view aacenc.h @ 10273:38147f0f94cc libavcodec

Specify maximum sample rate of MLP by defining the factor relative to 48000 instead of directly. This makes clear that the code assumes the maximum sample rate to be a multiple of 48000 and also removes the division from the MAX_BLOCKSIZE macros, which causes an issue with the Solaris assembler where "/" is a comment marker unless the --divide option is used.

/*
 * AAC encoder
 * Copyright (C) 2008 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
 */

#ifndef AVCODEC_AACENC_H
#define AVCODEC_AACENC_H

#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"

#include "aac.h"

#include "psymodel.h"

struct AACEncContext;

typedef struct AACCoefficientsEncoder {
    void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
                                  SingleChannelElement *sce, const float lambda);
    void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda);
    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
                                     int scale_idx, int cb, const float lambda);
    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
} AACCoefficientsEncoder;

extern AACCoefficientsEncoder ff_aac_coders[];

/**
 * AAC encoder context
 */
typedef struct AACEncContext {
    PutBitContext pb;
    FFTContext mdct1024;                         ///< long (1024 samples) frame transform context
    FFTContext mdct128;                          ///< short (128 samples) frame transform context
    DSPContext  dsp;
    DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
    int16_t* samples;                            ///< saved preprocessed input

    int samplerate_index;                        ///< MPEG-4 samplerate index

    ChannelElement *cpe;                         ///< channel elements
    FFPsyContext psy;
    struct FFPsyPreprocessContext* psypp;
    AACCoefficientsEncoder *coder;
    int cur_channel;
    int last_frame;
    float lambda;
    DECLARE_ALIGNED_16(int,   qcoefs[96][2]);    ///< quantized coefficients
    DECLARE_ALIGNED_16(float, scoefs[1024]);     ///< scaled coefficients
} AACEncContext;

#endif /* AVCODEC_AACENC_H */