X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fatrac1.c;h=d6c7053d7bd00e1c6a8b2b8a68e4f5ff2a66151a;hb=84465f2180308a3e998089517e76586563fd6162;hp=0b3a2449ae15e2c890adfe98bf460693c319d824;hpb=b11d40d12e18b425bcfbf062c2509d9491d0f440;p=ffmpeg diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index 0b3a2449ae1..d6c7053d7bd 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -3,25 +3,25 @@ * Copyright (c) 2009 Maxim Poliakovski * Copyright (c) 2009 Benjamin Larsson * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** - * @file libavcodec/atrac1.c + * @file * Atrac 1 compatible decoder. * This decoder handles raw ATRAC1 data and probably SDDS data. */ @@ -35,6 +35,8 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" +#include "fft.h" +#include "sinewin.h" #include "atrac.h" #include "atrac1data.h" @@ -58,11 +60,11 @@ 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 + 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; /** @@ -70,13 +72,13 @@ typedef struct { */ 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, 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]); + 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]); + DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; FFTContext mdct_ctx[3]; int channels; DSPContext dsp; @@ -98,7 +100,7 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, for (i = 0; i < transf_size / 2; i++) FFSWAP(float, spec[i], spec[transf_size - 1 - i]); } - ff_imdct_half(mdct_context, out, spec); + mdct_context->imdct_half(mdct_context, out, spec); } @@ -119,33 +121,33 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) 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; + /* 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; + /* 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; + 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); + 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); + /* 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, 16); - prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; - start_pos += block_size; - pos += block_size; - } + 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)); @@ -222,7 +224,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, 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]]; + float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]]; bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ /* check for bitstream overflow */ @@ -251,7 +253,7 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, } -void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) +static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) { float temp[256]; float iqmf_temp[512 + 46]; @@ -304,20 +306,15 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data, at1_subband_synthesis(q, su, q->out_samples[ch]); } - /* round, convert to 16bit and interleave */ + /* interleave; FIXME, should create/use a DSP function */ if (q->channels == 1) { /* mono */ - q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15), - 32700.0 / (1 << 15), AT1_SU_SAMPLES); + memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4); } else { /* stereo */ for (i = 0; i < AT1_SU_SAMPLES; i++) { - samples[i * 2] = av_clipf(q->out_samples[0][i], - -32700.0 / (1 << 15), - 32700.0 / (1 << 15)); - samples[i * 2 + 1] = av_clipf(q->out_samples[1][i], - -32700.0 / (1 << 15), - 32700.0 / (1 << 15)); + samples[i * 2] = q->out_samples[0][i]; + samples[i * 2 + 1] = q->out_samples[1][i]; } } @@ -330,7 +327,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) { AT1Ctx *q = avctx->priv_data; - avctx->sample_fmt = SAMPLE_FMT_FLT; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; q->channels = avctx->channels; @@ -339,7 +336,7 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) 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_sine_window_init(ff_sine_32, 32); + ff_init_ff_sine_windows(5); atrac_generate_tables(); @@ -369,9 +366,9 @@ static av_cold int atrac1_decode_end(AVCodecContext * avctx) { } -AVCodec atrac1_decoder = { +AVCodec ff_atrac1_decoder = { .name = "atrac1", - .type = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_ATRAC1, .priv_data_size = sizeof(AT1Ctx), .init = atrac1_decode_init,