X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fqcelpdec.c;h=d4a82ee9010de51d6ddfaf4f554e789013c4dbf2;hb=6cf7f30655e95e27fd0394b5a80970d6f9517015;hp=ce4ee842d1771d18e7e639cfd39515c7cd4af2bf;hpb=8e36385a63ee0311130266f8fd23fc5e375d50ce;p=ffmpeg diff --git a/libavcodec/qcelpdec.c b/libavcodec/qcelpdec.c index ce4ee842d17..d4a82ee9010 100644 --- a/libavcodec/qcelpdec.c +++ b/libavcodec/qcelpdec.c @@ -2,74 +2,293 @@ * QCELP decoder * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file qcelpdec.c + * @file * QCELP decoder * @author Reynaldo H. Verdejo Pinochet - * @remark FFmpeg merging spearheaded by Kenan Gillet + * @remark Libav merging spearheaded by Kenan Gillet + * @remark Development mentored by Benjamin Larson */ #include +#include "libavutil/channel_layout.h" +#include "libavutil/float_dsp.h" #include "avcodec.h" -#include "bitstream.h" - -#include "qcelp.h" +#include "internal.h" +#include "get_bits.h" #include "qcelpdata.h" - -#include "celp_math.h" #include "celp_filters.h" +#include "acelp_filters.h" +#include "acelp_vectors.h" +#include "lsp.h" #undef NDEBUG #include -static void weighted_vector_sumf(float *out, const float *in_a, - const float *in_b, float weight_coeff_a, - float weight_coeff_b, int length) -{ - int i; +typedef enum { + I_F_Q = -1, /**< insufficient frame quality */ + SILENCE, + RATE_OCTAVE, + RATE_QUARTER, + RATE_HALF, + RATE_FULL +} qcelp_packet_rate; - for(i=0; ipriv_data; int i; - avctx->sample_fmt = SAMPLE_FMT_FLT; + 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.; + 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; + + assert(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 { + assert(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; } /** - * Computes the scaled codebook vector Cdn From INDEX and GAIN + * Compute the scaled codebook vector Cdn From INDEX and GAIN * for all rates. * * The specification lacks some information here. @@ -89,18 +308,18 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx) { * @param gain array holding the 4 pitch subframe gain values * @param cdn_vector array for the generated scaled codebook vector */ -static void compute_svector(const QCELPContext *q, - const float *gain, - float *cdn_vector) { - int i, j, k; +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; + float *rnd, tmp_gain, fir_filter_value; - switch (q->framerate) { + switch (q->bitrate) { case RATE_FULL: for (i = 0; i < 16; i++) { tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO; - cindex = -q->cindex[i]; + cindex = -q->frame.cindex[i]; for (j = 0; j < 10; j++) *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127]; } @@ -108,43 +327,45 @@ static void compute_svector(const QCELPContext *q, case RATE_HALF: for (i = 0; i < 4; i++) { tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO; - cindex = -q->cindex[i]; + 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->lspv[4])<<14 | - (0x003F & q->lspv[3])<< 8 | - (0x0060 & q->lspv[2])<< 1 | - (0x0007 & q->lspv[1])<< 3 | - (0x0038 & q->lspv[0])>> 3 ; - rnd = q->rnd_fir_filter_mem + 20; + 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; + *rnd = (int16_t) cbseed; - // FIR filter + // 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]; + fir_filter_value += qcelp_rnd_fir_coefs[j] * + (rnd[-j] + rnd[-20+j]); - *cdn_vector++ = tmp_gain * fir_filter_value; + 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)); + 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; + cbseed = 521 * cbseed + 259; + *cdn_vector++ = tmp_gain * (int16_t) cbseed; } } break; @@ -156,6 +377,9 @@ static void compute_svector(const QCELPContext *q, *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127]; } break; + case SILENCE: + memset(cdn_vector, 0, 160 * sizeof(float)); + break; } } @@ -166,26 +390,15 @@ static void compute_svector(const QCELPContext *q, * @param v_in gain-controlled vector * @param v_ref vector to control gain of * - * FIXME: If v_ref is a zero vector, it energy is zero - * and the behavior of the gain control is - * undefined in the specs. - * - * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6 + * 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, j, len; - float scalefactor; - - for (i = 0, j = 0; i < 4; i++) { - scalefactor = ff_dot_productf(v_in + j, v_in + j, 40); - if (scalefactor) - scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor); - else - av_log_missing_feature(NULL, "Zero energy for gain control", 1); - for (len = j + 40; j < len; j++) - v_out[j] = scalefactor * v_in[j]; +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); } } @@ -201,7 +414,8 @@ static void apply_gain_ctrl(float *v_out, * @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 + * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 + * otherwise * * @return filter output vector */ @@ -209,24 +423,20 @@ 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; + 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]) - { + for (i = 0; i < 4; i++) { + if (gain[i]) { v_lag = memory + 143 + 40 * i - lag[i]; - for(v_len=v_in+40; v_inbitrate >= 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 { + assert(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 */ -void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc, - const int subframe_num) +static void interpolate_lpc(QCELPContext *q, const float *curr_lspf, + float *lpc, const int subframe_num) { float interpolated_lspf[10]; float weight; - if(q->framerate >= RATE_QUARTER) + if (q->bitrate >= RATE_QUARTER) weight = 0.25 * (subframe_num + 1); - else if(q->framerate == RATE_OCTAVE && !subframe_num) + else if (q->bitrate == RATE_OCTAVE && !subframe_num) weight = 0.625; else weight = 1.0; - if(weight != 1.0) - { - weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, - weight, 1.0 - weight, 10); - qcelp_lspf2lpc(interpolated_lspf, lpc); - }else if(q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num)) - qcelp_lspf2lpc(curr_lspf, lpc); + 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 int buf_size2framerate(const int buf_size) +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; + 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 -1; + 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 bufffer + * + * @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 is missing, guessing the 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); + 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); } -AVCodec qcelp_decoder = +static int qcelp_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) { - .name = "qcelp", - .type = CODEC_TYPE_AUDIO, - .id = CODEC_ID_QCELP, - .init = qcelp_decode_init, - .decode = qcelp_decode_frame, + 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) { + av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + 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; + + init_get_bits(&q->gb, buf, 8 * buf_size); + + 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 = CODEC_CAP_DR1, .priv_data_size = sizeof(QCELPContext), - .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"), };