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

webrtc/api/webrtcsdp_unittest.cc

 /*
  *  Copyright 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1877 matching lines @@
   std::string sdp_with_bundle = kSdpFullString;
   InjectAfter(kSessionTime,
               "a=group:BUNDLE audio_content_name video_content_name\r\n",
               &sdp_with_bundle);
   EXPECT_EQ(sdp_with_bundle, message);
 }
 
 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()));
   std::string message = webrtc::SdpSerialize(jdesc_, false);
   std::string sdp_with_bandwidth = kSdpFullString;
   InjectAfter("c=IN IP4 74.125.224.39\r\n",
               "b=AS:100\r\n",
               &sdp_with_bandwidth);
   InjectAfter("c=IN IP4 74.125.127.126\r\n",
               "b=AS:50\r\n",
@@ skipping 98 matching lines @@
   rtc::sprintfn(new_portstr, sizeof(new_portstr), "%d", kNewPort);
   rtc::replace_substrs(default_portstr, strlen(default_portstr),
                              new_portstr, strlen(new_portstr),
                              &expected_sdp);
 
   EXPECT_EQ(expected_sdp, message);
 }
 
 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));
   std::string message = webrtc::SdpSerialize(jsep_desc, false);
 
   std::string expected_sdp = kSdpString;
   expected_sdp.append(kSdpRtpDataChannelString);
   // Serializing data content shouldn't ignore bandwidth settings.
   InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
               "b=AS:100\r\n",
@@ skipping 195 matching lines @@
   InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n",
               "b=AS:100\r\n",
               &sdp_with_bandwidth);
   InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n",
               "b=AS:50\r\n",
               &sdp_with_bandwidth);
   EXPECT_TRUE(
       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()));
   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
   JsepSessionDescription jdesc_with_ice_options(kDummyString);
   std::string sdp_with_ice_options = kSdpFullString;
   InjectAfter(kSessionTime,
@@ skipping 374 matching lines @@
   JsepSessionDescription jdesc_with_bandwidth(kDummyString);
 
   EXPECT_FALSE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
 }
 
 TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
   AddSctpDataChannel();
   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;
   sdp_with_bandwidth.append(kSdpSctpDataChannelString);
   InjectAfter("a=mid:data_content_name\r\n",
               "b=AS:100\r\n",
               &sdp_with_bandwidth);
   JsepSessionDescription jdesc_with_bandwidth(kDummyString);
 
   // SCTP has congestion control, so we shouldn't limit the bandwidth
@@ skipping 467 matching lines @@
   EXPECT_TRUE(
       SdpDeserialize(kUnifiedPlanSdpFullString, &deserialized_description));
 
   EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
 }
 
 TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) {
   MakeUnifiedPlanDescription();
   TestSerialize(jdesc_, true);
 }