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Create a NetEq test for when the first incoming payload type is unknown

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This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2014-11-20 11:01:02 +00:00
parent 049e4ece30
commit ed91068bf1
1 changed files with 66 additions and 0 deletions
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66
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
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@ -587,4 +587,70 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
EXPECT_CALL(mock_decoder, Die());
}
// This test verifies that NetEq can handle the situation where the first
// incoming packet is rejected.
// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=4021.
TEST_F(NetEqImplTest, DISABLED_FirstPacketUnknown) {
UseNoMocks();
CreateInstance();
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
rtp_header.header.timestamp = 0x12345678;
rtp_header.header.ssrc = 0x87654321;
// Insert one packet. Note that we have not registered any payload type, so
// this packet will be rejected.
EXPECT_EQ(NetEq::kFail,
neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
kReceiveTime));
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
// Pull audio once.
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
int16_t output[kMaxOutputSize];
int samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
ASSERT_LE(samples_per_channel, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(kOutputPLC, type);
// Register the payload type.
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
// Insert 10 packets.
for (int i = 0; i < 10; ++i) {
rtp_header.header.sequenceNumber++;
rtp_header.header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
kReceiveTime));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
// Pull audio repeatedly and make sure we get normal output, that is not PLC.
for (int i = 0; i < 3; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
ASSERT_LE(samples_per_channel, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(kOutputNormal, type)
<< "NetEq did not decode the packets as expected.";
}
}
} // namespace webrtc