RTCRTPStreamStats.ssrc changed type to uint32_t.

webrtc/pc/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 1704 matching lines @@
       [&session_stats](const ChannelNamePairs&) {
         return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
       }));
   EXPECT_CALL(test_->session(), voice_channel())
       .WillRepeatedly(Return(&voice_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCInboundRTPStreamStats expected_audio(
       "RTCInboundRTPAudioStream_1", report->timestamp_us());
-  expected_audio.ssrc = "1";
+  expected_audio.ssrc = 1;
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.track_id =
       "RTCMediaStreamTrack_remote_audio_RemoteAudioTrackID_1";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_InboundAudio_42";
   expected_audio.packets_received = 2;
   expected_audio.bytes_received = 3;
   expected_audio.packets_lost = 42;
@@ skipping 62 matching lines @@
       [&session_stats](const ChannelNamePairs&) {
         return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
       }));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCInboundRTPStreamStats expected_video(
       "RTCInboundRTPVideoStream_1", report->timestamp_us());
-  expected_video.ssrc = "1";
+  expected_video.ssrc = 1;
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.track_id =
       "RTCMediaStreamTrack_remote_video_RemoteVideoTrackID_1";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_InboundVideo_42";
   expected_video.fir_count = 5;
   expected_video.pli_count = 6;
   expected_video.nack_count = 7;
@@ skipping 59 matching lines @@
       [&session_stats](const ChannelNamePairs&) {
         return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
       }));
   EXPECT_CALL(test_->session(), voice_channel())
       .WillRepeatedly(Return(&voice_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCOutboundRTPStreamStats expected_audio(
       "RTCOutboundRTPAudioStream_1", report->timestamp_us());
-  expected_audio.ssrc = "1";
+  expected_audio.ssrc = 1;
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.track_id =
       "RTCMediaStreamTrack_local_audio_LocalAudioTrackID_1";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
   expected_audio.packets_sent = 2;
   expected_audio.bytes_sent = 3;
   expected_audio.round_trip_time = 4.5;
@@ skipping 59 matching lines @@
       [&session_stats](const ChannelNamePairs&) {
         return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
       }));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCOutboundRTPStreamStats expected_video(
       "RTCOutboundRTPVideoStream_1", report->timestamp_us());
-  expected_video.ssrc = "1";
+  expected_video.ssrc = 1;
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.track_id =
       "RTCMediaStreamTrack_local_video_LocalVideoTrackID_1";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
@@ skipping 70 matching lines @@
       }));
   EXPECT_CALL(test_->session(), voice_channel())
       .WillRepeatedly(Return(&voice_channel));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCOutboundRTPStreamStats expected_audio(
       "RTCOutboundRTPAudioStream_1", report->timestamp_us());
-  expected_audio.ssrc = "1";
+  expected_audio.ssrc = 1;
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
   expected_audio.packets_sent = 2;
   expected_audio.bytes_sent = 3;
   // |expected_audio.round_trip_time| should be undefined.
 
   ASSERT_TRUE(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   RTCOutboundRTPStreamStats expected_video(
       "RTCOutboundRTPVideoStream_1", report->timestamp_us());
-  expected_video.ssrc = "1";
+  expected_video.ssrc = 1;
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
   expected_video.packets_sent = 5;
   expected_video.bytes_sent = 6;
@@ skipping 178 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc