Chromium Code Reviews| Index: webrtc/call/rtx_receive_stream_unittest.cc |
| diff --git a/webrtc/call/rtx_receive_stream_unittest.cc b/webrtc/call/rtx_receive_stream_unittest.cc |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..b4deb2e6f3502837b2f660b8e4d810029381ce9f |
| --- /dev/null |
| +++ b/webrtc/call/rtx_receive_stream_unittest.cc |
| @@ -0,0 +1,92 @@ |
| +/* |
| + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| + * |
| + * Use of this source code is governed by a BSD-style license |
| + * that can be found in the LICENSE file in the root of the source |
| + * tree. An additional intellectual property rights grant can be found |
| + * in the file PATENTS. All contributing project authors may |
| + * be found in the AUTHORS file in the root of the source tree. |
| + */ |
| + |
| +#include "webrtc/call/rtx_receive_stream.h" |
| +#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
| +#include "webrtc/test/gmock.h" |
| +#include "webrtc/test/gtest.h" |
| + |
| +namespace webrtc { |
| + |
| +namespace { |
| + |
| +class MockRtpPacketSink : public RtpPacketSinkInterface { |
| + public: |
| + MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived&)); |
| +}; |
| + |
| +constexpr int kMediaPayloadType = 100; |
| +constexpr int kRtxPayloadType = 98; |
| +constexpr int kMediaSSRC = 0x3333333; |
|
danilchap
2017/05/19 08:39:02
may be uint32_t for SSRC
nisse-webrtc
2017/05/19 10:59:14
Done.
|
| + |
|
danilchap
2017/05/19 08:39:03
might be helpful to add constants for the original
nisse-webrtc
2017/05/19 10:59:14
Added constant for the seqno.
|
| +constexpr uint8_t kRtxPacket[] = { |
| + 0x80, // Version 2, |
| + 98, // Payload type |
|
danilchap
2017/05/19 08:39:03
likely add dots at the end of the comments.
nisse-webrtc
2017/05/19 10:59:14
Done.
|
| + 0x12, 0x34, // Seqno, |
| + 0x11, 0x11, 0x11, 0x11, // Timestamp |
| + 0x22, 0x22, 0x22, 0x22, // SSRC |
| + // RTX header |
| + 0x56, 0x57, // Orig seqno |
| + // Payload |
| + 0xee, |
| +}; |
| + |
| +std::map<int, int> PTMapping() { |
|
danilchap
2017/05/19 08:39:03
it is easier to read if you do not abbreviate: so
nisse-webrtc
2017/05/19 10:59:14
Done.
|
| + std::map<int, int> m; |
|
danilchap
2017/05/19 08:39:02
I wonder if
return {{kRtxPayloadType, kMediaPayloa
|
| + m[kRtxPayloadType] = kMediaPayloadType; |
| + return m; |
| +} |
| + |
| +template <typename T> |
| +rtc::ArrayView<T> Truncate(rtc::ArrayView<T> a, size_t drop) { |
| + return a.subview(0, a.size() - drop); |
| +} |
| + |
| +} // namespace |
| + |
| +TEST(RtxReceiveStreamTest, RestoresPacketPayload) { |
| + testing::StrictMock<MockRtpPacketSink> media_sink; |
|
danilchap
2017/05/19 08:39:02
personally I prefer to add
using ::testing::_;
usi
nisse-webrtc
2017/05/19 10:59:14
Done for some of the symbols.
|
| + RtxReceiveStream rtx_sink(&media_sink, PTMapping(), kMediaSSRC); |
| + RtpPacketReceived rtx_packet; |
| + EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket))); |
| + |
| + EXPECT_CALL(media_sink, |
| + OnRtpPacket(testing::AllOf( |
|
danilchap
2017/05/19 08:39:02
Alternative suggestion (not sure it is better thou
nisse-webrtc
2017/05/19 10:59:14
I've changed to this alternative. It provides clea
|
| + testing::Truly([](const RtpPacketReceived& packet) { |
| + return packet.SequenceNumber() == 0x5657; |
| + }), |
| + testing::Truly([](const RtpPacketReceived& packet) { |
|
danilchap
2017/05/19 08:39:02
Or... yet another (again, not sure it is more read
|
| + return packet.Ssrc() == kMediaSSRC; |
| + }), |
| + testing::Truly([](const RtpPacketReceived& packet) { |
| + return packet.payload().size() == 1 && |
| + packet.payload()[0] == 0xee; |
| + })))); |
| + rtx_sink.OnRtpPacket(rtx_packet); |
| +} |
| + |
| +TEST(RtxReceiveStreamTest, IgnoresUnknownPayloadType) { |
| + testing::StrictMock<MockRtpPacketSink> media_sink; |
| + RtxReceiveStream rtx_sink(&media_sink, std::map<int, int>(), kMediaSSRC); |
| + RtpPacketReceived rtx_packet; |
| + EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket))); |
| + rtx_sink.OnRtpPacket(rtx_packet); |
| +} |
| + |
| +TEST(RtxReceiveStreamTest, IgnoresTruncatedPacket) { |
| + testing::StrictMock<MockRtpPacketSink> media_sink; |
| + RtxReceiveStream rtx_sink(&media_sink, PTMapping(), kMediaSSRC); |
| + RtpPacketReceived rtx_packet; |
| + EXPECT_TRUE( |
| + rtx_packet.Parse(Truncate(rtc::ArrayView<const uint8_t>(kRtxPacket), 2))); |
| + rtx_sink.OnRtpPacket(rtx_packet); |
| +} |
| + |
| +} // namespace webrtc |