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FFmpeg/libavcodec/takdec.c
Alexandra Hájková 381a4e31a6 tak: Convert to the new bitstream reader
2017-01-25 11:06:58 +01:00

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/*
* 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/internal.h"
#include "libavutil/samplefmt.h"
#define BITSTREAM_READER_LE
#include "audiodsp.h"
#include "avcodec.h"
#include "bitstream.h"
#include "internal.h"
#include "unary.h"
#include "tak.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
AudioDSPContext adsp;
TAKStreamInfo ti;
BitstreamContext bc; // 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_audiodsp_init(&s->adsp);
s->avctx = avctx;
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(BitstreamContext *bc, 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 = bitstream_read(bc, code.init);
if (x >= code.escape && bitstream_read_bit(bc)) {
x |= 1 << code.init;
if (x >= code.aescape) {
int scale = get_unary(bc, 1, 9);
if (scale == 9) {
int scale_bits = bitstream_read(bc, 3);
if (scale_bits > 0) {
if (scale_bits == 7) {
scale_bits += bitstream_read(bc, 5);
if (scale_bits > 29)
return AVERROR_INVALIDDATA;
}
scale = bitstream_read(bc, 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)
{
BitstreamContext *bc = &s->bc;
int i, mode, ret;
if (length > s->nb_samples)
return AVERROR_INVALIDDATA;
if (bitstream_read_bit(bc)) {
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 = bitstream_read(bc, 6);
for (i = 1; i < wlength; i++) {
int c = get_unary(bc, 1, 6);
switch (c) {
case 6:
mode = bitstream_read(bc, 6);
break;
case 5:
case 4:
case 3: {
/* mode += sign ? (1 - c) : (c - 1) */
int sign = bitstream_read_bit(bc);
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(bc, mode, decoded, len)) < 0)
return ret;
decoded += len;
}
} else {
mode = bitstream_read(bc, 6);
if ((ret = decode_segment(bc, mode, decoded, length)) < 0)
return ret;
}
return 0;
}
static int bits_esc4(BitstreamContext *bc)
{
if (bitstream_read_bit(bc))
return bitstream_read(bc, 4) + 1;
else
return 0;
}
static void decode_filter_coeffs(TAKDecContext *s, int filter_order, int size,
int filter_quant, int16_t *filter)
{
BitstreamContext *bc = &s->bc;
int i, j, a, b;
int filter_tmp[MAX_PREDICTORS];
int16_t predictors[MAX_PREDICTORS];
predictors[0] = bitstream_read_signed(bc, 10);
predictors[1] = bitstream_read_signed(bc, 10);
predictors[2] = bitstream_read_signed(bc, size) << (10 - size);
predictors[3] = bitstream_read_signed(bc, size) << (10 - size);
if (filter_order > 4) {
int av_uninit(code_size);
int code_size_base = size - bitstream_read_bit(bc);
for (i = 4; i < filter_order; i++) {
if (!(i & 3))
code_size = code_size_base - bitstream_read(bc, 2);
predictors[i] = bitstream_read_signed(bc, 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]);
BitstreamContext *bc = &s->bc;
int i, ret;
int dshift, size, filter_quant, filter_order;
memset(filter, 0, MAX_PREDICTORS * sizeof(*filter));
if (!bitstream_read_bit(bc))
return decode_residues(s, decoded, subframe_size);
filter_order = predictor_sizes[bitstream_read(bc, 4)];
if (prev_subframe_size > 0 && bitstream_read_bit(bc)) {
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 = bitstream_read(bc, 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 = bits_esc4(bc);
size = bitstream_read_bit(bc) + 6;
filter_quant = 10;
if (bitstream_read_bit(bc)) {
filter_quant -= bitstream_read(bc, 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->adsp.scalarproduct_int16(&s->residues[i], filter,
FFALIGN(filter_order, 16));
v = (av_clip_intp2(v >> filter_quant, 13) << 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;
BitstreamContext *bc = &s->bc;
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] = bits_esc4(bc);
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 bitstream_read_signed()
* instead of skipping it produces bit-exact output. This has been
* reported to the TAK author. */
*decoded++ = bitstream_read_signed(bc,
avctx->bits_per_coded_sample -
s->sample_shift[chan]);
s->lpc_mode[chan] = bitstream_read(bc, 2);
nb_subframes = bitstream_read(bc, 3) + 1;
i = 0;
if (nb_subframes > 1) {
if (bitstream_bits_left(bc) < (nb_subframes - 1) * 6)
return AVERROR_INVALIDDATA;
prev = 0;
for (; i < nb_subframes - 1; i++) {
int subframe_end = bitstream_read(bc, 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)
{
BitstreamContext *bc = &s->bc;
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 = bits_esc4(bc);
dfactor = bitstream_read_signed(bc, 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]);
int length2, order_half, filter_order, dval1, dval2;
int av_uninit(code_size);
memset(filter, 0, MAX_PREDICTORS * sizeof(*filter));
if (length < 256)
return AVERROR_INVALIDDATA;
dshift = bits_esc4(bc);
filter_order = 8 << bitstream_read_bit(bc);
dval1 = bitstream_read_bit(bc);
dval2 = bitstream_read_bit(bc);
for (i = 0; i < filter_order; i++) {
if (!(i & 3))
code_size = 14 - bitstream_read(bc, 3);
filter[i] = bitstream_read_signed(bc, 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->adsp.scalarproduct_int16(&s->residues[i], filter,
FFALIGN(filter_order, 16));
p1[i] = (av_clip_intp2(v >> 10, 13) << 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;
AVFrame *frame = data;
BitstreamContext *bc = &s->bc;
int chan, i, ret, hsize;
if (pkt->size < TAK_MIN_FRAME_HEADER_BYTES)
return AVERROR_INVALIDDATA;
bitstream_init8(bc, pkt->data, pkt->size);
if ((ret = ff_tak_decode_frame_header(avctx, bc, &s->ti, 0)) < 0)
return ret;
if (s->ti.flags & TAK_FRAME_FLAG_HAS_METADATA) {
avpriv_request_sample(avctx, "Frame metadata");
return AVERROR_PATCHWELCOME;
}
hsize = bitstream_tell(bc) / 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");
if (avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
}
if (s->ti.codec != TAK_CODEC_MONO_STEREO &&
s->ti.codec != TAK_CODEC_MULTICHANNEL) {
avpriv_report_missing_feature(avctx, "TAK codec type %d", 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;
frame->nb_samples = s->nb_samples;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 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);
if (buf_size < 0)
return buf_size;
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 *)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] = bitstream_read_signed(bc, 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 (bitstream_read_bit(bc)) {
// some kind of subframe length, but it seems to be unused
bitstream_skip(bc, 6);
}
s->dmode = bitstream_read(bc, 3);
if (ret = decorrelate(s, 0, 1, s->nb_samples - 1))
return ret;
}
} else if (s->ti.codec == TAK_CODEC_MULTICHANNEL) {
if (bitstream_read_bit(bc)) {
int ch_mask = 0;
chan = bitstream_read(bc, 4) + 1;
if (chan > avctx->channels)
return AVERROR_INVALIDDATA;
for (i = 0; i < chan; i++) {
int nbit = bitstream_read(bc, 4);
if (nbit >= avctx->channels)
return AVERROR_INVALIDDATA;
if (ch_mask & 1 << nbit)
return AVERROR_INVALIDDATA;
s->mcdparams[i].present = bitstream_read_bit(bc);
if (s->mcdparams[i].present) {
s->mcdparams[i].index = bitstream_read(bc, 2);
s->mcdparams[i].chan2 = bitstream_read(bc, 4);
if (s->mcdparams[i].chan2 >= avctx->channels) {
av_log(avctx, AV_LOG_ERROR,
"invalid channel 2 (%d) for %d channel(s)\n",
s->mcdparams[i].chan2, avctx->channels);
return AVERROR_INVALIDDATA;
}
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];
}
}
bitstream_align(bc);
bitstream_skip(bc, 24);
if (bitstream_bits_left(bc) < 0)
av_log(avctx, AV_LOG_DEBUG, "overread\n");
else if (bitstream_bits_left(bc) > 0)
av_log(avctx, AV_LOG_DEBUG, "underread\n");
if (avctx->err_recognition & AV_EF_CRCCHECK) {
if (ff_tak_check_crc(pkt->data + hsize,
bitstream_tell(bc) / 8 - hsize)) {
av_log(avctx, AV_LOG_ERROR, "CRC error\n");
if (avctx->err_recognition & AV_EF_EXPLODE)
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 *)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 *)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 *)frame->extended_data[chan];
for (i = 0; i < s->nb_samples; i++)
samples[i] <<= 8;
}
break;
}
*got_frame_ptr = 1;
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",
.long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
.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 = AV_CODEC_CAP_DR1,
.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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