RealAudio 14.4k encoder.

Patch by Francesco Lavra (firstnamelastname@interfree.it)


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@23579 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

Changelog

@@ -89,6 +89,7 @@ version 0.6:
 - 35% faster VP3/Theora decoding
 - faster AAC decoding
 - faster H.264 decoding
+- RealAudio 1.0 (14.4K) encoder
 
 
 

configure

@@ -1270,6 +1270,7 @@ png_decoder_select="zlib"
 png_encoder_select="zlib"
 qcelp_decoder_select="lsp"
 qdm2_decoder_select="mdct rdft"
+real_144_encoder_select="lpc"
 rv10_decoder_select="h263_decoder"
 rv10_encoder_select="h263_encoder"
 rv20_decoder_select="h263_decoder"

doc/general.texi

@@ -635,7 +635,7 @@ following image formats are supported:
 @item QCELP / PureVoice      @tab     @tab  X
 @item QDesign Music Codec 2  @tab     @tab  X
     @tab There are still some distortions.
-@item RealAudio 1.0 (14.4K)  @tab     @tab  X
+@item RealAudio 1.0 (14.4K)  @tab  X  @tab  X
     @tab Real 14400 bit/s codec
 @item RealAudio 2.0 (28.8K)  @tab     @tab  X
     @tab Real 28800 bit/s codec

libavcodec/Makefile

@@ -282,6 +282,7 @@ OBJS-$(CONFIG_QTRLE_DECODER)           += qtrle.o
 OBJS-$(CONFIG_QTRLE_ENCODER)           += qtrleenc.o
 OBJS-$(CONFIG_R210_DECODER)            += r210dec.o
 OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
+OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_math.o celp_filters.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o

libavcodec/allcodecs.c

@@ -247,7 +247,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (NELLYMOSER, nellymoser);
     REGISTER_DECODER (QCELP, qcelp);
     REGISTER_DECODER (QDM2, qdm2);
-    REGISTER_DECODER (RA_144, ra_144);
+    REGISTER_ENCDEC  (RA_144, ra_144);
     REGISTER_DECODER (RA_288, ra_288);
     REGISTER_DECODER (SHORTEN, shorten);
     REGISTER_DECODER (SIPR, sipr);

libavcodec/avcodec.h

@@ -30,8 +30,8 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 52
-#define LIBAVCODEC_VERSION_MINOR 75
-#define LIBAVCODEC_VERSION_MICRO  1
+#define LIBAVCODEC_VERSION_MINOR 76
+#define LIBAVCODEC_VERSION_MICRO  0
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
                                                LIBAVCODEC_VERSION_MINOR, \

libavcodec/ra144.h

@@ -23,13 +23,18 @@
 #define AVCODEC_RA144_H
 
 #include <stdint.h>
+#include "dsputil.h"
 
 #define NBLOCKS         4       ///< number of subblocks within a block
 #define BLOCKSIZE       40      ///< subblock size in 16-bit words
 #define BUFFERSIZE      146     ///< the size of the adaptive codebook
+#define FIXED_CB_SIZE   128     ///< size of fixed codebooks
+#define FRAMESIZE       20      ///< size of encoded frame
+#define LPC_ORDER       10      ///< order of LPC filter
 
 typedef struct {
     AVCodecContext *avctx;
+    DSPContext dsp;
 
     unsigned int     old_energy;        ///< previous frame energy
 
@@ -41,6 +46,8 @@ typedef struct {
 
     unsigned int     lpc_refl_rms[2];
 
+    int16_t curr_block[NBLOCKS * BLOCKSIZE];
+
     /** The current subblock padded by the last 10 values of the previous one. */
     int16_t curr_sblock[50];
 

libavcodec/ra144enc.c

+/*
+ * Real Audio 1.0 (14.4K) encoder
+ * Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
+ *
+ * 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 libavcodec/ra144enc.c
+ * Real Audio 1.0 (14.4K) encoder
+ * @author Francesco Lavra <francescolavra@interfree.it>
+ */
+
+#include <values.h>
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "lpc.h"
+#include "celp_filters.h"
+#include "ra144.h"
+
+
+static av_cold int ra144_encode_init(AVCodecContext * avctx)
+{
+    RA144Context *ractx;
+
+    if (avctx->sample_fmt != SAMPLE_FMT_S16) {
+        av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
+        return -1;
+    }
+    if (avctx->channels != 1) {
+        av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
+               avctx->channels);
+        return -1;
+    }
+    avctx->frame_size = NBLOCKS * BLOCKSIZE;
+    avctx->bit_rate = 8000;
+    ractx = avctx->priv_data;
+    ractx->lpc_coef[0] = ractx->lpc_tables[0];
+    ractx->lpc_coef[1] = ractx->lpc_tables[1];
+    ractx->avctx = avctx;
+    dsputil_init(&ractx->dsp, avctx);
+    return 0;
+}
+
+
+/**
+ * Quantizes a value by searching a sorted table for the element with the
+ * nearest value
+ *
+ * @param value value to quantize
+ * @param table array containing the quantization table
+ * @param size size of the quantization table
+ * @return index of the quantization table corresponding to the element with the
+ *         nearest value
+ */
+static int quantize(int value, const int16_t *table, unsigned int size)
+{
+    unsigned int low = 0, high = size - 1;
+
+    while (1) {
+        int index = (low + high) >> 1;
+        int error = table[index] - value;
+
+        if (index == low)
+            return table[high] + error > value ? low : high;
+        if (error > 0) {
+            high = index;
+        } else {
+            low = index;
+        }
+    }
+}
+
+
+/**
+ * Orthogonalizes a vector to another vector
+ *
+ * @param v vector to orthogonalize
+ * @param u vector against which orthogonalization is performed
+ */
+static void orthogonalize(float *v, const float *u)
+{
+    int i;
+    float num = 0, den = 0;
+
+    for (i = 0; i < BLOCKSIZE; i++) {
+        num += v[i] * u[i];
+        den += u[i] * u[i];
+    }
+    num /= den;
+    for (i = 0; i < BLOCKSIZE; i++)
+        v[i] -= num * u[i];
+}
+
+
+/**
+ * Calculates match score and gain of an LPC-filtered vector with respect to
+ * input data, possibly othogonalizing it to up to 2 other vectors
+ *
+ * @param work array used to calculate the filtered vector
+ * @param coefs coefficients of the LPC filter
+ * @param vect original vector
+ * @param ortho1 first vector against which orthogonalization is performed
+ * @param ortho2 second vector against which orthogonalization is performed
+ * @param data input data
+ * @param score pointer to variable where match score is returned
+ * @param gain pointer to variable where gain is returned
+ */
+static void get_match_score(float *work, const float *coefs, float *vect,
+                            const float *ortho1, const float *ortho2,
+                            const float *data, float *score, float *gain)
+{
+    float c, g;
+    int i;
+
+    ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
+    if (ortho1)
+        orthogonalize(work, ortho1);
+    if (ortho2)
+        orthogonalize(work, ortho2);
+    c = g = 0;
+    for (i = 0; i < BLOCKSIZE; i++) {
+        g += work[i] * work[i];
+        c += data[i] * work[i];
+    }
+    if (c <= 0) {
+        *score = 0;
+        return;
+    }
+    *gain = c / g;
+    *score = *gain * c;
+}
+
+
+/**
+ * Creates a vector from the adaptive codebook at a given lag value
+ *
+ * @param vect array where vector is stored
+ * @param cb adaptive codebook
+ * @param lag lag value
+ */
+static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
+{
+    int i;
+
+    cb += BUFFERSIZE - lag;
+    for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
+        vect[i] = cb[i];
+    if (lag < BLOCKSIZE)
+        for (i = 0; i < BLOCKSIZE - lag; i++)
+            vect[lag + i] = cb[i];
+}
+
+
+/**
+ * Searches the adaptive codebook for the best entry and gain and removes its
+ * contribution from input data
+ *
+ * @param adapt_cb array from which the adaptive codebook is extracted
+ * @param work array used to calculate LPC-filtered vectors
+ * @param coefs coefficients of the LPC filter
+ * @param data input data
+ * @return index of the best entry of the adaptive codebook
+ */
+static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
+                              const float *coefs, float *data)
+{
+    int i, best_vect;
+    float score, gain, best_score, best_gain;
+    float exc[BLOCKSIZE];
+
+    gain = best_score = 0;
+    for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
+        create_adapt_vect(exc, adapt_cb, i);
+        get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
+        if (score > best_score) {
+            best_score = score;
+            best_vect = i;
+            best_gain = gain;
+        }
+    }
+    if (!best_score)
+        return 0;
+
+    /**
+     * Re-calculate the filtered vector from the vector with maximum match score
+     * and remove its contribution from input data.
+     */
+    create_adapt_vect(exc, adapt_cb, best_vect);
+    ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
+    for (i = 0; i < BLOCKSIZE; i++)
+        data[i] -= best_gain * work[i];
+    return (best_vect - BLOCKSIZE / 2 + 1);
+}
+
+
+/**
+ * Finds the best vector of a fixed codebook by applying an LPC filter to
+ * codebook entries, possibly othogonalizing them to up to 2 other vectors and
+ * matching the results with input data
+ *
+ * @param work array used to calculate the filtered vectors
+ * @param coefs coefficients of the LPC filter
+ * @param cb fixed codebook
+ * @param ortho1 first vector against which orthogonalization is performed
+ * @param ortho2 second vector against which orthogonalization is performed
+ * @param data input data
+ * @param idx pointer to variable where the index of the best codebook entry is
+ *        returned
+ * @param gain pointer to variable where the gain of the best codebook entry is
+ *        returned
+ */
+static void find_best_vect(float *work, const float *coefs,
+                           const int8_t cb[][BLOCKSIZE], const float *ortho1,
+                           const float *ortho2, float *data, int *idx,
+                           float *gain)
+{
+    int i, j;
+    float g, score, best_score;
+    float vect[BLOCKSIZE];
+
+    *idx = *gain = best_score = 0;
+    for (i = 0; i < FIXED_CB_SIZE; i++) {
+        for (j = 0; j < BLOCKSIZE; j++)
+            vect[j] = cb[i][j];
+        get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
+        if (score > best_score) {
+            best_score = score;
+            *idx = i;
+            *gain = g;
+        }
+    }
+}
+
+
+/**
+ * Searches the two fixed codebooks for the best entry and gain
+ *
+ * @param work array used to calculate LPC-filtered vectors
+ * @param coefs coefficients of the LPC filter
+ * @param data input data
+ * @param cba_idx index of the best entry of the adaptive codebook
+ * @param cb1_idx pointer to variable where the index of the best entry of the
+ *        first fixed codebook is returned
+ * @param cb2_idx pointer to variable where the index of the best entry of the
+ *        second fixed codebook is returned
+ */
+static void fixed_cb_search(float *work, const float *coefs, float *data,
+                            int cba_idx, int *cb1_idx, int *cb2_idx)
+{
+    int i, ortho_cb1;
+    float gain;
+    float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
+    float vect[BLOCKSIZE];
+
+    /**
+     * The filtered vector from the adaptive codebook can be retrieved from
+     * work, because this function is called just after adaptive_cb_search().
+     */
+    if (cba_idx)
+        memcpy(cba_vect, work, sizeof(cba_vect));
+
+    find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
+                   data, cb1_idx, &gain);
+
+    /**
+     * Re-calculate the filtered vector from the vector with maximum match score
+     * and remove its contribution from input data.
+     */
+    if (gain) {
+        for (i = 0; i < BLOCKSIZE; i++)
+            vect[i] = ff_cb1_vects[*cb1_idx][i];
+        ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
+        if (cba_idx)
+            orthogonalize(work, cba_vect);
+        for (i = 0; i < BLOCKSIZE; i++)
+            data[i] -= gain * work[i];
+        memcpy(cb1_vect, work, sizeof(cb1_vect));
+        ortho_cb1 = 1;
+    } else
+        ortho_cb1 = 0;
+
+    find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
+                   ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
+}
+
+
+/**
+ * Encodes a subblock of the current frame
+ *
+ * @param ractx encoder context
+ * @param sblock_data input data of the subblock
+ * @param lpc_coefs coefficients of the LPC filter
+ * @param rms RMS of the reflection coefficients
+ * @param pb pointer to PutBitContext of the current frame
+ */
+static void ra144_encode_subblock(RA144Context *ractx,
+                                  const int16_t *sblock_data,
+                                  const int16_t *lpc_coefs, unsigned int rms,
+                                  PutBitContext *pb)
+{
+    float data[BLOCKSIZE], work[LPC_ORDER + BLOCKSIZE];
+    float coefs[LPC_ORDER];
+    float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
+    int16_t cba_vect[BLOCKSIZE];
+    int cba_idx, cb1_idx, cb2_idx, gain;
+    int i, n, m[3];
+    float g[3];
+    float error, best_error;
+
+    for (i = 0; i < LPC_ORDER; i++) {
+        work[i] = ractx->curr_sblock[BLOCKSIZE + i];
+        coefs[i] = lpc_coefs[i] * (1/4096.0);
+    }
+
+    /**
+     * Calculate the zero-input response of the LPC filter and subtract it from
+     * input data.
+     */
+    memset(data, 0, sizeof(data));
+    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
+                                 LPC_ORDER);
+    for (i = 0; i < BLOCKSIZE; i++) {
+        zero[i] = work[LPC_ORDER + i];
+        data[i] = sblock_data[i] - zero[i];
+    }
+
+    /**
+     * Codebook search is performed without taking into account the contribution
+     * of the previous subblock, since it has been just subtracted from input
+     * data.
+     */
+    memset(work, 0, LPC_ORDER * sizeof(*work));
+
+    cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
+                                 data);
+    if (cba_idx) {
+        /**
+         * The filtered vector from the adaptive codebook can be retrieved from
+         * work, see implementation of adaptive_cb_search().
+         */
+        memcpy(cba, work + LPC_ORDER, sizeof(cba));
+
+        ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
+        m[0] = (ff_irms(cba_vect) * rms) >> 12;
+    }
+    fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
+    for (i = 0; i < BLOCKSIZE; i++) {
+        cb1[i] = ff_cb1_vects[cb1_idx][i];
+        cb2[i] = ff_cb2_vects[cb2_idx][i];
+    }
+    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
+                                 LPC_ORDER);
+    memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
+    m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
+    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
+                                 LPC_ORDER);
+    memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
+    m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
+    best_error = FLT_MAX;
+    gain = 0;
+    for (n = 0; n < 256; n++) {
+        g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
+               (1/4096.0);
+        g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
+               (1/4096.0);
+        error = 0;
+        if (cba_idx) {
+            g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
+                   (1/4096.0);
+            for (i = 0; i < BLOCKSIZE; i++) {
+                data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
+                          g[2] * cb2[i];
+                error += (data[i] - sblock_data[i]) *
+                         (data[i] - sblock_data[i]);
+            }
+        } else {
+            for (i = 0; i < BLOCKSIZE; i++) {
+                data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
+                error += (data[i] - sblock_data[i]) *
+                         (data[i] - sblock_data[i]);
+            }
+        }
+        if (error < best_error) {
+            best_error = error;
+            gain = n;
+        }
+    }
+    put_bits(pb, 7, cba_idx);
+    put_bits(pb, 8, gain);
+    put_bits(pb, 7, cb1_idx);
+    put_bits(pb, 7, cb2_idx);
+    ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
+                          gain);
+}
+
+
+static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
+                              int buf_size, void *data)
+{
+    static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
+    static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
+    RA144Context *ractx;
+    PutBitContext pb;
+    int32_t lpc_data[NBLOCKS * BLOCKSIZE];
+    int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
+    int shift[LPC_ORDER];
+    int16_t block_coefs[NBLOCKS][LPC_ORDER];
+    int lpc_refl[LPC_ORDER];    /**< reflection coefficients of the frame */
+    unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
+    int energy = 0;
+    int i, idx;
+
+    if (buf_size < FRAMESIZE) {
+        av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+        return 0;
+    }
+    ractx = avctx->priv_data;
+
+    /**
+     * Since the LPC coefficients are calculated on a frame centered over the
+     * fourth subframe, to encode a given frame, data from the next frame is
+     * needed. In each call to this function, the previous frame (whose data are
+     * saved in the encoder context) is encoded, and data from the current frame
+     * are saved in the encoder context to be used in the next function call.
+     */
+    for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
+        lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
+        energy += (lpc_data[i] * lpc_data[i]) >> 4;
+    }
+    for (i = 2 * BLOCKSIZE + BLOCKSIZE / 2; i < NBLOCKS * BLOCKSIZE; i++) {
+        lpc_data[i] = *((int16_t *)data + i - 2 * BLOCKSIZE - BLOCKSIZE / 2) >>
+                      2;
+        energy += (lpc_data[i] * lpc_data[i]) >> 4;
+    }
+    energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
+                                    32)];
+
+    ff_lpc_calc_coefs(&ractx->dsp, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
+                      LPC_ORDER, 16, lpc_coefs, shift, 1, ORDER_METHOD_EST, 12,
+                      0);
+    for (i = 0; i < LPC_ORDER; i++)
+        block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
+                                        (12 - shift[LPC_ORDER - 1]));
+
+    /**
+     * TODO: apply perceptual weighting of the input speech through bandwidth
+     * expansion of the LPC filter.
+     */
+
+    if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
+        /**
+         * The filter is unstable: use the coefficients of the previous frame.
+         */
+        ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
+        ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx);
+    }
+    init_put_bits(&pb, frame, buf_size);
+    for (i = 0; i < LPC_ORDER; i++) {
+        idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
+        put_bits(&pb, bit_sizes[i], idx);
+        lpc_refl[i] = ff_lpc_refl_cb[i][idx];
+    }
+    ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
+    ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
+    refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
+    refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
+                            energy <= ractx->old_energy,
+                            ff_t_sqrt(energy * ractx->old_energy) >> 12);
+    refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
+    refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
+    ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
+    put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
+    for (i = 0; i < NBLOCKS; i++)
+        ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
+                              block_coefs[i], refl_rms[i], &pb);
+    flush_put_bits(&pb);
+    ractx->old_energy = energy;
+    ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
+    FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
+    for (i = 0; i < NBLOCKS * BLOCKSIZE; i++)
+        ractx->curr_block[i] = *((int16_t *)data + i) >> 2;
+    return FRAMESIZE;
+}
+
+
+AVCodec ra_144_encoder =
+{
+    "real_144",
+    CODEC_TYPE_AUDIO,
+    CODEC_ID_RA_144,
+    sizeof(RA144Context),
+    ra144_encode_init,
+    ra144_encode_frame,
+    .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"),
+};