Skip to content
GitLab
Projects
Groups
Snippets
/
Help
Help
Support
Community forum
Keyboard shortcuts
?
Submit feedback
Contribute to GitLab
Sign in / Register
Toggle navigation
Menu
Open sidebar
Arun Raghavan
webrtc-audio-processing
Commits
407bfbf6
Commit
407bfbf6
authored
Oct 13, 2015
by
Arun Raghavan
🐾
Browse files
build: Make build succeed without test and non-audio deps
parent
753eada3
Changes
6
Hide whitespace changes
Inline
Side-by-side
configure.ac
View file @
407bfbf6
...
...
@@ -24,8 +24,9 @@ AS_CASE(["x${with_ns_mode}"],
[NS_FIXED=0])
AM_CONDITIONAL(NS_FIXED, [test "x${NS_FIXED}" = "x1"])
COMMON_CFLAGS="-DWEBRTC_POSIX -DWEBRTC_LINUX -DNDEBUG -I\$(top_srcdir)"
COMMON_CXXFLAGS="-std=c++11 -DWEBRTC_POSIX -DWEBRTC_LINUX -DNDEBUG -I\$(top_srcdir)"
# FIXME: POSIX and LINUX should not be hard-coded
COMMON_CFLAGS="-DWEBRTC_AUDIO_PROCESSING_ONLY_BUILD -DWEBRTC_POSIX -DWEBRTC_LINUX -DNDEBUG -I\$(top_srcdir)"
COMMON_CXXFLAGS="-std=c++11 -DWEBRTC_AUDIO_PROCESSING_ONLY_BUILD -DWEBRTC_POSIX -DWEBRTC_LINUX -DNDEBUG -I\$(top_srcdir)"
AC_SUBST([COMMON_CFLAGS])
AC_SUBST([COMMON_CXXFLAGS])
...
...
webrtc/common_audio/channel_buffer.h
View file @
407bfbf6
...
...
@@ -16,7 +16,9 @@
#include
"webrtc/base/checks.h"
#include
"webrtc/base/scoped_ptr.h"
#include
"webrtc/common_audio/include/audio_util.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include
"webrtc/test/testsupport/gtest_prod_util.h"
#endif
namespace
webrtc
{
...
...
webrtc/common_audio/resampler/sinc_resampler.h
View file @
407bfbf6
...
...
@@ -17,7 +17,9 @@
#include
"webrtc/base/constructormagic.h"
#include
"webrtc/base/scoped_ptr.h"
#include
"webrtc/system_wrappers/interface/aligned_malloc.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include
"webrtc/test/testsupport/gtest_prod_util.h"
#endif
#include
"webrtc/typedefs.h"
namespace
webrtc
{
...
...
@@ -85,8 +87,10 @@ class SincResampler {
float
*
get_kernel_for_testing
()
{
return
kernel_storage_
.
get
();
}
private:
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
FRIEND_TEST_ALL_PREFIXES
(
SincResamplerTest
,
Convolve
);
FRIEND_TEST_ALL_PREFIXES
(
SincResamplerTest
,
ConvolveBenchmark
);
#endif
void
InitializeKernel
();
void
UpdateRegions
(
bool
second_load
);
...
...
webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
View file @
407bfbf6
...
...
@@ -95,8 +95,10 @@ class IntelligibilityEnhancer {
AudioSource
source_
;
};
friend
class
TransformCallback
;
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
FRIEND_TEST_ALL_PREFIXES
(
IntelligibilityEnhancerTest
,
TestErbCreation
);
FRIEND_TEST_ALL_PREFIXES
(
IntelligibilityEnhancerTest
,
TestSolveForGains
);
#endif
// Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source.
void
DispatchAudio
(
AudioSource
source
,
...
...
webrtc/modules/audio_processing/transient/transient_suppressor.h
View file @
407bfbf6
...
...
@@ -15,7 +15,9 @@
#include
<set>
#include
"webrtc/base/scoped_ptr.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include
"webrtc/test/testsupport/gtest_prod_util.h"
#endif
#include
"webrtc/typedefs.h"
namespace
webrtc
{
...
...
@@ -59,8 +61,10 @@ class TransientSuppressor {
bool
key_pressed
);
private:
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
FRIEND_TEST_ALL_PREFIXES
(
TransientSuppressorTest
,
TypingDetectionLogicWorksAsExpectedForMono
);
#endif
void
Suppress
(
float
*
in_ptr
,
float
*
spectral_mean
,
float
*
out_ptr
);
void
UpdateKeypress
(
bool
key_pressed
);
...
...
webrtc/modules/interface/module_common_types.h
View file @
407bfbf6
...
...
@@ -19,7 +19,9 @@
#include
"webrtc/base/constructormagic.h"
#include
"webrtc/common_types.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include
"webrtc/common_video/rotation.h"
#endif
#include
"webrtc/typedefs.h"
namespace
webrtc
{
...
...
@@ -233,6 +235,7 @@ enum RtpVideoCodecTypes {
kRtpVideoVp9
,
kRtpVideoH264
};
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
// Since RTPVideoHeader is used as a member of a union, it can't have a
// non-trivial default constructor.
struct
RTPVideoHeader
{
...
...
@@ -246,9 +249,12 @@ struct RTPVideoHeader {
RtpVideoCodecTypes
codec
;
RTPVideoTypeHeader
codecHeader
;
};
#endif
union
RTPTypeHeader
{
RTPAudioHeader
Audio
;
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
RTPVideoHeader
Video
;
#endif
};
struct
WebRtcRTPHeader
{
...
...
Write
Preview
Supports
Markdown
0%
Try again
or
attach a new file
.
Cancel
You are about to add
0
people
to the discussion. Proceed with caution.
Finish editing this message first!
Cancel
Please
register
or
sign in
to comment