diff options
Diffstat (limited to 'libavcodec/ra144.h')
| -rw-r--r-- | libavcodec/ra144.h | 22 |
1 files changed, 13 insertions, 9 deletions
diff --git a/libavcodec/ra144.h b/libavcodec/ra144.h index 81d6964abc..df747905b3 100644 --- a/libavcodec/ra144.h +++ b/libavcodec/ra144.h @@ -1,21 +1,21 @@ /* * Real Audio 1.0 (14.4K) - * Copyright (c) 2003 the ffmpeg project + * Copyright (c) 2003 The FFmpeg Project * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -25,16 +25,18 @@ #include <stdint.h> #include "lpc.h" #include "audio_frame_queue.h" +#include "audiodsp.h" #define NBLOCKS 4 ///< number of subblocks within a block #define BLOCKSIZE 40 ///< subblock size in 16-bit words #define BUFFERSIZE 146 ///< the size of the adaptive codebook #define FIXED_CB_SIZE 128 ///< size of fixed codebooks -#define FRAMESIZE 20 ///< size of encoded frame +#define FRAME_SIZE 20 ///< size of encoded frame #define LPC_ORDER 10 ///< order of LPC filter typedef struct RA144Context { AVCodecContext *avctx; + AudioDSPContext adsp; LPCContext lpc_ctx; AudioFrameQueue afq; int last_frame; @@ -56,7 +58,9 @@ typedef struct RA144Context { /** Adaptive codebook, its size is two units bigger to avoid a * buffer overflow. */ - uint16_t adapt_cb[146+2]; + int16_t adapt_cb[146+2]; + + DECLARE_ALIGNED(16, int16_t, buffer_a)[FFALIGN(BLOCKSIZE,16)]; } RA144Context; void ff_copy_and_dup(int16_t *target, const int16_t *source, int offset); @@ -68,8 +72,8 @@ unsigned int ff_rms(const int *data); int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy); unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy); -int ff_irms(const int16_t *data); -void ff_subblock_synthesis(RA144Context *ractx, const uint16_t *lpc_coefs, +int ff_irms(AudioDSPContext *adsp, const int16_t *data/*align 16*/); +void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain); |