Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index a05c00ba49b4c3d9dfcedfbec3c53e8e6c734932..bebeacecb0be8dca43c670cccee224e9693269c2 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -20,6 +20,7 @@ |
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" |
#include "webrtc/modules/pacing/paced_sender.h" |
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" |
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
#include "webrtc/test/gtest.h" |
#include "webrtc/test/mock_voe_channel_proxy.h" |
#include "webrtc/test/mock_voice_engine.h" |
@@ -109,6 +110,8 @@ struct ConfigHelper { |
.Times(1); |
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull())) |
.Times(1); // Destructor resets the event log |
+ EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)) |
+ .Times(1); |
return channel_proxy_; |
})); |
stream_config_.voe_channel_id = kChannelId; |
@@ -132,6 +135,7 @@ struct ConfigHelper { |
BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; } |
rtc::TaskQueue* worker_queue() { return &worker_queue_; } |
RtcEventLog* event_log() { return &event_log_; } |
+ RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; } |
void SetupMockForSendTelephoneEvent() { |
EXPECT_TRUE(channel_proxy_); |
@@ -186,6 +190,7 @@ struct ConfigHelper { |
testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_; |
CongestionController congestion_controller_; |
MockRtcEventLog event_log_; |
+ MockRtcpRttStats rtcp_rtt_stats_; |
testing::NiceMock<MockLimitObserver> limit_observer_; |
BitrateAllocator bitrate_allocator_; |
// |worker_queue| is defined last to ensure all pending tasks are cancelled |
@@ -215,7 +220,7 @@ TEST(AudioSendStreamTest, ConstructDestruct) { |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.congestion_controller(), helper.bitrate_allocator(), |
- helper.event_log()); |
+ helper.event_log(), helper.rtcp_rtt_stats()); |
} |
TEST(AudioSendStreamTest, SendTelephoneEvent) { |
@@ -223,7 +228,7 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) { |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.congestion_controller(), helper.bitrate_allocator(), |
- helper.event_log()); |
+ helper.event_log(), helper.rtcp_rtt_stats()); |
helper.SetupMockForSendTelephoneEvent(); |
EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType, |
kTelephoneEventCode, kTelephoneEventDuration)); |
@@ -234,7 +239,7 @@ TEST(AudioSendStreamTest, SetMuted) { |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.congestion_controller(), helper.bitrate_allocator(), |
- helper.event_log()); |
+ helper.event_log(), helper.rtcp_rtt_stats()); |
EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true)); |
send_stream.SetMuted(true); |
} |
@@ -244,7 +249,7 @@ TEST(AudioSendStreamTest, GetStats) { |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.congestion_controller(), helper.bitrate_allocator(), |
- helper.event_log()); |
+ helper.event_log(), helper.rtcp_rtt_stats()); |
helper.SetupMockForGetStats(); |
AudioSendStream::Stats stats = send_stream.GetStats(); |
EXPECT_EQ(kSsrc, stats.local_ssrc); |
@@ -274,7 +279,7 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) { |
internal::AudioSendStream send_stream( |
helper.config(), helper.audio_state(), helper.worker_queue(), |
helper.congestion_controller(), helper.bitrate_allocator(), |
- helper.event_log()); |
+ helper.event_log(), helper.rtcp_rtt_stats()); |
helper.SetupMockForGetStats(); |
EXPECT_FALSE(send_stream.GetStats().typing_noise_detected); |