/* * ALAC audio encoder * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net> * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" #include "internal.h" #include "lpc.h" #include "mathops.h" #define DEFAULT_FRAME_SIZE 4096 #define DEFAULT_SAMPLE_SIZE 16 #define MAX_CHANNELS 8 #define ALAC_EXTRADATA_SIZE 36 #define ALAC_FRAME_HEADER_SIZE 55 #define ALAC_FRAME_FOOTER_SIZE 3 #define ALAC_ESCAPE_CODE 0x1FF #define ALAC_MAX_LPC_ORDER 30 #define DEFAULT_MAX_PRED_ORDER 6 #define DEFAULT_MIN_PRED_ORDER 4 #define ALAC_MAX_LPC_PRECISION 9 #define ALAC_MAX_LPC_SHIFT 9 #define ALAC_CHMODE_LEFT_RIGHT 0 #define ALAC_CHMODE_LEFT_SIDE 1 #define ALAC_CHMODE_RIGHT_SIDE 2 #define ALAC_CHMODE_MID_SIDE 3 typedef struct RiceContext { int history_mult; int initial_history; int k_modifier; int rice_modifier; } RiceContext; typedef struct AlacLPCContext { int lpc_order; int lpc_coeff[ALAC_MAX_LPC_ORDER+1]; int lpc_quant; } AlacLPCContext; typedef struct AlacEncodeContext { int frame_size; /**< current frame size */ int verbatim; /**< current frame verbatim mode flag */ int compression_level; int min_prediction_order; int max_prediction_order; int max_coded_frame_size; int write_sample_size; int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE]; int32_t predictor_buf[DEFAULT_FRAME_SIZE]; int interlacing_shift; int interlacing_leftweight; PutBitContext pbctx; RiceContext rc; AlacLPCContext lpc[MAX_CHANNELS]; LPCContext lpc_ctx; AVCodecContext *avctx; } AlacEncodeContext; static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples) { int ch, i; for (ch = 0; ch < s->avctx->channels; ch++) { int32_t *bptr = s->sample_buf[ch]; const int16_t *sptr = input_samples[ch]; for (i = 0; i < s->frame_size; i++) bptr[i] = sptr[i]; } } static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size) { int divisor, q, r; k = FFMIN(k, s->rc.k_modifier); divisor = (1<<k) - 1; q = x / divisor; r = x % divisor; if (q > 8) { // write escape code and sample value directly put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); put_bits(&s->pbctx, write_sample_size, x); } else { if (q) put_bits(&s->pbctx, q, (1<<q) - 1); put_bits(&s->pbctx, 1, 0); if (k != 1) { if (r > 0) put_bits(&s->pbctx, k, r+1); else put_bits(&s->pbctx, k-1, 0); } } } static void write_frame_header(AlacEncodeContext *s) { int encode_fs = 0; if (s->frame_size < DEFAULT_FRAME_SIZE) encode_fs = 1; put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1 put_bits(&s->pbctx, 16, 0); // Seems to be zero put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim if (encode_fs) put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame } static void calc_predictor_params(AlacEncodeContext *s, int ch) { int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; int shift[MAX_LPC_ORDER]; int opt_order; if (s->compression_level == 1) { s->lpc[ch].lpc_order = 6; s->lpc[ch].lpc_quant = 6; s->lpc[ch].lpc_coeff[0] = 160; s->lpc[ch].lpc_coeff[1] = -190; s->lpc[ch].lpc_coeff[2] = 170; s->lpc[ch].lpc_coeff[3] = -130; s->lpc[ch].lpc_coeff[4] = 80; s->lpc[ch].lpc_coeff[5] = -25; } else { opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch], s->frame_size, s->min_prediction_order, s->max_prediction_order, ALAC_MAX_LPC_PRECISION, coefs, shift, FF_LPC_TYPE_LEVINSON, 0, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); s->lpc[ch].lpc_order = opt_order; s->lpc[ch].lpc_quant = shift[opt_order-1]; memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); } } static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) { int i, best; int32_t lt, rt; uint64_t sum[4]; uint64_t score[4]; /* calculate sum of 2nd order residual for each channel */ sum[0] = sum[1] = sum[2] = sum[3] = 0; for (i = 2; i < n; i++) { lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2]; rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2]; sum[2] += FFABS((lt + rt) >> 1); sum[3] += FFABS(lt - rt); sum[0] += FFABS(lt); sum[1] += FFABS(rt); } /* calculate score for each mode */ score[0] = sum[0] + sum[1]; score[1] = sum[0] + sum[3]; score[2] = sum[1] + sum[3]; score[3] = sum[2] + sum[3]; /* return mode with lowest score */ best = 0; for (i = 1; i < 4; i++) { if (score[i] < score[best]) best = i; } return best; } static void alac_stereo_decorrelation(AlacEncodeContext *s) { int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; int i, mode, n = s->frame_size; int32_t tmp; mode = estimate_stereo_mode(left, right, n); switch (mode) { case ALAC_CHMODE_LEFT_RIGHT: s->interlacing_leftweight = 0; s->interlacing_shift = 0; break; case ALAC_CHMODE_LEFT_SIDE: for (i = 0; i < n; i++) right[i] = left[i] - right[i]; s->interlacing_leftweight = 1; s->interlacing_shift = 0; break; case ALAC_CHMODE_RIGHT_SIDE: for (i = 0; i < n; i++) { tmp = right[i]; right[i] = left[i] - right[i]; left[i] = tmp + (right[i] >> 31); } s->interlacing_leftweight = 1; s->interlacing_shift = 31; break; default: for (i = 0; i < n; i++) { tmp = left[i]; left[i] = (tmp + right[i]) >> 1; right[i] = tmp - right[i]; } s->interlacing_leftweight = 1; s->interlacing_shift = 1; break; } } static void alac_linear_predictor(AlacEncodeContext *s, int ch) { int i; AlacLPCContext lpc = s->lpc[ch]; if (lpc.lpc_order == 31) { s->predictor_buf[0] = s->sample_buf[ch][0]; for (i = 1; i < s->frame_size; i++) { s->predictor_buf[i] = s->sample_buf[ch][i ] - s->sample_buf[ch][i - 1]; } return; } // generalised linear predictor if (lpc.lpc_order > 0) { int32_t *samples = s->sample_buf[ch]; int32_t *residual = s->predictor_buf; // generate warm-up samples residual[0] = samples[0]; for (i = 1; i <= lpc.lpc_order; i++) residual[i] = samples[i] - samples[i-1]; // perform lpc on remaining samples for (i = lpc.lpc_order + 1; i < s->frame_size; i++) { int sum = 1 << (lpc.lpc_quant - 1), res_val, j; for (j = 0; j < lpc.lpc_order; j++) { sum += (samples[lpc.lpc_order-j] - samples[0]) * lpc.lpc_coeff[j]; } sum >>= lpc.lpc_quant; sum += samples[0]; residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum, s->write_sample_size); res_val = residual[i]; if (res_val) { int index = lpc.lpc_order - 1; int neg = (res_val < 0); while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) { int val = samples[0] - samples[lpc.lpc_order - index]; int sign = (val ? FFSIGN(val) : 0); if (neg) sign *= -1; lpc.lpc_coeff[index] -= sign; val *= sign; res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index); index--; } } samples++; } } } static void alac_entropy_coder(AlacEncodeContext *s) { unsigned int history = s->rc.initial_history; int sign_modifier = 0, i, k; int32_t *samples = s->predictor_buf; for (i = 0; i < s->frame_size;) { int x; k = av_log2((history >> 9) + 3); x = -2 * (*samples) -1; x ^= x >> 31; samples++; i++; encode_scalar(s, x - sign_modifier, k, s->write_sample_size); history += x * s->rc.history_mult - ((history * s->rc.history_mult) >> 9); sign_modifier = 0; if (x > 0xFFFF) history = 0xFFFF; if (history < 128 && i < s->frame_size) { unsigned int block_size = 0; k = 7 - av_log2(history) + ((history + 16) >> 6); while (*samples == 0 && i < s->frame_size) { samples++; i++; block_size++; } encode_scalar(s, block_size, k, 16); sign_modifier = (block_size <= 0xFFFF); history = 0; } } } static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples) { int i, j; int prediction_type = 0; PutBitContext *pb = &s->pbctx; init_put_bits(pb, avpkt->data, avpkt->size); if (s->verbatim) { write_frame_header(s); /* samples are channel-interleaved in verbatim mode */ for (i = 0; i < s->frame_size; i++) for (j = 0; j < s->avctx->channels; j++) put_sbits(pb, 16, samples[j][i]); } else { init_sample_buffers(s, samples); write_frame_header(s); if (s->avctx->channels == 2) alac_stereo_decorrelation(s); put_bits(pb, 8, s->interlacing_shift); put_bits(pb, 8, s->interlacing_leftweight); for (i = 0; i < s->avctx->channels; i++) { calc_predictor_params(s, i); put_bits(pb, 4, prediction_type); put_bits(pb, 4, s->lpc[i].lpc_quant); put_bits(pb, 3, s->rc.rice_modifier); put_bits(pb, 5, s->lpc[i].lpc_order); // predictor coeff. table for (j = 0; j < s->lpc[i].lpc_order; j++) put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); } // apply lpc and entropy coding to audio samples for (i = 0; i < s->avctx->channels; i++) { alac_linear_predictor(s, i); // TODO: determine when this will actually help. for now it's not used. if (prediction_type == 15) { // 2nd pass 1st order filter for (j = s->frame_size - 1; j > 0; j--) s->predictor_buf[j] -= s->predictor_buf[j - 1]; } alac_entropy_coder(s); } } put_bits(pb, 3, 7); flush_put_bits(pb); 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; av_freep(&avctx->coded_frame); 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; /* TODO: Correctly implement multi-channel ALAC. It is similar to multi-channel AAC, in that it has a series of single-channel (SCE), channel-pair (CPE), and LFE elements. */ if (avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "only mono or stereo input is currently supported\n"); return AVERROR_PATCHWELCOME; } // 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, DEFAULT_SAMPLE_SIZE); // FIXME: consider wasted_bytes s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; 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, DEFAULT_SAMPLE_SIZE); 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 * DEFAULT_SAMPLE_SIZE); // 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; } avctx->coded_frame = avcodec_alloc_frame(); if (!avctx->coded_frame) { ret = AVERROR(ENOMEM); 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; int16_t **samples = (int16_t **)frame->extended_data; 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, DEFAULT_SAMPLE_SIZE); else max_frame_size = s->max_coded_frame_size; if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size))) return ret; /* use verbatim mode for compression_level 0 */ s->verbatim = !s->compression_level; out_bytes = write_frame(s, avpkt, samples); if (out_bytes > max_frame_size) { /* frame too large. use verbatim mode */ s->verbatim = 1; out_bytes = write_frame(s, avpkt, samples); } avpkt->size = out_bytes; *got_packet_ptr = 1; return 0; } AVCodec ff_alac_encoder = { .name = "alac", .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, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), };