Updated structures and functions for setting the max bitrate limit to take rtc::Optional<int>

webrtc/api/webrtcsdp_unittest.cc

Index: webrtc/api/webrtcsdp_unittest.cc
diff --git a/webrtc/api/webrtcsdp_unittest.cc b/webrtc/api/webrtcsdp_unittest.cc
index c52720453b7021042760dabc688efd5768dc94fd..c92c347d36076007abc388c032ad2a24136311a0 100644
--- a/webrtc/api/webrtcsdp_unittest.cc
+++ b/webrtc/api/webrtcsdp_unittest.cc
@@ -1895,10 +1895,10 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundle) {
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
   VideoContentDescription* vcd = static_cast<VideoContentDescription*>(
       GetFirstVideoContent(&desc_)->description);
-  vcd->set_bandwidth(100 * 1000);
+  vcd->set_bandwidth(rtc::Optional<int>(100 * 1000));
   AudioContentDescription* acd = static_cast<AudioContentDescription*>(
       GetFirstAudioContent(&desc_)->description);
-  acd->set_bandwidth(50 * 1000);
+  acd->set_bandwidth(rtc::Optional<int>(50 * 1000));
   ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
                                 jdesc_.session_id(),
                                 jdesc_.session_version()));
@@ -2017,7 +2017,7 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
 
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
   AddRtpDataChannel();
-  data_desc_->set_bandwidth(100*1000);
+  data_desc_->set_bandwidth(rtc::Optional<int>(100 * 1000));
   JsepSessionDescription jsep_desc(kDummyString);
 
   ASSERT_TRUE(jsep_desc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
@@ -2233,10 +2233,10 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
       SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
   VideoContentDescription* vcd = static_cast<VideoContentDescription*>(
       GetFirstVideoContent(&desc_)->description);
-  vcd->set_bandwidth(100 * 1000);
+  vcd->set_bandwidth(rtc::Optional<int>(100 * 1000));
   AudioContentDescription* acd = static_cast<AudioContentDescription*>(
       GetFirstAudioContent(&desc_)->description);
-  acd->set_bandwidth(50 * 1000);
+  acd->set_bandwidth(rtc::Optional<int>(50 * 1000));
   ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
                                 jdesc_.session_id(),
                                 jdesc_.session_version()));
@@ -2631,7 +2631,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
   JsepSessionDescription jdesc(kDummyString);
   DataContentDescription* dcd = static_cast<DataContentDescription*>(
      GetFirstDataContent(&desc_)->description);
-  dcd->set_bandwidth(100 * 1000);
+  dcd->set_bandwidth(rtc::Optional<int>(100 * 1000));
   ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
 
   std::string sdp_with_bandwidth = kSdpString;