| 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 ea70d304d79acb3e7526b3caf01dcd4f5d8d1a90..7f940fc767f85e4d6f391308d84a7e6213ce1eda 100644
|
| --- a/webrtc/audio/audio_send_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_send_stream_unittest.cc
|
| @@ -50,6 +50,13 @@ const int kTelephoneEventPayloadType = 123;
|
| const int kTelephoneEventCode = 45;
|
| const int kTelephoneEventDuration = 6789;
|
|
|
| +class MockLimitObserver : public BitrateAllocator::LimitObserver {
|
| + public:
|
| + MOCK_METHOD2(OnAllocationLimitsChanged,
|
| + void(uint32_t min_send_bitrate_bps,
|
| + uint32_t max_padding_bitrate_bps));
|
| +};
|
| +
|
| struct ConfigHelper {
|
| ConfigHelper()
|
| : simulated_clock_(123456),
|
| @@ -57,7 +64,8 @@ struct ConfigHelper {
|
| congestion_controller_(&simulated_clock_,
|
| &bitrate_observer_,
|
| &remote_bitrate_observer_,
|
| - &event_log_) {
|
| + &event_log_),
|
| + bitrate_allocator_(&limit_observer_) {
|
| using testing::Invoke;
|
| using testing::StrEq;
|
|
|
| @@ -116,6 +124,7 @@ struct ConfigHelper {
|
| CongestionController* congestion_controller() {
|
| return &congestion_controller_;
|
| }
|
| + BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
|
|
|
| void SetupMockForSendTelephoneEvent() {
|
| EXPECT_TRUE(channel_proxy_);
|
| @@ -170,6 +179,8 @@ struct ConfigHelper {
|
| testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
|
| CongestionController congestion_controller_;
|
| MockRtcEventLog event_log_;
|
| + testing::NiceMock<MockLimitObserver> limit_observer_;
|
| + BitrateAllocator bitrate_allocator_;
|
| };
|
| } // namespace
|
|
|
| @@ -192,13 +203,15 @@ TEST(AudioSendStreamTest, ConfigToString) {
|
| TEST(AudioSendStreamTest, ConstructDestruct) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| - helper.congestion_controller());
|
| + helper.congestion_controller(),
|
| + helper.bitrate_allocator());
|
| }
|
|
|
| TEST(AudioSendStreamTest, SendTelephoneEvent) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| - helper.congestion_controller());
|
| + helper.congestion_controller(),
|
| + helper.bitrate_allocator());
|
| helper.SetupMockForSendTelephoneEvent();
|
| EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
|
| kTelephoneEventCode, kTelephoneEventDuration));
|
| @@ -207,7 +220,8 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
|
| TEST(AudioSendStreamTest, SetMuted) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| - helper.congestion_controller());
|
| + helper.congestion_controller(),
|
| + helper.bitrate_allocator());
|
| EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
|
| send_stream.SetMuted(true);
|
| }
|
| @@ -215,7 +229,8 @@ TEST(AudioSendStreamTest, SetMuted) {
|
| TEST(AudioSendStreamTest, GetStats) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| - helper.congestion_controller());
|
| + helper.congestion_controller(),
|
| + helper.bitrate_allocator());
|
| helper.SetupMockForGetStats();
|
| AudioSendStream::Stats stats = send_stream.GetStats();
|
| EXPECT_EQ(kSsrc, stats.local_ssrc);
|
| @@ -243,7 +258,8 @@ TEST(AudioSendStreamTest, GetStats) {
|
| TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| - helper.congestion_controller());
|
| + helper.congestion_controller(),
|
| + helper.bitrate_allocator());
|
| helper.SetupMockForGetStats();
|
| EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
|
|
|
|
|