Index: webrtc/video/payload_router_unittest.cc |
diff --git a/webrtc/video/payload_router_unittest.cc b/webrtc/video/payload_router_unittest.cc |
index 62dba29c05e72873b977a40be9f6159e2fcc2706..5b6612124c25d60be4be424dbfa89eba7406180e 100644 |
--- a/webrtc/video/payload_router_unittest.cc |
+++ b/webrtc/video/payload_router_unittest.cc |
@@ -186,4 +186,25 @@ |
.WillOnce(Return(kTestMinPayloadLength)); |
EXPECT_EQ(kTestMinPayloadLength, payload_router.MaxPayloadLength()); |
} |
+ |
+TEST(PayloadRouterTest, SetTargetSendBitrates) { |
+ NiceMock<MockRtpRtcp> rtp_1; |
+ NiceMock<MockRtpRtcp> rtp_2; |
+ std::vector<RtpRtcp*> modules; |
+ modules.push_back(&rtp_1); |
+ modules.push_back(&rtp_2); |
+ PayloadRouter payload_router(modules, 42); |
+ std::vector<VideoStream> streams(2); |
+ streams[0].max_bitrate_bps = 10000; |
+ streams[1].max_bitrate_bps = 100000; |
+ payload_router.SetSendStreams(streams); |
+ |
+ const uint32_t bitrate_1 = 10000; |
+ const uint32_t bitrate_2 = 76543; |
+ EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) |
+ .Times(1); |
+ EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) |
+ .Times(1); |
+ payload_router.SetTargetSendBitrate(bitrate_1 + bitrate_2); |
+} |
} // namespace webrtc |