Add SafeMin() and SafeMax(), which accept args of different types
Specifically, they handle all combinations of two integer and two
floating-point arguments by picking a result type that is guaranteed
to be able to hold the result. This means callers no longer have to
deal with potentially dangerous casting to make all the arguments have
the same type, like they have to with std::min() and std::max().
Also, they're constexpr.
Mostly for illustrative purposes, this CL replaces a few std::min()
and std::max() calls with SafeMin() and SafeMax().
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2810483002
Cr-Commit-Position: refs/heads/master@{#17869}
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index 814eea9..c56a57b 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -21,6 +21,7 @@
#include "webrtc/base/gtest_prod_util.h"
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/base/protobuf_utils.h"
+#include "webrtc/base/safe_minmax.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
@@ -678,7 +679,7 @@
// Calculate expected delay estimate and acceptable regions. Further,
// limit them w.r.t. AEC delay estimation support.
const size_t samples_per_ms =
- std::min(static_cast<size_t>(16), frame_->samples_per_channel_ / 10);
+ rtc::SafeMin<size_t>(16u, frame_->samples_per_channel_ / 10);
int expected_median = std::min(std::max(delay_ms - system_delay_ms,
delay_min), delay_max);
int expected_median_high = std::min(