Index: webrtc/modules/audio_processing/test/debug_dump_test.cc |
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc |
index d2dd9c8b5a6416c69f2217972e9e7b4462ff39f4..005faa0f44c51f8188fcafb61f636d5565481252 100644 |
--- a/webrtc/modules/audio_processing/test/debug_dump_test.cc |
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc |
@@ -327,7 +327,8 @@ void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) { |
else |
apm_->set_stream_key_pressed(true); |
- ASSERT_EQ(input_config_.num_channels(), msg.input_channel_size()); |
+ ASSERT_EQ(input_config_.num_channels(), |
+ static_cast<size_t>(msg.input_channel_size())); |
ASSERT_EQ(input_config_.num_frames() * sizeof(float), |
msg.input_channel(0).size()); |
@@ -341,7 +342,8 @@ void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) { |
output_config_, output_->channels())); |
// Check that output of APM is bit-exact to the output in the dump. |
- ASSERT_EQ(output_config_.num_channels(), msg.output_channel_size()); |
+ ASSERT_EQ(output_config_.num_channels(), |
+ static_cast<size_t>(msg.output_channel_size())); |
ASSERT_EQ(output_config_.num_frames() * sizeof(float), |
msg.output_channel(0).size()); |
for (int i = 0; i < msg.output_channel_size(); ++i) { |
@@ -355,7 +357,8 @@ void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) { |
ASSERT_TRUE(apm_.get()); |
ASSERT_GT(msg.channel_size(), 0); |
- ASSERT_EQ(reverse_config_.num_channels(), msg.channel_size()); |
+ ASSERT_EQ(reverse_config_.num_channels(), |
+ static_cast<size_t>(msg.channel_size())); |
ASSERT_EQ(reverse_config_.num_frames() * sizeof(float), |
msg.channel(0).size()); |