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authortheuni <theuni-nospam-@xbmc.org>2011-01-24 16:05:21 -0500
committertheuni <theuni-nospam-@xbmc.org>2011-01-24 16:05:21 -0500
commitc51b1189e3d5353e842991f5859ddcea0f73e426 (patch)
treeef2cb8a6184699aa614f3655dca4ce661cdc108e /lib/ffmpeg/libavcodec/ra288.c
parentbe61ebdc9e897fe40c6f371111724de79ddee8d5 (diff)
Merged cptspiff's code-reshuffle branch.
Squashed commit due to build breakage during code-reshuffle history. Conflicts: xbmc/Util.cpp xbmc/cdrip/CDDARipper.cpp xbmc/filesystem/Directory.cpp xbmc/filesystem/File.cpp
Diffstat (limited to 'lib/ffmpeg/libavcodec/ra288.c')
-rw-r--r--lib/ffmpeg/libavcodec/ra288.c217
1 files changed, 217 insertions, 0 deletions
diff --git a/lib/ffmpeg/libavcodec/ra288.c b/lib/ffmpeg/libavcodec/ra288.c
new file mode 100644
index 0000000000..bfc62e1ffa
--- /dev/null
+++ b/lib/ffmpeg/libavcodec/ra288.c
@@ -0,0 +1,217 @@
+/*
+ * 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 "avcodec.h"
+#define ALT_BITSTREAM_READER_LE
+#include "get_bits.h"
+#include "ra288.h"
+#include "lpc.h"
+#include "celp_math.h"
+#include "celp_filters.h"
+
+#define MAX_BACKWARD_FILTER_ORDER 36
+#define MAX_BACKWARD_FILTER_LEN 40
+#define MAX_BACKWARD_FILTER_NONREC 35
+
+typedef struct {
+ float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
+ float gain_lpc[10]; ///< 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)
+{
+ avctx->sample_fmt = SAMPLE_FMT_FLT;
+ return 0;
+}
+
+static void apply_window(float *tgt, const float *m1, const float *m2, int n)
+{
+ while (n--)
+ *tgt++ = *m1++ * *m2++;
+}
+
+static void convolve(float *tgt, const float *src, int len, int n)
+{
+ for (; n >= 0; n--)
+ tgt[n] = ff_dot_productf(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 = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
+
+ sum = FFMAX(sum, 1);
+
+ /* shift and store */
+ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
+
+ gain_block[9] = 10 * log10(sum) - 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(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];
+ float work[MAX_BACKWARD_FILTER_ORDER + MAX_BACKWARD_FILTER_LEN + MAX_BACKWARD_FILTER_NONREC];
+
+ apply_window(work, window, hist, order + n + non_rec);
+
+ 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(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(order, n, non_rec, temp, hist, rec, window);
+
+ if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
+ apply_window(lpc, lpc, tab, order);
+
+ memmove(hist, hist + n, move_size*sizeof(*hist));
+}
+
+static int ra288_decode_frame(AVCodecContext * avctx, void *data,
+ int *data_size, AVPacket *avpkt)
+{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ float *out = data;
+ int i, j;
+ 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 0;
+ }
+
+ if (*data_size < 32*5*4)
+ return -1;
+
+ init_get_bits(&gb, buf, avctx->block_align * 8);
+
+ for (i=0; i < 32; i++) {
+ float gain = amptable[get_bits(&gb, 3)];
+ int cb_coef = get_bits(&gb, 6 + (i&1));
+
+ decode(ractx, gain, cb_coef);
+
+ for (j=0; j < 5; j++)
+ *(out++) = ractx->sp_hist[70 + 36 + j];
+
+ if ((i & 7) == 3) {
+ backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window,
+ ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
+
+ backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window,
+ ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
+ }
+ }
+
+ *data_size = (char *)out - (char *)data;
+ return avctx->block_align;
+}
+
+AVCodec ra_288_decoder =
+{
+ "real_288",
+ AVMEDIA_TYPE_AUDIO,
+ CODEC_ID_RA_288,
+ sizeof(RA288Context),
+ ra288_decode_init,
+ NULL,
+ NULL,
+ ra288_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
+};
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