * @param cdn_vector array for the generated scaled codebook vector |
| */ |
| static void compute_svector(QCELPContext *q, const float *gain, |
| float *cdn_vector) |
| { |
| int i, j, k; |
| uint16_t cbseed, cindex; |
| float *rnd, tmp_gain, fir_filter_value; |
| |
| switch (q->bitrate) { |
| case RATE_FULL: |
| for (i = 0; i < 16; i++) { |
| tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; |
| cindex = -q->frame.cindex[i]; |
| for (j = 0; j < 10; j++) |
| *cdn_vector++ = tmp_gain * |
| qcelp_rate_full_codebook[cindex++ & 127]; |
| } |
| break; |
| case RATE_HALF: |
| for (i = 0; i < 4; i++) { |
| tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO; |
| cindex = -q->frame.cindex[i]; |
| for (j = 0; j < 40; j++) |
| *cdn_vector++ = tmp_gain * |
| qcelp_rate_half_codebook[cindex++ & 127]; |
| } |
| break; |
| case RATE_QUARTER: |
| cbseed = (0x0003 & q->frame.lspv[4]) << 14 | |
| (0x003F & q->frame.lspv[3]) << 8 | |
| (0x0060 & q->frame.lspv[2]) << 1 | |
| (0x0007 & q->frame.lspv[1]) << 3 | |
| (0x0038 & q->frame.lspv[0]) >> 3; |
| rnd = q->rnd_fir_filter_mem + 20; |
| for (i = 0; i < 8; i++) { |
| tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); |
| for (k = 0; k < 20; k++) { |
| cbseed = 521 * cbseed + 259; |
| *rnd = (int16_t) cbseed; |
| |
| // FIR filter |
| fir_filter_value = 0.0; |
| for (j = 0; j < 10; j++) |
| fir_filter_value += qcelp_rnd_fir_coefs[j] * |
| (rnd[-j] + rnd[-20+j]); |
| |
| fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10]; |
| *cdn_vector++ = tmp_gain * fir_filter_value; |
| rnd++; |
| } |
| } |
| memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, |
| 20 * sizeof(float)); |
| break; |
| case RATE_OCTAVE: |
| cbseed = q->first16bits; |
| for (i = 0; i < 8; i++) { |
| tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0); |
| for (j = 0; j < 20; j++) { |
| cbseed = 521 * cbseed + 259; |
| *cdn_vector++ = tmp_gain * (int16_t) cbseed; |
| } |
| } |
| break; |
| case I_F_Q: |
| cbseed = -44; // random codebook index |
| for (i = 0; i < 4; i++) { |
| tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; |
| for (j = 0; j < 40; j++) |
| *cdn_vector++ = tmp_gain * |
| qcelp_rate_full_codebook[cbseed++ & 127]; |
| } |
| break; |
| case SILENCE: |
| memset(cdn_vector, 0, 160 * sizeof(float)); |
| break; |
| } |
| } |
| |
| /** |
| * Apply generic gain control. |
| * |
| * @param v_out output vector |
| * @param v_in gain-controlled vector |
| * @param v_ref vector to control gain of |
| * |
| * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6 |
| */ |
| static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in) |
| { |
| int i; |
| |
| for (i = 0; i < 160; i += 40) { |
| float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40); |
| ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40); |
| } |
| } |
| |
| /** |
| * Apply filter in pitch-subframe steps. |
| * |
| * @param memory buffer for the previous state of the filter |
| * - must be able to contain 303 elements |
| * - the 143 first elements are from the previous state |
| * - the next 160 are for output |
| * @param v_in input filter vector |
| * @param gain per-subframe gain array, each element is between 0.0 and 2.0 |
| * @param lag per-subframe lag array, each element is |
| * - between 16 and 143 if its corresponding pfrac is 0, |
| * - between 16 and 139 otherwise |
| * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 |
| * otherwise |
| * |
| * @return filter output vector |
| */ |
| static const float *do_pitchfilter(float memory[303], const float v_in[160], |
| const float gain[4], const uint8_t *lag, |
| const uint8_t pfrac[4]) |
| { |
| int i, j; |
| float *v_lag, *v_out; |
| const float *v_len; |
| |
| v_out = memory + 143; // Output vector starts at memory[143]. |
| |
| for (i = 0; i < 4; i++) { |
| if (gain[i]) { |
| v_lag = memory + 143 + 40 * i - lag[i]; |
| for (v_len = v_in + 40; v_in < v_len; v_in++) { |
| if (pfrac[i]) { // If it is a fractional lag... |
| for (j = 0, *v_out = 0.0; j < 4; j++) |
| *v_out += qcelp_hammsinc_table[j] * |
| (v_lag[j - 4] + v_lag[3 - j]); |
| } else |
| *v_out = *v_lag; |
| |
| *v_out = *v_in + gain[i] * *v_out; |
| |
| v_lag++; |
| v_out++; |
| } |
| } else { |
| memcpy(v_out, v_in, 40 * sizeof(float)); |
| v_in += 40; |
| v_out += 40; |
| } |
| } |
| |
| memmove(memory, memory + 160, 143 * sizeof(float)); |
| return memory + 143; |
| } |
| |
| /** |
| * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector. |
| * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2 |
| * |
| * @param q the context |
| * @param cdn_vector the scaled codebook vector |
| */ |
| static void apply_pitch_filters(QCELPContext *q, float *cdn_vector) |
| { |
| int i; |
| const float *v_synthesis_filtered, *v_pre_filtered; |
| |
| if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE || |
| (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) { |
| |
| if (q->bitrate >= RATE_HALF) { |
| // Compute gain & lag for the whole frame. |
| for (i = 0; i < 4; i++) { |
| q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0; |
| |
| q->pitch_lag[i] = q->frame.plag[i] + 16; |
| } |
| } else { |
| float max_pitch_gain; |
| |
| if (q->bitrate == I_F_Q) { |
| if (q->erasure_count < 3) |
| max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1); |
| else |
| max_pitch_gain = 0.0; |
| } else { |
| av_assert2(q->bitrate == SILENCE); |
| max_pitch_gain = 1.0; |
| } |
| for (i = 0; i < 4; i++) |
| q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain); |
| |
| memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac)); |
| } |
| |
| // pitch synthesis filter |
| v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem, |
| cdn_vector, q->pitch_gain, |
| q->pitch_lag, q->frame.pfrac); |
| |
| // pitch prefilter update |
| for (i = 0; i < 4; i++) |
| q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0); |
| |
| v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem, |
| v_synthesis_filtered, |
| q->pitch_gain, q->pitch_lag, |
| q->frame.pfrac); |
| |
| apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered); |
| } else { |
| memcpy(q->pitch_synthesis_filter_mem, |
| cdn_vector + 17, 143 * sizeof(float)); |
| memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float)); |
| memset(q->pitch_gain, 0, sizeof(q->pitch_gain)); |
| memset(q->pitch_lag, 0, sizeof(q->pitch_lag)); |
| } |
| } |
| |
| /** |
| * Reconstruct LPC coefficients from the line spectral pair frequencies |
| * and perform bandwidth expansion. |
| * |
| * @param lspf line spectral pair frequencies |
| * @param lpc linear predictive coding coefficients |
| * |
| * @note: bandwidth_expansion_coeff could be precalculated into a table |
| * but it seems to be slower on x86 |
| * |
| * TIA/EIA/IS-733 2.4.3.3.5 |
| */ |
| static void lspf2lpc(const float *lspf, float *lpc) |
| { |
| double lsp[10]; |
| double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF; |
| int i; |
| |
| for (i = 0; i < 10; i++) |
| lsp[i] = cos(M_PI * lspf[i]); |
| |
| ff_acelp_lspd2lpc(lsp, lpc, 5); |
| |
| for (i = 0; i < 10; i++) { |
| lpc[i] *= bandwidth_expansion_coeff; |
| bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF; |
| } |
| } |
| |
| /** |
| * Interpolate LSP frequencies and compute LPC coefficients |
| * for a given bitrate & pitch subframe. |
| * |
| * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2 |
| * |
| * @param q the context |
| * @param curr_lspf LSP frequencies vector of the current frame |
| * @param lpc float vector for the resulting LPC |
| * @param subframe_num frame number in decoded stream |
| */ |
| static void interpolate_lpc(QCELPContext *q, const float *curr_lspf, |
| float *lpc, const int subframe_num) |
| { |
| float interpolated_lspf[10]; |
| float weight; |
| |
| if (q->bitrate >= RATE_QUARTER) |
| weight = 0.25 * (subframe_num + 1); |
| else if (q->bitrate == RATE_OCTAVE && !subframe_num) |
| weight = 0.625; |
| else |
| weight = 1.0; |
| |
| if (weight != 1.0) { |
| ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, |
| weight, 1.0 - weight, 10); |
| lspf2lpc(interpolated_lspf, lpc); |
| } else if (q->bitrate >= RATE_QUARTER || |
| (q->bitrate == I_F_Q && !subframe_num)) |
| lspf2lpc(curr_lspf, lpc); |
| else if (q->bitrate == SILENCE && !subframe_num) |
| lspf2lpc(q->prev_lspf, lpc); |
| } |
| |
| static qcelp_packet_rate buf_size2bitrate(const int buf_size) |
| { |
| switch (buf_size) { |
| case 35: return RATE_FULL; |
| case 17: return RATE_HALF; |
| case 8: return RATE_QUARTER; |
| case 4: return RATE_OCTAVE; |
| case 1: return SILENCE; |
| } |
| |
| return I_F_Q; |
| } |
| |
| /** |
| * Determine the bitrate from the frame size and/or the first byte of the frame. |
| * |
| * @param avctx the AV codec context |
| * @param buf_size length of the buffer |
| * @param buf the buffer |
| * |
| * @return the bitrate on success, |
| * I_F_Q if the bitrate cannot be satisfactorily determined |
| * |
| * TIA/EIA/IS-733 2.4.8.7.1 |
| */ |
| static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, |
| const int buf_size, |
| const uint8_t **buf) |
| { |
| qcelp_packet_rate bitrate; |
| |
| if ((bitrate = buf_size2bitrate(buf_size)) >= 0) { |
| if (bitrate > **buf) { |
| QCELPContext *q = avctx->priv_data; |
| if (!q->warned_buf_mismatch_bitrate) { |
| av_log(avctx, AV_LOG_WARNING, |
| "Claimed bitrate and buffer size mismatch.\n"); |
| q->warned_buf_mismatch_bitrate = 1; |
| } |
| bitrate = **buf; |
| } else if (bitrate < **buf) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Buffer is too small for the claimed bitrate.\n"); |
| return I_F_Q; |
| } |
| (*buf)++; |
| } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) { |
| av_log(avctx, AV_LOG_WARNING, |
| "Bitrate byte missing, guessing bitrate from packet size.\n"); |
| } else |
| return I_F_Q; |
| |
| if (bitrate == SILENCE) { |
| // FIXME: Remove this warning when tested with samples. |
| avpriv_request_sample(avctx, "Blank frame handling"); |
| } |
| return bitrate; |
| } |
| |
| static void warn_insufficient_frame_quality(AVCodecContext *avctx, |
| const char *message) |
| { |
| av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", |
| avctx->frame_number, message); |
| } |
| |
| static void postfilter(QCELPContext *q, float *samples, float *lpc) |
| { |
| static const float pow_0_775[10] = { |
| 0.775000, 0.600625, 0.465484, 0.360750, 0.279582, |
| 0.216676, 0.167924, 0.130141, 0.100859, 0.078166 |
| }, pow_0_625[10] = { |
| 0.625000, 0.390625, 0.244141, 0.152588, 0.095367, |
| 0.059605, 0.037253, 0.023283, 0.014552, 0.009095 |
| }; |
| float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160]; |
| int n; |
| |
| for (n = 0; n < 10; n++) { |
| lpc_s[n] = lpc[n] * pow_0_625[n]; |
| lpc_p[n] = lpc[n] * pow_0_775[n]; |
| } |
| |
| ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s, |
| q->formant_mem + 10, 160, 10); |
| memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10); |
| ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10); |
| memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10); |
| |
| ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160); |
| |
| ff_adaptive_gain_control(samples, pole_out + 10, |
| avpriv_scalarproduct_float_c(q->formant_mem + 10, |
| q->formant_mem + 10, |
| 160), |
| 160, 0.9375, &q->postfilter_agc_mem); |
| } |
| |
| static int qcelp_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| const uint8_t *buf = avpkt->data; |
| int buf_size = avpkt->size; |
| QCELPContext *q = avctx->priv_data; |
| AVFrame *frame = data; |
| float *outbuffer; |
| int i, ret; |
| float quantized_lspf[10], lpc[10]; |
| float gain[16]; |
| float *formant_mem; |
| |
| /* get output buffer */ |
| frame->nb_samples = 160; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| outbuffer = (float *)frame->data[0]; |
| |
| if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) { |
| warn_insufficient_frame_quality(avctx, "Bitrate cannot be determined."); |
| goto erasure; |
| } |
| |
| if (q->bitrate == RATE_OCTAVE && |
| (q->first16bits = AV_RB16(buf)) == 0xFFFF) { |
| warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on."); |
| goto erasure; |
| } |
| |
| if (q->bitrate > SILENCE) { |
| const QCELPBitmap *bitmaps = qcelp_unpacking_bitmaps_per_rate[q->bitrate]; |
| const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] + |
| qcelp_unpacking_bitmaps_lengths[q->bitrate]; |
| uint8_t *unpacked_data = (uint8_t *)&q->frame; |
| |
| if ((ret = init_get_bits8(&q->gb, buf, buf_size)) < 0) |
| return ret; |
| |
| memset(&q->frame, 0, sizeof(QCELPFrame)); |
| |
| for (; bitmaps < bitmaps_end; bitmaps++) |
| unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos; |
| |
| // Check for erasures/blanks on rates 1, 1/4 and 1/8. |
| if (q->frame.reserved) { |
| warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area."); |
| goto erasure; |
| } |
| if (q->bitrate == RATE_QUARTER && |
| codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) { |
| warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed."); |
| goto erasure; |
| } |
| |
| if (q->bitrate >= RATE_HALF) { |
| for (i = 0; i < 4; i++) { |
| if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) { |
| warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter."); |
| goto erasure; |
| } |
| } |
| } |
| } |
| |
| decode_gain_and_index(q, gain); |
| compute_svector(q, gain, outbuffer); |
| |
| if (decode_lspf(q, quantized_lspf) < 0) { |
| warn_insufficient_frame_quality(avctx, "Badly received packets in frame."); |
| goto erasure; |
| } |
| |
| apply_pitch_filters(q, outbuffer); |
| |
| if (q->bitrate == I_F_Q) { |
| erasure: |
| q->bitrate = I_F_Q; |
| q->erasure_count++; |
| decode_gain_and_index(q, gain); |
| compute_svector(q, gain, outbuffer); |
| decode_lspf(q, quantized_lspf); |
| apply_pitch_filters(q, outbuffer); |
| } else |
| q->erasure_count = 0; |
| |
| formant_mem = q->formant_mem + 10; |
| for (i = 0; i < 4; i++) { |
| interpolate_lpc(q, quantized_lspf, lpc, i); |
| ff_celp_lp_synthesis_filterf(formant_mem, lpc, |
| outbuffer + i * 40, 40, 10); |
| formant_mem += 40; |
| } |
| |
| // postfilter, as per TIA/EIA/IS-733 2.4.8.6 |
| postfilter(q, outbuffer, lpc); |
| |
| memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float)); |
| |
| memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf)); |
| q->prev_bitrate = q->bitrate; |
| |
| *got_frame_ptr = 1; |
| |
| return buf_size; |
| } |
| |
| AVCodec ff_qcelp_decoder = { |
| .name = "qcelp", |
| .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_QCELP, |
| .init = qcelp_decode_init, |
| .decode = qcelp_decode_frame, |
| .capabilities = AV_CODEC_CAP_DR1, |
| .priv_data_size = sizeof(QCELPContext), |
| }; |
