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Unified Diff: webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc

Issue 1228843002: Update audio code to use size_t more correctly, (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Review comments Created 5 years, 4 months ago
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Index: webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index ea88f24a17cffcb45b9bf420b7b332ec31cfd879..5564e20267f39fbb6aa93765e7768e07341ed41f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -43,7 +43,7 @@ struct TestParameters {
class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
protected:
static const int kTimeStepMs = 10;
- static const int kMaxBlockSize = 480; // 10 ms @ 48 kHz.
+ static const size_t kMaxBlockSize = 480; // 10 ms @ 48 kHz.
static const uint8_t kPayloadTypeMono = 95;
static const uint8_t kPayloadTypeMulti = 96;
@@ -52,7 +52,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
sample_rate_hz_(GetParam().sample_rate),
samples_per_ms_(sample_rate_hz_ / 1000),
frame_size_ms_(GetParam().frame_size),
- frame_size_samples_(frame_size_ms_ * samples_per_ms_),
+ frame_size_samples_(
+ static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
output_size_samples_(10 * samples_per_ms_),
rtp_generator_mono_(samples_per_ms_),
rtp_generator_(samples_per_ms_),
@@ -212,7 +213,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
}
NetEqOutputType output_type;
// Get audio from mono instance.
- int samples_per_channel;
+ size_t samples_per_channel;
int num_channels;
EXPECT_EQ(NetEq::kOK,
neteq_mono_->GetAudio(kMaxBlockSize, output_,
@@ -242,8 +243,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
const int sample_rate_hz_;
const int samples_per_ms_;
const int frame_size_ms_;
- const int frame_size_samples_;
- const int output_size_samples_;
+ const size_t frame_size_samples_;
+ const size_t output_size_samples_;
NetEq* neteq_mono_;
NetEq* neteq_;
test::RtpGenerator rtp_generator_mono_;
@@ -256,8 +257,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
int16_t* output_multi_channel_;
WebRtcRTPHeader rtp_header_mono_;
WebRtcRTPHeader rtp_header_;
- int payload_size_bytes_;
- int multi_payload_size_bytes_;
+ size_t payload_size_bytes_;
+ size_t multi_payload_size_bytes_;
int last_send_time_;
int last_arrival_time_;
rtc::scoped_ptr<test::InputAudioFile> input_file_;
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