Refactoring of RTCP options in BaseChannel.

webrtc/api/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 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 83 matching lines @@
   *os << stats.ToString();
 }
 
 void PrintTo(const RTCTransportStats& stats, ::std::ostream* os) {
   *os << stats.ToString();
 }
 
 namespace {
 
 const int64_t kGetStatsReportTimeoutMs = 1000;
-const bool kDefaultRtcpEnabled = false;
+const bool kDefaultRtcpMuxRequired = true;
 const bool kDefaultSrtpRequired = true;
 
 struct CertificateInfo {
   rtc::scoped_refptr<rtc::RTCCertificate> certificate;
   std::vector<std::string> ders;
   std::vector<std::string> pems;
   std::vector<std::string> fingerprints;
 };
 
 std::unique_ptr<CertificateInfo> CreateFakeCertificateAndInfoFromDers(
@@ skipping 476 matching lines @@
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
   ExpectReportContainsCertificateInfo(report, *local_certinfo);
   ExpectReportContainsCertificateInfo(report, *remote_certinfo);
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
       test_->media_engine(), voice_media_channel, "VoiceContentName",
-      kDefaultRtcpEnabled, kDefaultSrtpRequired);
+      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
 
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
-      video_media_channel, "VideoContentName", kDefaultRtcpEnabled,
+      video_media_channel, "VideoContentName", kDefaultRtcpMuxRequired,
       kDefaultSrtpRequired);
 
   // Audio
   cricket::VoiceMediaInfo voice_media_info;
 
   RtpCodecParameters inbound_audio_codec;
   inbound_audio_codec.payload_type = 1;
   inbound_audio_codec.mime_type = "opus";
   inbound_audio_codec.clock_rate = 1337;
   voice_media_info.receive_codecs.insert(
@@ skipping 822 matching lines @@
   EXPECT_EQ(expected_remote_video_track,
             report->Get(expected_remote_video_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
       test_->media_engine(), voice_media_channel, "VoiceContentName",
-      kDefaultRtcpEnabled, kDefaultSrtpRequired);
+      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
 
   cricket::VoiceMediaInfo voice_media_info;
 
   voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
   voice_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   voice_media_info.receivers[0].local_stats[0].ssrc = 1;
   voice_media_info.receivers[0].packets_lost = 42;
   voice_media_info.receivers[0].packets_rcvd = 2;
   voice_media_info.receivers[0].bytes_rcvd = 3;
@@ skipping 51 matching lines @@
   EXPECT_EQ(audio, expected_audio);
 
   ASSERT_TRUE(report->Get(*expected_audio.transport_id));
   ASSERT_TRUE(report->Get(*expected_audio.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
-      video_media_channel, "VideoContentName", kDefaultRtcpEnabled,
+      video_media_channel, "VideoContentName", kDefaultRtcpMuxRequired,
       kDefaultSrtpRequired);
 
   cricket::VideoMediaInfo video_media_info;
 
   video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
   video_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   video_media_info.receivers[0].local_stats[0].ssrc = 1;
   video_media_info.receivers[0].packets_rcvd = 2;
   video_media_info.receivers[0].packets_lost = 42;
@@ skipping 59 matching lines @@
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*video.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
       test_->media_engine(), voice_media_channel, "VoiceContentName",
-      kDefaultRtcpEnabled, kDefaultSrtpRequired);
+      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
 
   cricket::VoiceMediaInfo voice_media_info;
 
   voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
   voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
   voice_media_info.senders[0].local_stats[0].ssrc = 1;
   voice_media_info.senders[0].packets_sent = 2;
   voice_media_info.senders[0].bytes_sent = 3;
   voice_media_info.senders[0].rtt_ms = 4500;
   voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
@@ skipping 46 matching lines @@
   EXPECT_EQ(audio, expected_audio);
 
   ASSERT_TRUE(report->Get(*expected_audio.transport_id));
   ASSERT_TRUE(report->Get(*expected_audio.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
-      video_media_channel, "VideoContentName", kDefaultRtcpEnabled,
+      video_media_channel, "VideoContentName", kDefaultRtcpMuxRequired,
       kDefaultSrtpRequired);
 
   cricket::VideoMediaInfo video_media_info;
 
   video_media_info.senders.push_back(cricket::VideoSenderInfo());
   video_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
   video_media_info.senders[0].local_stats[0].ssrc = 1;
   video_media_info.senders[0].firs_rcvd = 2;
   video_media_info.senders[0].plis_rcvd = 3;
   video_media_info.senders[0].nacks_rcvd = 4;
@@ skipping 58 matching lines @@
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*expected_video.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
       test_->media_engine(), voice_media_channel, "VoiceContentName",
-      kDefaultRtcpEnabled, kDefaultSrtpRequired);
+      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), nullptr,
-      video_media_channel, "VideoContentName", kDefaultRtcpEnabled,
+      video_media_channel, "VideoContentName", kDefaultRtcpMuxRequired,
       kDefaultSrtpRequired);
 
   cricket::VoiceMediaInfo voice_media_info;
   voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
   voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
   voice_media_info.senders[0].local_stats[0].ssrc = 1;
   voice_media_info.senders[0].packets_sent = 2;
   voice_media_info.senders[0].bytes_sent = 3;
   voice_media_info.senders[0].rtt_ms = -1;
   voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
@@ skipping 249 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc