From 57231e4d5b467833fb289439cd35a92513bb55c1 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Tue, 2 Oct 2012 13:43:19 +0000 Subject: tak: demuxer, parser, and decoder Signed-off-by: Paul B Mahol Signed-off-by: Justin Ruggles --- libavcodec/takdec.c | 929 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 929 insertions(+) create mode 100644 libavcodec/takdec.c (limited to 'libavcodec/takdec.c') diff --git a/libavcodec/takdec.c b/libavcodec/takdec.c new file mode 100644 index 0000000000..87fcf832ba --- /dev/null +++ b/libavcodec/takdec.c @@ -0,0 +1,929 @@ +/* + * TAK decoder + * Copyright (c) 2012 Paul B Mahol + * + * This file is part of Libav. + * + * 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. + * + * 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * TAK (Tom's lossless Audio Kompressor) decoder + * @author Paul B Mahol + */ + +#include "libavutil/samplefmt.h" +#include "tak.h" +#include "avcodec.h" +#include "dsputil.h" +#include "internal.h" +#include "unary.h" + +#define MAX_SUBFRAMES 8 // max number of subframes per channel +#define MAX_PREDICTORS 256 + +typedef struct MCDParam { + int8_t present; // decorrelation parameter availability for this channel + int8_t index; // index into array of decorrelation types + int8_t chan1; + int8_t chan2; +} MCDParam; + +typedef struct TAKDecContext { + AVCodecContext *avctx; // parent AVCodecContext + AVFrame frame; // AVFrame for decoded output + DSPContext dsp; + TAKStreamInfo ti; + GetBitContext gb; // bitstream reader initialized to start at the current frame + + int uval; + int nb_samples; // number of samples in the current frame + uint8_t *decode_buffer; + unsigned int decode_buffer_size; + int32_t *decoded[TAK_MAX_CHANNELS]; // decoded samples for each channel + + int8_t lpc_mode[TAK_MAX_CHANNELS]; + int8_t sample_shift[TAK_MAX_CHANNELS]; // shift applied to every sample in the channel + int subframe_scale; + + int8_t dmode; // channel decorrelation type in the current frame + + MCDParam mcdparams[TAK_MAX_CHANNELS]; // multichannel decorrelation parameters + + int16_t *residues; + unsigned int residues_buf_size; +} TAKDecContext; + +static const int8_t mc_dmodes[] = { 1, 3, 4, 6, }; + +static const uint16_t predictor_sizes[] = { + 4, 8, 12, 16, 24, 32, 48, 64, 80, 96, 128, 160, 192, 224, 256, 0, +}; + +static const struct CParam { + int init; + int escape; + int scale; + int aescape; + int bias; +} xcodes[50] = { + { 0x01, 0x0000001, 0x0000001, 0x0000003, 0x0000008 }, + { 0x02, 0x0000003, 0x0000001, 0x0000007, 0x0000006 }, + { 0x03, 0x0000005, 0x0000002, 0x000000E, 0x000000D }, + { 0x03, 0x0000003, 0x0000003, 0x000000D, 0x0000018 }, + { 0x04, 0x000000B, 0x0000004, 0x000001C, 0x0000019 }, + { 0x04, 0x0000006, 0x0000006, 0x000001A, 0x0000030 }, + { 0x05, 0x0000016, 0x0000008, 0x0000038, 0x0000032 }, + { 0x05, 0x000000C, 0x000000C, 0x0000034, 0x0000060 }, + { 0x06, 0x000002C, 0x0000010, 0x0000070, 0x0000064 }, + { 0x06, 0x0000018, 0x0000018, 0x0000068, 0x00000C0 }, + { 0x07, 0x0000058, 0x0000020, 0x00000E0, 0x00000C8 }, + { 0x07, 0x0000030, 0x0000030, 0x00000D0, 0x0000180 }, + { 0x08, 0x00000B0, 0x0000040, 0x00001C0, 0x0000190 }, + { 0x08, 0x0000060, 0x0000060, 0x00001A0, 0x0000300 }, + { 0x09, 0x0000160, 0x0000080, 0x0000380, 0x0000320 }, + { 0x09, 0x00000C0, 0x00000C0, 0x0000340, 0x0000600 }, + { 0x0A, 0x00002C0, 0x0000100, 0x0000700, 0x0000640 }, + { 0x0A, 0x0000180, 0x0000180, 0x0000680, 0x0000C00 }, + { 0x0B, 0x0000580, 0x0000200, 0x0000E00, 0x0000C80 }, + { 0x0B, 0x0000300, 0x0000300, 0x0000D00, 0x0001800 }, + { 0x0C, 0x0000B00, 0x0000400, 0x0001C00, 0x0001900 }, + { 0x0C, 0x0000600, 0x0000600, 0x0001A00, 0x0003000 }, + { 0x0D, 0x0001600, 0x0000800, 0x0003800, 0x0003200 }, + { 0x0D, 0x0000C00, 0x0000C00, 0x0003400, 0x0006000 }, + { 0x0E, 0x0002C00, 0x0001000, 0x0007000, 0x0006400 }, + { 0x0E, 0x0001800, 0x0001800, 0x0006800, 0x000C000 }, + { 0x0F, 0x0005800, 0x0002000, 0x000E000, 0x000C800 }, + { 0x0F, 0x0003000, 0x0003000, 0x000D000, 0x0018000 }, + { 0x10, 0x000B000, 0x0004000, 0x001C000, 0x0019000 }, + { 0x10, 0x0006000, 0x0006000, 0x001A000, 0x0030000 }, + { 0x11, 0x0016000, 0x0008000, 0x0038000, 0x0032000 }, + { 0x11, 0x000C000, 0x000C000, 0x0034000, 0x0060000 }, + { 0x12, 0x002C000, 0x0010000, 0x0070000, 0x0064000 }, + { 0x12, 0x0018000, 0x0018000, 0x0068000, 0x00C0000 }, + { 0x13, 0x0058000, 0x0020000, 0x00E0000, 0x00C8000 }, + { 0x13, 0x0030000, 0x0030000, 0x00D0000, 0x0180000 }, + { 0x14, 0x00B0000, 0x0040000, 0x01C0000, 0x0190000 }, + { 0x14, 0x0060000, 0x0060000, 0x01A0000, 0x0300000 }, + { 0x15, 0x0160000, 0x0080000, 0x0380000, 0x0320000 }, + { 0x15, 0x00C0000, 0x00C0000, 0x0340000, 0x0600000 }, + { 0x16, 0x02C0000, 0x0100000, 0x0700000, 0x0640000 }, + { 0x16, 0x0180000, 0x0180000, 0x0680000, 0x0C00000 }, + { 0x17, 0x0580000, 0x0200000, 0x0E00000, 0x0C80000 }, + { 0x17, 0x0300000, 0x0300000, 0x0D00000, 0x1800000 }, + { 0x18, 0x0B00000, 0x0400000, 0x1C00000, 0x1900000 }, + { 0x18, 0x0600000, 0x0600000, 0x1A00000, 0x3000000 }, + { 0x19, 0x1600000, 0x0800000, 0x3800000, 0x3200000 }, + { 0x19, 0x0C00000, 0x0C00000, 0x3400000, 0x6000000 }, + { 0x1A, 0x2C00000, 0x1000000, 0x7000000, 0x6400000 }, + { 0x1A, 0x1800000, 0x1800000, 0x6800000, 0xC000000 }, +}; + +static av_cold void tak_init_static_data(AVCodec *codec) +{ + ff_tak_init_crc(); +} + +static int set_bps_params(AVCodecContext *avctx) +{ + switch (avctx->bits_per_coded_sample) { + case 8: + avctx->sample_fmt = AV_SAMPLE_FMT_U8P; + break; + case 16: + avctx->sample_fmt = AV_SAMPLE_FMT_S16P; + break; + case 24: + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + break; + default: + av_log(avctx, AV_LOG_ERROR, "unsupported bits per sample: %d\n", + avctx->bits_per_coded_sample); + return AVERROR_INVALIDDATA; + } + avctx->bits_per_raw_sample = avctx->bits_per_coded_sample; + + return 0; +} + +static void set_sample_rate_params(AVCodecContext *avctx) +{ + TAKDecContext *s = avctx->priv_data; + int shift = 3 - (avctx->sample_rate / 11025); + shift = FFMAX(0, shift); + s->uval = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << shift; + s->subframe_scale = FFALIGN(avctx->sample_rate + 511 >> 9, 4) << 1; +} + +static av_cold int tak_decode_init(AVCodecContext *avctx) +{ + TAKDecContext *s = avctx->priv_data; + + ff_dsputil_init(&s->dsp, avctx); + + s->avctx = avctx; + avcodec_get_frame_defaults(&s->frame); + avctx->coded_frame = &s->frame; + + set_sample_rate_params(avctx); + + return set_bps_params(avctx); +} + +static void decode_lpc(int32_t *coeffs, int mode, int length) +{ + int i; + + if (length < 2) + return; + + if (mode == 1) { + int a1 = *coeffs++; + for (i = 0; i < length - 1 >> 1; i++) { + *coeffs += a1; + coeffs[1] += *coeffs; + a1 = coeffs[1]; + coeffs += 2; + } + if (length - 1 & 1) + *coeffs += a1; + } else if (mode == 2) { + int a1 = coeffs[1]; + int a2 = a1 + *coeffs; + coeffs[1] = a2; + if (length > 2) { + coeffs += 2; + for (i = 0; i < length - 2 >> 1; i++) { + int a3 = *coeffs + a1; + int a4 = a3 + a2; + *coeffs = a4; + a1 = coeffs[1] + a3; + a2 = a1 + a4; + coeffs[1] = a2; + coeffs += 2; + } + if (length & 1) + *coeffs += a1 + a2; + } + } else if (mode == 3) { + int a1 = coeffs[1]; + int a2 = a1 + *coeffs; + coeffs[1] = a2; + if (length > 2) { + int a3 = coeffs[2]; + int a4 = a3 + a1; + int a5 = a4 + a2; + coeffs += 3; + for (i = 0; i < length - 3; i++) { + a3 += *coeffs; + a4 += a3; + a5 += a4; + *coeffs = a5; + coeffs++; + } + } + } +} + +static int decode_segment(GetBitContext *gb, int mode, int32_t *decoded, + int len) +{ + struct CParam code; + int i; + + if (!mode) { + memset(decoded, 0, len * sizeof(*decoded)); + return 0; + } + + if (mode > FF_ARRAY_ELEMS(xcodes)) + return AVERROR_INVALIDDATA; + code = xcodes[mode - 1]; + + for (i = 0; i < len; i++) { + int x = get_bits_long(gb, code.init); + if (x >= code.escape && get_bits1(gb)) { + x |= 1 << code.init; + if (x >= code.aescape) { + int scale = get_unary(gb, 1, 9); + if (scale == 9) { + int scale_bits = get_bits(gb, 3); + if (scale_bits > 0) { + if (scale_bits == 7) { + scale_bits += get_bits(gb, 5); + if (scale_bits > 29) + return AVERROR_INVALIDDATA; + } + scale = get_bits_long(gb, scale_bits) + 1; + x += code.scale * scale; + } + x += code.bias; + } else + x += code.scale * scale - code.escape; + } else + x -= code.escape; + } + decoded[i] = (x >> 1) ^ -(x & 1); + } + + return 0; +} + +static int decode_residues(TAKDecContext *s, int32_t *decoded, int length) +{ + GetBitContext *gb = &s->gb; + int i, mode, ret; + + if (length > s->nb_samples) + return AVERROR_INVALIDDATA; + + if (get_bits1(gb)) { + int wlength, rval; + int coding_mode[128]; + + wlength = length / s->uval; + + rval = length - (wlength * s->uval); + + if (rval < s->uval / 2) + rval += s->uval; + else + wlength++; + + if (wlength <= 1 || wlength > 128) + return AVERROR_INVALIDDATA; + + coding_mode[0] = mode = get_bits(gb, 6); + + for (i = 1; i < wlength; i++) { + int c = get_unary(gb, 1, 6); + + switch (c) { + case 6: + mode = get_bits(gb, 6); + break; + case 5: + case 4: + case 3: { + /* mode += sign ? (1 - c) : (c - 1) */ + int sign = get_bits1(gb); + mode += (-sign ^ (c - 1)) + sign; + break; + } + case 2: + mode++; + break; + case 1: + mode--; + break; + } + coding_mode[i] = mode; + } + + i = 0; + while (i < wlength) { + int len = 0; + + mode = coding_mode[i]; + do { + if (i >= wlength - 1) + len += rval; + else + len += s->uval; + i++; + + if (i == wlength) + break; + } while (coding_mode[i] == mode); + + if ((ret = decode_segment(gb, mode, decoded, len)) < 0) + return ret; + decoded += len; + } + } else { + mode = get_bits(gb, 6); + if ((ret = decode_segment(gb, mode, decoded, length)) < 0) + return ret; + } + + return 0; +} + +static int get_bits_esc4(GetBitContext *gb) +{ + if (get_bits1(gb)) + return get_bits(gb, 4) + 1; + else + return 0; +} + +static void decode_filter_coeffs(TAKDecContext *s, int filter_order, int size, + int filter_quant, int16_t *filter) +{ + GetBitContext *gb = &s->gb; + int i, j, a, b; + int filter_tmp[MAX_PREDICTORS]; + int16_t predictors[MAX_PREDICTORS]; + + predictors[0] = get_sbits(gb, 10); + predictors[1] = get_sbits(gb, 10); + predictors[2] = get_sbits(gb, size) << (10 - size); + predictors[3] = get_sbits(gb, size) << (10 - size); + if (filter_order > 4) { + int av_uninit(code_size); + int code_size_base = size - get_bits1(gb); + + for (i = 4; i < filter_order; i++) { + if (!(i & 3)) + code_size = code_size_base - get_bits(gb, 2); + predictors[i] = get_sbits(gb, code_size) << (10 - size); + } + } + + filter_tmp[0] = predictors[0] << 6; + for (i = 1; i < filter_order; i++) { + int *p1 = &filter_tmp[0]; + int *p2 = &filter_tmp[i - 1]; + + for (j = 0; j < (i + 1) / 2; j++) { + int tmp = *p1 + (predictors[i] * *p2 + 256 >> 9); + *p2 = *p2 + (predictors[i] * *p1 + 256 >> 9); + *p1 = tmp; + p1++; + p2--; + } + + filter_tmp[i] = predictors[i] << 6; + } + + a = 1 << (32 - (15 - filter_quant)); + b = 1 << ((15 - filter_quant) - 1); + for (i = 0, j = filter_order - 1; i < filter_order / 2; i++, j--) { + filter[j] = a - ((filter_tmp[i] + b) >> (15 - filter_quant)); + filter[i] = a - ((filter_tmp[j] + b) >> (15 - filter_quant)); + } +} + +static int decode_subframe(TAKDecContext *s, int32_t *decoded, + int subframe_size, int prev_subframe_size) +{ + LOCAL_ALIGNED_16(int16_t, filter, [MAX_PREDICTORS]) = { 0, }; + GetBitContext *gb = &s->gb; + int i, ret; + int dshift, size, filter_quant, filter_order; + + if (!get_bits1(gb)) + return decode_residues(s, decoded, subframe_size); + + filter_order = predictor_sizes[get_bits(gb, 4)]; + + if (prev_subframe_size > 0 && get_bits1(gb)) { + if (filter_order > prev_subframe_size) + return AVERROR_INVALIDDATA; + + decoded -= filter_order; + subframe_size += filter_order; + + if (filter_order > subframe_size) + return AVERROR_INVALIDDATA; + } else { + int lpc_mode; + + if (filter_order > subframe_size) + return AVERROR_INVALIDDATA; + + lpc_mode = get_bits(gb, 2); + if (lpc_mode > 2) + return AVERROR_INVALIDDATA; + + if ((ret = decode_residues(s, decoded, filter_order)) < 0) + return ret; + + if (lpc_mode) + decode_lpc(decoded, lpc_mode, filter_order); + } + + dshift = get_bits_esc4(gb); + size = get_bits1(gb) + 6; + + filter_quant = 10; + if (get_bits1(gb)) { + filter_quant -= get_bits(gb, 3) + 1; + if (filter_quant < 3) + return AVERROR_INVALIDDATA; + } + + decode_filter_coeffs(s, filter_order, size, filter_quant, filter); + + if ((ret = decode_residues(s, &decoded[filter_order], + subframe_size - filter_order)) < 0) + return ret; + + av_fast_malloc(&s->residues, &s->residues_buf_size, + FFALIGN(subframe_size + 16, 16) * sizeof(*s->residues)); + if (!s->residues) + return AVERROR(ENOMEM); + memset(s->residues, 0, s->residues_buf_size); + + for (i = 0; i < filter_order; i++) + s->residues[i] = *decoded++ >> dshift; + + for (i = 0; i < subframe_size - filter_order; i++) { + int v = 1 << (filter_quant - 1); + + v += s->dsp.scalarproduct_int16(&s->residues[i], filter, + FFALIGN(filter_order, 16)); + + v = (av_clip(v >> filter_quant, -8192, 8191) << dshift) - *decoded; + *decoded++ = v; + s->residues[filter_order + i] = v >> dshift; + } + + emms_c(); + + return 0; +} + +static int decode_channel(TAKDecContext *s, int chan) +{ + AVCodecContext *avctx = s->avctx; + GetBitContext *gb = &s->gb; + int32_t *decoded = s->decoded[chan]; + int left = s->nb_samples - 1; + int i, prev, ret, nb_subframes; + int subframe_len[MAX_SUBFRAMES]; + + s->sample_shift[chan] = get_bits_esc4(gb); + if (s->sample_shift[chan] >= avctx->bits_per_coded_sample) + return AVERROR_INVALIDDATA; + + /* NOTE: TAK 2.2.0 appears to set the sample value to 0 if + * bits_per_coded_sample - sample_shift is 1, but this produces + * non-bit-exact output. Reading the 1 bit using get_sbits() instead + * of skipping it produces bit-exact output. This has been reported + * to the TAK author. */ + *decoded++ = get_sbits(gb, + avctx->bits_per_coded_sample - + s->sample_shift[chan]); + s->lpc_mode[chan] = get_bits(gb, 2); + nb_subframes = get_bits(gb, 3) + 1; + + i = 0; + if (nb_subframes > 1) { + if (get_bits_left(gb) < (nb_subframes - 1) * 6) + return AVERROR_INVALIDDATA; + + prev = 0; + for (; i < nb_subframes - 1; i++) { + int subframe_end = get_bits(gb, 6) * s->subframe_scale; + if (subframe_end <= prev) + return AVERROR_INVALIDDATA; + subframe_len[i] = subframe_end - prev; + left -= subframe_len[i]; + prev = subframe_end; + } + + if (left <= 0) + return AVERROR_INVALIDDATA; + } + subframe_len[i] = left; + + prev = 0; + for (i = 0; i < nb_subframes; i++) { + if ((ret = decode_subframe(s, decoded, subframe_len[i], prev)) < 0) + return ret; + decoded += subframe_len[i]; + prev = subframe_len[i]; + } + + return 0; +} + +static int decorrelate(TAKDecContext *s, int c1, int c2, int length) +{ + GetBitContext *gb = &s->gb; + int32_t *p1 = s->decoded[c1] + 1; + int32_t *p2 = s->decoded[c2] + 1; + int i; + int dshift, dfactor; + + switch (s->dmode) { + case 1: /* left/side */ + for (i = 0; i < length; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + p2[i] = a + b; + } + break; + case 2: /* side/right */ + for (i = 0; i < length; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + p1[i] = b - a; + } + break; + case 3: /* side/mid */ + for (i = 0; i < length; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + a -= b >> 1; + p1[i] = a; + p2[i] = a + b; + } + break; + case 4: /* side/left with scale factor */ + FFSWAP(int32_t*, p1, p2); + case 5: /* side/right with scale factor */ + dshift = get_bits_esc4(gb); + dfactor = get_sbits(gb, 10); + for (i = 0; i < length; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + b = dfactor * (b >> dshift) + 128 >> 8 << dshift; + p1[i] = b - a; + } + break; + case 6: + FFSWAP(int32_t*, p1, p2); + case 7: { + LOCAL_ALIGNED_16(int16_t, filter, [MAX_PREDICTORS]) = { 0 }; + int length2, order_half, filter_order, dval1, dval2; + int av_uninit(code_size); + + if (length < 256) + return AVERROR_INVALIDDATA; + + dshift = get_bits_esc4(gb); + filter_order = 8 << get_bits1(gb); + dval1 = get_bits1(gb); + dval2 = get_bits1(gb); + + for (i = 0; i < filter_order; i++) { + if (!(i & 3)) + code_size = 14 - get_bits(gb, 3); + filter[i] = get_sbits(gb, code_size); + } + + order_half = filter_order / 2; + length2 = length - (filter_order - 1); + + /* decorrelate beginning samples */ + if (dval1) { + for (i = 0; i < order_half; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + p1[i] = a + b; + } + } + + /* decorrelate ending samples */ + if (dval2) { + for (i = length2 + order_half; i < length; i++) { + int32_t a = p1[i]; + int32_t b = p2[i]; + p1[i] = a + b; + } + } + + av_fast_malloc(&s->residues, &s->residues_buf_size, + FFALIGN(length + 16, 16) * sizeof(*s->residues)); + if (!s->residues) + return AVERROR(ENOMEM); + memset(s->residues, 0, s->residues_buf_size); + + for (i = 0; i < length; i++) + s->residues[i] = p2[i] >> dshift; + + p1 += order_half; + + for (i = 0; i < length2; i++) { + int v = 1 << 9; + + v += s->dsp.scalarproduct_int16(&s->residues[i], filter, + FFALIGN(filter_order, 16)); + + p1[i] = (av_clip(v >> 10, -8192, 8191) << dshift) - p1[i]; + } + + emms_c(); + break; + } + } + + return 0; +} + +static int tak_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *pkt) +{ + TAKDecContext *s = avctx->priv_data; + GetBitContext *gb = &s->gb; + int chan, i, ret, hsize; + + if (pkt->size < TAK_MIN_FRAME_HEADER_BYTES) + return AVERROR_INVALIDDATA; + + init_get_bits(gb, pkt->data, pkt->size * 8); + + if ((ret = ff_tak_decode_frame_header(avctx, gb, &s->ti, 0)) < 0) + return ret; + + if (s->ti.flags & TAK_FRAME_FLAG_HAS_METADATA) { + av_log_missing_feature(avctx, "frame metadata", 1); + return AVERROR_PATCHWELCOME; + } + + hsize = get_bits_count(gb) / 8; + if (avctx->err_recognition & AV_EF_CRCCHECK) { + if (ff_tak_check_crc(pkt->data, hsize)) { + av_log(avctx, AV_LOG_ERROR, "CRC error\n"); + return AVERROR_INVALIDDATA; + } + } + + if (s->ti.codec != TAK_CODEC_MONO_STEREO && + s->ti.codec != TAK_CODEC_MULTICHANNEL) { + av_log(avctx, AV_LOG_ERROR, "unsupported codec: %d\n", s->ti.codec); + return AVERROR_PATCHWELCOME; + } + if (s->ti.data_type) { + av_log(avctx, AV_LOG_ERROR, + "unsupported data type: %d\n", s->ti.data_type); + return AVERROR_INVALIDDATA; + } + if (s->ti.codec == TAK_CODEC_MONO_STEREO && s->ti.channels > 2) { + av_log(avctx, AV_LOG_ERROR, + "invalid number of channels: %d\n", s->ti.channels); + return AVERROR_INVALIDDATA; + } + if (s->ti.channels > 6) { + av_log(avctx, AV_LOG_ERROR, + "unsupported number of channels: %d\n", s->ti.channels); + return AVERROR_INVALIDDATA; + } + + if (s->ti.frame_samples <= 0) { + av_log(avctx, AV_LOG_ERROR, "unsupported/invalid number of samples\n"); + return AVERROR_INVALIDDATA; + } + + if (s->ti.bps != avctx->bits_per_coded_sample) { + avctx->bits_per_coded_sample = s->ti.bps; + if ((ret = set_bps_params(avctx)) < 0) + return ret; + } + if (s->ti.sample_rate != avctx->sample_rate) { + avctx->sample_rate = s->ti.sample_rate; + set_sample_rate_params(avctx); + } + if (s->ti.ch_layout) + avctx->channel_layout = s->ti.ch_layout; + avctx->channels = s->ti.channels; + + s->nb_samples = s->ti.last_frame_samples ? s->ti.last_frame_samples + : s->ti.frame_samples; + + s->frame.nb_samples = s->nb_samples; + if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) + return ret; + + if (avctx->bits_per_coded_sample <= 16) { + int buf_size = av_samples_get_buffer_size(NULL, avctx->channels, + s->nb_samples, + AV_SAMPLE_FMT_S32P, 0); + av_fast_malloc(&s->decode_buffer, &s->decode_buffer_size, buf_size); + if (!s->decode_buffer) + return AVERROR(ENOMEM); + ret = av_samples_fill_arrays((uint8_t **)s->decoded, NULL, + s->decode_buffer, avctx->channels, + s->nb_samples, AV_SAMPLE_FMT_S32P, 0); + if (ret < 0) + return ret; + } else { + for (chan = 0; chan < avctx->channels; chan++) + s->decoded[chan] = (int32_t *)s->frame.extended_data[chan]; + } + + if (s->nb_samples < 16) { + for (chan = 0; chan < avctx->channels; chan++) { + int32_t *decoded = s->decoded[chan]; + for (i = 0; i < s->nb_samples; i++) + decoded[i] = get_sbits(gb, avctx->bits_per_coded_sample); + } + } else { + if (s->ti.codec == TAK_CODEC_MONO_STEREO) { + for (chan = 0; chan < avctx->channels; chan++) + if (ret = decode_channel(s, chan)) + return ret; + + if (avctx->channels == 2) { + if (get_bits1(gb)) { + // some kind of subframe length, but it seems to be unused + skip_bits(gb, 6); + } + + s->dmode = get_bits(gb, 3); + if (ret = decorrelate(s, 0, 1, s->nb_samples - 1)) + return ret; + } + } else if (s->ti.codec == TAK_CODEC_MULTICHANNEL) { + if (get_bits1(gb)) { + int ch_mask = 0; + + chan = get_bits(gb, 4) + 1; + if (chan > avctx->channels) + return AVERROR_INVALIDDATA; + + for (i = 0; i < chan; i++) { + int nbit = get_bits(gb, 4); + + if (nbit >= avctx->channels) + return AVERROR_INVALIDDATA; + + if (ch_mask & 1 << nbit) + return AVERROR_INVALIDDATA; + + s->mcdparams[i].present = get_bits1(gb); + if (s->mcdparams[i].present) { + s->mcdparams[i].index = get_bits(gb, 2); + s->mcdparams[i].chan2 = get_bits(gb, 4); + if (s->mcdparams[i].index == 1) { + if ((nbit == s->mcdparams[i].chan2) || + (ch_mask & 1 << s->mcdparams[i].chan2)) + return AVERROR_INVALIDDATA; + + ch_mask |= 1 << s->mcdparams[i].chan2; + } else if (!(ch_mask & 1 << s->mcdparams[i].chan2)) { + return AVERROR_INVALIDDATA; + } + } + s->mcdparams[i].chan1 = nbit; + + ch_mask |= 1 << nbit; + } + } else { + chan = avctx->channels; + for (i = 0; i < chan; i++) { + s->mcdparams[i].present = 0; + s->mcdparams[i].chan1 = i; + } + } + + for (i = 0; i < chan; i++) { + if (s->mcdparams[i].present && s->mcdparams[i].index == 1) + if (ret = decode_channel(s, s->mcdparams[i].chan2)) + return ret; + + if (ret = decode_channel(s, s->mcdparams[i].chan1)) + return ret; + + if (s->mcdparams[i].present) { + s->dmode = mc_dmodes[s->mcdparams[i].index]; + if (ret = decorrelate(s, + s->mcdparams[i].chan2, + s->mcdparams[i].chan1, + s->nb_samples - 1)) + return ret; + } + } + } + + for (chan = 0; chan < avctx->channels; chan++) { + int32_t *decoded = s->decoded[chan]; + + if (s->lpc_mode[chan]) + decode_lpc(decoded, s->lpc_mode[chan], s->nb_samples); + + if (s->sample_shift[chan] > 0) + for (i = 0; i < s->nb_samples; i++) + decoded[i] <<= s->sample_shift[chan]; + } + } + + align_get_bits(gb); + skip_bits(gb, 24); + if (get_bits_left(gb) < 0) + av_log(avctx, AV_LOG_DEBUG, "overread\n"); + else if (get_bits_left(gb) > 0) + av_log(avctx, AV_LOG_DEBUG, "underread\n"); + + if (avctx->err_recognition & AV_EF_CRCCHECK) { + if (ff_tak_check_crc(pkt->data + hsize, + get_bits_count(gb) / 8 - hsize)) { + av_log(avctx, AV_LOG_ERROR, "CRC error\n"); + return AVERROR_INVALIDDATA; + } + } + + /* convert to output buffer */ + switch (avctx->sample_fmt) { + case AV_SAMPLE_FMT_U8P: + for (chan = 0; chan < avctx->channels; chan++) { + uint8_t *samples = (uint8_t *)s->frame.extended_data[chan]; + int32_t *decoded = s->decoded[chan]; + for (i = 0; i < s->nb_samples; i++) + samples[i] = decoded[i] + 0x80; + } + break; + case AV_SAMPLE_FMT_S16P: + for (chan = 0; chan < avctx->channels; chan++) { + int16_t *samples = (int16_t *)s->frame.extended_data[chan]; + int32_t *decoded = s->decoded[chan]; + for (i = 0; i < s->nb_samples; i++) + samples[i] = decoded[i]; + } + break; + case AV_SAMPLE_FMT_S32P: + for (chan = 0; chan < avctx->channels; chan++) { + int32_t *samples = (int32_t *)s->frame.extended_data[chan]; + for (i = 0; i < s->nb_samples; i++) + samples[i] <<= 8; + } + break; + } + + *got_frame_ptr = 1; + *(AVFrame *)data = s->frame; + + return pkt->size; +} + +static av_cold int tak_decode_close(AVCodecContext *avctx) +{ + TAKDecContext *s = avctx->priv_data; + + av_freep(&s->decode_buffer); + av_freep(&s->residues); + + return 0; +} + +AVCodec ff_tak_decoder = { + .name = "tak", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_TAK, + .priv_data_size = sizeof(TAKDecContext), + .init = tak_decode_init, + .init_static_data = tak_init_static_data, + .close = tak_decode_close, + .decode = tak_decode_frame, + .capabilities = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"), + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_NONE }, +}; 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