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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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32 // This test transport verifies that no functions get called. 32 // This test transport verifies that no functions get called.
33 class TestTransport : public Transport, 33 class TestTransport : public Transport,
34 public NullRtpData { 34 public NullRtpData {
35 public: 35 public:
36 explicit TestTransport() 36 explicit TestTransport()
37 : rtcp_receiver_(NULL) { 37 : rtcp_receiver_(NULL) {
38 } 38 }
39 void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) { 39 void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
40 rtcp_receiver_ = rtcp_receiver; 40 rtcp_receiver_ = rtcp_receiver;
41 } 41 }
42 int SendPacket(int /*ch*/, const void* /*data*/, size_t /*len*/) override { 42 int SendPacket(const void* /*data*/, size_t /*len*/) override {
43 ADD_FAILURE(); // FAIL() gives a compile error. 43 ADD_FAILURE(); // FAIL() gives a compile error.
44 return -1; 44 return -1;
45 } 45 }
46 46
47 // Injects an RTCP packet into the receiver. 47 int SendRTCPPacket(const void* packet, size_t packet_len) override {
48 int SendRTCPPacket(int /* ch */,
49 const void* packet,
50 size_t packet_len) override {
51 ADD_FAILURE(); 48 ADD_FAILURE();
52 return 0; 49 return 0;
53 } 50 }
54 51
55 int OnReceivedPayloadData(const uint8_t* payloadData, 52 int OnReceivedPayloadData(const uint8_t* payloadData,
56 const size_t payloadSize, 53 const size_t payloadSize,
57 const WebRtcRTPHeader* rtpHeader) override { 54 const WebRtcRTPHeader* rtpHeader) override {
58 ADD_FAILURE(); 55 ADD_FAILURE();
59 return 0; 56 return 0;
60 } 57 }
61 RTCPReceiver* rtcp_receiver_; 58 RTCPReceiver* rtcp_receiver_;
62 }; 59 };
63 60
64 class RtcpReceiverTest : public ::testing::Test { 61 class RtcpReceiverTest : public ::testing::Test {
65 protected: 62 protected:
66 static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000; 63 static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
67 64
68 RtcpReceiverTest() 65 RtcpReceiverTest()
69 : over_use_detector_options_(), 66 : over_use_detector_options_(),
70 system_clock_(1335900000), 67 system_clock_(1335900000),
71 remote_bitrate_observer_(), 68 remote_bitrate_observer_(),
72 remote_bitrate_estimator_(new RemoteBitrateEstimatorSingleStream( 69 remote_bitrate_estimator_(new RemoteBitrateEstimatorSingleStream(
73 &remote_bitrate_observer_, 70 &remote_bitrate_observer_,
74 &system_clock_, 71 &system_clock_,
75 kRemoteBitrateEstimatorMinBitrateBps)) { 72 kRemoteBitrateEstimatorMinBitrateBps)) {
76 test_transport_ = new TestTransport(); 73 test_transport_ = new TestTransport();
77 74
78 RtpRtcp::Configuration configuration; 75 RtpRtcp::Configuration configuration;
79 configuration.id = 0;
80 configuration.audio = false; 76 configuration.audio = false;
81 configuration.clock = &system_clock_; 77 configuration.clock = &system_clock_;
82 configuration.outgoing_transport = test_transport_; 78 configuration.outgoing_transport = test_transport_;
83 configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get(); 79 configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
84 rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration); 80 rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
85 rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, false, NULL, NULL, 81 rtcp_receiver_ = new RTCPReceiver(&system_clock_, false, NULL, NULL, NULL,
86 NULL, rtp_rtcp_impl_); 82 rtp_rtcp_impl_);
87 test_transport_->SetRTCPReceiver(rtcp_receiver_); 83 test_transport_->SetRTCPReceiver(rtcp_receiver_);
88 } 84 }
89 ~RtcpReceiverTest() { 85 ~RtcpReceiverTest() {
90 delete rtcp_receiver_; 86 delete rtcp_receiver_;
91 delete rtp_rtcp_impl_; 87 delete rtp_rtcp_impl_;
92 delete test_transport_; 88 delete test_transport_;
93 } 89 }
94 90
95 // Injects an RTCP packet into the receiver. 91 // Injects an RTCP packet into the receiver.
96 // Returns 0 for OK, non-0 for failure. 92 // Returns 0 for OK, non-0 for failure.
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1027 rr2.WithReportBlock(rb2); 1023 rr2.WithReportBlock(rb2);
1028 rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build()); 1024 rtc::scoped_ptr<rtcp::RawPacket> p2(rr2.Build());
1029 EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length())); 1025 EXPECT_EQ(0, InjectRtcpPacket(p2->Buffer(), p2->Length()));
1030 EXPECT_TRUE(callback.Matches(kSourceSsrc, kSequenceNumber, kFractionLoss, 1026 EXPECT_TRUE(callback.Matches(kSourceSsrc, kSequenceNumber, kFractionLoss,
1031 kCumulativeLoss, kJitter)); 1027 kCumulativeLoss, kJitter));
1032 } 1028 }
1033 1029
1034 } // Anonymous namespace 1030 } // Anonymous namespace
1035 1031
1036 } // namespace webrtc 1032 } // namespace webrtc
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