| 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
|
|
|
|
|