| 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_);
|
|
|