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| author | Android Chromium Automerger <chromium-automerger@android> | 2014-04-10 22:38:37 +0000 |
|---|---|---|
| committer | Android Chromium Automerger <chromium-automerger@android> | 2014-04-10 22:38:37 +0000 |
| commit | 54b48fa13d401b118b94ca2fe675d8b7e001187d (patch) | |
| tree | 7cf44113635b342fcc09593a5af7121cef62d149 /modules/video_coding | |
| parent | c71dd0dcd9ff4ccf721127f79fd9a41cee6b13ce (diff) | |
| parent | 55bc2810c06fe624311518c4502af5ca8a5c085c (diff) | |
| download | webrtc-54b48fa13d401b118b94ca2fe675d8b7e001187d.tar.gz | |
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 55bc2810c06fe624311518c4502af5ca8a5c085c
This commit was generated by merge_from_chromium.py.
Change-Id: I9ccff119d81dc5f08bc0a54253907449806630ca
Diffstat (limited to 'modules/video_coding')
| -rw-r--r-- | modules/video_coding/main/test/mt_rx_tx_test.cc | 2 | ||||
| -rw-r--r-- | modules/video_coding/main/test/rtp_player.cc | 2 |
2 files changed, 2 insertions, 2 deletions
diff --git a/modules/video_coding/main/test/mt_rx_tx_test.cc b/modules/video_coding/main/test/mt_rx_tx_test.cc index 5cad5d40..8f3963f9 100644 --- a/modules/video_coding/main/test/mt_rx_tx_test.cc +++ b/modules/video_coding/main/test/mt_rx_tx_test.cc @@ -157,7 +157,7 @@ int MTRxTxTest(CmdArgs& args) configuration.outgoing_transport = outgoingTransport; RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration); scoped_ptr<RTPPayloadRegistry> registry(new RTPPayloadRegistry( - -1, RTPPayloadStrategy::CreateStrategy(false))); + RTPPayloadStrategy::CreateStrategy(false))); scoped_ptr<RtpReceiver> rtp_receiver( RtpReceiver::CreateVideoReceiver(-1, Clock::GetRealTimeClock(), &dataCallback, NULL, registry.get())); diff --git a/modules/video_coding/main/test/rtp_player.cc b/modules/video_coding/main/test/rtp_player.cc index 4c157a71..e314a7c9 100644 --- a/modules/video_coding/main/test/rtp_player.cc +++ b/modules/video_coding/main/test/rtp_player.cc @@ -273,7 +273,7 @@ class SsrcHandlers { LostPackets* lost_packets) : rtp_header_parser_(RtpHeaderParser::Create()), rtp_payload_registry_(new RTPPayloadRegistry( - 0, RTPPayloadStrategy::CreateStrategy(false))), + RTPPayloadStrategy::CreateStrategy(false))), rtp_module_(), payload_sink_(), ssrc_(ssrc), |