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The referenced bug is fixed and issues like this haven't been observed for a long time. Follow up CL to https://webrtc-review.googlesource.com/c/src/+/268769/comments/8777e26e_0356bffb Bug: chromium:615050 Change-Id: I8f9e5db5a1b9c787867598e973a367a5c5e367df Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269761 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37699}
48 lines
1.6 KiB
C++
48 lines
1.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "common_audio/resampler/include/push_resampler.h"
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#include "rtc_base/checks.h" // RTC_DCHECK_IS_ON
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#include "test/gtest.h"
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#include "test/testsupport/rtc_expect_death.h"
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// Quality testing of PushResampler is done in audio/remix_resample_unittest.cc.
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namespace webrtc {
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TEST(PushResamplerTest, VerifiesInputParameters) {
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PushResampler<int16_t> resampler;
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EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
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EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
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EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 8));
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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TEST(PushResamplerDeathTest, VerifiesBadInputParameters1) {
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PushResampler<int16_t> resampler;
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RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
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"src_sample_rate_hz");
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}
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TEST(PushResamplerDeathTest, VerifiesBadInputParameters2) {
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PushResampler<int16_t> resampler;
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RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
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"dst_sample_rate_hz");
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}
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TEST(PushResamplerDeathTest, VerifiesBadInputParameters3) {
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PushResampler<int16_t> resampler;
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RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0),
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"num_channels");
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}
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#endif
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} // namespace webrtc
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