X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fatrac1.c;h=202ed8f5ea994c37c423ac8813659fc730b4efb2;hb=0fb49b597be60efb84bb39c09576baecc1471eeb;hp=d08056f5a9ea99025d7d4ddf3e56e83a097c10c2;hpb=1e1898c00fc430c9e1489c6b7ad38f6461451afd;p=ffmpeg diff --git a/libavcodec/atrac1.c b/libavcodec/atrac1.c index d08056f5a9e..202ed8f5ea9 100644 --- a/libavcodec/atrac1.c +++ b/libavcodec/atrac1.c @@ -23,7 +23,7 @@ /** * @file libavcodec/atrac1.c * Atrac 1 compatible decoder. - * This decoder handles raw ATRAC1 data. + * This decoder handles raw ATRAC1 data and probably SDDS data. */ /* Many thanks to Tim Craig for all the help! */ @@ -35,6 +35,7 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" +#include "fft.h" #include "atrac.h" #include "atrac1data.h" @@ -57,14 +58,12 @@ typedef struct { int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band int num_bfus; ///< number of Block Floating Units - int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU - int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU 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; /** @@ -72,20 +71,18 @@ 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]); - MDCTContext mdct_ctx[3]; + DECLARE_ALIGNED(16, float, out_samples)[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; + FFTContext mdct_ctx[3]; int channels; DSPContext dsp; } AT1Ctx; -DECLARE_ALIGNED_16(static float, short_window[32]); - /** 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}; @@ -94,11 +91,9 @@ 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) { - MDCTContext* mdct_context; + FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; int transf_size = 1 << nbits; - mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; - if (rev_spec) { int i; for (i = 0; i < transf_size / 2; i++) @@ -111,9 +106,12 @@ static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, 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; + 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]; @@ -121,42 +119,38 @@ static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ num_blocks = 1 << 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; + 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; + /* 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; + } - if (num_blocks == 1) { - /* long blocks */ - at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num); - pos += block_size; // move to the next mdct block in the spectrum + 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 long blocks */ - q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16], - &su->spectrum[0][ref_pos], short_window, 0, 16); - memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); - } else { - /* short blocks */ - float *prev_buf; - start_pos = 0; - prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; - for (; num_blocks != 0; num_blocks--) { - at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num); - - /* overlap and window between short blocks */ - q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, - &su->spectrum[0][ref_pos + start_pos], short_window, 0, 16); - - prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; - start_pos += 32; // use hardcoded block_size - pos += 32; - } + /* 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; } @@ -197,6 +191,8 @@ 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)]; @@ -210,15 +206,15 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, /* get word length index (idwl) for each BFU */ for (i = 0; i < su->num_bfus; i++) - su->idwls[i] = get_bits(gb, 4); + idwls[i] = get_bits(gb, 4); /* get scalefactor index (idsf) for each BFU */ for (i = 0; i < su->num_bfus; i++) - su->idsfs[i] = get_bits(gb, 6); + idsfs[i] = get_bits(gb, 6); /* zero idwl/idsf for empty BFUs */ for (i = su->num_bfus; i < AT1_MAX_BFU; i++) - su->idwls[i] = su->idsfs[i] = 0; + 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++) { @@ -226,9 +222,9 @@ static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, int pos; int num_specs = specs_per_bfu[bfu_num]; - int word_len = !!su->idwls[bfu_num] + su->idwls[bfu_num]; - float scale_factor = sf_table[su->idsfs[bfu_num]]; - bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ + 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) @@ -256,7 +252,7 @@ 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]; @@ -344,7 +340,7 @@ 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(short_window, 32); + ff_init_ff_sine_windows(5); atrac_generate_tables(); @@ -363,13 +359,24 @@ static av_cold int atrac1_decode_init(AVCodecContext *avctx) 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 = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_ATRAC1, .priv_data_size = sizeof(AT1Ctx), .init = atrac1_decode_init, - .close = NULL, + .close = atrac1_decode_end, .decode = atrac1_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), };