| /* |
| * SIPR / ACELP.NET decoder |
| * |
| * Copyright (c) 2008 Vladimir Voroshilov |
| * Copyright (c) 2009 Vitor Sessak |
| * |
| * 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 |
| */ |
| |
| #include <math.h> |
| #include <stdint.h> |
| #include <string.h> |
| |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/mathematics.h" |
| #include "avcodec.h" |
| #define BITSTREAM_READER_LE |
| #include "get_bits.h" |
| #include "internal.h" |
| |
| #include "lsp.h" |
| #include "acelp_vectors.h" |
| #include "acelp_pitch_delay.h" |
| #include "acelp_filters.h" |
| #include "celp_filters.h" |
| |
| #define MAX_SUBFRAME_COUNT 5 |
| |
| #include "sipr.h" |
| #include "siprdata.h" |
| |
| typedef struct SiprModeParam { |
| const char *mode_name; |
| uint16_t bits_per_frame; |
| uint8_t subframe_count; |
| uint8_t frames_per_packet; |
| float pitch_sharp_factor; |
| |
| /* bitstream parameters */ |
| uint8_t number_of_fc_indexes; |
| uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor |
| |
| /** size in bits of the i-th stage vector of quantizer */ |
| uint8_t vq_indexes_bits[5]; |
| |
| /** size in bits of the adaptive-codebook index for every subframe */ |
| uint8_t pitch_delay_bits[5]; |
| |
| uint8_t gp_index_bits; |
| uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes |
| uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes |
| } SiprModeParam; |
| |
| static const SiprModeParam modes[MODE_COUNT] = { |
| [MODE_16k] = { |
| .mode_name = "16k", |
| .bits_per_frame = 160, |
| .subframe_count = SUBFRAME_COUNT_16k, |
| .frames_per_packet = 1, |
| .pitch_sharp_factor = 0.00, |
| |
| .number_of_fc_indexes = 10, |
| .ma_predictor_bits = 1, |
| .vq_indexes_bits = {7, 8, 7, 7, 7}, |
| .pitch_delay_bits = {9, 6}, |
| .gp_index_bits = 4, |
| .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5}, |
| .gc_index_bits = 5 |
| }, |
| |
| [MODE_8k5] = { |
| .mode_name = "8k5", |
| .bits_per_frame = 152, |
| .subframe_count = 3, |
| .frames_per_packet = 1, |
| .pitch_sharp_factor = 0.8, |
| |
| .number_of_fc_indexes = 3, |
| .ma_predictor_bits = 0, |
| .vq_indexes_bits = {6, 7, 7, 7, 5}, |
| .pitch_delay_bits = {8, 5, 5}, |
| .gp_index_bits = 0, |
| .fc_index_bits = {9, 9, 9}, |
| .gc_index_bits = 7 |
| }, |
| |
| [MODE_6k5] = { |
| .mode_name = "6k5", |
| .bits_per_frame = 232, |
| .subframe_count = 3, |
| .frames_per_packet = 2, |
| .pitch_sharp_factor = 0.8, |
| |
| .number_of_fc_indexes = 3, |
| .ma_predictor_bits = 0, |
| .vq_indexes_bits = {6, 7, 7, 7, 5}, |
| .pitch_delay_bits = {8, 5, 5}, |
| .gp_index_bits = 0, |
| .fc_index_bits = {5, 5, 5}, |
| .gc_index_bits = 7 |
| }, |
| |
| [MODE_5k0] = { |
| .mode_name = "5k0", |
| .bits_per_frame = 296, |
| .subframe_count = 5, |
| .frames_per_packet = 2, |
| .pitch_sharp_factor = 0.85, |
| |
| .number_of_fc_indexes = 1, |
| .ma_predictor_bits = 0, |
| .vq_indexes_bits = {6, 7, 7, 7, 5}, |
| .pitch_delay_bits = {8, 5, 8, 5, 5}, |
| .gp_index_bits = 0, |
| .fc_index_bits = {10}, |
| .gc_index_bits = 7 |
| } |
| }; |
| |
| const float ff_pow_0_5[] = { |
| 1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4), |
| 1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8), |
| 1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12), |
| 1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16) |
| }; |
| |
| static void dequant(float *out, const int *idx, const float * const cbs[]) |
| { |
| int i; |
| int stride = 2; |
| int num_vec = 5; |
| |
| for (i = 0; i < num_vec; i++) |
| memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float)); |
| |
| } |
| |
| static void lsf_decode_fp(float *lsfnew, float *lsf_history, |
| const SiprParameters *parm) |
| { |
| int i; |
| float lsf_tmp[LP_FILTER_ORDER]; |
| |
| dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks); |
| |
| for (i = 0; i < LP_FILTER_ORDER; i++) |
| lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i]; |
| |
| ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1); |
| |
| /* Note that a minimum distance is not enforced between the last value and |
| the previous one, contrary to what is done in ff_acelp_reorder_lsf() */ |
| ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1); |
| lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI); |
| |
| memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history)); |
| |
| for (i = 0; i < LP_FILTER_ORDER - 1; i++) |
| lsfnew[i] = cos(lsfnew[i]); |
| lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI; |
| } |
| |
| /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */ |
| static void pitch_sharpening(int pitch_lag_int, float beta, |
| float *fixed_vector) |
| { |
| int i; |
| |
| for (i = pitch_lag_int; i < SUBFR_SIZE; i++) |
| fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int]; |
| } |
| |
| /** |
| * Extract decoding parameters from the input bitstream. |
| * @param parms parameters structure |
| * @param pgb pointer to initialized GetBitContext structure |
| */ |
| static void decode_parameters(SiprParameters* parms, GetBitContext *pgb, |
| const SiprModeParam *p) |
| { |
| int i, j; |
| |
| if (p->ma_predictor_bits) |
| parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits); |
| |
| for (i = 0; i < 5; i++) |
| parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]); |
| |
| for (i = 0; i < p->subframe_count; i++) { |
| parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]); |
| if (p->gp_index_bits) |
| parms->gp_index[i] = get_bits(pgb, p->gp_index_bits); |
| |
| for (j = 0; j < p->number_of_fc_indexes; j++) |
| parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]); |
| |
| parms->gc_index[i] = get_bits(pgb, p->gc_index_bits); |
| } |
| } |
| |
| static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az, |
| int num_subfr) |
| { |
| double lsfint[LP_FILTER_ORDER]; |
| int i,j; |
| float t, t0 = 1.0 / num_subfr; |
| |
| t = t0 * 0.5; |
| for (i = 0; i < num_subfr; i++) { |
| for (j = 0; j < LP_FILTER_ORDER; j++) |
| lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j]; |
| |
| ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER); |
| Az += LP_FILTER_ORDER; |
| t += t0; |
| } |
| } |
| |
| /** |
| * Evaluate the adaptive impulse response. |
| */ |
| static void eval_ir(const float *Az, int pitch_lag, float *freq, |
| float pitch_sharp_factor) |
| { |
| float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1]; |
| int i; |
| |
| tmp1[0] = 1.0; |
| for (i = 0; i < LP_FILTER_ORDER; i++) { |
| tmp1[i+1] = Az[i] * ff_pow_0_55[i]; |
| tmp2[i ] = Az[i] * ff_pow_0_7 [i]; |
| } |
| memset(tmp1 + 11, 0, 37 * sizeof(float)); |
| |
| ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE, |
| LP_FILTER_ORDER); |
| |
| pitch_sharpening(pitch_lag, pitch_sharp_factor, freq); |
| } |
| |
| /** |
| * Evaluate the convolution of a vector with a sparse vector. |
| */ |
| static void convolute_with_sparse(float *out, const AMRFixed *pulses, |
| const float *shape, int length) |
| { |
| int i, j; |
| |
| memset(out, 0, length*sizeof(float)); |
| for (i = 0; i < pulses->n; i++) |
| for (j = pulses->x[i]; j < length; j++) |
| out[j] += pulses->y[i] * shape[j - pulses->x[i]]; |
| } |
| |
| /** |
| * Apply postfilter, very similar to AMR one. |
| */ |
| static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples) |
| { |
| float buf[SUBFR_SIZE + LP_FILTER_ORDER]; |
| float *pole_out = buf + LP_FILTER_ORDER; |
| float lpc_n[LP_FILTER_ORDER]; |
| float lpc_d[LP_FILTER_ORDER]; |
| int i; |
| |
| for (i = 0; i < LP_FILTER_ORDER; i++) { |
| lpc_d[i] = lpc[i] * ff_pow_0_75[i]; |
| lpc_n[i] = lpc[i] * ff_pow_0_5 [i]; |
| }; |
| |
| memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem, |
| LP_FILTER_ORDER*sizeof(float)); |
| |
| ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE, |
| LP_FILTER_ORDER); |
| |
| memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, |
| LP_FILTER_ORDER*sizeof(float)); |
| |
| ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE); |
| |
| memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0, |
| LP_FILTER_ORDER*sizeof(*pole_out)); |
| |
| memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER, |
| LP_FILTER_ORDER*sizeof(*pole_out)); |
| |
| ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE, |
| LP_FILTER_ORDER); |
| |
| } |
| |
| static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses, |
| SiprMode mode, int low_gain) |
| { |
| int i; |
| |
| switch (mode) { |
| case MODE_6k5: |
| for (i = 0; i < 3; i++) { |
| fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i; |
| fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1; |
| } |
| fixed_sparse->n = 3; |
| break; |
| case MODE_8k5: |
| for (i = 0; i < 3; i++) { |
| fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i; |
| fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i; |
| |
| fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0; |
| |
| fixed_sparse->y[2*i + 1] = |
| (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ? |
| -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i]; |
| } |
| |
| fixed_sparse->n = 6; |
| break; |
| case MODE_5k0: |
| default: |
| if (low_gain) { |
| int offset = (pulses[0] & 0x200) ? 2 : 0; |
| int val = pulses[0]; |
| |
| for (i = 0; i < 3; i++) { |
| int index = (val & 0x7) * 6 + 4 - i*2; |
| |
| fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1; |
| fixed_sparse->x[i] = index; |
| |
| val >>= 3; |
| } |
| fixed_sparse->n = 3; |
| } else { |
| int pulse_subset = (pulses[0] >> 8) & 1; |
| |
| fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset; |
| fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1; |
| |
| fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1; |
| fixed_sparse->y[1] = -fixed_sparse->y[0]; |
| fixed_sparse->n = 2; |
| } |
| break; |
| } |
| } |
| |
| static void decode_frame(SiprContext *ctx, SiprParameters *params, |
| float *out_data) |
| { |
| int i, j; |
| int subframe_count = modes[ctx->mode].subframe_count; |
| int frame_size = subframe_count * SUBFR_SIZE; |
| float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT]; |
| float *excitation; |
| float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER]; |
| float lsf_new[LP_FILTER_ORDER]; |
| float *impulse_response = ir_buf + LP_FILTER_ORDER; |
| float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for |
| // memory alignment |
| int t0_first = 0; |
| AMRFixed fixed_cb; |
| |
| memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float)); |
| lsf_decode_fp(lsf_new, ctx->lsf_history, params); |
| |
| sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count); |
| |
| memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float)); |
| |
| excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL; |
| |
| for (i = 0; i < subframe_count; i++) { |
| float *pAz = Az + i*LP_FILTER_ORDER; |
| float fixed_vector[SUBFR_SIZE]; |
| int T0,T0_frac; |
| float pitch_gain, gain_code, avg_energy; |
| |
| ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i, |
| ctx->mode == MODE_5k0, 6); |
| |
| if (i == 0 || (i == 2 && ctx->mode == MODE_5k0)) |
| t0_first = T0; |
| |
| ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0), |
| ff_b60_sinc, 6, |
| 2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER, |
| SUBFR_SIZE); |
| |
| decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode, |
| ctx->past_pitch_gain < 0.8); |
| |
| eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor); |
| |
| convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response, |
| SUBFR_SIZE); |
| |
| avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector, |
| fixed_vector, |
| SUBFR_SIZE)) / |
| SUBFR_SIZE; |
| |
| ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0]; |
| |
| gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1], |
| avg_energy, ctx->energy_history, |
| 34 - 15.0/(0.05*M_LN10/M_LN2), |
| pred); |
| |
| ff_weighted_vector_sumf(excitation, excitation, fixed_vector, |
| pitch_gain, gain_code, SUBFR_SIZE); |
| |
| pitch_gain *= 0.5 * pitch_gain; |
| pitch_gain = FFMIN(pitch_gain, 0.4); |
| |
| ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain; |
| ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain); |
| gain_code *= ctx->gain_mem; |
| |
| for (j = 0; j < SUBFR_SIZE; j++) |
| fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j]; |
| |
| if (ctx->mode == MODE_5k0) { |
| postfilter_5k0(ctx, pAz, fixed_vector); |
| |
| ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE, |
| pAz, excitation, SUBFR_SIZE, |
| LP_FILTER_ORDER); |
| } |
| |
| ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector, |
| SUBFR_SIZE, LP_FILTER_ORDER); |
| |
| excitation += SUBFR_SIZE; |
| } |
| |
| memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER, |
| LP_FILTER_ORDER * sizeof(float)); |
| |
| if (ctx->mode == MODE_5k0) { |
| for (i = 0; i < subframe_count; i++) { |
| float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE, |
| ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE, |
| SUBFR_SIZE); |
| ff_adaptive_gain_control(&synth[i * SUBFR_SIZE], |
| &synth[i * SUBFR_SIZE], energy, |
| SUBFR_SIZE, 0.9, &ctx->postfilter_agc); |
| } |
| |
| memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size, |
| LP_FILTER_ORDER*sizeof(float)); |
| } |
| memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL, |
| (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float)); |
| |
| ff_acelp_apply_order_2_transfer_function(out_data, synth, |
| (const float[2]) {-1.99997 , 1.000000000}, |
| (const float[2]) {-1.93307352, 0.935891986}, |
| 0.939805806, |
| ctx->highpass_filt_mem, |
| frame_size); |
| } |
| |
| static av_cold int sipr_decoder_init(AVCodecContext * avctx) |
| { |
| SiprContext *ctx = avctx->priv_data; |
| int i; |
| |
| switch (avctx->block_align) { |
| case 20: ctx->mode = MODE_16k; break; |
| case 19: ctx->mode = MODE_8k5; break; |
| case 29: ctx->mode = MODE_6k5; break; |
| case 37: ctx->mode = MODE_5k0; break; |
| default: |
| if (avctx->bit_rate > 12200) ctx->mode = MODE_16k; |
| else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5; |
| else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5; |
| else ctx->mode = MODE_5k0; |
| av_log(avctx, AV_LOG_WARNING, |
| "Invalid block_align: %d. Mode %s guessed based on bitrate: %"PRId64"\n", |
| avctx->block_align, modes[ctx->mode].mode_name, (int64_t)avctx->bit_rate); |
| } |
| |
| av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name); |
| |
| if (ctx->mode == MODE_16k) { |
| ff_sipr_init_16k(ctx); |
| ctx->decode_frame = ff_sipr_decode_frame_16k; |
| } else { |
| ctx->decode_frame = decode_frame; |
| } |
| |
| for (i = 0; i < LP_FILTER_ORDER; i++) |
| ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1)); |
| |
| for (i = 0; i < 4; i++) |
| ctx->energy_history[i] = -14; |
| |
| avctx->channels = 1; |
| avctx->channel_layout = AV_CH_LAYOUT_MONO; |
| avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| |
| return 0; |
| } |
| |
| static int sipr_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| SiprContext *ctx = avctx->priv_data; |
| AVFrame *frame = data; |
| const uint8_t *buf=avpkt->data; |
| SiprParameters parm; |
| const SiprModeParam *mode_par = &modes[ctx->mode]; |
| GetBitContext gb; |
| float *samples; |
| int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE; |
| int i, ret; |
| |
| ctx->avctx = avctx; |
| if (avpkt->size < (mode_par->bits_per_frame >> 3)) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Error processing packet: packet size (%d) too small\n", |
| avpkt->size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* get output buffer */ |
| frame->nb_samples = mode_par->frames_per_packet * subframe_size * |
| mode_par->subframe_count; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| samples = (float *)frame->data[0]; |
| |
| init_get_bits(&gb, buf, mode_par->bits_per_frame); |
| |
| for (i = 0; i < mode_par->frames_per_packet; i++) { |
| decode_parameters(&parm, &gb, mode_par); |
| |
| ctx->decode_frame(ctx, &parm, samples); |
| |
| samples += subframe_size * mode_par->subframe_count; |
| } |
| |
| *got_frame_ptr = 1; |
| |
| return mode_par->bits_per_frame >> 3; |
| } |
| |
| AVCodec ff_sipr_decoder = { |
| .name = "sipr", |
| .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_SIPR, |
| .priv_data_size = sizeof(SiprContext), |
| .init = sipr_decoder_init, |
| .decode = sipr_decode_frame, |
| .capabilities = AV_CODEC_CAP_DR1, |
| }; |