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)); |