Bug 1430255 - P2 - update RTPSources timebase, gtests;r?mjf draft
authorNico Grunbaum
Sun, 21 Jan 2018 15:13:13 -0800
changeset 751945 98d5cb88c30e33b83491af1a75ee6ebd59f27841
parent 751944 7247420884f847b6469ba286903ba42e9645a8be
push id98098
push userna-g@nostrum.com
push dateWed, 07 Feb 2018 07:02:32 +0000
reviewersmjf
bugs1430255
milestone60.0a1
Bug 1430255 - P2 - update RTPSources timebase, gtests;r?mjf MozReview-Commit-ID: 2Kl8squlrxC
media/webrtc/signaling/gtest/mediaconduit_unittests.cpp
media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
--- a/media/webrtc/signaling/gtest/mediaconduit_unittests.cpp
+++ b/media/webrtc/signaling/gtest/mediaconduit_unittests.cpp
@@ -373,22 +373,22 @@ class TransportConduitTest : public ::te
     mVideoTransport = nullptr;
   }
 
   //1. Dump audio samples to dummy external transport
   void TestDummyAudioAndTransport()
   {
     //get pointer to AudioSessionConduit
     int err=0;
-    mAudioSession = mozilla::AudioSessionConduit::Create();
+    mAudioSession = mozilla::AudioSessionConduit::Create(TimeStamp::Now());
     if( !mAudioSession ) {
       ASSERT_NE(mAudioSession, (void*)nullptr);
     }
 
-    mAudioSession2 = mozilla::AudioSessionConduit::Create();
+    mAudioSession2 = mozilla::AudioSessionConduit::Create(TimeStamp::Now());
     if( !mAudioSession2 ) {
       ASSERT_NE(mAudioSession2, (void*)nullptr);
     }
 
     WebrtcMediaTransport* xport = new WebrtcMediaTransport();
     ASSERT_NE(xport, (void*)nullptr);
     xport->SetAudioSession(mAudioSession, mAudioSession2);
     mAudioTransport = xport;
--- a/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
+++ b/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
@@ -246,17 +246,17 @@ class TransportInfo {
   TransportLayerLoopback *loopback_;
   TransportLayerDtls *dtls_;
 };
 
 class TestAgent {
  public:
   TestAgent() :
       audio_config_(109, "opus", 48000, 960, 2, 64000, false),
-      audio_conduit_(mozilla::AudioSessionConduit::Create()),
+      audio_conduit_(mozilla::AudioSessionConduit::Create(TimeStamp::Now())),
       audio_pipeline_(),
       use_bundle_(false) {
   }
 
   static void ConnectRtp(TestAgent *client, TestAgent *server) {
     TransportInfo::InitAndConnect(client->audio_rtp_transport_,
                                   server->audio_rtp_transport_);
   }