| Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
|
| index 4bbcc326e9a86f0d0743c116693c33b6c0b9a063..291dded3b22b8f5b32149aaff0c5c4b23e22a7c2 100644
|
| --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
|
| +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
|
| @@ -208,79 +208,6 @@ TEST_F(RtpRtcpAudioTest, Basic) {
|
| EXPECT_EQ(test_timestamp, timestamp);
|
| }
|
|
|
| -TEST_F(RtpRtcpAudioTest, RED) {
|
| - CodecInst voice_codec;
|
| - memset(&voice_codec, 0, sizeof(voice_codec));
|
| - voice_codec.pltype = 96;
|
| - voice_codec.plfreq = 8000;
|
| - memcpy(voice_codec.plname, "PCMU", 5);
|
| -
|
| - EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
|
| - EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
|
| - voice_codec.plname,
|
| - voice_codec.pltype,
|
| - voice_codec.plfreq,
|
| - voice_codec.channels,
|
| - (voice_codec.rate < 0) ? 0 : voice_codec.rate));
|
| - EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
|
| - voice_codec.rate = test_rate;
|
| - EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
|
| - voice_codec.plname,
|
| - voice_codec.pltype,
|
| - voice_codec.plfreq,
|
| - voice_codec.channels,
|
| - (voice_codec.rate < 0) ? 0 : voice_codec.rate));
|
| -
|
| - module1->SetSSRC(test_ssrc);
|
| - module1->SetStartTimestamp(test_timestamp);
|
| - EXPECT_EQ(0, module1->SetSendingStatus(true));
|
| -
|
| - voice_codec.pltype = 127;
|
| - voice_codec.plfreq = 8000;
|
| - memcpy(voice_codec.plname, "RED", 4);
|
| -
|
| - EXPECT_EQ(0, module1->SetSendREDPayloadType(voice_codec.pltype));
|
| - int8_t red = 0;
|
| - EXPECT_EQ(0, module1->SendREDPayloadType(&red));
|
| - EXPECT_EQ(voice_codec.pltype, red);
|
| - EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
|
| - voice_codec.plname,
|
| - voice_codec.pltype,
|
| - voice_codec.plfreq,
|
| - voice_codec.channels,
|
| - (voice_codec.rate < 0) ? 0 : voice_codec.rate));
|
| - EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
|
| - voice_codec.plname,
|
| - voice_codec.pltype,
|
| - voice_codec.plfreq,
|
| - voice_codec.channels,
|
| - (voice_codec.rate < 0) ? 0 : voice_codec.rate));
|
| -
|
| - RTPFragmentationHeader fragmentation;
|
| - fragmentation.fragmentationVectorSize = 2;
|
| - fragmentation.fragmentationLength = new size_t[2];
|
| - fragmentation.fragmentationLength[0] = 4;
|
| - fragmentation.fragmentationLength[1] = 4;
|
| - fragmentation.fragmentationOffset = new size_t[2];
|
| - fragmentation.fragmentationOffset[0] = 0;
|
| - fragmentation.fragmentationOffset[1] = 4;
|
| - fragmentation.fragmentationTimeDiff = new uint16_t[2];
|
| - fragmentation.fragmentationTimeDiff[0] = 0;
|
| - fragmentation.fragmentationTimeDiff[1] = 0;
|
| - fragmentation.fragmentationPlType = new uint8_t[2];
|
| - fragmentation.fragmentationPlType[0] = 96;
|
| - fragmentation.fragmentationPlType[1] = 96;
|
| -
|
| - const uint8_t test[5] = "test";
|
| - // Send a RTP packet.
|
| - EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 160, -1,
|
| - test, 4, &fragmentation, nullptr,
|
| - nullptr));
|
| -
|
| - EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
|
| - EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
|
| -}
|
| -
|
| TEST_F(RtpRtcpAudioTest, DTMF) {
|
| CodecInst voice_codec;
|
| memset(&voice_codec, 0, sizeof(voice_codec));
|
|
|