c1eb560a5c
* The old resampler was found to have a wraparound bug. * Remove support for the old resampler from PushResampler. * Use PushResampler in AudioCodingModule. * The old resampler must still be removed from the file utility. BUG=webrtc:1867,webrtc:827 TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio R=henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1590004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
29 lines
1.1 KiB
C++
29 lines
1.1 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 "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.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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TEST(PushResamplerTest, VerifiesInputParameters) {
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PushResampler resampler;
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EXPECT_EQ(-1, resampler.InitializeIfNeeded(-1, 16000, 1));
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EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, -1, 1));
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EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 0));
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EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 3));
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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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}
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} // namespace webrtc
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