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Stefan Westerfeld
audiowmark
Commits
aedf2f2e
Commit
aedf2f2e
authored
Jan 08, 2019
by
Stefan Westerfeld
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Started implementation for reading and writing a sync block.
Signed-off-by:
Stefan Westerfeld
<
stefan@space.twc.de
>
parent
4ba1cad0
Changes
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audiowmark.cc
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src/audiowmark.cc
View file @
aedf2f2e
...
...
@@ -33,6 +33,9 @@ namespace Params
static
int
block_size
=
32
;
// block size for mix step (non-linear bit storage)
static
int
have_key
=
0
;
static
size_t
payload_size
=
128
;
// number of payload bits for the watermark
static
int
sync_bits
=
6
;
static
int
sync_frames_per_bit
=
32
;
}
void
...
...
@@ -460,6 +463,46 @@ mark_data (const WavData& wav_data, const vector<vector<complex<float>>>& fft_ou
}
}
size_t
mark_sync_frame_count
()
{
return
Params
::
sync_bits
*
Params
::
sync_frames_per_bit
;
}
void
mark_sync
(
const
WavData
&
wav_data
,
const
vector
<
vector
<
complex
<
float
>>>&
fft_out
,
vector
<
vector
<
complex
<
float
>>>&
fft_delta_spect
)
{
const
int
frame_count
=
fft_out
.
size
()
/
wav_data
.
n_channels
();
// sync block always written in linear order (no mix)
for
(
int
f
=
0
;
f
<
frame_count
;
f
++
)
{
for
(
int
ch
=
0
;
ch
<
wav_data
.
n_channels
();
ch
++
)
{
size_t
index
=
f
*
wav_data
.
n_channels
()
+
ch
;
vector
<
int
>
up
;
vector
<
int
>
down
;
get_up_down
(
f
,
up
,
down
);
// FIXME use extra random stream for sync
const
int
data_bit
=
(
f
/
Params
::
sync_frames_per_bit
)
&
1
;
/* write 010101 */
const
double
data_bit_sign
=
data_bit
>
0
?
1
:
-
1
;
for
(
auto
u
:
up
)
{
const
float
mag_factor
=
pow
(
abs
(
fft_out
[
index
][
u
]),
-
Params
::
water_delta
*
data_bit_sign
);
fft_delta_spect
[
index
][
u
]
=
fft_out
[
index
][
u
]
*
(
mag_factor
-
1
);
}
for
(
auto
d
:
down
)
{
const
float
mag_factor
=
pow
(
abs
(
fft_out
[
index
][
d
]),
Params
::
water_delta
*
data_bit_sign
);
fft_delta_spect
[
index
][
d
]
=
fft_out
[
index
][
d
]
*
(
mag_factor
-
1
);
}
}
}
}
vector
<
vector
<
complex
<
float
>>>
get_frame_range
(
const
WavData
&
wav_data
,
const
vector
<
vector
<
complex
<
float
>>>&
src
,
size_t
start
,
size_t
count
)
{
...
...
@@ -539,10 +582,16 @@ add_watermark (const string& infile, const string& outfile, const string& bits)
fft_delta_spect
.
push_back
(
vector
<
complex
<
float
>>
(
fft_out
.
back
().
size
()));
}
}
/*
vector<vector<complex<float>>> fft_out_range = get_frame_range (wav_data, fft_out, 0, mark_data_frame_count());
vector<vector<complex<float>>> fft_delta_spect_range = get_frame_range (wav_data, fft_delta_spect, 0, mark_data_frame_count());
mark_data (wav_data, fft_out_range, fft_delta_spect_range, bitvec);
copy_frame_range (wav_data, fft_delta_spect_range, fft_delta_spect, 0);
*/
vector
<
vector
<
complex
<
float
>>>
fft_out_range
=
get_frame_range
(
wav_data
,
fft_out
,
0
,
mark_sync_frame_count
());
vector
<
vector
<
complex
<
float
>>>
fft_delta_spect_range
=
get_frame_range
(
wav_data
,
fft_delta_spect
,
0
,
mark_sync_frame_count
());
mark_sync
(
wav_data
,
fft_out_range
,
fft_delta_spect_range
);
copy_frame_range
(
wav_data
,
fft_delta_spect_range
,
fft_delta_spect
,
0
);
/* generate synthesis window */
// we want overlapping synthesis windows, so the window affects the last, the current and the next frame
...
...
@@ -737,9 +786,69 @@ linear_decode (const WavData& wav_data, vector<vector<complex<float>>>& fft_out,
return
normalize_soft_bits
(
soft_bit_vec
);
}
void
sync_decode
(
const
WavData
&
wav_data
,
vector
<
vector
<
complex
<
float
>>>&
fft_out
,
vector
<
vector
<
complex
<
float
>>>&
fft_orig_out
)
{
// FIXME: is copypasted
const
int
frame_count
=
mark_sync_frame_count
();
double
umag
=
0
,
dmag
=
0
;
double
sync_quality
=
0
;
bool
sync_match
=
true
;
for
(
int
f
=
0
;
f
<
frame_count
;
f
++
)
{
for
(
int
ch
=
0
;
ch
<
wav_data
.
n_channels
();
ch
++
)
{
const
size_t
index
=
f
*
wav_data
.
n_channels
()
+
ch
;
vector
<
int
>
up
;
vector
<
int
>
down
;
get_up_down
(
f
,
up
,
down
);
const
double
min_db
=
-
96
;
for
(
auto
u
:
up
)
{
umag
+=
db_from_factor
(
abs
(
fft_out
[
index
][
u
]),
min_db
);
if
(
index
<
fft_orig_out
.
size
())
umag
-=
db_from_factor
(
abs
(
fft_orig_out
[
index
][
u
]),
min_db
);
}
for
(
auto
d
:
down
)
{
dmag
+=
db_from_factor
(
abs
(
fft_out
[
index
][
d
]),
min_db
);
if
(
index
<
fft_orig_out
.
size
())
dmag
-=
db_from_factor
(
abs
(
fft_orig_out
[
index
][
d
]),
min_db
);
}
}
if
((
f
%
Params
::
sync_frames_per_bit
)
==
(
Params
::
sync_frames_per_bit
-
1
))
{
const
int
data_bit
=
(
umag
<
dmag
)
?
0
:
1
;
const
int
expect_data_bit
=
(
f
/
Params
::
sync_frames_per_bit
)
&
1
;
/* expect 010101 */
if
(
data_bit
!=
expect_data_bit
)
sync_match
=
false
;
//const double norm = (1 + Params::water_delta) / (1 - Params::water_delta) - 1;
//const double norm = (1 + Params::water_delta) * (1 + Params::water_delta) - 1;
const
double
norm
=
1
;
printf
(
"%d %f
\n
"
,
data_bit
,
/*fabs*/
(
umag
/
dmag
-
1
)
/
norm
);
sync_quality
+=
fabs
(
umag
/
dmag
-
1
)
/
norm
;
umag
=
0
;
dmag
=
0
;
}
}
sync_quality
/=
Params
::
sync_bits
;
printf
(
"sync_match = %d
\n
"
,
sync_match
);
printf
(
"sync_quality = %f
\n
"
,
sync_quality
);
}
int
decode_and_report
(
const
WavData
&
wav_data
,
const
string
&
orig_pattern
,
vector
<
vector
<
complex
<
float
>>>&
fft_out
,
vector
<
vector
<
complex
<
float
>>>&
fft_orig_out
)
{
sync_decode
(
wav_data
,
fft_out
,
fft_orig_out
);
return
0
;
vector
<
float
>
soft_bit_vec
;
if
(
Params
::
mix
)
{
...
...
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