Index: webrtc/audio/audio_receive_stream_unittest.cc |
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc |
index 9d6aa9bdb4c4f02552e0a0fc28a6338ee92f89cc..2e789f92c280713eccae2ba8ebf1b7d216c8cdf0 100644 |
--- a/webrtc/audio/audio_receive_stream_unittest.cc |
+++ b/webrtc/audio/audio_receive_stream_unittest.cc |
@@ -85,6 +85,7 @@ struct ConfigHelper { |
EXPECT_FALSE(channel_proxy_); |
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1); |
+ EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1); |
EXPECT_CALL(*channel_proxy_, |
SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId)) |
.Times(1); |
@@ -112,6 +113,7 @@ struct ConfigHelper { |
stream_config_.voe_channel_id = kChannelId; |
stream_config_.rtp.local_ssrc = kLocalSsrc; |
stream_config_.rtp.remote_ssrc = kRemoteSsrc; |
+ stream_config_.rtp.nack.rtp_history_ms = 300; |
stream_config_.rtp.extensions.push_back( |
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
stream_config_.rtp.extensions.push_back( |
@@ -234,9 +236,9 @@ TEST(AudioReceiveStreamTest, ConfigToString) { |
RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
config.voe_channel_id = kChannelId; |
EXPECT_EQ( |
- "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{uri: " |
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}], " |
- "transport_cc: off}, " |
+ "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, " |
+ "nack: {rtp_history_ms: 0}, extensions: [{uri: " |
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, " |
"rtcp_send_transport: nullptr, " |
"voe_channel_id: 2}", |
config.ToString()); |