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tak: demuxer, parser, and decoder

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
pull/8/head
Paul B Mahol 12 years ago committed by Justin Ruggles
parent 096abfa150
commit 57231e4d5b
16 changed files with 1579 additions and 2 deletions
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  1. 1
      Changelog
  2. 1
      configure
  3. 2
      doc/general.texi
  4. 3
      libavcodec/Makefile
  5. 2
      libavcodec/allcodecs.c
  6. 1
      libavcodec/avcodec.h
  7. 7
      libavcodec/codec_desc.c
  8. 150
      libavcodec/tak.c
  9. 166
      libavcodec/tak.h
  10. 128
      libavcodec/tak_parser.c
  11. 929
      libavcodec/takdec.c
  12. 2
      libavcodec/version.h
  13. 1
      libavformat/Makefile
  14. 1
      libavformat/allformats.c
  15. 185
      libavformat/takdec.c
  16. 2
      libavformat/version.h

1
Changelog
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@ -7,6 +7,7 @@ version <next>:
- audio volume filter
- deprecated the avconv -vol option. the volume filter is to be used instead.
- multi-channel ALAC encoding up to 7.1
- TAK demuxer, parser, and decoder
version 9_beta2:

1
configure vendored
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@ -1675,6 +1675,7 @@ 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"

2
doc/general.texi
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@ -273,6 +273,7 @@ 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
@ -800,6 +801,7 @@ following image formats are supported:
@item SMPTE 302M AES3 audio @tab @tab X
@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.

3
libavcodec/Makefile
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@ -349,6 +349,7 @@ 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-$(CONFIG_THEORA_DECODER) += xiph.o
@ -555,6 +556,7 @@ OBJS-$(CONFIG_RTP_MUXER) += mpeg4audio.o xiph.o
OBJS-$(CONFIG_RTPDEC) += mjpeg.o
OBJS-$(CONFIG_SPDIF_DEMUXER) += aacadtsdec.o mpeg4audio.o
OBJS-$(CONFIG_SPDIF_MUXER) += dca.o
OBJS-$(CONFIG_TAK_DEMUXER) += tak.o
OBJS-$(CONFIG_WEBM_MUXER) += mpeg4audio.o mpegaudiodata.o \
xiph.o flac.o flacdata.o
OBJS-$(CONFIG_WTV_DEMUXER) += mpeg4audio.o mpegaudiodata.o
@ -631,6 +633,7 @@ 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

2
libavcodec/allcodecs.c
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@ -295,6 +295,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (SHORTEN, shorten);
REGISTER_DECODER (SIPR, sipr);
REGISTER_DECODER (SMACKAUD, smackaud);
REGISTER_DECODER (TAK, tak);
REGISTER_DECODER (TRUEHD, truehd);
REGISTER_DECODER (TRUESPEECH, truespeech);
REGISTER_DECODER (TTA, tta);
@ -431,6 +432,7 @@ 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);

1
libavcodec/avcodec.h
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@ -407,6 +407,7 @@ enum AVCodecID {
AV_CODEC_ID_ILBC,
AV_CODEC_ID_OPUS,
AV_CODEC_ID_COMFORT_NOISE,
AV_CODEC_ID_TAK,
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.

7
libavcodec/codec_desc.c
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@ -2119,6 +2119,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("RFC 3389 Comfort Noise"),
.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 */
{

150
libavcodec/tak.c
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@ -0,0 +1,150 @@
/*
* TAK common code
* 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
*/
#include "libavutil/bswap.h"
#include "libavutil/crc.h"
#include "libavutil/intreadwrite.h"
#include "tak.h"
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_bits64(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 > 0 && value <= 18)
channel_mask |= 1 << (value - 1);
}
}
}
s->ch_layout = channel_mask;
s->frame_samples = tak_get_nb_samples(s->sample_rate, frame_type);
}
int ff_tak_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
TAKStreamInfo *ti, int log_level_offset)
{
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;
}
ti->flags = get_bits(gb, TAK_FRAME_HEADER_FLAGS_BITS);
ti->frame_num = get_bits(gb, TAK_FRAME_HEADER_NO_BITS);
if (ti->flags & TAK_FRAME_FLAG_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 (ti->flags & TAK_FRAME_FLAG_HAS_INFO) {
avpriv_tak_parse_streaminfo(gb, ti);
if (get_bits(gb, 6))
skip_bits(gb, 25);
align_get_bits(gb);
}
skip_bits(gb, 24);
return 0;
}

166
libavcodec/tak.h
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@ -0,0 +1,166 @@
/*
* TAK decoder/demuxer common code
* 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/demuxer common functions
*/
#ifndef AVCODEC_TAK_H
#define AVCODEC_TAK_H
#include <stdint.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_FRAME_FLAG_IS_LAST 0x1
#define TAK_FRAME_FLAG_HAS_INFO 0x2
#define TAK_FRAME_FLAG_HAS_METADATA 0x4
#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 TAKCodecType {
TAK_CODEC_MONO_STEREO = 2,
TAK_CODEC_MULTICHANNEL = 4
};
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 flags;
enum TAKCodecType 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 storage for parsed information
*/
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 */

128
libavcodec/tak_parser.c
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@ -0,0 +1,128 @@
/*
* TAK parser
* Copyright (c) 2012 Michael Niedermayer
*
* 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 parser
**/
#include "tak.h"
#include "parser.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,
};

929
libavcodec/takdec.c
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@ -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 },
};

2
libavcodec/version.h
Unescape View File

@ -27,7 +27,7 @@
*/
#define LIBAVCODEC_VERSION_MAJOR 54
#define LIBAVCODEC_VERSION_MINOR 33
#define LIBAVCODEC_VERSION_MINOR 34
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \

1
libavformat/Makefile
Unescape View File

@ -304,6 +304,7 @@ OBJS-$(CONFIG_SRT_MUXER) += rawenc.o
OBJS-$(CONFIG_STR_DEMUXER) += psxstr.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_TMV_DEMUXER) += tmv.o

1
libavformat/allformats.c
Unescape View File

@ -214,6 +214,7 @@ void av_register_all(void)
REGISTER_MUXDEMUX (SRT, srt);
REGISTER_DEMUXER (STR, str);
REGISTER_MUXDEMUX (SWF, swf);
REGISTER_DEMUXER (TAK, tak);
REGISTER_MUXER (TG2, tg2);
REGISTER_MUXER (TGP, tgp);
REGISTER_DEMUXER (THP, thp);

185
libavformat/takdec.c
Unescape View File

@ -0,0 +1,185 @@
/*
* Raw TAK demuxer
* 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
*/
#include "libavcodec/tak.h"
#include "avformat.h"
#include "internal.h"
#include "rawdec.h"
#include "apetag.h"
typedef struct TAKDemuxContext {
int mlast_frame;
int64_t data_end;
} 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;
tc->mlast_frame = 0;
if (avio_rl32(pb) != MKTAG('t', 'B', 'a', 'K')) {
avio_seek(pb, -4, SEEK_CUR);
return 0;
}
while (!pb->eof_reached) {
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: {
int64_t curpos = avio_tell(pb);
if (pb->seekable) {
ff_ape_parse_tag(s);
avio_seek(pb, curpos, SEEK_SET);
}
tc->data_end += curpos;
return 0;
}
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->data_end = get_bits64(&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, left;
left = tc->data_end - avio_tell(s->pb);
size = FFMIN(left, 1024);
if (size <= 0)
return AVERROR_EOF;
ret = av_get_packet(pb, pkt, size);
if (ret < 0)
return ret;
pkt->stream_index = 0;
} 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,
};

2
libavformat/version.h
Unescape View File

@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVFORMAT_VERSION_MAJOR 54
#define LIBAVFORMAT_VERSION_MINOR 19
#define LIBAVFORMAT_VERSION_MINOR 20
#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \

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