Mercurial > libavcodec.hg
view atrac1.c @ 12241:c7f6ddcc5c01 libavcodec
VP8: optimize DC-only chroma case in the same way as luma.
Add MMX idct_dc_add4uv function for this case.
~40% faster chroma idct.
author | darkshikari |
---|---|
date | Fri, 23 Jul 2010 06:02:52 +0000 |
parents | 8b6f3d3b55cb |
children |
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/* * Atrac 1 compatible decoder * Copyright (c) 2009 Maxim Poliakovski * Copyright (c) 2009 Benjamin Larsson * * 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 */ /** * @file * Atrac 1 compatible decoder. * This decoder handles raw ATRAC1 data and probably SDDS data. */ /* Many thanks to Tim Craig for all the help! */ #include <math.h> #include <stddef.h> #include <stdio.h> #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" #include "fft.h" #include "atrac.h" #include "atrac1data.h" #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 #define AT1_MAX_CHANNELS 2 #define AT1_QMF_BANDS 3 #define IDX_LOW_BAND 0 #define IDX_MID_BAND 1 #define IDX_HIGH_BAND 2 /** * Sound unit struct, one unit is used per channel */ typedef struct { int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band int num_bfus; ///< number of Block Floating Units float* spectrum[2]; DECLARE_ALIGNED(16, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer DECLARE_ALIGNED(16, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer DECLARE_ALIGNED(16, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter DECLARE_ALIGNED(16, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter DECLARE_ALIGNED(16, float, last_qmf_delay)[256+23]; ///< delay line for the last stacked QMF filter } AT1SUCtx; /** * The atrac1 context, holds all needed parameters for decoding */ typedef struct { AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit DECLARE_ALIGNED(16, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer DECLARE_ALIGNED(16, float, low)[256]; DECLARE_ALIGNED(16, float, mid)[256]; DECLARE_ALIGNED(16, float, high)[512]; float* bands[3]; DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; FFTContext mdct_ctx[3]; int channels; DSPContext dsp; } AT1Ctx; /** size of the transform in samples in the long mode for each QMF band */ static const uint16_t samples_per_band[3] = {128, 128, 256}; static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec) { FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; int transf_size = 1 << nbits; if (rev_spec) { int i; for (i = 0; i < transf_size / 2; i++) FFSWAP(float, spec[i], spec[transf_size - 1 - i]); } ff_imdct_half(mdct_context, out, spec); } static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) { int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; unsigned int start_pos, ref_pos = 0, pos = 0; for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { float *prev_buf; int j; band_samples = samples_per_band[band_num]; log2_block_count = su->log2_block_count[band_num]; /* number of mdct blocks in the current QMF band: 1 - for long mode */ /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ num_blocks = 1 << log2_block_count; if (num_blocks == 1) { /* mdct block size in samples: 128 (long mode, low & mid bands), */ /* 256 (long mode, high band) and 32 (short mode, all bands) */ block_size = band_samples >> log2_block_count; /* calc transform size in bits according to the block_size_mode */ nbits = mdct_long_nbits[band_num] - log2_block_count; if (nbits != 5 && nbits != 7 && nbits != 8) return -1; } else { block_size = 32; nbits = 5; } start_pos = 0; prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; for (j=0; j < num_blocks; j++) { at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num); /* overlap and window */ q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, &su->spectrum[0][ref_pos + start_pos], ff_sine_32, 0, 16); prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; start_pos += block_size; pos += block_size; } if (num_blocks == 1) memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); ref_pos += band_samples; } /* Swap buffers so the mdct overlap works */ FFSWAP(float*, su->spectrum[0], su->spectrum[1]); return 0; } /** * Parse the block size mode byte */ static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) { int log2_block_count_tmp, i; for (i = 0; i < 2; i++) { /* low and mid band */ log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp & 1) return -1; log2_block_cnt[i] = 2 - log2_block_count_tmp; } /* high band */ log2_block_count_tmp = get_bits(gb, 2); if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) return -1; log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; skip_bits(gb, 2); return 0; } static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU_SAMPLES]) { int bits_used, band_num, bfu_num, i; uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU /* parse the info byte (2nd byte) telling how much BFUs were coded */ su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; /* calc number of consumed bits: num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ bits_used = su->num_bfus * 10 + 32 + bfu_amount_tab2[get_bits(gb, 2)] + (bfu_amount_tab3[get_bits(gb, 3)] << 1); /* get word length index (idwl) for each BFU */ for (i = 0; i < su->num_bfus; i++) idwls[i] = get_bits(gb, 4); /* get scalefactor index (idsf) for each BFU */ for (i = 0; i < su->num_bfus; i++) idsfs[i] = get_bits(gb, 6); /* zero idwl/idsf for empty BFUs */ for (i = su->num_bfus; i < AT1_MAX_BFU; i++) idwls[i] = idsfs[i] = 0; /* read in the spectral data and reconstruct MDCT spectrum of this channel */ for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { int pos; int num_specs = specs_per_bfu[bfu_num]; int word_len = !!idwls[bfu_num] + idwls[bfu_num]; float scale_factor = sf_table[idsfs[bfu_num]]; bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ /* check for bitstream overflow */ if (bits_used > AT1_SU_MAX_BITS) return -1; /* get the position of the 1st spec according to the block size mode */ pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; if (word_len) { float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); for (i = 0; i < num_specs; i++) { /* read in a quantized spec and convert it to * signed int and then inverse quantization */ spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; } } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */ memset(&spec[pos], 0, num_specs * sizeof(float)); } } } return 0; } static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) { float temp[256]; float iqmf_temp[512 + 46]; /* combine low and middle bands */ atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); /* delay the signal of the high band by 23 samples */ memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); /* combine (low + middle) and high bands */ atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); } static int atrac1_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AT1Ctx *q = avctx->priv_data; int ch, ret, i; GetBitContext gb; float* samples = data; if (buf_size < 212 * q->channels) { av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); return -1; } for (ch = 0; ch < q->channels; ch++) { AT1SUCtx* su = &q->SUs[ch]; init_get_bits(&gb, &buf[212 * ch], 212 * 8); /* parse block_size_mode, 1st byte */ ret = at1_parse_bsm(&gb, su->log2_block_count); if (ret < 0) return ret; ret = at1_unpack_dequant(&gb, su, q->spec); if (ret < 0) return ret; ret = at1_imdct_block(su, q); if (ret < 0) return ret; at1_subband_synthesis(q, su, q->out_samples[ch]); } /* interleave; FIXME, should create/use a DSP function */ if (q->channels == 1) { /* mono */ memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4); } else { /* stereo */ for (i = 0; i < AT1_SU_SAMPLES; i++) { samples[i * 2] = q->out_samples[0][i]; samples[i * 2 + 1] = q->out_samples[1][i]; } } *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); return avctx->block_align; } static av_cold int atrac1_decode_init(AVCodecContext *avctx) { AT1Ctx *q = avctx->priv_data; avctx->sample_fmt = SAMPLE_FMT_FLT; q->channels = avctx->channels; /* Init the mdct transforms */ ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); ff_init_ff_sine_windows(5); atrac_generate_tables(); dsputil_init(&q->dsp, avctx); q->bands[0] = q->low; q->bands[1] = q->mid; q->bands[2] = q->high; /* Prepare the mdct overlap buffers */ q->SUs[0].spectrum[0] = q->SUs[0].spec1; q->SUs[0].spectrum[1] = q->SUs[0].spec2; q->SUs[1].spectrum[0] = q->SUs[1].spec1; q->SUs[1].spectrum[1] = q->SUs[1].spec2; return 0; } static av_cold int atrac1_decode_end(AVCodecContext * avctx) { AT1Ctx *q = avctx->priv_data; ff_mdct_end(&q->mdct_ctx[0]); ff_mdct_end(&q->mdct_ctx[1]); ff_mdct_end(&q->mdct_ctx[2]); return 0; } AVCodec atrac1_decoder = { .name = "atrac1", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_ATRAC1, .priv_data_size = sizeof(AT1Ctx), .init = atrac1_decode_init, .close = atrac1_decode_end, .decode = atrac1_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), };