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 2939453e4f9a7011a6db48a6e88e341ef72f3774..1b400262ae0c633b187cbb8b41c6fa3c78ad5cdf 100644 |
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc |
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc |
@@ -30,12 +30,12 @@ class TargetDelayTest : public ::testing::Test { |
void SetUp() { |
EXPECT_TRUE(acm_.get() != NULL); |
- CodecInst codec; |
- ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1)); |
ASSERT_EQ(0, acm_->InitializeReceiver()); |
- ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec)); |
+ constexpr int pltype = 108; |
+ ASSERT_EQ(true, |
+ acm_->RegisterReceiveCodec(pltype, {"L16", kSampleRateHz, 1})); |
- rtp_info_.header.payloadType = codec.pltype; |
+ rtp_info_.header.payloadType = pltype; |
rtp_info_.header.timestamp = 0; |
rtp_info_.header.ssrc = 0x12345678; |
rtp_info_.header.markerBit = false; |