Use suffixed {uint,int}{8,16,32,64}_t types.

talk/app/webrtc/webrtcsession_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 1447 matching lines @@
   ASSERT_EQ(1u, voice_channel_->send_streams().size());
   EXPECT_TRUE(kAudioTrack1 == voice_channel_->send_streams()[0].id);
 
   // Create new offer without send streams.
   mediastream_signaling_.SendNothing();
   offer = CreateOffer();
 
   // Verify the session id is the same and the session version is
   // increased.
   EXPECT_EQ(session_id_orig, offer->session_id());
-  EXPECT_LT(rtc::FromString<uint64>(session_version_orig),
-            rtc::FromString<uint64>(offer->session_version()));
+  EXPECT_LT(rtc::FromString<uint64_t>(session_version_orig),
+            rtc::FromString<uint64_t>(offer->session_version()));
 
   SetLocalDescriptionWithoutError(offer);
   EXPECT_EQ(0u, video_channel_->send_streams().size());
   EXPECT_EQ(0u, voice_channel_->send_streams().size());
 
   mediastream_signaling_.SendAudioVideoStream2();
   answer = CreateRemoteAnswer(session_->local_description());
   SetRemoteDescriptionWithoutError(answer);
 
   // Make sure the receive streams have not changed.
@@ skipping 38 matching lines @@
   offer = CreateOffer();
   SetRemoteDescriptionWithoutError(offer);
 
   // Answer by turning off all send streams.
   mediastream_signaling_.SendNothing();
   answer = CreateAnswer(NULL);
 
   // Verify the session id is the same and the session version is
   // increased.
   EXPECT_EQ(session_id_orig, answer->session_id());
-  EXPECT_LT(rtc::FromString<uint64>(session_version_orig),
-            rtc::FromString<uint64>(answer->session_version()));
+  EXPECT_LT(rtc::FromString<uint64_t>(session_version_orig),
+            rtc::FromString<uint64_t>(answer->session_version()));
   SetLocalDescriptionWithoutError(answer);
 
   ASSERT_EQ(2u, video_channel_->recv_streams().size());
   EXPECT_TRUE(kVideoTrack1 == video_channel_->recv_streams()[0].id);
   EXPECT_TRUE(kVideoTrack2 == video_channel_->recv_streams()[1].id);
   ASSERT_EQ(2u, voice_channel_->recv_streams().size());
   EXPECT_TRUE(kAudioTrack1 == voice_channel_->recv_streams()[0].id);
   EXPECT_TRUE(kAudioTrack2 == voice_channel_->recv_streams()[1].id);
 
   // Make sure we have no send streams.
@@ skipping 1560 matching lines @@
   SetLocalDescriptionWithoutError(offer);
 }
 
 TEST_F(WebRtcSessionTest, SetAudioPlayout) {
   Init();
   mediastream_signaling_.SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->recv_streams().size());
-  uint32 receive_ssrc  = channel->recv_streams()[0].first_ssrc();
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
   double left_vol, right_vol;
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(1, left_vol);
   EXPECT_EQ(1, right_vol);
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   session_->SetAudioPlayout(receive_ssrc, false, renderer.get());
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(0, left_vol);
   EXPECT_EQ(0, right_vol);
   EXPECT_EQ(0, renderer->channel_id());
   session_->SetAudioPlayout(receive_ssrc, true, NULL);
   EXPECT_TRUE(channel->GetOutputScaling(receive_ssrc, &left_vol, &right_vol));
   EXPECT_EQ(1, left_vol);
   EXPECT_EQ(1, right_vol);
   EXPECT_EQ(-1, renderer->channel_id());
 }
 
 TEST_F(WebRtcSessionTest, SetAudioSend) {
   Init();
   mediastream_signaling_.SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
-  uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
 
   cricket::AudioOptions options;
   options.echo_cancellation.Set(true);
 
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   session_->SetAudioSend(send_ssrc, false, options, renderer.get());
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
   EXPECT_EQ(0, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // This will trigger SetSink(NULL) to the |renderer|.
   session_->SetAudioSend(send_ssrc, true, options, NULL);
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
   bool value;
   EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
   EXPECT_TRUE(value);
   EXPECT_EQ(-1, renderer->channel_id());
   EXPECT_TRUE(renderer->sink() == NULL);
 }
 
 TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
   Init();
   mediastream_signaling_.SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
-  uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
 
   rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
   cricket::AudioOptions options;
   session_->SetAudioSend(send_ssrc, true, options, renderer.get());
   EXPECT_TRUE(renderer->sink() != NULL);
 
   // Delete the |renderer| and it will trigger OnClose() to the sink, and this
   // will invalidate the |renderer_| pointer in the sink and prevent getting a
   // SetSink(NULL) callback afterwards.
   renderer.reset();
 
   // This will trigger SetSink(NULL) if no OnClose() callback.
   session_->SetAudioSend(send_ssrc, true, options, NULL);
 }
 
 TEST_F(WebRtcSessionTest, SetVideoPlayout) {
   Init();
   mediastream_signaling_.SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_LT(0u, channel->renderers().size());
   EXPECT_TRUE(channel->renderers().begin()->second == NULL);
   ASSERT_EQ(1u, channel->recv_streams().size());
-  uint32 receive_ssrc  = channel->recv_streams()[0].first_ssrc();
+  uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc();
   cricket::FakeVideoRenderer renderer;
   session_->SetVideoPlayout(receive_ssrc, true, &renderer);
   EXPECT_TRUE(channel->renderers().begin()->second == &renderer);
   session_->SetVideoPlayout(receive_ssrc, false, &renderer);
   EXPECT_TRUE(channel->renderers().begin()->second == NULL);
 }
 
 TEST_F(WebRtcSessionTest, SetVideoSend) {
   Init();
   mediastream_signaling_.SendAudioVideoStream1();
   CreateAndSetRemoteOfferAndLocalAnswer();
   cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
   ASSERT_TRUE(channel != NULL);
   ASSERT_EQ(1u, channel->send_streams().size());
-  uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
+  uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
   cricket::VideoOptions* options = NULL;
   session_->SetVideoSend(send_ssrc, false, options);
   EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
   session_->SetVideoSend(send_ssrc, true, options);
   EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
 }
 
 TEST_F(WebRtcSessionTest, CanNotInsertDtmf) {
   TestCanInsertDtmf(false);
@@ skipping 13 matching lines @@
 
   // Insert DTMF
   const int expected_flags = DF_SEND;
   const int expected_duration = 90;
   session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
   session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
   session_->InsertDtmf(kAudioTrack1, 2, expected_duration);
 
   // Verify
   ASSERT_EQ(3U, channel->dtmf_info_queue().size());
-  const uint32 send_ssrc  = channel->send_streams()[0].first_ssrc();
+  const uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
   EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
                               expected_duration, expected_flags));
   EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
                               expected_duration, expected_flags));
   EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
                               expected_duration, expected_flags));
 }
 
 // This test verifies the |initiator| flag when session initiates the call.
 TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
@@ skipping 783 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));