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

Issue 387933005: Cast: Refactor RTCP handling (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: fix test Created 6 years, 5 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 "base/bind.h" 7 #include "base/bind.h"
8 #include "base/bind_helpers.h" 8 #include "base/bind_helpers.h"
9 #include "base/memory/scoped_ptr.h" 9 #include "base/memory/scoped_ptr.h"
10 #include "base/test/simple_test_tick_clock.h" 10 #include "base/test/simple_test_tick_clock.h"
11 #include "media/base/media.h" 11 #include "media/base/media.h"
12 #include "media/cast/cast_config.h" 12 #include "media/cast/cast_config.h"
13 #include "media/cast/cast_environment.h" 13 #include "media/cast/cast_environment.h"
14 #include "media/cast/net/cast_transport_config.h" 14 #include "media/cast/net/cast_transport_config.h"
15 #include "media/cast/net/cast_transport_sender_impl.h" 15 #include "media/cast/net/cast_transport_sender_impl.h"
16 #include "media/cast/net/rtcp/rtcp.h" 16 #include "media/cast/net/rtcp/rtcp_receiver.h"
17 #include "media/cast/sender/audio_sender.h" 17 #include "media/cast/sender/audio_sender.h"
18 #include "media/cast/test/fake_single_thread_task_runner.h" 18 #include "media/cast/test/fake_single_thread_task_runner.h"
19 #include "media/cast/test/utility/audio_utility.h" 19 #include "media/cast/test/utility/audio_utility.h"
20 #include "testing/gtest/include/gtest/gtest.h" 20 #include "testing/gtest/include/gtest/gtest.h"
21 21
22 namespace media { 22 namespace media {
23 namespace cast { 23 namespace cast {
24 24
25 class TestPacketSender : public PacketSender { 25 class TestPacketSender : public PacketSender {
26 public: 26 public:
27 TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {} 27 TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
28 28
29 virtual bool SendPacket(PacketRef packet, 29 virtual bool SendPacket(PacketRef packet,
30 const base::Closure& cb) OVERRIDE { 30 const base::Closure& cb) OVERRIDE {
31 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { 31 if (RtcpReceiver::IsRtcpPacket(&packet->data[0], packet->data.size())) {
32 ++number_of_rtcp_packets_; 32 ++number_of_rtcp_packets_;
33 } else { 33 } else {
34 // Check that at least one RTCP packet was sent before the first RTP 34 // Check that at least one RTCP packet was sent before the first RTP
35 // packet. This confirms that the receiver will have the necessary lip 35 // packet. This confirms that the receiver will have the necessary lip
36 // sync info before it has to calculate the playout time of the first 36 // sync info before it has to calculate the playout time of the first
37 // frame. 37 // frame.
38 if (number_of_rtp_packets_ == 0) 38 if (number_of_rtp_packets_ == 0)
39 EXPECT_LE(1, number_of_rtcp_packets_); 39 EXPECT_LE(1, number_of_rtcp_packets_);
40 ++number_of_rtp_packets_; 40 ++number_of_rtp_packets_;
41 } 41 }
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132 base::TimeDelta max_rtcp_timeout = 132 base::TimeDelta max_rtcp_timeout =
133 base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2); 133 base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
134 testing_clock_->Advance(max_rtcp_timeout); 134 testing_clock_->Advance(max_rtcp_timeout);
135 task_runner_->RunTasks(); 135 task_runner_->RunTasks();
136 EXPECT_LE(1, transport_.number_of_rtp_packets()); 136 EXPECT_LE(1, transport_.number_of_rtp_packets());
137 EXPECT_LE(1, transport_.number_of_rtcp_packets()); 137 EXPECT_LE(1, transport_.number_of_rtcp_packets());
138 } 138 }
139 139
140 } // namespace cast 140 } // namespace cast
141 } // namespace media 141 } // namespace media
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