mirror of
https://github.com/xenia-project/FFmpeg.git
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2912e87a6c
Signed-off-by: Mans Rullgard <mans@mansr.com>
538 lines
16 KiB
C
538 lines
16 KiB
C
/**
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* ALAC audio encoder
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* Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "put_bits.h"
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#include "dsputil.h"
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#include "lpc.h"
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#include "mathops.h"
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#define DEFAULT_FRAME_SIZE 4096
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#define DEFAULT_SAMPLE_SIZE 16
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#define MAX_CHANNELS 8
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#define ALAC_EXTRADATA_SIZE 36
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#define ALAC_FRAME_HEADER_SIZE 55
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#define ALAC_FRAME_FOOTER_SIZE 3
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#define ALAC_ESCAPE_CODE 0x1FF
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#define ALAC_MAX_LPC_ORDER 30
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#define DEFAULT_MAX_PRED_ORDER 6
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#define DEFAULT_MIN_PRED_ORDER 4
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#define ALAC_MAX_LPC_PRECISION 9
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#define ALAC_MAX_LPC_SHIFT 9
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#define ALAC_CHMODE_LEFT_RIGHT 0
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#define ALAC_CHMODE_LEFT_SIDE 1
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#define ALAC_CHMODE_RIGHT_SIDE 2
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#define ALAC_CHMODE_MID_SIDE 3
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typedef struct RiceContext {
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int history_mult;
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int initial_history;
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int k_modifier;
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int rice_modifier;
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} RiceContext;
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typedef struct AlacLPCContext {
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int lpc_order;
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int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
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int lpc_quant;
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} AlacLPCContext;
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typedef struct AlacEncodeContext {
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int compression_level;
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int min_prediction_order;
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int max_prediction_order;
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int max_coded_frame_size;
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int write_sample_size;
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int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
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int32_t predictor_buf[DEFAULT_FRAME_SIZE];
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int interlacing_shift;
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int interlacing_leftweight;
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PutBitContext pbctx;
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RiceContext rc;
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AlacLPCContext lpc[MAX_CHANNELS];
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LPCContext lpc_ctx;
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AVCodecContext *avctx;
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} AlacEncodeContext;
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static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples)
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{
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int ch, i;
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for(ch=0;ch<s->avctx->channels;ch++) {
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const int16_t *sptr = input_samples + ch;
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for(i=0;i<s->avctx->frame_size;i++) {
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s->sample_buf[ch][i] = *sptr;
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sptr += s->avctx->channels;
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}
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}
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}
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static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
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{
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int divisor, q, r;
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k = FFMIN(k, s->rc.k_modifier);
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divisor = (1<<k) - 1;
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q = x / divisor;
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r = x % divisor;
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if(q > 8) {
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// write escape code and sample value directly
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put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
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put_bits(&s->pbctx, write_sample_size, x);
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} else {
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if(q)
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put_bits(&s->pbctx, q, (1<<q) - 1);
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put_bits(&s->pbctx, 1, 0);
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if(k != 1) {
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if(r > 0)
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put_bits(&s->pbctx, k, r+1);
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else
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put_bits(&s->pbctx, k-1, 0);
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}
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}
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}
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static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
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{
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put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
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put_bits(&s->pbctx, 16, 0); // Seems to be zero
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put_bits(&s->pbctx, 1, 1); // Sample count is in the header
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put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
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put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
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put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
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}
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static void calc_predictor_params(AlacEncodeContext *s, int ch)
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{
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int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
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int shift[MAX_LPC_ORDER];
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int opt_order;
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if (s->compression_level == 1) {
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s->lpc[ch].lpc_order = 6;
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s->lpc[ch].lpc_quant = 6;
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s->lpc[ch].lpc_coeff[0] = 160;
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s->lpc[ch].lpc_coeff[1] = -190;
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s->lpc[ch].lpc_coeff[2] = 170;
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s->lpc[ch].lpc_coeff[3] = -130;
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s->lpc[ch].lpc_coeff[4] = 80;
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s->lpc[ch].lpc_coeff[5] = -25;
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} else {
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opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch],
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s->avctx->frame_size,
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s->min_prediction_order,
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s->max_prediction_order,
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ALAC_MAX_LPC_PRECISION, coefs, shift,
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AV_LPC_TYPE_LEVINSON, 0,
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ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
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s->lpc[ch].lpc_order = opt_order;
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s->lpc[ch].lpc_quant = shift[opt_order-1];
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memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
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}
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}
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static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
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{
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int i, best;
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int32_t lt, rt;
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uint64_t sum[4];
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uint64_t score[4];
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/* calculate sum of 2nd order residual for each channel */
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sum[0] = sum[1] = sum[2] = sum[3] = 0;
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for(i=2; i<n; i++) {
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lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
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rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
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sum[2] += FFABS((lt + rt) >> 1);
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sum[3] += FFABS(lt - rt);
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sum[0] += FFABS(lt);
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sum[1] += FFABS(rt);
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}
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/* calculate score for each mode */
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score[0] = sum[0] + sum[1];
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score[1] = sum[0] + sum[3];
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score[2] = sum[1] + sum[3];
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score[3] = sum[2] + sum[3];
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/* return mode with lowest score */
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best = 0;
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for(i=1; i<4; i++) {
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if(score[i] < score[best]) {
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best = i;
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}
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}
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return best;
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}
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static void alac_stereo_decorrelation(AlacEncodeContext *s)
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{
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int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
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int i, mode, n = s->avctx->frame_size;
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int32_t tmp;
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mode = estimate_stereo_mode(left, right, n);
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switch(mode)
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{
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case ALAC_CHMODE_LEFT_RIGHT:
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s->interlacing_leftweight = 0;
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s->interlacing_shift = 0;
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break;
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case ALAC_CHMODE_LEFT_SIDE:
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for(i=0; i<n; i++) {
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right[i] = left[i] - right[i];
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}
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s->interlacing_leftweight = 1;
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s->interlacing_shift = 0;
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break;
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case ALAC_CHMODE_RIGHT_SIDE:
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for(i=0; i<n; i++) {
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tmp = right[i];
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right[i] = left[i] - right[i];
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left[i] = tmp + (right[i] >> 31);
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}
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s->interlacing_leftweight = 1;
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s->interlacing_shift = 31;
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break;
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default:
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for(i=0; i<n; i++) {
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tmp = left[i];
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left[i] = (tmp + right[i]) >> 1;
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right[i] = tmp - right[i];
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}
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s->interlacing_leftweight = 1;
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s->interlacing_shift = 1;
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break;
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}
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}
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static void alac_linear_predictor(AlacEncodeContext *s, int ch)
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{
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int i;
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AlacLPCContext lpc = s->lpc[ch];
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if(lpc.lpc_order == 31) {
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s->predictor_buf[0] = s->sample_buf[ch][0];
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for(i=1; i<s->avctx->frame_size; i++)
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s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
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return;
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}
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// generalised linear predictor
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if(lpc.lpc_order > 0) {
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int32_t *samples = s->sample_buf[ch];
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int32_t *residual = s->predictor_buf;
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// generate warm-up samples
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residual[0] = samples[0];
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for(i=1;i<=lpc.lpc_order;i++)
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residual[i] = samples[i] - samples[i-1];
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// perform lpc on remaining samples
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for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
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int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
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for (j = 0; j < lpc.lpc_order; j++) {
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sum += (samples[lpc.lpc_order-j] - samples[0]) *
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lpc.lpc_coeff[j];
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}
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sum >>= lpc.lpc_quant;
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sum += samples[0];
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residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
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s->write_sample_size);
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res_val = residual[i];
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if(res_val) {
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int index = lpc.lpc_order - 1;
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int neg = (res_val < 0);
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while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
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int val = samples[0] - samples[lpc.lpc_order - index];
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int sign = (val ? FFSIGN(val) : 0);
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if(neg)
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sign*=-1;
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lpc.lpc_coeff[index] -= sign;
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val *= sign;
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res_val -= ((val >> lpc.lpc_quant) *
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(lpc.lpc_order - index));
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index--;
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}
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}
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samples++;
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}
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}
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}
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static void alac_entropy_coder(AlacEncodeContext *s)
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{
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unsigned int history = s->rc.initial_history;
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int sign_modifier = 0, i, k;
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int32_t *samples = s->predictor_buf;
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for(i=0;i < s->avctx->frame_size;) {
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int x;
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k = av_log2((history >> 9) + 3);
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x = -2*(*samples)-1;
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x ^= (x>>31);
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samples++;
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i++;
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encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
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history += x * s->rc.history_mult
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- ((history * s->rc.history_mult) >> 9);
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sign_modifier = 0;
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if(x > 0xFFFF)
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history = 0xFFFF;
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if((history < 128) && (i < s->avctx->frame_size)) {
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unsigned int block_size = 0;
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k = 7 - av_log2(history) + ((history + 16) >> 6);
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while((*samples == 0) && (i < s->avctx->frame_size)) {
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samples++;
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i++;
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block_size++;
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}
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encode_scalar(s, block_size, k, 16);
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sign_modifier = (block_size <= 0xFFFF);
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history = 0;
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}
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}
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}
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static void write_compressed_frame(AlacEncodeContext *s)
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{
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int i, j;
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if(s->avctx->channels == 2)
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alac_stereo_decorrelation(s);
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put_bits(&s->pbctx, 8, s->interlacing_shift);
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put_bits(&s->pbctx, 8, s->interlacing_leftweight);
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for(i=0;i<s->avctx->channels;i++) {
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calc_predictor_params(s, i);
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put_bits(&s->pbctx, 4, 0); // prediction type : currently only type 0 has been RE'd
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put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
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put_bits(&s->pbctx, 3, s->rc.rice_modifier);
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put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
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// predictor coeff. table
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for(j=0;j<s->lpc[i].lpc_order;j++) {
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put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
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}
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}
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// apply lpc and entropy coding to audio samples
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for(i=0;i<s->avctx->channels;i++) {
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alac_linear_predictor(s, i);
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alac_entropy_coder(s);
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}
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}
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static av_cold int alac_encode_init(AVCodecContext *avctx)
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{
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AlacEncodeContext *s = avctx->priv_data;
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int ret;
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uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
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avctx->frame_size = DEFAULT_FRAME_SIZE;
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avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
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if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
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av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
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return -1;
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}
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// Set default compression level
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if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
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s->compression_level = 2;
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else
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s->compression_level = av_clip(avctx->compression_level, 0, 2);
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// Initialize default Rice parameters
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s->rc.history_mult = 40;
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s->rc.initial_history = 10;
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s->rc.k_modifier = 14;
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s->rc.rice_modifier = 4;
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s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
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s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
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AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
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AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
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AV_WB32(alac_extradata+12, avctx->frame_size);
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AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
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AV_WB8 (alac_extradata+21, avctx->channels);
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AV_WB32(alac_extradata+24, s->max_coded_frame_size);
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AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
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AV_WB32(alac_extradata+32, avctx->sample_rate);
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// Set relevant extradata fields
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if(s->compression_level > 0) {
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AV_WB8(alac_extradata+18, s->rc.history_mult);
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AV_WB8(alac_extradata+19, s->rc.initial_history);
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AV_WB8(alac_extradata+20, s->rc.k_modifier);
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}
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s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
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if(avctx->min_prediction_order >= 0) {
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if(avctx->min_prediction_order < MIN_LPC_ORDER ||
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avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
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av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
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return -1;
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}
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s->min_prediction_order = avctx->min_prediction_order;
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}
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s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
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if(avctx->max_prediction_order >= 0) {
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if(avctx->max_prediction_order < MIN_LPC_ORDER ||
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avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
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av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
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return -1;
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}
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s->max_prediction_order = avctx->max_prediction_order;
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}
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if(s->max_prediction_order < s->min_prediction_order) {
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av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
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s->min_prediction_order, s->max_prediction_order);
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return -1;
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}
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avctx->extradata = alac_extradata;
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avctx->extradata_size = ALAC_EXTRADATA_SIZE;
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avctx->coded_frame = avcodec_alloc_frame();
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avctx->coded_frame->key_frame = 1;
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s->avctx = avctx;
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ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order,
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AV_LPC_TYPE_LEVINSON);
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return ret;
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}
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static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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int buf_size, void *data)
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{
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AlacEncodeContext *s = avctx->priv_data;
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PutBitContext *pb = &s->pbctx;
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int i, out_bytes, verbatim_flag = 0;
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if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
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av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
|
|
return -1;
|
|
}
|
|
|
|
if(buf_size < 2*s->max_coded_frame_size) {
|
|
av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
|
|
return -1;
|
|
}
|
|
|
|
verbatim:
|
|
init_put_bits(pb, frame, buf_size);
|
|
|
|
if((s->compression_level == 0) || verbatim_flag) {
|
|
// Verbatim mode
|
|
const int16_t *samples = data;
|
|
write_frame_header(s, 1);
|
|
for(i=0; i<avctx->frame_size*avctx->channels; i++) {
|
|
put_sbits(pb, 16, *samples++);
|
|
}
|
|
} else {
|
|
init_sample_buffers(s, data);
|
|
write_frame_header(s, 0);
|
|
write_compressed_frame(s);
|
|
}
|
|
|
|
put_bits(pb, 3, 7);
|
|
flush_put_bits(pb);
|
|
out_bytes = put_bits_count(pb) >> 3;
|
|
|
|
if(out_bytes > s->max_coded_frame_size) {
|
|
/* frame too large. use verbatim mode */
|
|
if(verbatim_flag || (s->compression_level == 0)) {
|
|
/* still too large. must be an error. */
|
|
av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
|
|
return -1;
|
|
}
|
|
verbatim_flag = 1;
|
|
goto verbatim;
|
|
}
|
|
|
|
return out_bytes;
|
|
}
|
|
|
|
static av_cold int alac_encode_close(AVCodecContext *avctx)
|
|
{
|
|
AlacEncodeContext *s = avctx->priv_data;
|
|
ff_lpc_end(&s->lpc_ctx);
|
|
av_freep(&avctx->extradata);
|
|
avctx->extradata_size = 0;
|
|
av_freep(&avctx->coded_frame);
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_alac_encoder = {
|
|
"alac",
|
|
AVMEDIA_TYPE_AUDIO,
|
|
CODEC_ID_ALAC,
|
|
sizeof(AlacEncodeContext),
|
|
alac_encode_init,
|
|
alac_encode_frame,
|
|
alac_encode_close,
|
|
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE},
|
|
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
|
};
|