Commit d7a47392 authored by Paul B Mahol's avatar Paul B Mahol

TAK demuxer, decoder and parser

Signed-off-by: 's avatarPaul B Mahol <onemda@gmail.com>
parent 208a5d13
......@@ -8,6 +8,7 @@ version next:
- Opus encoder using libopus
- ffprobe -select_streams option
- Pinnacle TARGA CineWave YUV16 decoder
- TAK demuxer, decoder and parser
version 1.0:
......
......@@ -1824,6 +1824,7 @@ sap_muxer_select="rtp_muxer rtp_protocol"
sdp_demuxer_select="rtpdec"
smoothstreaming_muxer_select="ismv_muxer"
spdif_muxer_select="aac_parser"
tak_demuxer_select="tak_parser"
tg2_muxer_select="mov_muxer"
tgp_muxer_select="mov_muxer"
w64_demuxer_deps="wav_demuxer"
......
......@@ -304,6 +304,7 @@ library:
@item raw video @tab X @tab X
@item raw id RoQ @tab X @tab
@item raw Shorten @tab @tab X
@item raw TAK @tab @tab X
@item raw TrueHD @tab X @tab X
@item raw VC-1 @tab @tab X
@item raw PCM A-law @tab X @tab X
......@@ -863,6 +864,7 @@ following image formats are supported:
@tab experimental codec
@item Speex @tab E @tab E
@tab supported through external library libspeex
@item TAK (Tom's lossless Audio Kompressor) @tab @tab X
@item True Audio (TTA) @tab @tab X
@item TrueHD @tab @tab X
@tab Used in HD-DVD and Blu-Ray discs.
......
......@@ -400,6 +400,7 @@ OBJS-$(CONFIG_SVQ3_DECODER) += svq3.o svq13.o h263.o h264.o \
h264_loopfilter.o h264_direct.o \
h264_sei.o h264_ps.o h264_refs.o \
h264_cavlc.o h264_cabac.o cabac.o
OBJS-$(CONFIG_TAK_DECODER) += takdec.o tak.o
OBJS-$(CONFIG_TARGA_DECODER) += targa.o
OBJS-$(CONFIG_TARGA_ENCODER) += targaenc.o rle.o
OBJS-$(CONFIG_TARGA_Y216_DECODER) += targa_y216dec.o
......@@ -713,6 +714,7 @@ OBJS-$(CONFIG_MPEGVIDEO_PARSER) += mpegvideo_parser.o \
OBJS-$(CONFIG_PNM_PARSER) += pnm_parser.o pnm.o
OBJS-$(CONFIG_RV30_PARSER) += rv34_parser.o
OBJS-$(CONFIG_RV40_PARSER) += rv34_parser.o
OBJS-$(CONFIG_TAK_PARSER) += tak_parser.o tak.o
OBJS-$(CONFIG_VC1_PARSER) += vc1_parser.o vc1.o vc1data.o \
msmpeg4.o msmpeg4data.o mpeg4video.o \
h263.o
......
......@@ -328,6 +328,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (SMACKAUD, smackaud);
REGISTER_ENCDEC (SONIC, sonic);
REGISTER_ENCODER (SONIC_LS, sonic_ls);
REGISTER_DECODER (TAK, tak);
REGISTER_DECODER (TRUEHD, truehd);
REGISTER_DECODER (TRUESPEECH, truespeech);
REGISTER_DECODER (TTA, tta);
......@@ -486,6 +487,7 @@ void avcodec_register_all(void)
REGISTER_PARSER (PNM, pnm);
REGISTER_PARSER (RV30, rv30);
REGISTER_PARSER (RV40, rv40);
REGISTER_PARSER (TAK, tak);
REGISTER_PARSER (VC1, vc1);
REGISTER_PARSER (VORBIS, vorbis);
REGISTER_PARSER (VP3, vp3);
......
......@@ -431,6 +431,7 @@ enum AVCodecID {
AV_CODEC_ID_SONIC_LS = MKBETAG('S','O','N','L'),
AV_CODEC_ID_PAF_AUDIO = MKBETAG('P','A','F','A'),
AV_CODEC_ID_OPUS = MKBETAG('O','P','U','S'),
AV_CODEC_ID_TAK = MKBETAG('t','B','a','K'),
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.
......
......@@ -2252,6 +2252,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("Opus (Opus Interactive Audio Codec)"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_TAK,
.type = AVMEDIA_TYPE_AUDIO,
.name = "tak",
.long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
.props = AV_CODEC_PROP_LOSSLESS,
},
/* subtitle codecs */
{
......
/*
* TAK common code
* Copyright (c) 2012 Paul B Mahol
*
* 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
*/
#include "tak.h"
#include "libavutil/crc.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/bswap.h"
static const int64_t tak_channel_layouts[] = {
0,
AV_CH_FRONT_LEFT,
AV_CH_FRONT_RIGHT,
AV_CH_FRONT_CENTER,
AV_CH_LOW_FREQUENCY,
AV_CH_BACK_LEFT,
AV_CH_BACK_RIGHT,
AV_CH_FRONT_LEFT_OF_CENTER,
AV_CH_FRONT_RIGHT_OF_CENTER,
AV_CH_BACK_CENTER,
AV_CH_SIDE_LEFT,
AV_CH_SIDE_RIGHT,
AV_CH_TOP_CENTER,
AV_CH_TOP_FRONT_LEFT,
AV_CH_TOP_FRONT_CENTER,
AV_CH_TOP_FRONT_RIGHT,
AV_CH_TOP_BACK_LEFT,
AV_CH_TOP_BACK_CENTER,
AV_CH_TOP_BACK_RIGHT,
};
static const uint16_t frame_duration_type_quants[] = {
3, 4, 6, 8, 4096, 8192, 16384, 512, 1024, 2048,
};
static int tak_get_nb_samples(int sample_rate, enum TAKFrameSizeType type)
{
int nb_samples, max_nb_samples;
if (type <= TAK_FST_250ms) {
nb_samples = sample_rate * frame_duration_type_quants[type] >>
TAK_FRAME_DURATION_QUANT_SHIFT;
max_nb_samples = 16384;
} else if (type < FF_ARRAY_ELEMS(frame_duration_type_quants)) {
nb_samples = frame_duration_type_quants[type];
max_nb_samples = sample_rate * frame_duration_type_quants[TAK_FST_250ms] >>
TAK_FRAME_DURATION_QUANT_SHIFT;
} else {
return AVERROR_INVALIDDATA;
}
if (nb_samples <= 0 || nb_samples > max_nb_samples)
return AVERROR_INVALIDDATA;
return nb_samples;
}
static int crc_init = 0;
#if CONFIG_SMALL
#define CRC_TABLE_SIZE 257
#else
#define CRC_TABLE_SIZE 1024
#endif
static AVCRC crc_24[CRC_TABLE_SIZE];
av_cold void ff_tak_init_crc(void)
{
if (!crc_init) {
av_crc_init(crc_24, 0, 24, 0x864CFBU, sizeof(crc_24));
crc_init = 1;
}
}
int ff_tak_check_crc(const uint8_t *buf, unsigned int buf_size)
{
uint32_t crc, CRC;
if (buf_size < 4)
return AVERROR_INVALIDDATA;
buf_size -= 3;
CRC = av_bswap32(AV_RL24(buf + buf_size)) >> 8;
crc = av_crc(crc_24, 0xCE04B7U, buf, buf_size);
if (CRC != crc)
return AVERROR_INVALIDDATA;
return 0;
}
void avpriv_tak_parse_streaminfo(GetBitContext *gb, TAKStreamInfo *s)
{
uint64_t channel_mask = 0;
int frame_type, i;
s->codec = get_bits(gb, TAK_ENCODER_CODEC_BITS);
skip_bits(gb, TAK_ENCODER_PROFILE_BITS);
frame_type = get_bits(gb, TAK_SIZE_FRAME_DURATION_BITS);
s->samples = get_bits_longlong(gb, TAK_SIZE_SAMPLES_NUM_BITS);
s->data_type = get_bits(gb, TAK_FORMAT_DATA_TYPE_BITS);
s->sample_rate = get_bits(gb, TAK_FORMAT_SAMPLE_RATE_BITS) + TAK_SAMPLE_RATE_MIN;
s->bps = get_bits(gb, TAK_FORMAT_BPS_BITS) + TAK_BPS_MIN;
s->channels = get_bits(gb, TAK_FORMAT_CHANNEL_BITS) + TAK_CHANNELS_MIN;
if (get_bits1(gb)) {
skip_bits(gb, TAK_FORMAT_VALID_BITS);
if (get_bits1(gb)) {
for (i = 0; i < s->channels; i++) {
int value = get_bits(gb, TAK_FORMAT_CH_LAYOUT_BITS);
if (value < FF_ARRAY_ELEMS(tak_channel_layouts))
channel_mask |= tak_channel_layouts[value];
}
}
}
s->ch_layout = channel_mask;
s->frame_samples = tak_get_nb_samples(s->sample_rate, frame_type);
}
#define FRAME_IS_LAST 1
#define FRAME_HAVE_INFO 2
#define FRAME_HAVE_METADATA 4
int ff_tak_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
TAKStreamInfo *ti, int log_level_offset)
{
int flags;
if (get_bits(gb, TAK_FRAME_HEADER_SYNC_ID_BITS) != TAK_FRAME_HEADER_SYNC_ID) {
av_log(avctx, AV_LOG_ERROR + log_level_offset, "missing sync id\n");
return AVERROR_INVALIDDATA;
}
flags = get_bits(gb, TAK_FRAME_HEADER_FLAGS_BITS);
ti->frame_num = get_bits(gb, TAK_FRAME_HEADER_NO_BITS);
if (flags & FRAME_IS_LAST) {
ti->last_frame_samples = get_bits(gb, TAK_FRAME_HEADER_SAMPLE_COUNT_BITS) + 1;
skip_bits(gb, 2);
} else {
ti->last_frame_samples = 0;
}
if (flags & FRAME_HAVE_INFO) {
avpriv_tak_parse_streaminfo(gb, ti);
if (get_bits(gb, 6))
skip_bits(gb, 25);
align_get_bits(gb);
}
if (flags & FRAME_HAVE_METADATA)
return AVERROR_INVALIDDATA;
skip_bits(gb, 24);
return 0;
}
/*
* TAK decoder/demuxer common code
* Copyright (c) 2012 Paul B Mahol
*
* 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
* TAK (Tom's lossless Audio Kompressor) decoder/demuxer common functions
*/
#ifndef AVCODEC_TAK_H
#define AVCODEC_TAK_H
#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "avcodec.h"
#define TAK_FORMAT_DATA_TYPE_BITS 3
#define TAK_FORMAT_SAMPLE_RATE_BITS 18
#define TAK_FORMAT_BPS_BITS 5
#define TAK_FORMAT_CHANNEL_BITS 4
#define TAK_FORMAT_VALID_BITS 5
#define TAK_FORMAT_CH_LAYOUT_BITS 6
#define TAK_SIZE_FRAME_DURATION_BITS 4
#define TAK_SIZE_SAMPLES_NUM_BITS 35
#define TAK_LAST_FRAME_POS_BITS 40
#define TAK_LAST_FRAME_SIZE_BITS 24
#define TAK_ENCODER_CODEC_BITS 6
#define TAK_ENCODER_PROFILE_BITS 4
#define TAK_ENCODER_VERSION_BITS 24
#define TAK_SAMPLE_RATE_MIN 6000
#define TAK_CHANNELS_MIN 1
#define TAK_BPS_MIN 8
#define TAK_FRAME_HEADER_FLAGS_BITS 3
#define TAK_FRAME_HEADER_SYNC_ID 0xA0FF
#define TAK_FRAME_HEADER_SYNC_ID_BITS 16
#define TAK_FRAME_HEADER_SAMPLE_COUNT_BITS 14
#define TAK_FRAME_HEADER_NO_BITS 21
#define TAK_FRAME_DURATION_QUANT_SHIFT 5
#define TAK_CRC24_BITS 24
#define TAK_MAX_CHANNELS ( 1 << TAK_FORMAT_CHANNEL_BITS )
#define TAK_MIN_FRAME_HEADER_BITS ( TAK_FRAME_HEADER_SYNC_ID_BITS + \
TAK_FRAME_HEADER_FLAGS_BITS + \
TAK_FRAME_HEADER_NO_BITS + \
TAK_CRC24_BITS )
#define TAK_MIN_FRAME_HEADER_LAST_BITS ( TAK_MIN_FRAME_HEADER_BITS + 2 + \
TAK_FRAME_HEADER_SAMPLE_COUNT_BITS )
#define TAK_ENCODER_BITS ( TAK_ENCODER_CODEC_BITS + \
TAK_ENCODER_PROFILE_BITS )
#define TAK_SIZE_BITS ( TAK_SIZE_SAMPLES_NUM_BITS + \
TAK_SIZE_FRAME_DURATION_BITS )
#define TAK_FORMAT_BITS ( TAK_FORMAT_DATA_TYPE_BITS + \
TAK_FORMAT_SAMPLE_RATE_BITS + \
TAK_FORMAT_BPS_BITS + \
TAK_FORMAT_CHANNEL_BITS + 1 + \
TAK_FORMAT_VALID_BITS + 1 + \
TAK_FORMAT_CH_LAYOUT_BITS * \
TAK_MAX_CHANNELS )
#define TAK_STREAMINFO_BITS ( TAK_ENCODER_BITS + \
TAK_SIZE_BITS + \
TAK_FORMAT_BITS )
#define TAK_MAX_FRAME_HEADER_BITS ( TAK_MIN_FRAME_HEADER_LAST_BITS + \
TAK_STREAMINFO_BITS + 31 )
#define TAK_STREAMINFO_BYTES (( TAK_STREAMINFO_BITS + 7 ) / 8)
#define TAK_MAX_FRAME_HEADER_BYTES (( TAK_MAX_FRAME_HEADER_BITS + 7 ) / 8)
#define TAK_MIN_FRAME_HEADER_BYTES (( TAK_MIN_FRAME_HEADER_BITS + 7 ) / 8)
enum TAKMetaDataType {
TAK_METADATA_END = 0,
TAK_METADATA_STREAMINFO,
TAK_METADATA_SEEKTABLE,
TAK_METADATA_SIMPLE_WAVE_DATA,
TAK_METADATA_ENCODER,
TAK_METADATA_PADDING,
TAK_METADATA_MD5,
TAK_METADATA_LAST_FRAME,
};
enum TAKFrameSizeType {
TAK_FST_94ms = 0,
TAK_FST_125ms,
TAK_FST_188ms,
TAK_FST_250ms,
TAK_FST_4096,
TAK_FST_8192,
TAK_FST_16384,
TAK_FST_512,
TAK_FST_1024,
TAK_FST_2048,
};
typedef struct TAKStreamInfo {
int codec;
int data_type;
int sample_rate;
int channels;
int bps;
int frame_num;
int frame_samples;
int last_frame_samples;
uint64_t ch_layout;
int64_t samples;
} TAKStreamInfo;
void ff_tak_init_crc(void);
int ff_tak_check_crc(const uint8_t *buf, unsigned int buf_size);
/**
* Parse the Streaminfo metadata block
* @param[in] gb pointer to GetBitContext
* @param[out] s where parsed information is stored
*/
void avpriv_tak_parse_streaminfo(GetBitContext *gb, TAKStreamInfo *s);
/**
* Validate and decode a frame header.
* @param avctx AVCodecContext to use as av_log() context
* @param[in] gb GetBitContext from which to read frame header
* @param[out] s frame information
* @param log_level_offset log level offset. can be used to silence error messages.
* @return non-zero on error, 0 if ok
*/
int ff_tak_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
TAKStreamInfo *s, int log_level_offset);
#endif /* AVCODEC_TAK_H */
/*
* TAK parser
* Copyright (c) 2012 Michael Niedermayer
*
* 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
* TAK parser
**/
#include "parser.h"
#include "tak.h"
typedef struct TAKParseContext {
ParseContext pc;
TAKStreamInfo ti;
int index;
} TAKParseContext;
static av_cold int tak_init(AVCodecParserContext *s)
{
ff_tak_init_crc();
return 0;
}
static int tak_parse(AVCodecParserContext *s, AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
TAKParseContext *t = s->priv_data;
ParseContext *pc = &t->pc;
int next = END_NOT_FOUND;
GetBitContext gb;
int consumed = 0;
int needed = buf_size ? TAK_MAX_FRAME_HEADER_BYTES : 8;
if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) {
TAKStreamInfo ti;
init_get_bits(&gb, buf, buf_size);
if (!ff_tak_decode_frame_header(avctx, &gb, &ti, 127))
s->duration = t->ti.last_frame_samples ? t->ti.last_frame_samples :
t->ti.frame_samples;
*poutbuf = buf;
*poutbuf_size = buf_size;
return buf_size;
}
while (buf_size || t->index + needed <= pc->index) {
if (buf_size && t->index + TAK_MAX_FRAME_HEADER_BYTES > pc->index) {
int tmp_buf_size = FFMIN(2 * TAK_MAX_FRAME_HEADER_BYTES, buf_size);
const uint8_t *tmp_buf = buf;
ff_combine_frame(pc, END_NOT_FOUND, &tmp_buf, &tmp_buf_size);
consumed += tmp_buf_size;
buf += tmp_buf_size;
buf_size -= tmp_buf_size;
}
for (; t->index + needed <= pc->index; t->index++) {
if (pc->buffer[ t->index ] == 0xFF &&
pc->buffer[ t->index + 1 ] == 0xA0) {
TAKStreamInfo ti;
init_get_bits(&gb, pc->buffer + t->index,
8 * (pc->index - t->index));
if (!ff_tak_decode_frame_header(avctx, &gb,
pc->frame_start_found ? &ti : &t->ti, 127) &&
!ff_tak_check_crc(pc->buffer + t->index,
get_bits_count(&gb) / 8)) {
if (!pc->frame_start_found) {
pc->frame_start_found = 1;
s->duration = t->ti.last_frame_samples ?
t->ti.last_frame_samples :
t->ti.frame_samples;
} else {
pc->frame_start_found = 0;
next = t->index - pc->index;
t->index = 0;
goto found;
}
}
}
}
}
found:
if (consumed && !buf_size && next == END_NOT_FOUND ||
ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size + consumed;
}
if (next != END_NOT_FOUND) {
next += consumed;
pc->overread = FFMAX(0, -next);
}
*poutbuf = buf;
*poutbuf_size = buf_size;
return next;
}
AVCodecParser ff_tak_parser = {
.codec_ids = { AV_CODEC_ID_TAK },
.priv_data_size = sizeof(TAKParseContext),
.parser_init = tak_init,
.parser_parse = tak_parse,
.parser_close = ff_parse_close,
};
/*
* TAK decoder
* Copyright (c) 2012 Paul B Mahol
*
* 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
* TAK (Tom's lossless Audio Kompressor) decoder
* @author Paul B Mahol
*/
#include "tak.h"
#include "avcodec.h"
#include "unary.h"
#include "dsputil.h"
#define MAX_SUBFRAMES 8 ///< max number of subframes per channel
#define MAX_PREDICTORS 256
typedef struct MCDParam {
int8_t present; ///< is decorrelation parameters available 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 nb_samples; ///< number of samples in the current frame
int32_t *decode_buffer;
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
int32_t xred;
int size;
int ared;
int filter_order;
int16_t predictors[MAX_PREDICTORS];
int nb_subframes; ///< number of subframes in the current frame
int16_t subframe_len[MAX_SUBFRAMES]; ///< subframe length in samples
int subframe_scale;
int8_t dmode; ///< channel decorrelation type in the current frame
int8_t dshift;
int16_t dfactor;
int8_t dval1;
int8_t dval2;
MCDParam mcdparams[TAK_MAX_CHANNELS]; ///< multichannel decorrelation parameters
int wlength;
int uval;
int rval;
int8_t coding_mode[128];
DECLARE_ALIGNED(16, int16_t, filter)[MAX_PREDICTORS];
DECLARE_ALIGNED(16, int16_t, residues)[544];
} 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 void tak_set_bps(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;
}
}
static int get_shift(int sample_rate)
{
int shift;
if (sample_rate < 11025)
shift = 3;
else if (sample_rate < 22050)
shift = 2;
else if (sample_rate < 44100)
shift = 1;
else
shift = 0;
return shift;
}
static int get_scale(int sample_rate, int shift)
{
return FFALIGN(sample_rate + 511 >> 9, 4) << shift;
}
static av_cold int tak_decode_init(AVCodecContext *avctx)
{
TAKDecContext *s = avctx->priv_data;
ff_tak_init_crc();
ff_dsputil_init(&s->dsp, avctx);
s->avctx = avctx;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
s->uval = get_scale(avctx->sample_rate, get_shift(avctx->sample_rate));
s->subframe_scale = get_scale(avctx->sample_rate, 1);
tak_set_bps(avctx);
return 0;
}
static int get_code(GetBitContext *gb, int nbits)
{
if (nbits == 1) {
skip_bits1(gb);
return 0;
} else {
return get_sbits(gb, nbits);
}
}
static void decode_lpc(int32_t *coeffs, int mode, int length)
{
int i, a1, a2, a3, a4, a5;
if (length < 2)
return;
if (mode == 1) {
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) {
a1 = coeffs[1];
a2 = a1 + *coeffs;
coeffs[1] = a2;
if (length > 2) {
coeffs += 2;
for (i = 0; i < (length - 2 >> 1); i++) {
a3 = *coeffs + a1;
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) {
a1 = coeffs[1];
a2 = a1 + *coeffs;
coeffs[1] = a2;
if (length > 2) {
a3 = coeffs[2];
a4 = a3 + a1;
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(TAKDecContext *s, int8_t value, int32_t *dst, int len)
{
GetBitContext *gb = &s->gb;
if (!value) {
memset(dst, 0, len * 4);
} else {
int x, y, z, i = 0;
value--;
do {
while (1) {
x = get_bits_long(gb, xcodes[value].init);
if (x >= xcodes[value].escape)
break;
dst[i++] = (x >> 1) ^ -(x & 1);
if (i >= len)
return 0;
}
y = get_bits1(gb);
x = (y << xcodes[value].init) | x;
if (x >= xcodes[value].aescape) {
int c = get_unary(gb, 1, 9);
if (c == 9) {
int d;
z = x + xcodes[value].bias;
d = get_bits(gb, 3);
if (d == 7) {
d = get_bits(gb, 5) + 7;
if (d > 29)
return AVERROR_INVALIDDATA;
}
if (d)
z += xcodes[value].scale * (get_bits_long(gb, d) + 1);
} else {
z = xcodes[value].scale * c + x - xcodes[value].escape;
}
} else {
z = x - (xcodes[value].escape & -y);
}
dst[i++] = (z >> 1) ^ -(z & 1);
} while (i < len);
}
return 0;
}
static int xget(TAKDecContext *s, int d, int q)
{
int x;
x = d / q;
s->rval = d - (x * q);
if (s->rval < q / 2) {
s->rval += q;
} else {
x++;
}
if (x <= 1 || x > 128)
return -1;
return x;
}
static int get_len(TAKDecContext *s, int b)
{
if (b >= s->wlength - 1)
return s->rval;
else
return s->uval;
}
static int decode_coeffs(TAKDecContext *s, int32_t *dst, int length)
{
GetBitContext *gb = &s->gb;
int i, v, ret;
if (length > s->nb_samples)
return AVERROR_INVALIDDATA;
if (get_bits1(gb)) {
if ((s->wlength = xget(s, length, s->uval)) < 0)
return AVERROR_INVALIDDATA;
s->coding_mode[0] = v = get_bits(gb, 6);
if (s->coding_mode[0] > FF_ARRAY_ELEMS(xcodes))
return AVERROR_INVALIDDATA;
for (i = 1; i < s->wlength; i++) {
int c = get_unary(gb, 1, 6);
if (c > 5) {
v = get_bits(gb, 6);
} else if (c > 2) {
int t = get_bits1(gb);
v += (-t ^ (c - 1)) + t;
} else {
v += (-(c & 1) ^ (((c & 1) + c) >> 1)) + (c & 1);
}
if (v > FF_ARRAY_ELEMS(xcodes))
return AVERROR_INVALIDDATA;
s->coding_mode[i] = v;
}
i = 0;
while (i < s->wlength) {
int len = 0;
v = s->coding_mode[i];
do {
len += get_len(s, i);
i++;
if (i == s->wlength)
break;
} while (v == s->coding_mode[i]);
if ((ret = decode_segment(s, v, dst, len)) < 0)
return ret;
dst += len;
}
} else {
v = get_bits(gb, 6);
if (v > FF_ARRAY_ELEMS(xcodes))
return AVERROR_INVALIDDATA;
if ((ret = decode_segment(s, v, dst, length)) < 0)
return ret;
}
return 0;
}
static int get_b(GetBitContext *gb)
{
if (get_bits1(gb))
return get_bits(gb, 4) + 1;
else
return 0;
}
static int decode_subframe(TAKDecContext *s, int32_t *ptr, int subframe_size,
int prev_subframe_size)
{
GetBitContext *gb = &s->gb;
int tmp, x, y, i, j, ret = 0;
int tfilter[MAX_PREDICTORS];
if (get_bits1(gb)) {
s->filter_order = predictor_sizes[get_bits(gb, 4)];
if (prev_subframe_size > 0 && get_bits1(gb)) {
if (s->filter_order > prev_subframe_size)
return AVERROR_INVALIDDATA;
ptr -= s->filter_order;
subframe_size += s->filter_order;
if (s->filter_order > subframe_size)
return AVERROR_INVALIDDATA;
} else {
int lpc;
if (s->filter_order > subframe_size)
return AVERROR_INVALIDDATA;
lpc = get_bits(gb, 2);
if (lpc > 2)
return AVERROR_INVALIDDATA;
if ((ret = decode_coeffs(s, ptr, s->filter_order)) < 0)
return ret;
decode_lpc(ptr, lpc, s->filter_order);
}
s->xred = get_b(gb);
s->size = get_bits1(gb) + 5;
if (get_bits1(gb)) {
s->ared = get_bits(gb, 3) + 1;
if (s->ared > 7)
return AVERROR_INVALIDDATA;
} else {
s->ared = 0;
}
s->predictors[0] = get_code(gb, 10);
s->predictors[1] = get_code(gb, 10);
s->predictors[2] = get_code(gb, s->size + 1) << (9 - s->size);
s->predictors[3] = get_code(gb, s->size + 1) << (9 - s->size);
if (s->filter_order > 4) {
tmp = s->size + 1 - get_bits1(gb);
for (i = 4; i < s->filter_order; i++) {
if (!(i & 3))
x = tmp - get_bits(gb, 2);
s->predictors[i] = get_code(gb, x) << (9 - s->size);
}
}
tfilter[0] = s->predictors[0] << 6;
for (i = 1; i < s->filter_order; i++) {
int32_t *p1 = &tfilter[0];
int32_t *p2 = &tfilter[i - 1];
for (j = 0; j < (i + 1) / 2; j++) {
x = *p1 + (s->predictors[i] * *p2 + 256 >> 9);
*p2 += s->predictors[i] * *p1 + 256 >> 9;
*p1++ = x;
p2--;
}
tfilter[i] = s->predictors[i] << 6;
}
x = -1 << (32 - (s->ared + 5));
y = 1 << ((s->ared + 5) - 1);
for (i = 0, j = s->filter_order - 1; i < s->filter_order / 2; i++, j--) {
tmp = y + tfilter[j];
s->filter[j] = -(x & -(y + tfilter[i] >> 31) |
(y + tfilter[i]) >> (s->ared + 5));
s->filter[i] = -(x & -(tmp >> 31) | (tmp >> s->ared + 5));
}
if ((ret = decode_coeffs(s, &ptr[s->filter_order],
subframe_size - s->filter_order)) < 0)
return ret;
for (i = 0; i < s->filter_order; i++)
s->residues[i] = *ptr++ >> s->xred;
y = FF_ARRAY_ELEMS(s->residues) - s->filter_order;
x = subframe_size - s->filter_order;
while (x > 0) {
tmp = FFMIN(y, x);
for (i = 0; i < tmp; i++) {
int v, w, m;
v = 1 << (10 - s->ared - 1);
if (!(s->filter_order & 15)) {
v += s->dsp.scalarproduct_int16(&s->residues[i], s->filter,
s->filter_order);
} else if (s->filter_order & 4) {
for (j = 0; j < s->filter_order; j += 4) {
v += s->residues[i + j + 3] * s->filter[j + 3] +
s->residues[i + j + 2] * s->filter[j + 2] +
s->residues[i + j + 1] * s->filter[j + 1] +
s->residues[i + j ] * s->filter[j ];
}
} else {
for (j = 0; j < s->filter_order; j += 8) {
v += s->residues[i + j + 7] * s->filter[j + 7] +
s->residues[i + j + 6] * s->filter[j + 6] +
s->residues[i + j + 5] * s->filter[j + 5] +
s->residues[i + j + 4] * s->filter[j + 4] +
s->residues[i + j + 3] * s->filter[j + 3] +
s->residues[i + j + 2] * s->filter[j + 2] +
s->residues[i + j + 1] * s->filter[j + 1] +
s->residues[i + j ] * s->filter[j ];
}
}
m = (-1 << (32 - (10 - s->ared))) & -(v >> 31) | (v >> 10 - s->ared);
m = av_clip(m, -8192, 8191);
w = (m << s->xred) - *ptr;
*ptr++ = w;
s->residues[s->filter_order + i] = w >> s->xred;
}
x -= tmp;
if (x > 0)
memcpy(s->residues, &s->residues[y], 2 * s->filter_order);
}
emms_c();
} else {
ret = decode_coeffs(s, ptr, subframe_size);
}
return ret;
}
static int decode_channel(TAKDecContext *s, int chan)
{
AVCodecContext *avctx = s->avctx;
GetBitContext *gb = &s->gb;
int32_t *dst = s->decoded[chan];
int i = 0, ret, prev = 0;
int left = s->nb_samples - 1;
s->sample_shift[chan] = get_b(gb);
if (s->sample_shift[chan] >= avctx->bits_per_coded_sample)
return AVERROR_INVALIDDATA;
*dst++ = get_code(gb, avctx->bits_per_coded_sample - s->sample_shift[chan]);
s->lpc_mode[chan] = get_bits(gb, 2);
s->nb_subframes = get_bits(gb, 3) + 1;
if (s->nb_subframes > 1) {
if (get_bits_left(gb) < (s->nb_subframes - 1) * 6)
return AVERROR_INVALIDDATA;
for (; i < s->nb_subframes - 1; i++) {
int v = get_bits(gb, 6);
s->subframe_len[i] = (v - prev) * s->subframe_scale;
if (s->subframe_len[i] <= 0)
return AVERROR_INVALIDDATA;
left -= s->subframe_len[i];
prev = v;
}
if (left <= 0)
return AVERROR_INVALIDDATA;
}
s->subframe_len[i] = left;
prev = 0;
for (i = 0; i < s->nb_subframes; i++) {
if ((ret = decode_subframe(s, dst, s->subframe_len[i], prev)) < 0)
return ret;
dst += s->subframe_len[i];
prev = s->subframe_len[i];
}
return 0;
}
static int decorrelate(TAKDecContext *s, int c1, int c2, int length)
{
GetBitContext *gb = &s->gb;
uint32_t *p1 = s->decoded[c1] + 1;
uint32_t *p2 = s->decoded[c2] + 1;
int a, b, i, x, tmp;
if (s->dmode > 3) {
s->dshift = get_b(gb);
if (s->dmode > 5) {
if (get_bits1(gb))
s->filter_order = 16;
else
s->filter_order = 8;
s->dval1 = get_bits1(gb);
s->dval2 = get_bits1(gb);
for (i = 0; i < s->filter_order; i++) {
if (!(i & 3))
x = 14 - get_bits(gb, 3);
s->filter[i] = get_code(gb, x);
}
} else {
s->dfactor = get_code(gb, 10);
}
}
switch (s->dmode) {
case 1:
for (i = 0; i < length; i++, p1++, p2++)
*p2 += *p1;
break;
case 2:
for (i = 0; i < length; i++, p1++, p2++)
*p1 = *p2 - *p1;
break;
case 3:
for (i = 0; i < length; i++, p1++, p2++) {
x = (*p2 & 1) + 2 * *p1;
a = -*p2 + x;
b = *p2 + x;
*p1 = a & 0x80000000 | (a >> 1);
*p2 = b & 0x80000000 | (b >> 1);
}
break;
case 4:
FFSWAP(uint32_t *, p1, p2);
case 5:
if (s->dshift)
tmp = -1 << (32 - s->dshift);
else
tmp = 0;
for (i = 0; i < length; i++, p1++, p2++) {
x = s->dfactor * (tmp & -(*p2 >> 31) | (*p2 >> s->dshift)) + 128;
*p1 = ((-(x >> 31) & 0xFF000000 | (x >> 8)) << s->dshift) - *p1;
}
break;
case 6:
FFSWAP(uint32_t *, p1, p2);
case 7:
if (length < 256)
return AVERROR_INVALIDDATA;
a = s->filter_order / 2;
b = length - (s->filter_order - 1);
if (s->dval1) {
for (i = 0; i < a; i++)
p1[i] += p2[i];
}
if (s->dval2) {
x = a + b;
for (i = 0; i < length - x; i++)
p1[x + i] += p2[x + i];
}
for (i = 0; i < s->filter_order; i++)
s->residues[i] = *p2++ >> s->dshift;
p1 += a;
x = FF_ARRAY_ELEMS(s->residues) - s->filter_order;
for (; b > 0; b -= tmp) {
tmp = FFMIN(b, x);
for (i = 0; i < tmp; i++)
s->residues[s->filter_order + i] = *p2++ >> s->dshift;
for (i = 0; i < tmp; i++) {
int v, w, m;
v = 1 << 9;
if (s->filter_order == 16) {
v += s->dsp.scalarproduct_int16(&s->residues[i], s->filter,
s->filter_order);
} else {
v += s->residues[i + 7] * s->filter[7] +
s->residues[i + 6] * s->filter[6] +
s->residues[i + 5] * s->filter[5] +
s->residues[i + 4] * s->filter[4] +
s->residues[i + 3] * s->filter[3] +
s->residues[i + 2] * s->filter[2] +
s->residues[i + 1] * s->filter[1] +
s->residues[i ] * s->filter[0];
}
m = (-1 << 22) & -(v >> 31) | (v >> 10);
m = av_clip(m, -8192, 8191);
w = (m << s->dshift) - *p1;
*p1++ = w;
}
memcpy(s->residues, &s->residues[tmp], 2 * s->filter_order);
}
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;
int32_t *p;
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 (avctx->err_recognition & AV_EF_CRCCHECK) {
hsize = get_bits_count(gb) / 8;
if (ff_tak_check_crc(pkt->data, hsize)) {
av_log(avctx, AV_LOG_ERROR, "CRC error\n");
return AVERROR_INVALIDDATA;
}
}
if (s->ti.codec != 2 && s->ti.codec != 4) {
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 == 2 && 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;
tak_set_bps(avctx);
}
if (s->ti.sample_rate != avctx->sample_rate) {
avctx->sample_rate = s->ti.sample_rate;
s->uval = get_scale(avctx->sample_rate, get_shift(avctx->sample_rate));
s->subframe_scale = get_scale(avctx->sample_rate, 1);
}
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 = avctx->get_buffer(avctx, &s->frame)) < 0)
return ret;
if (avctx->bits_per_coded_sample <= 16) {
av_fast_malloc(&s->decode_buffer, &s->decode_buffer_size,
sizeof(*s->decode_buffer) * FFALIGN(s->nb_samples, 8) *
avctx->channels + FF_INPUT_BUFFER_PADDING_SIZE);
if (!s->decode_buffer)
return AVERROR(ENOMEM);
for (chan = 0; chan < avctx->channels; chan++)
s->decoded[chan] = s->decode_buffer +
chan * FFALIGN(s->nb_samples, 8);
} else {
for (chan = 0; chan < avctx->channels; chan++)
s->decoded[chan] = (int32_t *)s->frame.data[chan];
}
if (s->nb_samples < 16) {
for (chan = 0; chan < avctx->channels; chan++) {
p = s->decoded[chan];
for (i = 0; i < s->nb_samples; i++)
*p++ = get_code(gb, avctx->bits_per_coded_sample);
}
} else {
if (s->ti.codec == 2) {
for (chan = 0; chan < avctx->channels; chan++) {
if (ret = decode_channel(s, chan))
return ret;
}
if (avctx->channels == 2) {
s->nb_subframes = get_bits(gb, 1) + 1;
if (s->nb_subframes > 1)
s->subframe_len[1] = get_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 == 4) {
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++) {
p = s->decoded[chan];
decode_lpc(p, s->lpc_mode[chan], s->nb_samples);
if (s->sample_shift[chan] > 0) {
for (i = 0; i < s->nb_samples; i++)
*p++ <<= 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->bits_per_coded_sample) {
case 8:
for (chan = 0; chan < avctx->channels; chan++) {
uint8_t *samples = (uint8_t *)s->frame.data[chan];
p = s->decoded[chan];
for (i = 0; i < s->nb_samples; i++, p++)
*samples++ = *p + 0x80;
}
break;
case 16:
for (chan = 0; chan < avctx->channels; chan++) {
int16_t *samples = (int16_t *)s->frame.data[chan];
p = s->decoded[chan];
for (i = 0; i < s->nb_samples; i++, p++)
*samples++ = *p;
}
break;
case 24:
for (chan = 0; chan < avctx->channels; chan++) {
int32_t *samples = (int32_t *)s->frame.data[chan];
for (i = 0; i < s->nb_samples; i++)
*samples++ <<= 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);
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,
.close = tak_decode_close,
.decode = tak_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("TAK (Tom's lossless Audio Kompressor)"),
};
......@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 54
#define LIBAVCODEC_VERSION_MINOR 64
#define LIBAVCODEC_VERSION_MINOR 65
#define LIBAVCODEC_VERSION_MICRO 100
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
......
......@@ -333,6 +333,7 @@ OBJS-$(CONFIG_STR_DEMUXER) += psxstr.o
OBJS-$(CONFIG_SUBVIEWER_DEMUXER) += subviewerdec.o
OBJS-$(CONFIG_SWF_DEMUXER) += swfdec.o swf.o
OBJS-$(CONFIG_SWF_MUXER) += swfenc.o swf.o
OBJS-$(CONFIG_TAK_DEMUXER) += takdec.o apetag.o img2.o rawdec.o
OBJS-$(CONFIG_THP_DEMUXER) += thp.o
OBJS-$(CONFIG_TIERTEXSEQ_DEMUXER) += tiertexseq.o
OBJS-$(CONFIG_MKVTIMESTAMP_V2_MUXER) += mkvtimestamp_v2.o
......
......@@ -233,6 +233,7 @@ void av_register_all(void)
REGISTER_DEMUXER (STR, str);
REGISTER_DEMUXER (SUBVIEWER, subviewer);
REGISTER_MUXDEMUX (SWF, swf);
REGISTER_DEMUXER (TAK, tak);
REGISTER_MUXER (TG2, tg2);
REGISTER_MUXER (TGP, tgp);
REGISTER_DEMUXER (THP, thp);
......
/*
* Raw TAK demuxer
* Copyright (c) 2012 Paul B Mahol
*
* 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
*/
#include "libavcodec/tak.h"
#include "avformat.h"
#include "internal.h"
#include "rawdec.h"
#include "apetag.h"
typedef struct TAKDemuxContext {
int mlast_frame;
int64_t left;
} TAKDemuxContext;
static int tak_probe(AVProbeData *p)
{
if (!memcmp(p->buf, "tBaK", 4))
return AVPROBE_SCORE_MAX / 2;
return 0;
}
static int tak_read_header(AVFormatContext *s)
{
TAKDemuxContext *tc = s->priv_data;
AVIOContext *pb = s->pb;
GetBitContext gb;
AVStream *st;
uint8_t *buffer = NULL;
int ret;
st = avformat_new_stream(s, 0);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_TAK;
st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
tc->mlast_frame = 0;
if (avio_rl32(pb) != MKTAG('t', 'B', 'a', 'K')) {
avio_seek(pb, -4, SEEK_CUR);
return 0;
}
while (!url_feof(pb)) {
enum TAKMetaDataType type;
int size;
type = avio_r8(pb) & 0x7f;
size = avio_rl24(pb);
switch (type) {
case TAK_METADATA_STREAMINFO:
case TAK_METADATA_LAST_FRAME:
case TAK_METADATA_ENCODER:
buffer = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
if (!buffer)
return AVERROR(ENOMEM);
if (avio_read(pb, buffer, size) != size) {
av_freep(&buffer);
return AVERROR(EIO);
}
init_get_bits(&gb, buffer, size * 8);
break;
case TAK_METADATA_MD5: {
uint8_t md5[16];
int i;
if (size != 19)
return AVERROR_INVALIDDATA;
avio_read(pb, md5, 16);
avio_skip(pb, 3);
av_log(s, AV_LOG_VERBOSE, "MD5=");
for (i = 0; i < 16; i++)
av_log(s, AV_LOG_VERBOSE, "%02x", md5[i]);
av_log(s, AV_LOG_VERBOSE, "\n");
break;
}
case TAK_METADATA_END:
if (pb->seekable) {
int64_t curpos = avio_tell(pb);
ff_ape_parse_tag(s);
avio_seek(pb, curpos, SEEK_SET);
}
return 0;
break;
default:
ret = avio_skip(pb, size);
if (ret < 0)
return ret;
}
if (type == TAK_METADATA_STREAMINFO) {
TAKStreamInfo ti;
avpriv_tak_parse_streaminfo(&gb, &ti);
if (ti.samples > 0)
st->duration = ti.samples;
st->codec->bits_per_coded_sample = ti.bps;
if (ti.ch_layout)
st->codec->channel_layout = ti.ch_layout;
st->codec->sample_rate = ti.sample_rate;
st->codec->channels = ti.channels;
st->start_time = 0;
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
st->codec->extradata = buffer;
st->codec->extradata_size = size;
buffer = NULL;
} else if (type == TAK_METADATA_LAST_FRAME) {
if (size != 11)
return AVERROR_INVALIDDATA;
tc->mlast_frame = 1;
tc->left = get_bits_longlong(&gb, TAK_LAST_FRAME_POS_BITS) +
get_bits(&gb, TAK_LAST_FRAME_SIZE_BITS);
av_freep(&buffer);
} else if (type == TAK_METADATA_ENCODER) {
av_log(s, AV_LOG_VERBOSE, "encoder version: %0X\n",
get_bits_long(&gb, TAK_ENCODER_VERSION_BITS));
av_freep(&buffer);
}
}
return AVERROR_EOF;
}
static int raw_read_packet(AVFormatContext *s, AVPacket *pkt)
{
TAKDemuxContext *tc = s->priv_data;
int ret;
if (tc->mlast_frame) {
AVIOContext *pb = s->pb;
int64_t size;
size = FFMIN(tc->left, 1024);
if (!size)
return AVERROR_EOF;
ret = av_get_packet(pb, pkt, size);
if (ret < 0)
return ret;
pkt->stream_index = 0;
pkt->pos = avio_tell(pb);
tc->left -= ret;
} else {
ret = ff_raw_read_partial_packet(s, pkt);
}
return ret;
}
AVInputFormat ff_tak_demuxer = {
.name = "tak",
.long_name = NULL_IF_CONFIG_SMALL("raw TAK"),
.priv_data_size = sizeof(TAKDemuxContext),
.read_probe = tak_probe,
.read_header = tak_read_header,
.read_packet = raw_read_packet,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "tak",
.raw_codec_id = AV_CODEC_ID_TAK,
};
......@@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVFORMAT_VERSION_MAJOR 54
#define LIBAVFORMAT_VERSION_MINOR 29
#define LIBAVFORMAT_VERSION_MICRO 105
#define LIBAVFORMAT_VERSION_MINOR 30
#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \
......
......@@ -10,6 +10,9 @@ fate-lossless-monkeysaudio: CMD = md5 -i $(SAMPLES)/lossless-audio/luckynight-pa
FATE_SAMPLES_LOSSLESS_AUDIO += fate-lossless-shorten
fate-lossless-shorten: CMD = md5 -i $(SAMPLES)/lossless-audio/luckynight-partial.shn -f s16le
FATE_SAMPLES_LOSSLESS_AUDIO += fate-lossless-tak
fate-lossless-tak: CMD = crc -i $(SAMPLES)/lossless-audio/luckynight-partial.tak -f s16le
FATE_SAMPLES_LOSSLESS_AUDIO += fate-lossless-tta
fate-lossless-tta: CMD = crc -i $(SAMPLES)/lossless-audio/inside.tta
......
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