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Unified Diff: webrtc/modules/audio_processing/audio_processing_unittest.cc

Issue 1865633005: Don't always downsample to 16kHz in the reverse stream in APM (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Simplify if-statement Created 4 years, 8 months ago
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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
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