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Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )

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Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled.  I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Committed: https://crrev.com/60c4e0ae8f124f08372645a95042f4a1246d7aa3
Cr-Commit-Position: refs/heads/master@{#12925}

Committed: https://crrev.com/5771beb265129082d31736259b7dc6ca037cff4d
Cr-Commit-Position: refs/heads/master@{#12926}

Committed: https://crrev.com/54e1c6a500e390e543bce7b78fae65eb9bb14ab6
Cr-Commit-Position: refs/heads/master@{#12927}

Review URL: https://codereview.webrtc.org/2014973002 .

Cr-Commit-Position: refs/heads/master@{#12928}
This commit is contained in:
Tommi 2016-05-26 22:21:31 +02:00
parent 54e1c6a500
commit f9d2fe983f
1 changed files with 25 additions and 4 deletions
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29
webrtc/common_audio/resampler/push_resampler_unittest.cc
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@ -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 @@ namespace webrtc {
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