Index: webrtc/modules/audio_coding/test/target_delay_unittest.cc |
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc |
index 99c1c2da1ee05a919fd01e40a9cd93693a53e58a..5de5bf262b289877018eafb0b04cf2bfe9593b90 100644 |
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc |
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc |
@@ -150,8 +150,10 @@ class TargetDelayTest : public ::testing::Test { |
// Pull audio equivalent to the amount of audio in one RTP packet. |
void Pull() { |
AudioFrame frame; |
+ bool muted; |
for (int k = 0; k < kNum10msPerFrame; ++k) { // Pull one frame. |
- ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame)); |
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame, &muted)); |
+ ASSERT_FALSE(muted); |
// Had to use ASSERT_TRUE, ASSERT_EQ generated error. |
ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_); |
ASSERT_EQ(1u, frame.num_channels_); |