1 files changed, 99 insertions, 0 deletions
diff --git a/ffmpeg1/libavcodec/mpc.c b/ffmpeg1/libavcodec/mpc.c
new file mode 100644
index 0000000..3bd2d35
--- /dev/null
+++ b/ffmpeg1/libavcodec/mpc.c
@@ -0,0 +1,99 @@
+/*
+ * Musepack decoder core
+ * Copyright (c) 2006 Konstantin Shishkov
+ *
+ * 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
+ * Musepack decoder core
+ * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
+ * divided into 32 subbands.
+ */
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "mpegaudiodsp.h"
+#include "mpegaudio.h"
+
+#include "mpc.h"
+#include "mpcdata.h"
+
+void ff_mpc_init(void)
+{
+    ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
+}
+
+/**
+ * Process decoded Musepack data and produce PCM
+ */
+static void mpc_synth(MPCContext *c, int16_t **out, int channels)
+{
+    int dither_state = 0;
+    int i, ch;
+
+    for(ch = 0;  ch < channels; ch++){
+        for(i = 0; i < SAMPLES_PER_BAND; i++) {
+            ff_mpa_synth_filter_fixed(&c->mpadsp,
+                                c->synth_buf[ch], &(c->synth_buf_offset[ch]),
+                                ff_mpa_synth_window_fixed, &dither_state,
+                                out[ch] + 32 * i, 1,
+                                c->sb_samples[ch][i]);
+        }
+    }
+}
+
+void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
+                                 int channels)
+{
+    int i, j, ch;
+    Band *bands = c->bands;
+    int off;
+    float mul;
+
+    /* dequantize */
+    memset(c->sb_samples, 0, sizeof(c->sb_samples));
+    off = 0;
+    for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
+        for(ch = 0; ch < 2; ch++){
+            if(bands[i].res[ch]){
+                j = 0;
+                mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0] & 0xFF];
+                for(; j < 12; j++)
+                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
+                mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1] & 0xFF];
+                for(; j < 24; j++)
+                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
+                mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2] & 0xFF];
+                for(; j < 36; j++)
+                    c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
+            }
+        }
+        if(bands[i].msf){
+            int t1, t2;
+            for(j = 0; j < SAMPLES_PER_BAND; j++){
+                t1 = c->sb_samples[0][j][i];
+                t2 = c->sb_samples[1][j][i];
+                c->sb_samples[0][j][i] = t1 + t2;
+                c->sb_samples[1][j][i] = t1 - t2;
+            }
+        }
+    }
+
+    mpc_synth(c, out, channels);
+}