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Fixed Rtp/Rtcp tests

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R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1627005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org 2013-06-07 16:03:19 +00:00
parent 6367fe885a
commit 1064cf06b0
1 changed files with 8 additions and 11 deletions
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19
webrtc/voice_engine/test/auto_test/voe_extended_test.cc
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@ -4607,13 +4607,12 @@ int VoEExtendedTest::TestRTP_RTCP() {
TEST_MUSTPASS(network->DeRegisterExternalTransport(1));
TEST_MUSTPASS(voe_base_->DeleteChannel(0));
TEST_MUSTPASS(voe_base_->DeleteChannel(1));
voice_channel_transport.reset(NULL);
TEST_MUSTPASS(voe_base_->CreateChannel());
voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
voice_channel_transport->SetLocalReceiver(12345);
voice_channel_transport->SetSendDestination("127.0.0.1", 12347);
voice_channel_transport->SetLocalReceiver(12347);
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartSend(0));
@ -4774,13 +4773,13 @@ int VoEExtendedTest::TestRTP_RTCP() {
TEST_MUSTPASS(voe_base_->StopPlayout(0));
TEST_MUSTPASS(voe_base_->StopReceive(0));
TEST_MUSTPASS(voe_base_->DeleteChannel(0));
voice_channel_transport.reset(NULL);
SleepMs(100);
TEST_MUSTPASS(voe_base_->CreateChannel());
voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
voice_channel_transport->SetSendDestination("127.0.0.1", 12345);
voice_channel_transport->SetLocalReceiver(12345);
@ -4831,12 +4830,10 @@ int VoEExtendedTest::TestRTP_RTCP() {
TEST_MUSTPASS((NTPHigh == NTPHigh2) && (NTPLow == NTPLow2));
TEST_MUSTPASS(timestamp == timestamp2);
TEST_MUSTPASS(playoutTimestamp == playoutTimestamp2);
CodecInst cinst;
#ifdef WEBRTC_CODEC_RED
//The following test is related to defect 4985 and 4986
TEST_LOG("Turn FEC and VAD on and wait for 4 seconds and ensure that "
"the jitter is still small...");
CodecInst cinst;
#if (!defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID))
cinst.pltype = 104;
strcpy(cinst.plname, "isac");
@ -4860,7 +4857,7 @@ int VoEExtendedTest::TestRTP_RTCP() {
TEST_MUSTPASS(voe_base_->StartSend(0));
TEST_MUSTPASS(voe_base_->StartReceive(0));
TEST_MUSTPASS(voe_base_->StartPlayout(0));
TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, -1));
TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, 126));
MARK();
TEST_MUSTPASS(codec->SetVADStatus(0,true));
SleepMs(4000);
@ -4873,8 +4870,8 @@ int VoEExtendedTest::TestRTP_RTCP() {
TEST_MUSTPASS(jitter2 > 1000)
TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, false));
MARK();
//4985 and 4986 end
#endif // #ifdef WEBRTC_CODEC_RED
TEST(GetRTPStatistics);
ANL();
// Statistics summarized on local side based on received RTP packets.
@ -5029,9 +5026,9 @@ int VoEExtendedTest::TestRTP_RTCP() {
// We have to re-register the audio codec payload type as stopReceive will
// clean the database
TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
voice_channel_transport.reset(NULL);
voice_channel_transport.reset(new VoiceChannelTransport(network, 0));
voice_channel_transport->SetSendDestination("127.0.0.1", 8000);
voice_channel_transport->SetLocalReceiver(8000);