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Unified Diff: webrtc/modules/audio_processing/test/audio_processing_unittest.cc

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 4 months ago
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Index: webrtc/modules/audio_processing/test/audio_processing_unittest.cc
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index 8384c364c122a323aa50a3d498b7a7367111b081..d82ea31c24b1713f6f87ac0e76e00f19df33dc3e 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -129,21 +129,23 @@ void VerifyChannelsAreEqual(int16_t* stereo, int samples_per_channel) {
}
void SetFrameTo(AudioFrame* frame, int16_t value) {
- for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
+ for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+ ++i) {
frame->data_[i] = value;
}
}
void SetFrameTo(AudioFrame* frame, int16_t left, int16_t right) {
ASSERT_EQ(2, frame->num_channels_);
- for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+ for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
frame->data_[i] = left;
frame->data_[i + 1] = right;
}
}
void ScaleFrame(AudioFrame* frame, float scale) {
- for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) {
+ for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+ ++i) {
frame->data_[i] = FloatS16ToS16(frame->data_[i] * scale);
}
}
@@ -676,13 +678,18 @@ void ApmTest::ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
}
// Calculate expected delay estimate and acceptable regions. Further,
// limit them w.r.t. AEC delay estimation support.
- const int samples_per_ms = std::min(16, frame_->samples_per_channel_ / 10);
+ const size_t samples_per_ms =
+ std::min(static_cast<size_t>(16), 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(std::max(
- expected_median + 96 / samples_per_ms, delay_min), delay_max);
- int expected_median_low = std::min(std::max(
- expected_median - 96 / samples_per_ms, delay_min), delay_max);
+ int expected_median_high = std::min(
+ std::max(expected_median + static_cast<int>(96 / samples_per_ms),
+ delay_min),
+ delay_max);
+ int expected_median_low = std::min(
+ std::max(expected_median - static_cast<int>(96 / samples_per_ms),
+ delay_min),
+ delay_max);
// Verify delay metrics.
int median;
int std;
@@ -998,8 +1005,8 @@ TEST_F(ApmTest, DISABLED_EchoCancellationReportsCorrectDelays) {
2,
false);
// Sampling frequency dependent variables.
- const int num_ms_per_block = std::max(4,
- 640 / frame_->samples_per_channel_);
+ const int num_ms_per_block =
+ std::max(4, static_cast<int>(640 / frame_->samples_per_channel_));
const int delay_min_ms = -kLookaheadBlocks * num_ms_per_block;
const int delay_max_ms = (kMaxDelayBlocks - 1) * num_ms_per_block;
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