| Index: webrtc/common_audio/resampler/push_resampler_unittest.cc
|
| diff --git a/webrtc/common_audio/resampler/push_resampler_unittest.cc b/webrtc/common_audio/resampler/push_resampler_unittest.cc
|
| index 4449f4c633109afe24a61770268b9ed25e737b08..58880cc1b74a6cd479d748d94c278a93a5951eae 100644
|
| --- a/webrtc/common_audio/resampler/push_resampler_unittest.cc
|
| +++ b/webrtc/common_audio/resampler/push_resampler_unittest.cc
|
| @@ -9,6 +9,7 @@
|
| */
|
|
|
| #include "testing/gtest/include/gtest/gtest.h"
|
| +#include "webrtc/base/checks.h" // force defintion of RTC_DCHECK_IS_ON
|
| #include "webrtc/common_audio/resampler/include/push_resampler.h"
|
|
|
| // Quality testing of PushResampler is handled through output_mixer_unittest.cc.
|
| @@ -17,12 +18,32 @@
|
|
|
| TEST(PushResamplerTest, VerifiesInputParameters) {
|
| PushResampler<int16_t> resampler;
|
| - EXPECT_EQ(-1, resampler.InitializeIfNeeded(-1, 16000, 1));
|
| - EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, -1, 1));
|
| - EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 0));
|
| - EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 3));
|
| EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
|
| EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
|
| }
|
|
|
| +#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
|
| +TEST(PushResamplerTest, VerifiesBadInputParameters1) {
|
| + PushResampler<int16_t> resampler;
|
| + EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
|
| + "src_sample_rate_hz");
|
| +}
|
| +
|
| +TEST(PushResamplerTest, VerifiesBadInputParameters2) {
|
| + PushResampler<int16_t> resampler;
|
| + EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
|
| + "dst_sample_rate_hz");
|
| +}
|
| +
|
| +TEST(PushResamplerTest, VerifiesBadInputParameters3) {
|
| + PushResampler<int16_t> resampler;
|
| + EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), "num_channels");
|
| +}
|
| +
|
| +TEST(PushResamplerTest, VerifiesBadInputParameters4) {
|
| + PushResampler<int16_t> resampler;
|
| + EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 3), "num_channels");
|
| +}
|
| +#endif
|
| +
|
| } // namespace webrtc
|
|
|