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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
53 lines
1.8 KiB
C++
53 lines
1.8 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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// Quality testing of PushResampler is handled through output_mixer_unittest.cc.
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namespace webrtc {
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// The below tests are temporarily disabled on WEBRTC_WIN due to problems
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// with clang debug builds.
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// TODO(tommi): Re-enable when we've figured out what the problem is.
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// http://crbug.com/615050
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#if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG)
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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(PushResamplerTest, VerifiesBadInputParameters1) {
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PushResampler<int16_t> resampler;
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EXPECT_DEATH(resampler.InitializeIfNeeded(-1, 16000, 1),
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"src_sample_rate_hz");
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}
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TEST(PushResamplerTest, VerifiesBadInputParameters2) {
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PushResampler<int16_t> resampler;
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EXPECT_DEATH(resampler.InitializeIfNeeded(16000, -1, 1),
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"dst_sample_rate_hz");
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}
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TEST(PushResamplerTest, VerifiesBadInputParameters3) {
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PushResampler<int16_t> resampler;
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EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0), "num_channels");
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}
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#endif
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#endif
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} // namespace webrtc
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