diff options
Diffstat (limited to 'ffmpeg/libavcodec/alacenc.c')
| -rw-r--r-- | ffmpeg/libavcodec/alacenc.c | 656 |
1 files changed, 656 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/alacenc.c b/ffmpeg/libavcodec/alacenc.c new file mode 100644 index 0000000..4ee558c --- /dev/null +++ b/ffmpeg/libavcodec/alacenc.c @@ -0,0 +1,656 @@ +/* + * 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 "internal.h" +#include "lpc.h" +#include "mathops.h" +#include "alac_data.h" + +#define DEFAULT_FRAME_SIZE 4096 +#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; + int extra_bits; + int32_t sample_buf[2][DEFAULT_FRAME_SIZE]; + int32_t predictor_buf[DEFAULT_FRAME_SIZE]; + int interlacing_shift; + int interlacing_leftweight; + PutBitContext pbctx; + RiceContext rc; + AlacLPCContext lpc[2]; + LPCContext lpc_ctx; + AVCodecContext *avctx; +} AlacEncodeContext; + + +static void init_sample_buffers(AlacEncodeContext *s, int channels, + uint8_t const *samples[2]) +{ + int ch, i; + int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 - + s->avctx->bits_per_raw_sample; + +#define COPY_SAMPLES(type) do { \ + for (ch = 0; ch < channels; ch++) { \ + int32_t *bptr = s->sample_buf[ch]; \ + const type *sptr = (const type *)samples[ch]; \ + for (i = 0; i < s->frame_size; i++) \ + bptr[i] = sptr[i] >> shift; \ + } \ + } while (0) + + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) + COPY_SAMPLES(int32_t); + else + COPY_SAMPLES(int16_t); +} + +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_element_header(AlacEncodeContext *s, + enum AlacRawDataBlockType element, + int instance) +{ + int encode_fs = 0; + + if (s->frame_size < DEFAULT_FRAME_SIZE) + encode_fs = 1; + + put_bits(&s->pbctx, 3, element); // element type + put_bits(&s->pbctx, 4, instance); // element instance + put_bits(&s->pbctx, 12, 0); // unused header bits + put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header + put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit) + 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 void write_element(AlacEncodeContext *s, + enum AlacRawDataBlockType element, int instance, + const uint8_t *samples0, const uint8_t *samples1) +{ + uint8_t const *samples[2] = { samples0, samples1 }; + int i, j, channels; + int prediction_type = 0; + PutBitContext *pb = &s->pbctx; + + channels = element == TYPE_CPE ? 2 : 1; + + if (s->verbatim) { + write_element_header(s, element, instance); + /* samples are channel-interleaved in verbatim mode */ + if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { + int shift = 32 - s->avctx->bits_per_raw_sample; + int32_t const *samples_s32[2] = { (const int32_t *)samples0, + (const int32_t *)samples1 }; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s32[j][i] >> shift); + } else { + int16_t const *samples_s16[2] = { (const int16_t *)samples0, + (const int16_t *)samples1 }; + for (i = 0; i < s->frame_size; i++) + for (j = 0; j < channels; j++) + put_sbits(pb, s->avctx->bits_per_raw_sample, + samples_s16[j][i]); + } + } else { + s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits + + channels - 1; + + init_sample_buffers(s, channels, samples); + write_element_header(s, element, instance); + + if (channels == 2) + alac_stereo_decorrelation(s); + else + s->interlacing_shift = s->interlacing_leftweight = 0; + put_bits(pb, 8, s->interlacing_shift); + put_bits(pb, 8, s->interlacing_leftweight); + + for (i = 0; i < 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]); + } + + // write extra bits if needed + if (s->extra_bits) { + uint32_t mask = (1 << s->extra_bits) - 1; + for (i = 0; i < s->frame_size; i++) { + for (j = 0; j < channels; j++) { + put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask); + s->sample_buf[j][i] >>= s->extra_bits; + } + } + } + + // apply lpc and entropy coding to audio samples + for (i = 0; i < 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); + } + } +} + +static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, + uint8_t * const *samples) +{ + PutBitContext *pb = &s->pbctx; + const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1]; + const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1]; + int ch, element, sce, cpe; + + init_put_bits(pb, avpkt->data, avpkt->size); + + ch = element = sce = cpe = 0; + while (ch < s->avctx->channels) { + if (ch_elements[element] == TYPE_CPE) { + write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]], + samples[ch_map[ch + 1]]); + cpe++; + ch += 2; + } else { + write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL); + sce++; + ch++; + } + element++; + } + + put_bits(pb, 3, TYPE_END); + 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; + + 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; + } + + 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; + + 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", + .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 }, + .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), +}; 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