mirror of
https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-24 03:39:45 +00:00
662 lines
21 KiB
C
662 lines
21 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 "internal.h"
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#include "lpc.h"
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#include "mathops.h"
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#include "alac_data.h"
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#define DEFAULT_FRAME_SIZE 4096
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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 frame_size; /**< current frame size */
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int verbatim; /**< current frame verbatim mode flag */
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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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int extra_bits;
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int32_t sample_buf[2][DEFAULT_FRAME_SIZE];
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int32_t predictor_buf[2][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[2];
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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, int channels,
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uint8_t const *samples[2])
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{
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int ch, i;
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int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 -
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s->avctx->bits_per_raw_sample;
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#define COPY_SAMPLES(type) do { \
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for (ch = 0; ch < channels; ch++) { \
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int32_t *bptr = s->sample_buf[ch]; \
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const type *sptr = (const type *)samples[ch]; \
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for (i = 0; i < s->frame_size; i++) \
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bptr[i] = sptr[i] >> shift; \
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} \
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} while (0)
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if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P)
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COPY_SAMPLES(int32_t);
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else
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COPY_SAMPLES(int16_t);
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}
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static void encode_scalar(AlacEncodeContext *s, int x,
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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_element_header(AlacEncodeContext *s,
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enum AlacRawDataBlockType element,
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int instance)
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{
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int encode_fs = 0;
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if (s->frame_size < DEFAULT_FRAME_SIZE)
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encode_fs = 1;
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put_bits(&s->pbctx, 3, element); // element type
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put_bits(&s->pbctx, 4, instance); // element instance
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put_bits(&s->pbctx, 12, 0); // unused header bits
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put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header
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put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit)
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put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim
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if (encode_fs)
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put_bits32(&s->pbctx, s->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->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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FF_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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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->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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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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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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int32_t *residual = s->predictor_buf[ch];
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if (lpc.lpc_order == 31) {
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residual[0] = s->sample_buf[ch][0];
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for (i = 1; i < s->frame_size; i++) {
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residual[i] = s->sample_buf[ch][i ] -
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s->sample_buf[ch][i - 1];
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}
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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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// 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] = sign_extend(samples[i] - samples[i-1], s->write_sample_size);
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// perform lpc on remaining samples
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for (i = lpc.lpc_order + 1; i < s->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) * (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, int ch)
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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[ch];
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for (i = 0; i < s->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->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->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_element(AlacEncodeContext *s,
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enum AlacRawDataBlockType element, int instance,
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const uint8_t *samples0, const uint8_t *samples1)
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{
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uint8_t const *samples[2] = { samples0, samples1 };
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int i, j, channels;
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int prediction_type = 0;
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PutBitContext *pb = &s->pbctx;
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channels = element == TYPE_CPE ? 2 : 1;
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if (s->verbatim) {
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write_element_header(s, element, instance);
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/* samples are channel-interleaved in verbatim mode */
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if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
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int shift = 32 - s->avctx->bits_per_raw_sample;
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int32_t const *samples_s32[2] = { (const int32_t *)samples0,
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(const int32_t *)samples1 };
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for (i = 0; i < s->frame_size; i++)
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for (j = 0; j < channels; j++)
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put_sbits(pb, s->avctx->bits_per_raw_sample,
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samples_s32[j][i] >> shift);
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} else {
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int16_t const *samples_s16[2] = { (const int16_t *)samples0,
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(const int16_t *)samples1 };
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for (i = 0; i < s->frame_size; i++)
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for (j = 0; j < channels; j++)
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put_sbits(pb, s->avctx->bits_per_raw_sample,
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samples_s16[j][i]);
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}
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} else {
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s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits +
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channels - 1;
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init_sample_buffers(s, channels, samples);
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write_element_header(s, element, instance);
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// extract extra bits if needed
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if (s->extra_bits) {
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uint32_t mask = (1 << s->extra_bits) - 1;
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for (j = 0; j < channels; j++) {
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int32_t *extra = s->predictor_buf[j];
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int32_t *smp = s->sample_buf[j];
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for (i = 0; i < s->frame_size; i++) {
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extra[i] = smp[i] & mask;
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smp[i] >>= s->extra_bits;
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}
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}
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}
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if (channels == 2)
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alac_stereo_decorrelation(s);
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else
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s->interlacing_shift = s->interlacing_leftweight = 0;
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put_bits(pb, 8, s->interlacing_shift);
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put_bits(pb, 8, s->interlacing_leftweight);
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for (i = 0; i < channels; i++) {
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calc_predictor_params(s, i);
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put_bits(pb, 4, prediction_type);
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put_bits(pb, 4, s->lpc[i].lpc_quant);
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put_bits(pb, 3, s->rc.rice_modifier);
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put_bits(pb, 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(pb, 16, s->lpc[i].lpc_coeff[j]);
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}
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// write extra bits if needed
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if (s->extra_bits) {
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for (i = 0; i < s->frame_size; i++) {
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for (j = 0; j < channels; j++) {
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put_bits(pb, s->extra_bits, s->predictor_buf[j][i]);
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}
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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 < channels; i++) {
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alac_linear_predictor(s, i);
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// TODO: determine when this will actually help. for now it's not used.
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if (prediction_type == 15) {
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// 2nd pass 1st order filter
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int32_t *residual = s->predictor_buf[i];
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for (j = s->frame_size - 1; j > 0; j--)
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residual[j] -= residual[j - 1];
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}
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alac_entropy_coder(s, i);
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}
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}
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}
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static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
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uint8_t * const *samples)
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{
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PutBitContext *pb = &s->pbctx;
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const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1];
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const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1];
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int ch, element, sce, cpe;
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init_put_bits(pb, avpkt->data, avpkt->size);
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ch = element = sce = cpe = 0;
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while (ch < s->avctx->channels) {
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if (ch_elements[element] == TYPE_CPE) {
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write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]],
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samples[ch_map[ch + 1]]);
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cpe++;
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ch += 2;
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} else {
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write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL);
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sce++;
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ch++;
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}
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element++;
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}
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put_bits(pb, 3, TYPE_END);
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flush_put_bits(pb);
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|
|
return put_bits_count(pb) >> 3;
|
|
}
|
|
|
|
static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps)
|
|
{
|
|
int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE);
|
|
return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8;
|
|
}
|
|
|
|
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;
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int alac_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AlacEncodeContext *s = avctx->priv_data;
|
|
int ret;
|
|
uint8_t *alac_extradata;
|
|
|
|
avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
|
|
|
|
if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) {
|
|
if (avctx->bits_per_raw_sample != 24)
|
|
av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n");
|
|
avctx->bits_per_raw_sample = 24;
|
|
} else {
|
|
avctx->bits_per_raw_sample = 16;
|
|
s->extra_bits = 0;
|
|
}
|
|
|
|
// Set default compression level
|
|
if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
|
|
s->compression_level = 2;
|
|
else
|
|
s->compression_level = av_clip(avctx->compression_level, 0, 2);
|
|
|
|
// Initialize default Rice parameters
|
|
s->rc.history_mult = 40;
|
|
s->rc.initial_history = 10;
|
|
s->rc.k_modifier = 14;
|
|
s->rc.rice_modifier = 4;
|
|
|
|
s->max_coded_frame_size = get_max_frame_size(avctx->frame_size,
|
|
avctx->channels,
|
|
avctx->bits_per_raw_sample);
|
|
|
|
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
|
|
if (!avctx->extradata) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto error;
|
|
}
|
|
avctx->extradata_size = ALAC_EXTRADATA_SIZE;
|
|
|
|
alac_extradata = avctx->extradata;
|
|
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
|
|
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
|
|
AV_WB32(alac_extradata+12, avctx->frame_size);
|
|
AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample);
|
|
AV_WB8 (alac_extradata+21, avctx->channels);
|
|
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
|
|
AV_WB32(alac_extradata+28,
|
|
avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate
|
|
AV_WB32(alac_extradata+32, avctx->sample_rate);
|
|
|
|
// Set relevant extradata fields
|
|
if (s->compression_level > 0) {
|
|
AV_WB8(alac_extradata+18, s->rc.history_mult);
|
|
AV_WB8(alac_extradata+19, s->rc.initial_history);
|
|
AV_WB8(alac_extradata+20, s->rc.k_modifier);
|
|
}
|
|
|
|
s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
|
|
if (avctx->min_prediction_order >= 0) {
|
|
if (avctx->min_prediction_order < MIN_LPC_ORDER ||
|
|
avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n",
|
|
avctx->min_prediction_order);
|
|
ret = AVERROR(EINVAL);
|
|
goto error;
|
|
}
|
|
|
|
s->min_prediction_order = avctx->min_prediction_order;
|
|
}
|
|
|
|
s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
|
|
if (avctx->max_prediction_order >= 0) {
|
|
if (avctx->max_prediction_order < MIN_LPC_ORDER ||
|
|
avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n",
|
|
avctx->max_prediction_order);
|
|
ret = AVERROR(EINVAL);
|
|
goto error;
|
|
}
|
|
|
|
s->max_prediction_order = avctx->max_prediction_order;
|
|
}
|
|
|
|
if (s->max_prediction_order < s->min_prediction_order) {
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"invalid prediction orders: min=%d max=%d\n",
|
|
s->min_prediction_order, s->max_prediction_order);
|
|
ret = AVERROR(EINVAL);
|
|
goto error;
|
|
}
|
|
|
|
s->avctx = avctx;
|
|
|
|
if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size,
|
|
s->max_prediction_order,
|
|
FF_LPC_TYPE_LEVINSON)) < 0) {
|
|
goto error;
|
|
}
|
|
|
|
return 0;
|
|
error:
|
|
alac_encode_close(avctx);
|
|
return ret;
|
|
}
|
|
|
|
static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
AlacEncodeContext *s = avctx->priv_data;
|
|
int out_bytes, max_frame_size, ret;
|
|
|
|
s->frame_size = frame->nb_samples;
|
|
|
|
if (frame->nb_samples < DEFAULT_FRAME_SIZE)
|
|
max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
|
|
avctx->bits_per_raw_sample);
|
|
else
|
|
max_frame_size = s->max_coded_frame_size;
|
|
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0)
|
|
return ret;
|
|
|
|
/* use verbatim mode for compression_level 0 */
|
|
if (s->compression_level) {
|
|
s->verbatim = 0;
|
|
s->extra_bits = avctx->bits_per_raw_sample - 16;
|
|
} else {
|
|
s->verbatim = 1;
|
|
s->extra_bits = 0;
|
|
}
|
|
|
|
out_bytes = write_frame(s, avpkt, frame->extended_data);
|
|
|
|
if (out_bytes > max_frame_size) {
|
|
/* frame too large. use verbatim mode */
|
|
s->verbatim = 1;
|
|
s->extra_bits = 0;
|
|
out_bytes = write_frame(s, avpkt, frame->extended_data);
|
|
}
|
|
|
|
avpkt->size = out_bytes;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_alac_encoder = {
|
|
.name = "alac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_ALAC,
|
|
.priv_data_size = sizeof(AlacEncodeContext),
|
|
.init = alac_encode_init,
|
|
.encode2 = alac_encode_frame,
|
|
.close = alac_encode_close,
|
|
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
|
|
.channel_layouts = ff_alac_channel_layouts,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
|
|
AV_SAMPLE_FMT_S16P,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|