SIPR decoder for modes 8k5, 6k5 and 5k0.

Patch by Vladimir Voroshilov and myself.

Originally committed as revision 21125 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 608 insertions, 0 deletions

diff --git a/libavcodec/sipr.c b/libavcodec/sipr.c
new file mode 100644
index 0000000000..7bfa3ab9f1
--- /dev/null
+++ b/libavcodec/sipr.c
@@ -0,0 +1,608 @@
+/*
+ * 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 "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "dsputil.h"
+
+#include "lsp.h"
+#include "celp_math.h"
+#include "acelp_vectors.h"
+#include "acelp_pitch_delay.h"
+#include "acelp_filters.h"
+#include "celp_filters.h"
+
+#define LSFQ_DIFF_MIN        (0.0125 * M_PI)
+
+#define LP_FILTER_ORDER      10
+
+/** Number of past samples needed for excitation interpolation */
+#define L_INTERPOL           (LP_FILTER_ORDER + 1)
+
+/**  Subframe size for all modes except 16k */
+#define SUBFR_SIZE           48
+
+#include "siprdata.h"
+
+typedef enum {
+    MODE_16k,
+    MODE_8k5,
+    MODE_6k5,
+    MODE_5k0,
+    MODE_COUNT
+} SiprMode;
+
+typedef struct {
+    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;
+
+    /** 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_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,
+        .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,
+        .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,
+        .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
+    }
+};
+
+typedef struct {
+    AVCodecContext *avctx;
+    DSPContext dsp;
+
+    SiprModeParam m;
+    SiprMode mode;
+
+    float past_pitch_gain;
+    float lsf_history[LP_FILTER_ORDER];
+
+    float excitation[L_INTERPOL + PITCH_DELAY_MAX + 5*SUBFR_SIZE];
+
+    DECLARE_ALIGNED_16(float, synth_buf[LP_FILTER_ORDER + 5*SUBFR_SIZE + 6]);
+
+    float lsp_history[LP_FILTER_ORDER];
+    float gain_mem;
+    float energy_history[4];
+    float highpass_filt_mem[2];
+    float postfilter_mem[PITCH_DELAY_MAX + LP_FILTER_ORDER];
+
+    /* 5k0 */
+    float tilt_mem;
+    float postfilter_agc;
+    float postfilter_mem5k0[PITCH_DELAY_MAX + LP_FILTER_ORDER];
+    float postfilter_syn5k0[LP_FILTER_ORDER + SUBFR_SIZE*5];
+} SiprContext;
+
+typedef struct {
+    int vq_indexes[5];
+    int pitch_delay[5];        ///< pitch delay
+    int gp_index[5];           ///< adaptive-codebook gain indexes
+    int16_t fc_indexes[5][10]; ///< fixed-codebook indexes
+    int gc_index[5];           ///< fixed-codebook gain indexes
+} SiprParameters;
+
+
+static void dequant(float *out, const int *idx, const float *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];
+}
+
+/**
+ * Extracts 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;
+
+    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]);
+        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 lsp2lpc_sipr(const double *lsp, float *Az)
+{
+    int lp_half_order = LP_FILTER_ORDER >> 1;
+    double buf[lp_half_order + 1];
+    double pa[lp_half_order + 1];
+    double *qa = buf + 1;
+    int i,j;
+
+    qa[-1] = 0.0;
+
+    ff_lsp2polyf(lsp    , pa, lp_half_order    );
+    ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1);
+
+    for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) {
+        double paf =  pa[i]            * (1 + lsp[LP_FILTER_ORDER - 1]);
+        double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]);
+        Az[i-1]  = (paf + qaf) * 0.5;
+        Az[j-1]  = (paf - qaf) * 0.5;
+    }
+
+    Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) *
+        pa[lp_half_order] * 0.5;
+
+    Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1];
+}
+
+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];
+
+        lsp2lpc_sipr(lsfint, Az);
+        Az += LP_FILTER_ORDER;
+        t += t0;
+    }
+}
+
+/**
+ * Evaluates the adaptative 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.;
+    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);
+}
+
+/**
+ * Evaluates 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] *    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 frame_size = ctx->m.subframe_count * SUBFR_SIZE;
+    float Az[LP_FILTER_ORDER * ctx->m.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, ctx->m.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 < ctx->m.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, ctx->m.pitch_sharp_factor);
+
+        convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
+                              SUBFR_SIZE);
+
+        avg_energy =
+            (0.01 + ff_dot_productf(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/(log2f(10.0) * 0.05),
+                                          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 < ctx->m.subframe_count; i++) {
+            float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
+                                           ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
+                                           SUBFR_SIZE);
+            ff_adaptative_gain_control(&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));
+    }
+    memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
+           (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
+
+    ff_acelp_apply_order_2_transfer_function(synth,
+                                             (const float[2]) {-1.99997   , 1.000000000},
+                                             (const float[2]) {-1.93307352, 0.935891986},
+                                             0.939805806,
+                                             ctx->highpass_filt_mem,
+                                             frame_size);
+
+    ctx->dsp.vector_clipf(out_data, synth, -1, 32767./(1<<15), frame_size);
+
+}
+
+static av_cold int sipr_decoder_init(AVCodecContext * avctx)
+{
+    SiprContext *ctx = avctx->priv_data;
+    int i;
+
+    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;
+
+    ctx->m = modes[ctx->mode];
+    av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", ctx->m.mode_name);
+
+    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->sample_fmt = SAMPLE_FMT_FLT;
+
+    if (ctx->mode == MODE_16k) {
+        av_log(avctx, AV_LOG_ERROR, "decoding 16kbps SIPR files is not "
+                                    "supported yet.\n");
+        return -1;
+    }
+
+    dsputil_init(&ctx->dsp, avctx);
+
+    return 0;
+}
+
+static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
+                             int *data_size, AVPacket *avpkt)
+{
+    SiprContext *ctx = avctx->priv_data;
+    const uint8_t *buf=avpkt->data;
+    SiprParameters parm;
+    GetBitContext gb;
+    float *data = datap;
+    int i;
+
+    ctx->avctx = avctx;
+    if (avpkt->size < (ctx->m.bits_per_frame >> 3)) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Error processing packet: packet size (%d) too small\n",
+               avpkt->size);
+
+        *data_size = 0;
+        return -1;
+    }
+    if (*data_size < SUBFR_SIZE * ctx->m.subframe_count * sizeof(float)) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Error processing packet: output buffer (%d) too small\n",
+               *data_size);
+
+        *data_size = 0;
+        return -1;
+    }
+
+    init_get_bits(&gb, buf, ctx->m.bits_per_frame);
+
+    for (i = 0; i < ctx->m.frames_per_packet; i++) {
+        decode_parameters(&parm, &gb, &ctx->m);
+        decode_frame(ctx, &parm, data);
+
+        data += SUBFR_SIZE * ctx->m.subframe_count;
+    }
+
+    *data_size = ctx->m.frames_per_packet * SUBFR_SIZE *
+        ctx->m.subframe_count * sizeof(float);
+
+    return ctx->m.bits_per_frame >> 3;
+};
+
+AVCodec sipr_decoder = {
+    "sipr",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_SIPR,
+    sizeof(SiprContext),
+    sipr_decoder_init,
+    NULL,
+    NULL,
+    sipr_decode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
+};