Index: webrtc/modules/audio_processing/audio_processing_unittest.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc |
index 948c5efd93f4a3fc77a5974304445d88d9fa935d..ded75c8652079e4c2a523f98a5f2ace80c1d0ce3 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc |
@@ -54,7 +54,12 @@ bool write_ref_data = false; |
const google::protobuf::int32 kChannels[] = {1, 2}; |
const int kSampleRates[] = {8000, 16000, 32000, 48000}; |
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
+// Android doesn't support 48kHz. |
+const int kProcessSampleRates[] = {8000, 16000, 32000}; |
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
const int kProcessSampleRates[] = {8000, 16000, 32000, 48000}; |
+#endif |
enum StreamDirection { kForward = 0, kReverse }; |
@@ -2692,7 +2697,7 @@ INSTANTIATE_TEST_CASE_P( |
std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0), |
std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20), |
std::tr1::make_tuple(16000, 16000, 48000, 16000, 40, 20), |
- std::tr1::make_tuple(16000, 16000, 32000, 16000, 50, 20), |
+ std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20), |
std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0))); |
#elif defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
@@ -2748,7 +2753,7 @@ INSTANTIATE_TEST_CASE_P( |
std::tr1::make_tuple(16000, 32000, 32000, 32000, 25, 0), |
std::tr1::make_tuple(16000, 32000, 16000, 32000, 25, 20), |
std::tr1::make_tuple(16000, 16000, 48000, 16000, 35, 20), |
- std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20), |
+ std::tr1::make_tuple(16000, 16000, 32000, 16000, 35, 20), |
std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0))); |
#endif |