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/ )
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} Committed: https://crrev.com/f9d2fe983fe196373850c55acd3dc3824add480e Cr-Commit-Position: refs/heads/master@{#12928} Review URL: https://codereview.webrtc.org/2014973002 . Cr-Commit-Position: refs/heads/master@{#12929}
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@ -12,11 +12,33 @@
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#include <string.h>
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#include "webrtc/base/checks.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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namespace webrtc {
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namespace {
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// These checks were factored out into a non-templatized function
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// due to problems with clang on Windows in debug builds.
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// For some reason having the DCHECKs inline in the template code
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// caused the compiler to generate code that threw off the linker.
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void CheckValidInitParams(int src_sample_rate_hz, int dst_sample_rate_hz,
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size_t num_channels) {
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RTC_DCHECK_GT(src_sample_rate_hz, 0);
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RTC_DCHECK_GT(dst_sample_rate_hz, 0);
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RTC_DCHECK_GT(num_channels, 0u);
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RTC_DCHECK_LE(num_channels, 2u);
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}
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void CheckExpectedBufferSizes(size_t num_channels, int src_sample_rate,
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int dst_sample_rate) {
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const size_t src_size_10ms = src_sample_rate * num_channels / 100;
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const size_t dst_size_10ms = dst_sample_rate * num_channels / 100;
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RTC_CHECK_EQ(src_length, src_size_10ms);
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RTC_CHECK_GE(dst_capacity, dst_size_10ms);
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}
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}
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template <typename T>
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PushResampler<T>::PushResampler()
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@ -33,15 +55,19 @@ template <typename T>
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int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
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int dst_sample_rate_hz,
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size_t num_channels) {
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CheckValidInitParams(src_sample_rate_hz, dst_sample_rate_hz, num_channels);
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if (src_sample_rate_hz == src_sample_rate_hz_ &&
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dst_sample_rate_hz == dst_sample_rate_hz_ &&
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num_channels == num_channels_)
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num_channels == num_channels_) {
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// No-op if settings haven't changed.
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return 0;
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}
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if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
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num_channels <= 0 || num_channels > 2)
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if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 || num_channels <= 0 ||
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num_channels > 2) {
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return -1;
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}
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src_sample_rate_hz_ = src_sample_rate_hz;
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dst_sample_rate_hz_ = dst_sample_rate_hz;
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@ -68,10 +94,8 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
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template <typename T>
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int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
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size_t dst_capacity) {
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const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
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const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
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if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
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return -1;
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CheckExpectedBufferSizes(num_channels_, src_sample_rate_hz_,
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dst_sample_rate_hz_)
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if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
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// The old resampler provides this memcpy facility in the case of matching
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@ -22,7 +22,10 @@ TEST(PushResamplerTest, VerifiesInputParameters) {
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EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
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}
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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
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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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#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) && \
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!defined(WEBRTC_WIN)
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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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