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Side by Side Diff: media/cast/sender/video_sender_unittest.cc

Issue 655713003: Standardize usage of virtual/override/final in media/ (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 2 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include <stdint.h> 5 #include <stdint.h>
6 6
7 #include <vector> 7 #include <vector>
8 8
9 #include "base/bind.h" 9 #include "base/bind.h"
10 #include "base/memory/scoped_ptr.h" 10 #include "base/memory/scoped_ptr.h"
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57 } 57 }
58 58
59 class TestPacketSender : public PacketSender { 59 class TestPacketSender : public PacketSender {
60 public: 60 public:
61 TestPacketSender() 61 TestPacketSender()
62 : number_of_rtp_packets_(0), 62 : number_of_rtp_packets_(0),
63 number_of_rtcp_packets_(0), 63 number_of_rtcp_packets_(0),
64 paused_(false) {} 64 paused_(false) {}
65 65
66 // A singular packet implies a RTCP packet. 66 // A singular packet implies a RTCP packet.
67 virtual bool SendPacket(PacketRef packet, 67 bool SendPacket(PacketRef packet, const base::Closure& cb) override {
68 const base::Closure& cb) override {
69 if (paused_) { 68 if (paused_) {
70 stored_packet_ = packet; 69 stored_packet_ = packet;
71 callback_ = cb; 70 callback_ = cb;
72 return false; 71 return false;
73 } 72 }
74 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { 73 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
75 ++number_of_rtcp_packets_; 74 ++number_of_rtcp_packets_;
76 } else { 75 } else {
77 // Check that at least one RTCP packet was sent before the first RTP 76 // Check that at least one RTCP packet was sent before the first RTP
78 // packet. This confirms that the receiver will have the necessary lip 77 // packet. This confirms that the receiver will have the necessary lip
79 // sync info before it has to calculate the playout time of the first 78 // sync info before it has to calculate the playout time of the first
80 // frame. 79 // frame.
81 if (number_of_rtp_packets_ == 0) 80 if (number_of_rtp_packets_ == 0)
82 EXPECT_LE(1, number_of_rtcp_packets_); 81 EXPECT_LE(1, number_of_rtcp_packets_);
83 ++number_of_rtp_packets_; 82 ++number_of_rtp_packets_;
84 } 83 }
85 return true; 84 return true;
86 } 85 }
87 86
88 virtual int64 GetBytesSent() override { 87 int64 GetBytesSent() override { return 0; }
89 return 0;
90 }
91 88
92 int number_of_rtp_packets() const { return number_of_rtp_packets_; } 89 int number_of_rtp_packets() const { return number_of_rtp_packets_; }
93 90
94 int number_of_rtcp_packets() const { return number_of_rtcp_packets_; } 91 int number_of_rtcp_packets() const { return number_of_rtcp_packets_; }
95 92
96 void SetPause(bool paused) { 93 void SetPause(bool paused) {
97 paused_ = paused; 94 paused_ = paused;
98 if (!paused && stored_packet_.get()) { 95 if (!paused && stored_packet_.get()) {
99 SendPacket(stored_packet_, callback_); 96 SendPacket(stored_packet_, callback_);
100 callback_.Run(); 97 callback_.Run();
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534 cast_feedback.ack_frame_id = 0; 531 cast_feedback.ack_frame_id = 0;
535 video_sender_->OnReceivedCastFeedback(cast_feedback); 532 video_sender_->OnReceivedCastFeedback(cast_feedback);
536 533
537 transport_.SetPause(false); 534 transport_.SetPause(false);
538 RunTasks(33); 535 RunTasks(33);
539 EXPECT_EQ(0, transport_.number_of_rtp_packets()); 536 EXPECT_EQ(0, transport_.number_of_rtp_packets());
540 } 537 }
541 538
542 } // namespace cast 539 } // namespace cast
543 } // namespace media 540 } // namespace media
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