making act segmenter

1 files changed, 574 insertions, 0 deletions

diff --git a/ffmpeg/libavcodec/sipr.c b/ffmpeg/libavcodec/sipr.c
new file mode 100644
index 0000000..35e8bf5
--- /dev/null
+++ b/ffmpeg/libavcodec/sipr.c
@@ -0,0 +1,574 @@
+/*
+ * 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 {
+    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 *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.;
+    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: %d\n",
+               avctx->block_align, modes[ctx->mode].mode_name, 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 -1;
+    }
+
+    /* 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",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_SIPR,
+    .priv_data_size = sizeof(SiprContext),
+    .init           = sipr_decoder_init,
+    .decode         = sipr_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
+};