1 files changed, 48 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc b/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc
new file mode 100644
index 0000000000..91f2233aad
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/resampler/push_resampler_unittest.cc
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_audio/resampler/include/push_resampler.h"
+
+#include "rtc_base/checks.h"  // RTC_DCHECK_IS_ON
+#include "test/gtest.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+// Quality testing of PushResampler is done in audio/remix_resample_unittest.cc.
+
+namespace webrtc {
+
+TEST(PushResamplerTest, VerifiesInputParameters) {
+  PushResampler<int16_t> resampler;
+  EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
+  EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
+  EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 8));
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) {
+  PushResampler<int16_t> resampler;
+  RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
+                   "src_sample_rate_hz");
+}
+
+TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) {
+  PushResampler<int16_t> resampler;
+  RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
+                   "dst_sample_rate_hz");
+}
+
+TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) {
+  PushResampler<int16_t> resampler;
+  RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0),
+                   "num_channels");
+}
+#endif
+
+}  // namespace webrtc