build: Make build succeed without test and non-audio deps

5 files changed, 18 insertions, 0 deletions

diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h
index 6050090..a308d7b 100644
--- a/webrtc/common_audio/channel_buffer.h
+++ b/webrtc/common_audio/channel_buffer.h
@@ -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 {
 
diff --git a/webrtc/common_audio/resampler/sinc_resampler.h b/webrtc/common_audio/resampler/sinc_resampler.h
index 189112b..95ce44e 100644
--- a/webrtc/common_audio/resampler/sinc_resampler.h
+++ b/webrtc/common_audio/resampler/sinc_resampler.h
@@ -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);
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
index 1e9e35a..0215426 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
@@ -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,
diff --git a/webrtc/modules/audio_processing/transient/transient_suppressor.h b/webrtc/modules/audio_processing/transient/transient_suppressor.h
index 5a6f117..b62dcde 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppressor.h
+++ b/webrtc/modules/audio_processing/transient/transient_suppressor.h
@@ -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);
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 67019ca..18699d8 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -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 {