psymodel: Remove the single channel analysis function

[ffmpeg] / libavcodec / ac3dec.h

diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h
index a5b1111123d5c63822a33f011ba3ee308a16f1cf..590bee6b6d73c974d7d48f34703f275c37a2cb24 100644 (file)
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -2,20 +2,20 @@
  * Common code between the AC-3 and E-AC-3 decoders
  * Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -52,16 +52,12 @@
 
 #include "libavutil/lfg.h"
 #include "ac3.h"
+#include "ac3dsp.h"
 #include "get_bits.h"
 #include "dsputil.h"
 #include "fft.h"
 #include "fmtconvert.h"
 
-/* override ac3.h to include coupling channel */
-#undef AC3_MAX_CHANNELS
-#define AC3_MAX_CHANNELS 7
-#define CPL_CH 0
-
 #define AC3_OUTPUT_LFEON  8
 
 #define SPX_MAX_BANDS    17
@@ -79,6 +75,7 @@ typedef struct {
     int bit_rate;                           ///< stream bit rate, in bits-per-second
     int sample_rate;                        ///< sample frequency, in Hz
     int num_blocks;                         ///< number of audio blocks
+    int bitstream_mode;                     ///< bitstream mode                         (bsmod)
     int channel_mode;                       ///< channel mode                           (acmod)
     int channel_layout;                     ///< channel layout
     int lfe_on;                             ///< lfe channel in use
@@ -191,17 +188,18 @@ typedef struct {
 
 ///@defgroup opt optimization
     DSPContext dsp;                         ///< for optimization
+    AC3DSPContext ac3dsp;
     FmtConvertContext fmt_conv;             ///< optimized conversion functions
     float mul_bias;                         ///< scaling for float_to_int16 conversion
 ///@}
 
 ///@defgroup arrays aligned arrays
-    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///> fixed-point transform coefficients
-    DECLARE_ALIGNED(16, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
-    DECLARE_ALIGNED(16, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
-    DECLARE_ALIGNED(16, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
-    DECLARE_ALIGNED(16, float, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
-    DECLARE_ALIGNED(16, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
+    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///< fixed-point transform coefficients
+    DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
+    DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
+    DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
+    DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
+    DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
 ///@}
 } AC3DecodeContext;