| 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
|
|
|