Reenable WebRTC PushResampler format checks on Windows clang debug builds
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}
diff --git a/common_audio/resampler/push_resampler.cc b/common_audio/resampler/push_resampler.cc
index 837fec6..810d778 100644
--- a/common_audio/resampler/push_resampler.cc
+++ b/common_audio/resampler/push_resampler.cc
@@ -20,42 +20,6 @@
#include "rtc_base/checks.h"
namespace webrtc {
-namespace {
-// These checks were factored out into a non-templatized function
-// due to problems with clang on Windows in debug builds.
-// For some reason having the DCHECKs inline in the template code
-// caused the compiler to generate code that threw off the linker.
-// TODO(tommi): Re-enable when we've figured out what the problem is.
-// http://crbug.com/615050
-void CheckValidInitParams(int src_sample_rate_hz,
- int dst_sample_rate_hz,
- size_t num_channels) {
-// The below checks are temporarily disabled on WEBRTC_WIN due to problems
-// with clang debug builds.
-#if !defined(WEBRTC_WIN) && defined(__clang__)
- RTC_DCHECK_GT(src_sample_rate_hz, 0);
- RTC_DCHECK_GT(dst_sample_rate_hz, 0);
- RTC_DCHECK_GT(num_channels, 0);
-#endif
-}
-
-void CheckExpectedBufferSizes(size_t src_length,
- size_t dst_capacity,
- size_t num_channels,
- int src_sample_rate,
- int dst_sample_rate) {
-// The below checks are temporarily disabled on WEBRTC_WIN due to problems
-// with clang debug builds.
-// TODO(tommi): Re-enable when we've figured out what the problem is.
-// http://crbug.com/615050
-#if !defined(WEBRTC_WIN) && defined(__clang__)
- const size_t src_size_10ms = (src_sample_rate / 100) * num_channels;
- const size_t dst_size_10ms = (dst_sample_rate / 100) * num_channels;
- RTC_DCHECK_EQ(src_length, src_size_10ms);
- RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
-#endif
-}
-} // namespace
template <typename T>
PushResampler<T>::PushResampler()
@@ -68,7 +32,11 @@
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
size_t num_channels) {
- CheckValidInitParams(src_sample_rate_hz, dst_sample_rate_hz, num_channels);
+ // These checks used to be factored out of this template function due to
+ // Windows debug build issues with clang. http://crbug.com/615050
+ RTC_DCHECK_GT(src_sample_rate_hz, 0);
+ RTC_DCHECK_GT(dst_sample_rate_hz, 0);
+ RTC_DCHECK_GT(num_channels, 0);
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
@@ -109,8 +77,12 @@
size_t src_length,
T* dst,
size_t dst_capacity) {
- CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_,
- src_sample_rate_hz_, dst_sample_rate_hz_);
+ // These checks used to be factored out of this template function due to
+ // Windows debug build issues with clang. http://crbug.com/615050
+ const size_t src_size_10ms = (src_sample_rate_hz_ / 100) * num_channels_;
+ const size_t dst_size_10ms = (dst_sample_rate_hz_ / 100) * num_channels_;
+ RTC_DCHECK_EQ(src_length, src_size_10ms);
+ RTC_DCHECK_GE(dst_capacity, dst_size_10ms);
if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
// The old resampler provides this memcpy facility in the case of matching
diff --git a/common_audio/resampler/push_resampler_unittest.cc b/common_audio/resampler/push_resampler_unittest.cc
index f785ed9..91f2233 100644
--- a/common_audio/resampler/push_resampler_unittest.cc
+++ b/common_audio/resampler/push_resampler_unittest.cc
@@ -18,11 +18,6 @@
namespace webrtc {
-// The below tests are temporarily disabled on WEBRTC_WIN due to problems
-// with clang debug builds.
-// TODO(tommi): Re-enable when we've figured out what the problem is.
-// http://crbug.com/615050
-#if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG)
TEST(PushResamplerTest, VerifiesInputParameters) {
PushResampler<int16_t> resampler;
EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
@@ -48,8 +43,6 @@
RTC_EXPECT_DEATH(resampler.InitializeIfNeeded(16000, 16000, 0),
"num_channels");
}
-
-#endif
#endif
} // namespace webrtc
