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1
/**
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 * ALAC audio encoder
3
 * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>
4
 *
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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
8
 * 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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 */
21

    
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#include "avcodec.h"
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#include "get_bits.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"
28

    
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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
35

    
36
#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
42

    
43
#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
47

    
48
typedef struct RiceContext {
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    int history_mult;
50
    int initial_history;
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    int k_modifier;
52
    int rice_modifier;
53
} RiceContext;
54

    
55
typedef struct LPCContext {
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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;
59
} LPCContext;
60

    
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typedef struct AlacEncodeContext {
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    int compression_level;
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    int min_prediction_order;
64
    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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    LPCContext lpc[MAX_CHANNELS];
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    DSPContext dspctx;
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    AVCodecContext *avctx;
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} AlacEncodeContext;
77

    
78

    
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static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
80
{
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    int ch, i;
82

    
83
    for(ch=0;ch<s->avctx->channels;ch++) {
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        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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}
91

    
92
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
93
{
94
    int divisor, q, r;
95

    
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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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101
    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);
109

    
110
        if(k != 1) {
111
            if(r > 0)
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                put_bits(&s->pbctx, k, r+1);
113
            else
114
                put_bits(&s->pbctx, k-1, 0);
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        }
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    }
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}
118

    
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static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
120
{
121
    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_bits(&s->pbctx, 32, s->avctx->frame_size);          // No. of samples in the frame
127
}
128

    
129
static void calc_predictor_params(AlacEncodeContext *s, int ch)
130
{
131
    int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
132
    int shift[MAX_LPC_ORDER];
133
    int opt_order;
134

    
135
    opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order,
136
                                   ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
137

    
138
    s->lpc[ch].lpc_order = opt_order;
139
    s->lpc[ch].lpc_quant = shift[opt_order-1];
140
    memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
141
}
142

    
143
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
144
{
145
    int i, best;
146
    int32_t lt, rt;
147
    uint64_t sum[4];
148
    uint64_t score[4];
149

    
150
    /* calculate sum of 2nd order residual for each channel */
151
    sum[0] = sum[1] = sum[2] = sum[3] = 0;
152
    for(i=2; i<n; i++) {
153
        lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
154
        rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
155
        sum[2] += FFABS((lt + rt) >> 1);
156
        sum[3] += FFABS(lt - rt);
157
        sum[0] += FFABS(lt);
158
        sum[1] += FFABS(rt);
159
    }
160

    
161
    /* calculate score for each mode */
162
    score[0] = sum[0] + sum[1];
163
    score[1] = sum[0] + sum[3];
164
    score[2] = sum[1] + sum[3];
165
    score[3] = sum[2] + sum[3];
166

    
167
    /* return mode with lowest score */
168
    best = 0;
169
    for(i=1; i<4; i++) {
170
        if(score[i] < score[best]) {
171
            best = i;
172
        }
173
    }
174
    return best;
175
}
176

    
177
static void alac_stereo_decorrelation(AlacEncodeContext *s)
178
{
179
    int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
180
    int i, mode, n = s->avctx->frame_size;
181
    int32_t tmp;
182

    
183
    mode = estimate_stereo_mode(left, right, n);
184

    
185
    switch(mode)
186
    {
187
        case ALAC_CHMODE_LEFT_RIGHT:
188
            s->interlacing_leftweight = 0;
189
            s->interlacing_shift = 0;
190
            break;
191

    
192
        case ALAC_CHMODE_LEFT_SIDE:
193
            for(i=0; i<n; i++) {
194
                right[i] = left[i] - right[i];
195
            }
196
            s->interlacing_leftweight = 1;
197
            s->interlacing_shift = 0;
198
            break;
199

    
200
        case ALAC_CHMODE_RIGHT_SIDE:
201
            for(i=0; i<n; i++) {
202
                tmp = right[i];
203
                right[i] = left[i] - right[i];
204
                left[i] = tmp + (right[i] >> 31);
205
            }
206
            s->interlacing_leftweight = 1;
207
            s->interlacing_shift = 31;
208
            break;
209

    
210
        default:
211
            for(i=0; i<n; i++) {
212
                tmp = left[i];
213
                left[i] = (tmp + right[i]) >> 1;
214
                right[i] = tmp - right[i];
215
            }
216
            s->interlacing_leftweight = 1;
217
            s->interlacing_shift = 1;
218
            break;
219
    }
220
}
221

    
222
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
223
{
224
    int i;
225
    LPCContext lpc = s->lpc[ch];
226

    
227
    if(lpc.lpc_order == 31) {
228
        s->predictor_buf[0] = s->sample_buf[ch][0];
229

    
230
        for(i=1; i<s->avctx->frame_size; i++)
231
            s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
232

    
233
        return;
234
    }
235

    
236
    // generalised linear predictor
237

    
238
    if(lpc.lpc_order > 0) {
239
        int32_t *samples  = s->sample_buf[ch];
240
        int32_t *residual = s->predictor_buf;
241

    
242
        // generate warm-up samples
243
        residual[0] = samples[0];
244
        for(i=1;i<=lpc.lpc_order;i++)
245
            residual[i] = samples[i] - samples[i-1];
246

    
247
        // perform lpc on remaining samples
248
        for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
249
            int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
250

    
251
            for (j = 0; j < lpc.lpc_order; j++) {
252
                sum += (samples[lpc.lpc_order-j] - samples[0]) *
253
                        lpc.lpc_coeff[j];
254
            }
255

    
256
            sum >>= lpc.lpc_quant;
257
            sum += samples[0];
258
            residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
259
                                      s->write_sample_size);
260
            res_val = residual[i];
261

    
262
            if(res_val) {
263
                int index = lpc.lpc_order - 1;
264
                int neg = (res_val < 0);
265

    
266
                while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
267
                    int val = samples[0] - samples[lpc.lpc_order - index];
268
                    int sign = (val ? FFSIGN(val) : 0);
269

    
270
                    if(neg)
271
                        sign*=-1;
272

    
273
                    lpc.lpc_coeff[index] -= sign;
274
                    val *= sign;
275
                    res_val -= ((val >> lpc.lpc_quant) *
276
                            (lpc.lpc_order - index));
277
                    index--;
278
                }
279
            }
280
            samples++;
281
        }
282
    }
283
}
284

    
285
static void alac_entropy_coder(AlacEncodeContext *s)
286
{
287
    unsigned int history = s->rc.initial_history;
288
    int sign_modifier = 0, i, k;
289
    int32_t *samples = s->predictor_buf;
290

    
291
    for(i=0;i < s->avctx->frame_size;) {
292
        int x;
293

    
294
        k = av_log2((history >> 9) + 3);
295

    
296
        x = -2*(*samples)-1;
297
        x ^= (x>>31);
298

    
299
        samples++;
300
        i++;
301

    
302
        encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
303

    
304
        history += x * s->rc.history_mult
305
                   - ((history * s->rc.history_mult) >> 9);
306

    
307
        sign_modifier = 0;
308
        if(x > 0xFFFF)
309
            history = 0xFFFF;
310

    
311
        if((history < 128) && (i < s->avctx->frame_size)) {
312
            unsigned int block_size = 0;
313

    
314
            k = 7 - av_log2(history) + ((history + 16) >> 6);
315

    
316
            while((*samples == 0) && (i < s->avctx->frame_size)) {
317
                samples++;
318
                i++;
319
                block_size++;
320
            }
321
            encode_scalar(s, block_size, k, 16);
322

    
323
            sign_modifier = (block_size <= 0xFFFF);
324

    
325
            history = 0;
326
        }
327

    
328
    }
329
}
330

    
331
static void write_compressed_frame(AlacEncodeContext *s)
332
{
333
    int i, j;
334

    
335
    if(s->avctx->channels == 2)
336
        alac_stereo_decorrelation(s);
337
    put_bits(&s->pbctx, 8, s->interlacing_shift);
338
    put_bits(&s->pbctx, 8, s->interlacing_leftweight);
339

    
340
    for(i=0;i<s->avctx->channels;i++) {
341

    
342
        calc_predictor_params(s, i);
343

    
344
        put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd
345
        put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
346

    
347
        put_bits(&s->pbctx, 3, s->rc.rice_modifier);
348
        put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
349
        // predictor coeff. table
350
        for(j=0;j<s->lpc[i].lpc_order;j++) {
351
            put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
352
        }
353
    }
354

    
355
    // apply lpc and entropy coding to audio samples
356

    
357
    for(i=0;i<s->avctx->channels;i++) {
358
        alac_linear_predictor(s, i);
359
        alac_entropy_coder(s);
360
    }
361
}
362

    
363
static av_cold int alac_encode_init(AVCodecContext *avctx)
364
{
365
    AlacEncodeContext *s    = avctx->priv_data;
366
    uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
367

    
368
    avctx->frame_size      = DEFAULT_FRAME_SIZE;
369
    avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
370

    
371
    if(avctx->sample_fmt != SAMPLE_FMT_S16) {
372
        av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
373
        return -1;
374
    }
375

    
376
    // Set default compression level
377
    if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
378
        s->compression_level = 1;
379
    else
380
        s->compression_level = av_clip(avctx->compression_level, 0, 1);
381

    
382
    // Initialize default Rice parameters
383
    s->rc.history_mult    = 40;
384
    s->rc.initial_history = 10;
385
    s->rc.k_modifier      = 14;
386
    s->rc.rice_modifier   = 4;
387

    
388
    s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
389
                               avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample)>>3;
390

    
391
    s->write_sample_size  = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
392

    
393
    AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);
394
    AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));
395
    AV_WB32(alac_extradata+12, avctx->frame_size);
396
    AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
397
    AV_WB8 (alac_extradata+21, avctx->channels);
398
    AV_WB32(alac_extradata+24, s->max_coded_frame_size);
399
    AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
400
    AV_WB32(alac_extradata+32, avctx->sample_rate);
401

    
402
    // Set relevant extradata fields
403
    if(s->compression_level > 0) {
404
        AV_WB8(alac_extradata+18, s->rc.history_mult);
405
        AV_WB8(alac_extradata+19, s->rc.initial_history);
406
        AV_WB8(alac_extradata+20, s->rc.k_modifier);
407
    }
408

    
409
    s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
410
    if(avctx->min_prediction_order >= 0) {
411
        if(avctx->min_prediction_order < MIN_LPC_ORDER ||
412
           avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
413
            av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
414
                return -1;
415
        }
416

    
417
        s->min_prediction_order = avctx->min_prediction_order;
418
    }
419

    
420
    s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
421
    if(avctx->max_prediction_order >= 0) {
422
        if(avctx->max_prediction_order < MIN_LPC_ORDER ||
423
           avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
424
            av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
425
                return -1;
426
        }
427

    
428
        s->max_prediction_order = avctx->max_prediction_order;
429
    }
430

    
431
    if(s->max_prediction_order < s->min_prediction_order) {
432
        av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
433
               s->min_prediction_order, s->max_prediction_order);
434
        return -1;
435
    }
436

    
437
    avctx->extradata = alac_extradata;
438
    avctx->extradata_size = ALAC_EXTRADATA_SIZE;
439

    
440
    avctx->coded_frame = avcodec_alloc_frame();
441
    avctx->coded_frame->key_frame = 1;
442

    
443
    s->avctx = avctx;
444
    dsputil_init(&s->dspctx, avctx);
445

    
446
    return 0;
447
}
448

    
449
static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
450
                             int buf_size, void *data)
451
{
452
    AlacEncodeContext *s = avctx->priv_data;
453
    PutBitContext *pb = &s->pbctx;
454
    int i, out_bytes, verbatim_flag = 0;
455

    
456
    if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
457
        av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
458
        return -1;
459
    }
460

    
461
    if(buf_size < 2*s->max_coded_frame_size) {
462
        av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
463
        return -1;
464
    }
465

    
466
verbatim:
467
    init_put_bits(pb, frame, buf_size);
468

    
469
    if((s->compression_level == 0) || verbatim_flag) {
470
        // Verbatim mode
471
        int16_t *samples = data;
472
        write_frame_header(s, 1);
473
        for(i=0; i<avctx->frame_size*avctx->channels; i++) {
474
            put_sbits(pb, 16, *samples++);
475
        }
476
    } else {
477
        init_sample_buffers(s, data);
478
        write_frame_header(s, 0);
479
        write_compressed_frame(s);
480
    }
481

    
482
    put_bits(pb, 3, 7);
483
    flush_put_bits(pb);
484
    out_bytes = put_bits_count(pb) >> 3;
485

    
486
    if(out_bytes > s->max_coded_frame_size) {
487
        /* frame too large. use verbatim mode */
488
        if(verbatim_flag || (s->compression_level == 0)) {
489
            /* still too large. must be an error. */
490
            av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
491
            return -1;
492
        }
493
        verbatim_flag = 1;
494
        goto verbatim;
495
    }
496

    
497
    return out_bytes;
498
}
499

    
500
static av_cold int alac_encode_close(AVCodecContext *avctx)
501
{
502
    av_freep(&avctx->extradata);
503
    avctx->extradata_size = 0;
504
    av_freep(&avctx->coded_frame);
505
    return 0;
506
}
507

    
508
AVCodec alac_encoder = {
509
    "alac",
510
    CODEC_TYPE_AUDIO,
511
    CODEC_ID_ALAC,
512
    sizeof(AlacEncodeContext),
513
    alac_encode_init,
514
    alac_encode_frame,
515
    alac_encode_close,
516
    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
517
    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
518
};