Index: webrtc/video/payload_router_unittest.cc |
diff --git a/webrtc/video/payload_router_unittest.cc b/webrtc/video/payload_router_unittest.cc |
index 41e173bf5fb930669be163d0749ea2d39717aeb6..5b6612124c25d60be4be424dbfa89eba7406180e 100644 |
--- a/webrtc/video/payload_router_unittest.cc |
+++ b/webrtc/video/payload_router_unittest.cc |
@@ -25,8 +25,9 @@ using ::testing::Return; |
namespace webrtc { |
TEST(PayloadRouterTest, SendOnOneModule) { |
- MockRtpRtcp rtp; |
+ NiceMock<MockRtpRtcp> rtp; |
std::vector<RtpRtcp*> modules(1, &rtp); |
+ std::vector<VideoStream> streams(1); |
uint8_t payload = 'a'; |
int8_t payload_type = 96; |
@@ -38,7 +39,7 @@ TEST(PayloadRouterTest, SendOnOneModule) { |
encoded_image._length = 1; |
PayloadRouter payload_router(modules, payload_type); |
- payload_router.SetSendingRtpModules(modules.size()); |
+ payload_router.SetSendStreams(streams); |
EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type, |
encoded_image._timeStamp, |
@@ -71,7 +72,8 @@ TEST(PayloadRouterTest, SendOnOneModule) { |
.Times(1); |
EXPECT_EQ(0, payload_router.Encoded(encoded_image, nullptr, nullptr)); |
- payload_router.SetSendingRtpModules(0); |
+ streams.clear(); |
+ payload_router.SetSendStreams(streams); |
EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, payload_type, |
encoded_image._timeStamp, |
encoded_image.capture_time_ms_, &payload, |
@@ -81,11 +83,12 @@ TEST(PayloadRouterTest, SendOnOneModule) { |
} |
TEST(PayloadRouterTest, SendSimulcast) { |
- MockRtpRtcp rtp_1; |
- MockRtpRtcp rtp_2; |
+ NiceMock<MockRtpRtcp> rtp_1; |
+ NiceMock<MockRtpRtcp> rtp_2; |
std::vector<RtpRtcp*> modules; |
modules.push_back(&rtp_1); |
modules.push_back(&rtp_2); |
+ std::vector<VideoStream> streams(2); |
int8_t payload_type = 96; |
uint8_t payload = 'a'; |
@@ -97,7 +100,7 @@ TEST(PayloadRouterTest, SendSimulcast) { |
encoded_image._length = 1; |
PayloadRouter payload_router(modules, payload_type); |
- payload_router.SetSendingRtpModules(modules.size()); |
+ payload_router.SetSendStreams(streams); |
CodecSpecificInfo codec_info_1; |
memset(&codec_info_1, 0, sizeof(CodecSpecificInfo)); |
@@ -138,7 +141,8 @@ TEST(PayloadRouterTest, SendSimulcast) { |
EXPECT_EQ(-1, payload_router.Encoded(encoded_image, &codec_info_2, nullptr)); |
// Invalid simulcast index. |
- payload_router.SetSendingRtpModules(1); |
+ streams.pop_back(); // Remove a stream. |
+ payload_router.SetSendStreams(streams); |
payload_router.set_active(true); |
EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _)) |
.Times(0); |
@@ -152,15 +156,16 @@ TEST(PayloadRouterTest, MaxPayloadLength) { |
// Without any limitations from the modules, verify we get the max payload |
// length for IP/UDP/SRTP with a MTU of 150 bytes. |
const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4; |
- MockRtpRtcp rtp_1; |
- MockRtpRtcp rtp_2; |
+ 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); |
EXPECT_EQ(kDefaultMaxLength, PayloadRouter::DefaultMaxPayloadLength()); |
- payload_router.SetSendingRtpModules(modules.size()); |
+ std::vector<VideoStream> streams(2); |
+ payload_router.SetSendStreams(streams); |
// Modules return a higher length than the default value. |
EXPECT_CALL(rtp_1, MaxDataPayloadLength()) |
@@ -183,29 +188,23 @@ TEST(PayloadRouterTest, MaxPayloadLength) { |
} |
TEST(PayloadRouterTest, SetTargetSendBitrates) { |
- MockRtpRtcp rtp_1; |
- MockRtpRtcp rtp_2; |
+ 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); |
- payload_router.SetSendingRtpModules(modules.size()); |
+ 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; |
- std::vector<uint32_t> bitrates(2, bitrate_1); |
- bitrates[1] = bitrate_2; |
EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) |
.Times(1); |
EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) |
.Times(1); |
- payload_router.SetTargetSendBitrates(bitrates); |
- |
- bitrates.resize(1); |
- EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1)) |
- .Times(1); |
- EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2)) |
- .Times(0); |
- payload_router.SetTargetSendBitrates(bitrates); |
+ payload_router.SetTargetSendBitrate(bitrate_1 + bitrate_2); |
} |
} // namespace webrtc |