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Stefan Westerfeld
audiowmark
Commits
d1276514
Commit
d1276514
authored
Feb 19, 2020
by
Stefan Westerfeld
Browse files
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Merge branch 'limiter'
parents
4a2f0b2d
b67de297
Changes
11
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11 changed files
with
296 additions
and
8 deletions
+296
-8
NEWS
NEWS
+4
-0
configure.ac
configure.ac
+1
-1
Makefile.am
src/Makefile.am
+6
-2
TODO
src/TODO
+0
-5
audiowmark.cc
src/audiowmark.cc
+4
-0
limiter.cc
src/limiter.cc
+115
-0
limiter.hh
src/limiter.hh
+31
-0
testlimiter.cc
src/testlimiter.cc
+122
-0
wmadd.cc
src/wmadd.cc
+8
-0
wmcommon.cc
src/wmcommon.cc
+1
-0
wmcommon.hh
src/wmcommon.hh
+4
-0
No files found.
NEWS
View file @
d1276514
Overview of Changes in audiowmark-0.2.1:
* add limiter to avoid clipping during watermark generation
Overview of Changes in audiowmark-0.2.0:
* support input/output streams
...
...
configure.ac
View file @
d1276514
AC_INIT([audiowmark], [0.2.
0
])
AC_INIT([audiowmark], [0.2.
1
])
AC_CONFIG_SRCDIR([src/audiowmark.cc])
AC_CONFIG_AUX_DIR([build-aux])
AC_CONFIG_MACRO_DIR([m4])
...
...
src/Makefile.am
View file @
d1276514
...
...
@@ -3,13 +3,14 @@ bin_PROGRAMS = audiowmark
COMMON_SRC
=
utils.hh utils.cc convcode.hh convcode.cc random.hh random.cc wavdata.cc wavdata.hh
\
audiostream.cc audiostream.hh sfinputstream.cc sfinputstream.hh stdoutwavoutputstream.cc stdoutwavoutputstream.hh
\
sfoutputstream.cc sfoutputstream.hh rawinputstream.cc rawinputstream.hh rawoutputstream.cc rawoutputstream.hh
\
rawconverter.cc rawconverter.hh mp3inputstream.cc mp3inputstream.hh wmcommon.cc wmcommon.hh fft.cc fft.hh
rawconverter.cc rawconverter.hh mp3inputstream.cc mp3inputstream.hh wmcommon.cc wmcommon.hh fft.cc fft.hh
\
limiter.cc limiter.hh
COMMON_LIBS
=
$(SNDFILE_LIBS)
$(FFTW_LIBS)
$(LIBGCRYPT_LIBS)
$(LIBMPG123_LIBS)
audiowmark_SOURCES
=
audiowmark.cc wmget.cc wmadd.cc
$(COMMON_SRC)
audiowmark_LDFLAGS
=
$(COMMON_LIBS)
noinst_PROGRAMS
=
testconvcode testrandom testmp3 teststream
noinst_PROGRAMS
=
testconvcode testrandom testmp3 teststream
testlimiter
testconvcode_SOURCES
=
testconvcode.cc
$(COMMON_SRC)
testconvcode_LDFLAGS
=
$(COMMON_LIBS)
...
...
@@ -22,3 +23,6 @@ testmp3_LDFLAGS = $(COMMON_LIBS)
teststream_SOURCES
=
teststream.cc
$(COMMON_SRC)
teststream_LDFLAGS
=
$(COMMON_LIBS)
testlimiter_SOURCES
=
testlimiter.cc
$(COMMON_SRC)
testlimiter_LDFLAGS
=
$(COMMON_LIBS)
src/TODO
View file @
d1276514
possible improvements:
- dynamic bit strength
streaming:
- final documentation
- merge mp3 code
- fix virtual constructor FIXMEs
src/audiowmark.cc
View file @
d1276514
...
...
@@ -192,6 +192,10 @@ parse_options (int *argc_p,
{
Params
::
test_no_sync
=
true
;
}
else
if
(
check_arg
(
argc
,
argv
,
&
i
,
"--test-no-limiter"
))
{
Params
::
test_no_limiter
=
true
;
}
else
if
(
check_arg
(
argc
,
argv
,
&
i
,
"--test-truncate"
,
&
opt_arg
))
{
Params
::
test_truncate
=
atoi
(
opt_arg
);
...
...
src/limiter.cc
0 → 100644
View file @
d1276514
#include "limiter.hh"
#include <assert.h>
#include <math.h>
#include <stdio.h>
using
std
::
vector
;
using
std
::
max
;
Limiter
::
Limiter
(
int
n_channels
,
int
sample_rate
)
:
n_channels
(
n_channels
),
sample_rate
(
sample_rate
)
{
}
void
Limiter
::
set_block_size_ms
(
int
ms
)
{
block_size
=
sample_rate
*
ms
/
1000
;
}
void
Limiter
::
set_ceiling
(
float
new_ceiling
)
{
ceiling
=
new_ceiling
;
}
vector
<
float
>
Limiter
::
process
(
const
vector
<
float
>&
samples
)
{
assert
(
block_size
>=
1
);
assert
(
samples
.
size
()
%
n_channels
==
0
);
// process should be called with whole frames
buffer
.
insert
(
buffer
.
end
(),
samples
.
begin
(),
samples
.
end
());
/* need at least two complete blocks in buffer to produce output */
const
uint
buffered_blocks
=
buffer
.
size
()
/
n_channels
/
block_size
;
if
(
buffered_blocks
<
2
)
return
{};
const
uint
blocks_todo
=
buffered_blocks
-
1
;
vector
<
float
>
out
(
blocks_todo
*
block_size
*
n_channels
);
for
(
uint
b
=
0
;
b
<
blocks_todo
;
b
++
)
process_block
(
&
buffer
[
b
*
block_size
*
n_channels
],
&
out
[
b
*
block_size
*
n_channels
]);
buffer
.
erase
(
buffer
.
begin
(),
buffer
.
begin
()
+
blocks_todo
*
block_size
*
n_channels
);
return
out
;
}
float
Limiter
::
block_max
(
const
float
*
in
)
{
float
maximum
=
ceiling
;
for
(
uint
x
=
0
;
x
<
block_size
*
n_channels
;
x
++
)
maximum
=
max
(
maximum
,
fabs
(
in
[
x
]));
return
maximum
;
}
void
Limiter
::
process_block
(
const
float
*
in
,
float
*
out
)
{
if
(
block_max_last
<
ceiling
)
block_max_last
=
ceiling
;
if
(
block_max_current
<
ceiling
)
block_max_current
=
block_max
(
in
);
if
(
block_max_next
<
ceiling
)
block_max_next
=
block_max
(
in
+
block_size
*
n_channels
);
const
float
scale_start
=
ceiling
/
max
(
block_max_last
,
block_max_current
);
const
float
scale_end
=
ceiling
/
max
(
block_max_current
,
block_max_next
);
const
float
scale_step
=
(
scale_end
-
scale_start
)
/
block_size
;
for
(
size_t
i
=
0
;
i
<
block_size
;
i
++
)
{
const
float
scale
=
scale_start
+
i
*
scale_step
;
// debug_scale (scale);
for
(
uint
c
=
0
;
c
<
n_channels
;
c
++
)
out
[
i
*
n_channels
+
c
]
=
in
[
i
*
n_channels
+
c
]
*
scale
;
}
block_max_last
=
block_max_current
;
block_max_current
=
block_max_next
;
block_max_next
=
0
;
}
void
Limiter
::
debug_scale
(
float
scale
)
{
static
int
debug_scale_samples
=
0
;
if
(
debug_scale_samples
%
(
sample_rate
/
1000
)
==
0
)
printf
(
"%f %f
\n
"
,
double
(
debug_scale_samples
)
/
sample_rate
,
scale
);
debug_scale_samples
++
;
}
vector
<
float
>
Limiter
::
flush
()
{
vector
<
float
>
out
;
vector
<
float
>
zblock
(
1024
*
n_channels
);
size_t
todo
=
buffer
.
size
();
while
(
todo
>
0
)
{
vector
<
float
>
block
=
process
(
zblock
);
if
(
block
.
size
()
>
todo
)
block
.
resize
(
todo
);
out
.
insert
(
out
.
end
(),
block
.
begin
(),
block
.
end
());
todo
-=
block
.
size
();
}
return
out
;
}
src/limiter.hh
0 → 100644
View file @
d1276514
#ifndef AUDIOWMARK_LIMITER_HH
#define AUDIOWMARK_LIMITER_HH
#include <vector>
#include <sys/types.h>
class
Limiter
{
float
ceiling
=
1
;
float
block_max_last
=
0
;
float
block_max_current
=
0
;
float
block_max_next
=
0
;
uint
block_size
=
0
;
uint
n_channels
=
0
;
uint
sample_rate
=
0
;
std
::
vector
<
float
>
buffer
;
void
process_block
(
const
float
*
in
,
float
*
out
);
float
block_max
(
const
float
*
in
);
void
debug_scale
(
float
scale
);
public
:
Limiter
(
int
n_channels
,
int
sample_rate
);
void
set_block_size_ms
(
int
value_ms
);
void
set_ceiling
(
float
ceiling
);
std
::
vector
<
float
>
process
(
const
std
::
vector
<
float
>&
samples
);
std
::
vector
<
float
>
flush
();
};
#endif
/* AUDIOWMARK_LIMITER_HH */
src/testlimiter.cc
0 → 100644
View file @
d1276514
#include <string>
#include <vector>
#include <sndfile.h>
#include <assert.h>
#include <math.h>
#include <string.h>
#include <sys/time.h>
#include "sfinputstream.hh"
#include "sfoutputstream.hh"
#include "utils.hh"
#include "limiter.hh"
using
std
::
string
;
using
std
::
vector
;
using
std
::
max
;
using
std
::
min
;
static
double
gettime
()
{
timeval
tv
;
gettimeofday
(
&
tv
,
0
);
return
tv
.
tv_sec
+
tv
.
tv_usec
/
1000000.0
;
}
int
perf
()
{
Limiter
limiter
(
2
,
44100
);
limiter
.
set_block_size_ms
(
1000
);
vector
<
float
>
samples
(
2
*
1024
);
int
n_frames
=
0
;
double
start
=
gettime
();
for
(
int
i
=
0
;
i
<
100000
;
i
++
)
{
n_frames
+=
samples
.
size
()
/
2
;
vector
<
float
>
out_samples
=
limiter
.
process
(
samples
);
}
double
end
=
gettime
();
printf
(
"%f ns/frame
\n
"
,
(
end
-
start
)
*
1000
*
1000
*
1000
/
n_frames
);
return
0
;
}
int
impulses
()
{
Limiter
limiter
(
2
,
44100
);
limiter
.
set_block_size_ms
(
3
);
limiter
.
set_ceiling
(
0.9
);
vector
<
float
>
in_all
,
out_all
;
int
pos
=
0
;
for
(
int
block
=
0
;
block
<
10
;
block
++
)
{
vector
<
float
>
in_samples
;
for
(
int
i
=
0
;
i
<
1024
;
i
++
)
{
double
d
=
(
pos
++
%
441
)
==
440
?
1.0
:
0.5
;
in_samples
.
push_back
(
d
);
in_samples
.
push_back
(
d
);
/* stereo */
}
vector
<
float
>
out_samples
=
limiter
.
process
(
in_samples
);
in_all
.
insert
(
in_all
.
end
(),
in_samples
.
begin
(),
in_samples
.
end
());
out_all
.
insert
(
out_all
.
end
(),
out_samples
.
begin
(),
out_samples
.
end
());
}
vector
<
float
>
out_samples
=
limiter
.
flush
();
out_all
.
insert
(
out_all
.
end
(),
out_samples
.
begin
(),
out_samples
.
end
());
assert
(
in_all
.
size
()
==
out_all
.
size
());
for
(
size_t
i
=
0
;
i
<
out_all
.
size
();
i
+=
2
)
{
assert
(
out_all
[
i
]
==
out_all
[
i
+
1
]);
/* stereo */
printf
(
"%f %f
\n
"
,
in_all
[
i
],
out_all
[
i
]);
}
return
0
;
}
int
main
(
int
argc
,
char
**
argv
)
{
if
(
argc
==
2
&&
strcmp
(
argv
[
1
],
"perf"
)
==
0
)
return
perf
();
if
(
argc
==
2
&&
strcmp
(
argv
[
1
],
"impulses"
)
==
0
)
return
impulses
();
SFInputStream
in
;
SFOutputStream
out
;
Error
err
=
in
.
open
(
argv
[
1
]);
if
(
err
)
{
fprintf
(
stderr
,
"testlimiter: open input failed: %s
\n
"
,
err
.
message
());
return
1
;
}
err
=
out
.
open
(
argv
[
2
],
in
.
n_channels
(),
in
.
sample_rate
(),
16
,
in
.
n_frames
());
if
(
err
)
{
fprintf
(
stderr
,
"testlimiter: open output failed: %s
\n
"
,
err
.
message
());
return
1
;
}
Limiter
limiter
(
in
.
n_channels
(),
in
.
sample_rate
());
limiter
.
set_block_size_ms
(
1000
);
limiter
.
set_ceiling
(
0.9
);
vector
<
float
>
in_samples
;
do
{
in
.
read_frames
(
in_samples
,
1024
);
for
(
auto
&
s
:
in_samples
)
s
*=
1.1
;
vector
<
float
>
out_samples
=
limiter
.
process
(
in_samples
);
out
.
write_frames
(
out_samples
);
}
while
(
in_samples
.
size
());
out
.
write_frames
(
limiter
.
flush
());
}
src/wmadd.cc
View file @
d1276514
...
...
@@ -6,6 +6,7 @@
#include "wmcommon.hh"
#include "fft.hh"
#include "convcode.hh"
#include "limiter.hh"
#include "sfinputstream.hh"
#include "sfoutputstream.hh"
#include "mp3inputstream.hh"
...
...
@@ -643,6 +644,10 @@ add_watermark (const string& infile, const string& outfile, const string& bits)
if
(
!
wm_resampler
.
init_ok
())
return
1
;
Limiter
limiter
(
n_channels
,
in_stream
->
sample_rate
());
limiter
.
set_block_size_ms
(
Params
::
limiter_block_size_ms
);
limiter
.
set_ceiling
(
Params
::
limiter_ceiling
);
/* for signal to noise ratio */
double
snr_delta_power
=
0
;
double
snr_signal_power
=
0
;
...
...
@@ -687,6 +692,9 @@ add_watermark (const string& infile, const string& outfile, const string& bits)
for
(
size_t
i
=
0
;
i
<
samples
.
size
();
i
++
)
samples
[
i
]
+=
orig_samples
[
i
];
if
(
!
Params
::
test_no_limiter
)
samples
=
limiter
.
process
(
samples
);
size_t
max_write_frames
=
total_input_frames
-
total_output_frames
;
if
(
samples
.
size
()
>
max_write_frames
*
n_channels
)
samples
.
resize
(
max_write_frames
*
n_channels
);
...
...
src/wmcommon.cc
View file @
d1276514
...
...
@@ -10,6 +10,7 @@ bool Params::snr = false; // compute/show snr while adding waterma
int
Params
::
have_key
=
0
;
int
Params
::
test_cut
=
0
;
// for sync test
bool
Params
::
test_no_sync
=
false
;
// disable sync
bool
Params
::
test_no_limiter
=
false
;
// disable limiter
int
Params
::
test_truncate
=
0
;
Format
Params
::
input_format
=
Format
::
AUTO
;
...
...
src/wmcommon.hh
View file @
d1276514
...
...
@@ -37,8 +37,12 @@ public:
static
constexpr
size_t
frames_pad_start
=
250
;
// padding at start, in case track starts with silence
static
constexpr
int
mark_sample_rate
=
44100
;
// watermark generation and detection sample rate
static
constexpr
double
limiter_block_size_ms
=
1000
;
static
constexpr
double
limiter_ceiling
=
0.99
;
static
int
test_cut
;
// for sync test
static
bool
test_no_sync
;
static
bool
test_no_limiter
;
static
int
test_truncate
;
static
Format
input_format
;
...
...
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