webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc - Issue 2310943002: Add time line for acked bitrate.

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 2310943002: Add time line for acked bitrate. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 3 months ago

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Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

index de5aaf3bad9e9e974d276ac88070cd3c6233c249..6c9197f0f38659f735a3882a9360080bb86e9b93 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -136,7 +136,8 @@ class MockSendPacketObserver : public SendPacketObserver {

class MockTransportFeedbackObserver : public TransportFeedbackObserver {

public:

MOCK_METHOD3(AddPacket, void(uint16_t, size_t, int));

- MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));

+ MOCK_METHOD1(OnTransportFeedback,

+ std::vector<PacketInfo>(const rtcp::TransportFeedback&));

};

class RtpSenderTest : public ::testing::Test {