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Unified Diff: webrtc/modules/audio_coding/test/target_delay_unittest.cc

Issue 1985743002: Propagate muted parameter to VoE::Channel (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Resurrect the PlayoutData10Ms(int, AudioFrame*) method Created 4 years, 7 months ago
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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_);
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