diff ffmpeg/libavcodec/ra288.c @ 10:6840f77b83aa

commit

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/ffmpeg/libavcodec/ra288.c	Sun Apr 21 10:55:35 2013 +0200
@@ -0,0 +1,239 @@
+/*
+ * RealAudio 2.0 (28.8K)
+ * Copyright (c) 2003 the ffmpeg project
+ *
+ * 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 "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/internal.h"
+#include "avcodec.h"
+#include "internal.h"
+#define BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "ra288.h"
+#include "lpc.h"
+#include "celp_filters.h"
+
+#define MAX_BACKWARD_FILTER_ORDER  36
+#define MAX_BACKWARD_FILTER_LEN    40
+#define MAX_BACKWARD_FILTER_NONREC 35
+
+#define RA288_BLOCK_SIZE        5
+#define RA288_BLOCKS_PER_FRAME 32
+
+typedef struct {
+    AVFloatDSPContext fdsp;
+    DECLARE_ALIGNED(32, float,   sp_lpc)[FFALIGN(36, 16)];   ///< LPC coefficients for speech data (spec: A)
+    DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)];   ///< LPC coefficients for gain        (spec: GB)
+
+    /** speech data history                                      (spec: SB).
+     *  Its first 70 coefficients are updated only at backward filtering.
+     */
+    float sp_hist[111];
+
+    /// speech part of the gain autocorrelation                  (spec: REXP)
+    float sp_rec[37];
+
+    /** log-gain history                                         (spec: SBLG).
+     *  Its first 28 coefficients are updated only at backward filtering.
+     */
+    float gain_hist[38];
+
+    /// recursive part of the gain autocorrelation               (spec: REXPLG)
+    float gain_rec[11];
+} RA288Context;
+
+static av_cold int ra288_decode_init(AVCodecContext *avctx)
+{
+    RA288Context *ractx = avctx->priv_data;
+
+    avctx->channels       = 1;
+    avctx->channel_layout = AV_CH_LAYOUT_MONO;
+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
+
+    if (avctx->block_align <= 0) {
+        av_log(avctx, AV_LOG_ERROR, "unsupported block align\n");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+
+    return 0;
+}
+
+static void convolve(float *tgt, const float *src, int len, int n)
+{
+    for (; n >= 0; n--)
+        tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
+
+}
+
+static void decode(RA288Context *ractx, float gain, int cb_coef)
+{
+    int i;
+    double sumsum;
+    float sum, buffer[5];
+    float *block = ractx->sp_hist + 70 + 36; // current block
+    float *gain_block = ractx->gain_hist + 28;
+
+    memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
+
+    /* block 46 of G.728 spec */
+    sum = 32.;
+    for (i=0; i < 10; i++)
+        sum -= gain_block[9-i] * ractx->gain_lpc[i];
+
+    /* block 47 of G.728 spec */
+    sum = av_clipf(sum, 0, 60);
+
+    /* block 48 of G.728 spec */
+    /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
+    sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
+
+    for (i=0; i < 5; i++)
+        buffer[i] = codetable[cb_coef][i] * sumsum;
+
+    sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
+
+    sum = FFMAX(sum, 5. / (1<<24));
+
+    /* shift and store */
+    memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
+
+    gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
+
+    ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
+}
+
+/**
+ * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
+ *
+ * @param order   filter order
+ * @param n       input length
+ * @param non_rec number of non-recursive samples
+ * @param out     filter output
+ * @param hist    pointer to the input history of the filter
+ * @param out     pointer to the non-recursive part of the output
+ * @param out2    pointer to the recursive part of the output
+ * @param window  pointer to the windowing function table
+ */
+static void do_hybrid_window(RA288Context *ractx,
+                             int order, int n, int non_rec, float *out,
+                             float *hist, float *out2, const float *window)
+{
+    int i;
+    float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
+    float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
+    LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
+                                            MAX_BACKWARD_FILTER_LEN   +
+                                            MAX_BACKWARD_FILTER_NONREC, 16)]);
+
+    av_assert2(order>=0);
+
+    ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
+
+    convolve(buffer1, work + order    , n      , order);
+    convolve(buffer2, work + order + n, non_rec, order);
+
+    for (i=0; i <= order; i++) {
+        out2[i] = out2[i] * 0.5625 + buffer1[i];
+        out [i] = out2[i]          + buffer2[i];
+    }
+
+    /* Multiply by the white noise correcting factor (WNCF). */
+    *out *= 257./256.;
+}
+
+/**
+ * Backward synthesis filter, find the LPC coefficients from past speech data.
+ */
+static void backward_filter(RA288Context *ractx,
+                            float *hist, float *rec, const float *window,
+                            float *lpc, const float *tab,
+                            int order, int n, int non_rec, int move_size)
+{
+    float temp[MAX_BACKWARD_FILTER_ORDER+1];
+
+    do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
+
+    if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+        ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
+
+    memmove(hist, hist + n, move_size*sizeof(*hist));
+}
+
+static int ra288_decode_frame(AVCodecContext * avctx, void *data,
+                              int *got_frame_ptr, AVPacket *avpkt)
+{
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    float *out;
+    int i, ret;
+    RA288Context *ractx = avctx->priv_data;
+    GetBitContext gb;
+
+    if (buf_size < avctx->block_align) {
+        av_log(avctx, AV_LOG_ERROR,
+               "Error! Input buffer is too small [%d<%d]\n",
+               buf_size, avctx->block_align);
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* get output buffer */
+    frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+    out = (float *)frame->data[0];
+
+    init_get_bits(&gb, buf, avctx->block_align * 8);
+
+    for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
+        float gain = amptable[get_bits(&gb, 3)];
+        int cb_coef = get_bits(&gb, 6 + (i&1));
+
+        decode(ractx, gain, cb_coef);
+
+        memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
+        out += RA288_BLOCK_SIZE;
+
+        if ((i & 7) == 3) {
+            backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
+                            ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
+
+            backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
+                            ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
+        }
+    }
+
+    *got_frame_ptr = 1;
+
+    return avctx->block_align;
+}
+
+AVCodec ff_ra_288_decoder = {
+    .name           = "real_288",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_RA_288,
+    .priv_data_size = sizeof(RA288Context),
+    .init           = ra288_decode_init,
+    .decode         = ra288_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
+};