Index: 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 8703d6ed324819a0a041e823a88bf90db6f9e7ca..074aefe96115b07ec11fc195dcfb8021dbc37d54 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/bitrate_controller/include/mock/mock_bitrate_controller.h" |
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" |
#include "webrtc/modules/pacing/packet_router.h" |
@@ -66,7 +67,8 @@ struct ConfigHelper { |
: simulated_clock_(123456), |
congestion_controller_(&simulated_clock_, |
&bitrate_observer_, |
- &remote_bitrate_observer_) { |
+ &remote_bitrate_observer_, |
+ &event_log_) { |
using testing::Invoke; |
EXPECT_CALL(voice_engine_, |
@@ -98,6 +100,12 @@ struct ConfigHelper { |
.WillOnce(Return(&packet_router_)); |
EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects()) |
.Times(1); |
+ testing::Expectation expect_set = |
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())) |
the sun
2016/03/30 15:20:38
Could check for argument to be &event_log_ ?
ivoc
2016/03/31 09:06:56
Good idea, done.
|
+ .Times(1); |
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
+ .Times(1) |
+ .After(expect_set); |
return channel_proxy_; |
})); |
stream_config_.voe_channel_id = kChannelId; |
@@ -117,6 +125,7 @@ struct ConfigHelper { |
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_; } |
@@ -156,6 +165,7 @@ struct ConfigHelper { |
testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; |
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_; |
@@ -221,7 +231,8 @@ TEST(AudioReceiveStreamTest, ConfigToString) { |
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, "") { |
@@ -240,7 +251,8 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) { |
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); |
@@ -261,7 +273,8 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) { |
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); |