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authorKostya Shishkov <kostya.shishkov@gmail.com>2007-11-02 06:40:42 +0000
committerKostya Shishkov <kostya.shishkov@gmail.com>2007-11-02 06:40:42 +0000
commit24e649afe3e47ce33729a5ccd83a7993c70f50bb (patch)
treeba4f50778a2fa46a654d98e901306916140b35af /libavcodec/mpc.h
parent76c443c6d4c6b414eea6b0826ff6a0361795234d (diff)
Split Musepack decoder into SV7 decoder and synth core
(for upcoming SV8 decoder) Originally committed as revision 10905 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/mpc.h')
-rw-r--r--libavcodec/mpc.h80
1 files changed, 80 insertions, 0 deletions
diff --git a/libavcodec/mpc.h b/libavcodec/mpc.h
new file mode 100644
index 0000000000..ce438e3af9
--- /dev/null
+++ b/libavcodec/mpc.h
@@ -0,0 +1,80 @@
+/*
+ * Musepack decoder
+ * 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 mpc.h Musepack decoder
+ * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
+ * divided into 32 subbands.
+ */
+
+#ifndef FFMPEG_MPC_H
+#define FFMPEG_MPC_H
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+#include "random.h"
+
+#ifdef CONFIG_MPEGAUDIO_HP
+#define USE_HIGHPRECISION
+#endif
+#include "mpegaudio.h"
+
+#include "mpcdata.h"
+
+#define BANDS 32
+#define SAMPLES_PER_BAND 36
+#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
+
+/** Subband structure - hold all variables for each subband */
+typedef struct {
+ int msf; ///< mid-stereo flag
+ int res[2];
+ int scfi[2];
+ int scf_idx[2][3];
+ int Q[2];
+}Band;
+
+typedef struct {
+ DSPContext dsp;
+ GetBitContext gb;
+ int IS, MSS, gapless;
+ int lastframelen;
+ int maxbands, last_max_band;
+ int last_bits_used;
+ int oldDSCF[2][BANDS];
+ Band bands[BANDS];
+ int Q[2][MPC_FRAME_SIZE];
+ int cur_frame, frames;
+ uint8_t *bits;
+ int buf_size;
+ AVRandomState rnd;
+ int frames_to_skip;
+ /* for synthesis */
+ DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
+ int synth_buf_offset[MPA_MAX_CHANNELS];
+ DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
+} MPCContext;
+
+extern void ff_mpc_init();
+extern void ff_mpc_dequantize_and_synth(MPCContext *c, int maxband, void *dst);
+
+#endif /* FFMPEG_MPC_H */