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[chromium-blink-merge.git] / media / cast / audio_receiver / audio_receiver_unittest.cc
blobbcb7e3412186d6cbcf1f1a0efc2ed12940c9449e
1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #include "base/bind.h"
6 #include "base/memory/ref_counted.h"
7 #include "base/memory/scoped_ptr.h"
8 #include "base/test/simple_test_tick_clock.h"
9 #include "media/cast/audio_receiver/audio_receiver.h"
10 #include "media/cast/cast_defines.h"
11 #include "media/cast/cast_environment.h"
12 #include "media/cast/logging/simple_event_subscriber.h"
13 #include "media/cast/rtcp/test_rtcp_packet_builder.h"
14 #include "media/cast/test/fake_single_thread_task_runner.h"
15 #include "media/cast/transport/pacing/mock_paced_packet_sender.h"
16 #include "testing/gmock/include/gmock/gmock.h"
18 namespace media {
19 namespace cast {
21 using ::testing::_;
23 namespace {
25 const int64 kStartMillisecond = INT64_C(12345678900000);
26 const uint32 kFirstFrameId = 1234;
28 class FakeAudioClient {
29 public:
30 FakeAudioClient() : num_called_(0) {}
31 virtual ~FakeAudioClient() {}
33 void SetNextExpectedResult(uint32 expected_frame_id,
34 const base::TimeTicks& expected_playout_time) {
35 expected_frame_id_ = expected_frame_id;
36 expected_playout_time_ = expected_playout_time;
39 void DeliverEncodedAudioFrame(
40 scoped_ptr<transport::EncodedFrame> audio_frame) {
41 ASSERT_FALSE(!audio_frame)
42 << "If at shutdown: There were unsatisfied requests enqueued.";
43 EXPECT_EQ(expected_frame_id_, audio_frame->frame_id);
44 EXPECT_EQ(expected_playout_time_, audio_frame->reference_time);
45 num_called_++;
48 int number_times_called() const { return num_called_; }
50 private:
51 int num_called_;
52 uint32 expected_frame_id_;
53 base::TimeTicks expected_playout_time_;
55 DISALLOW_COPY_AND_ASSIGN(FakeAudioClient);
58 } // namespace
60 class AudioReceiverTest : public ::testing::Test {
61 protected:
62 AudioReceiverTest() {
63 // Configure the audio receiver to use PCM16.
64 audio_config_.rtp_payload_type = 127;
65 audio_config_.frequency = 16000;
66 audio_config_.channels = 1;
67 audio_config_.codec = transport::kPcm16;
68 audio_config_.use_external_decoder = true;
69 audio_config_.feedback_ssrc = 1234;
70 testing_clock_ = new base::SimpleTestTickClock();
71 testing_clock_->Advance(
72 base::TimeDelta::FromMilliseconds(kStartMillisecond));
73 task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
75 cast_environment_ = new CastEnvironment(
76 scoped_ptr<base::TickClock>(testing_clock_).Pass(),
77 task_runner_,
78 task_runner_,
79 task_runner_);
81 receiver_.reset(new AudioReceiver(cast_environment_, audio_config_,
82 &mock_transport_));
85 virtual ~AudioReceiverTest() {}
87 virtual void SetUp() {
88 payload_.assign(kMaxIpPacketSize, 0);
89 rtp_header_.is_key_frame = true;
90 rtp_header_.frame_id = kFirstFrameId;
91 rtp_header_.packet_id = 0;
92 rtp_header_.max_packet_id = 0;
93 rtp_header_.reference_frame_id = rtp_header_.frame_id;
94 rtp_header_.rtp_timestamp = 0;
97 void FeedOneFrameIntoReceiver() {
98 receiver_->OnReceivedPayloadData(
99 payload_.data(), payload_.size(), rtp_header_);
102 AudioReceiverConfig audio_config_;
103 std::vector<uint8> payload_;
104 RtpCastHeader rtp_header_;
105 base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.
106 transport::MockPacedPacketSender mock_transport_;
107 scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
108 scoped_refptr<CastEnvironment> cast_environment_;
109 FakeAudioClient fake_audio_client_;
111 // Important for the AudioReceiver to be declared last, since its dependencies
112 // must remain alive until after its destruction.
113 scoped_ptr<AudioReceiver> receiver_;
116 TEST_F(AudioReceiverTest, GetOnePacketEncodedFrame) {
117 SimpleEventSubscriber event_subscriber;
118 cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
120 EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)).Times(1);
122 // Enqueue a request for an audio frame.
123 receiver_->GetEncodedAudioFrame(
124 base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
125 base::Unretained(&fake_audio_client_)));
127 // The request should not be satisfied since no packets have been received.
128 task_runner_->RunTasks();
129 EXPECT_EQ(0, fake_audio_client_.number_times_called());
131 // Deliver one audio frame to the receiver and expect to get one frame back.
132 fake_audio_client_.SetNextExpectedResult(kFirstFrameId,
133 testing_clock_->NowTicks());
134 FeedOneFrameIntoReceiver();
135 task_runner_->RunTasks();
136 EXPECT_EQ(1, fake_audio_client_.number_times_called());
138 std::vector<FrameEvent> frame_events;
139 event_subscriber.GetFrameEventsAndReset(&frame_events);
141 ASSERT_TRUE(!frame_events.empty());
142 EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
143 EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type);
144 EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
145 EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
147 cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
150 TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
151 EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
152 .WillRepeatedly(testing::Return(true));
154 // Enqueue a request for an audio frame.
155 const FrameEncodedCallback frame_encoded_callback =
156 base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
157 base::Unretained(&fake_audio_client_));
158 receiver_->GetEncodedAudioFrame(frame_encoded_callback);
159 task_runner_->RunTasks();
160 EXPECT_EQ(0, fake_audio_client_.number_times_called());
162 // Receive one audio frame and expect to see the first request satisfied.
163 fake_audio_client_.SetNextExpectedResult(kFirstFrameId,
164 testing_clock_->NowTicks());
165 FeedOneFrameIntoReceiver();
166 task_runner_->RunTasks();
167 EXPECT_EQ(1, fake_audio_client_.number_times_called());
169 TestRtcpPacketBuilder rtcp_packet;
171 uint32 ntp_high;
172 uint32 ntp_low;
173 ConvertTimeTicksToNtp(testing_clock_->NowTicks(), &ntp_high, &ntp_low);
174 rtcp_packet.AddSrWithNtp(audio_config_.feedback_ssrc, ntp_high, ntp_low,
175 rtp_header_.rtp_timestamp);
177 testing_clock_->Advance(base::TimeDelta::FromMilliseconds(20));
179 receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
181 // Enqueue a second request for an audio frame, but it should not be
182 // fulfilled yet.
183 receiver_->GetEncodedAudioFrame(frame_encoded_callback);
184 task_runner_->RunTasks();
185 EXPECT_EQ(1, fake_audio_client_.number_times_called());
187 // Receive one audio frame out-of-order: Make sure that we are not continuous
188 // and that the RTP timestamp represents a time in the future.
189 rtp_header_.is_key_frame = false;
190 rtp_header_.frame_id = kFirstFrameId + 2;
191 rtp_header_.reference_frame_id = 0;
192 rtp_header_.rtp_timestamp = 960;
193 fake_audio_client_.SetNextExpectedResult(
194 kFirstFrameId + 2,
195 testing_clock_->NowTicks() + base::TimeDelta::FromMilliseconds(100));
196 FeedOneFrameIntoReceiver();
198 // Frame 2 should not come out at this point in time.
199 task_runner_->RunTasks();
200 EXPECT_EQ(1, fake_audio_client_.number_times_called());
202 // Enqueue a third request for an audio frame.
203 receiver_->GetEncodedAudioFrame(frame_encoded_callback);
204 task_runner_->RunTasks();
205 EXPECT_EQ(1, fake_audio_client_.number_times_called());
207 // After 100 ms has elapsed, Frame 2 is emitted (to satisfy the second
208 // request) because a decision was made to skip over the no-show Frame 1.
209 testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
210 task_runner_->RunTasks();
211 EXPECT_EQ(2, fake_audio_client_.number_times_called());
213 // Receive Frame 3 and expect it to fulfill the third request immediately.
214 rtp_header_.frame_id = kFirstFrameId + 3;
215 rtp_header_.reference_frame_id = rtp_header_.frame_id - 1;
216 rtp_header_.rtp_timestamp = 1280;
217 fake_audio_client_.SetNextExpectedResult(kFirstFrameId + 3,
218 testing_clock_->NowTicks());
219 FeedOneFrameIntoReceiver();
220 task_runner_->RunTasks();
221 EXPECT_EQ(3, fake_audio_client_.number_times_called());
223 // Move forward another 100 ms and run any pending tasks (there should be
224 // none). Expect no additional frames where emitted.
225 testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
226 task_runner_->RunTasks();
227 EXPECT_EQ(3, fake_audio_client_.number_times_called());
230 } // namespace cast
231 } // namespace media