commit 14d5dbe5b35c560e978dab0c02c48cb16d28273d
author ivoc <ivoc@webrtc.org> Mon Jul 04 07:06:55 2016 -0700
committer Commit bot <commit-bot@chromium.org> Mon Jul 04 14:07:03 2016 +0000
tree aa73a0a6a99a86ab13489ea0358f7235c7bf1ae9
parent 77ad394fa6680aa9fb36d0bcac330dd84b29dfc2

Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"

The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}

webrtc/audio/audio_receive_stream_unittest.cc

diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index aed1d1a..dd66cc6 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -15,6 +15,7 @@
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
@@ -70,7 +71,8 @@
         decoder_factory_(new rtc::RefCountedObject<MockAudioDecoderFactory>),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
-                               &remote_bitrate_observer_) {
+                               &remote_bitrate_observer_,
+                               &event_log_) {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
@@ -109,6 +111,12 @@
               .Times(1);
           EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
               .WillOnce(ReturnRef(decoder_factory_));
+          testing::Expectation expect_set =
+              EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
+              .Times(1)
+              .After(expect_set);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
@@ -130,6 +138,7 @@
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
+  MockRtcEventLog* event_log() { return &event_log_; }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
@@ -171,6 +180,7 @@
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
   MockCongestionController congestion_controller_;
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
+  MockRtcEventLog event_log_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioReceiveStream::Config stream_config_;
@@ -248,7 +258,8 @@
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
 }
 
 MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
@@ -267,7 +278,8 @@
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
   internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
   const int kTransportSequenceNumberValue = 1234;
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
@@ -295,7 +307,8 @@
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
   internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
 
   std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
   EXPECT_CALL(*helper.channel_proxy(),
@@ -307,7 +320,8 @@
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
@@ -349,7 +363,8 @@
 TEST(AudioReceiveStreamTest, SetGain) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
   EXPECT_CALL(*helper.channel_proxy(),
       SetChannelOutputVolumeScaling(FloatEq(0.765f)));
   recv_stream.SetGain(0.765f);