Configure VoE NACK through AudioSendStream::Config.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 4d9d465c65a832a4ae7aede7f7853895945c724b..dcccdde729d41348d3e9356f3d9c4f318e16ac5b 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -74,6 +74,7 @@ struct ConfigHelper {
           EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
           EXPECT_CALL(*channel_proxy_,
@@ -97,6 +98,7 @@ struct ConfigHelper {
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
+    stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
@@ -179,8 +181,8 @@ TEST(AudioSendStreamTest, ConfigToString) {
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
-      "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
-      "red_payload_type: 17}",
+      "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, "
+      "cng_payload_type: 42, red_payload_type: 17}",
       config.ToString());
 }