diff options
Diffstat (limited to 'media/cast/audio_receiver/audio_receiver_unittest.cc')
-rw-r--r-- | media/cast/audio_receiver/audio_receiver_unittest.cc | 261 |
1 files changed, 0 insertions, 261 deletions
diff --git a/media/cast/audio_receiver/audio_receiver_unittest.cc b/media/cast/audio_receiver/audio_receiver_unittest.cc deleted file mode 100644 index e53c1b9..0000000 --- a/media/cast/audio_receiver/audio_receiver_unittest.cc +++ /dev/null @@ -1,261 +0,0 @@ -// Copyright 2013 The Chromium Authors. All rights reserved. -// Use of this source code is governed by a BSD-style license that can be -// found in the LICENSE file. - -#include <deque> -#include <utility> - -#include "base/bind.h" -#include "base/memory/ref_counted.h" -#include "base/memory/scoped_ptr.h" -#include "base/test/simple_test_tick_clock.h" -#include "media/cast/audio_receiver/audio_receiver.h" -#include "media/cast/cast_defines.h" -#include "media/cast/cast_environment.h" -#include "media/cast/logging/simple_event_subscriber.h" -#include "media/cast/rtcp/test_rtcp_packet_builder.h" -#include "media/cast/test/fake_single_thread_task_runner.h" -#include "media/cast/test/utility/default_config.h" -#include "media/cast/transport/pacing/mock_paced_packet_sender.h" -#include "testing/gmock/include/gmock/gmock.h" - -using ::testing::_; - -namespace media { -namespace cast { - -namespace { - -const uint32 kFirstFrameId = 1234; -const int kPlayoutDelayMillis = 300; - -class FakeAudioClient { - public: - FakeAudioClient() : num_called_(0) {} - virtual ~FakeAudioClient() {} - - void AddExpectedResult(uint32 expected_frame_id, - const base::TimeTicks& expected_playout_time) { - expected_results_.push_back( - std::make_pair(expected_frame_id, expected_playout_time)); - } - - void DeliverEncodedAudioFrame( - scoped_ptr<transport::EncodedFrame> audio_frame) { - SCOPED_TRACE(::testing::Message() << "num_called_ is " << num_called_); - ASSERT_FALSE(!audio_frame) - << "If at shutdown: There were unsatisfied requests enqueued."; - ASSERT_FALSE(expected_results_.empty()); - EXPECT_EQ(expected_results_.front().first, audio_frame->frame_id); - EXPECT_EQ(expected_results_.front().second, audio_frame->reference_time); - expected_results_.pop_front(); - num_called_++; - } - - int number_times_called() const { return num_called_; } - - private: - std::deque<std::pair<uint32, base::TimeTicks> > expected_results_; - int num_called_; - - DISALLOW_COPY_AND_ASSIGN(FakeAudioClient); -}; - -} // namespace - -class AudioReceiverTest : public ::testing::Test { - protected: - AudioReceiverTest() { - // Configure the audio receiver to use PCM16. - audio_config_ = GetDefaultAudioReceiverConfig(); - audio_config_.rtp_max_delay_ms = kPlayoutDelayMillis; - audio_config_.frequency = 16000; - audio_config_.channels = 1; - audio_config_.codec.audio = transport::kPcm16; - testing_clock_ = new base::SimpleTestTickClock(); - testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks()); - start_time_ = testing_clock_->NowTicks(); - task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_); - - cast_environment_ = new CastEnvironment( - scoped_ptr<base::TickClock>(testing_clock_).Pass(), - task_runner_, - task_runner_, - task_runner_); - - receiver_.reset(new AudioReceiver(cast_environment_, audio_config_, - &mock_transport_)); - } - - virtual ~AudioReceiverTest() {} - - virtual void SetUp() { - payload_.assign(kMaxIpPacketSize, 0); - rtp_header_.is_key_frame = true; - rtp_header_.frame_id = kFirstFrameId; - rtp_header_.packet_id = 0; - rtp_header_.max_packet_id = 0; - rtp_header_.reference_frame_id = rtp_header_.frame_id; - rtp_header_.rtp_timestamp = 0; - } - - void FeedOneFrameIntoReceiver() { - receiver_->OnReceivedPayloadData( - payload_.data(), payload_.size(), rtp_header_); - } - - void FeedLipSyncInfoIntoReceiver() { - const base::TimeTicks now = testing_clock_->NowTicks(); - const int64 rtp_timestamp = (now - start_time_) * - audio_config_.frequency / base::TimeDelta::FromSeconds(1); - CHECK_LE(0, rtp_timestamp); - uint32 ntp_seconds; - uint32 ntp_fraction; - ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction); - TestRtcpPacketBuilder rtcp_packet; - rtcp_packet.AddSrWithNtp(audio_config_.incoming_ssrc, - ntp_seconds, ntp_fraction, - static_cast<uint32>(rtp_timestamp)); - receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass()); - } - - FrameReceiverConfig audio_config_; - std::vector<uint8> payload_; - RtpCastHeader rtp_header_; - base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment. - base::TimeTicks start_time_; - transport::MockPacedPacketSender mock_transport_; - scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_; - scoped_refptr<CastEnvironment> cast_environment_; - FakeAudioClient fake_audio_client_; - - // Important for the AudioReceiver to be declared last, since its dependencies - // must remain alive until after its destruction. - scoped_ptr<AudioReceiver> receiver_; -}; - -TEST_F(AudioReceiverTest, ReceivesOneFrame) { - SimpleEventSubscriber event_subscriber; - cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber); - - EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)) - .WillRepeatedly(testing::Return(true)); - - FeedLipSyncInfoIntoReceiver(); - task_runner_->RunTasks(); - - // Enqueue a request for an audio frame. - receiver_->GetEncodedAudioFrame( - base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame, - base::Unretained(&fake_audio_client_))); - - // The request should not be satisfied since no packets have been received. - task_runner_->RunTasks(); - EXPECT_EQ(0, fake_audio_client_.number_times_called()); - - // Deliver one audio frame to the receiver and expect to get one frame back. - const base::TimeDelta target_playout_delay = - base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis); - fake_audio_client_.AddExpectedResult( - kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay); - FeedOneFrameIntoReceiver(); - task_runner_->RunTasks(); - EXPECT_EQ(1, fake_audio_client_.number_times_called()); - - std::vector<FrameEvent> frame_events; - event_subscriber.GetFrameEventsAndReset(&frame_events); - - ASSERT_TRUE(!frame_events.empty()); - EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type); - EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type); - EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id); - EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp); - - cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber); -} - -TEST_F(AudioReceiverTest, ReceivesFramesSkippingWhenAppropriate) { - EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _)) - .WillRepeatedly(testing::Return(true)); - - const uint32 rtp_advance_per_frame = - audio_config_.frequency / audio_config_.max_frame_rate; - const base::TimeDelta time_advance_per_frame = - base::TimeDelta::FromSeconds(1) / audio_config_.max_frame_rate; - - FeedLipSyncInfoIntoReceiver(); - task_runner_->RunTasks(); - const base::TimeTicks first_frame_capture_time = testing_clock_->NowTicks(); - - // Enqueue a request for an audio frame. - const FrameEncodedCallback frame_encoded_callback = - base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame, - base::Unretained(&fake_audio_client_)); - receiver_->GetEncodedAudioFrame(frame_encoded_callback); - task_runner_->RunTasks(); - EXPECT_EQ(0, fake_audio_client_.number_times_called()); - - // Receive one audio frame and expect to see the first request satisfied. - const base::TimeDelta target_playout_delay = - base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis); - fake_audio_client_.AddExpectedResult( - kFirstFrameId, first_frame_capture_time + target_playout_delay); - rtp_header_.rtp_timestamp = 0; - FeedOneFrameIntoReceiver(); - task_runner_->RunTasks(); - EXPECT_EQ(1, fake_audio_client_.number_times_called()); - - // Enqueue a second request for an audio frame, but it should not be - // fulfilled yet. - receiver_->GetEncodedAudioFrame(frame_encoded_callback); - task_runner_->RunTasks(); - EXPECT_EQ(1, fake_audio_client_.number_times_called()); - - // Receive one audio frame out-of-order: Make sure that we are not continuous - // and that the RTP timestamp represents a time in the future. - rtp_header_.is_key_frame = false; - rtp_header_.frame_id = kFirstFrameId + 2; - rtp_header_.reference_frame_id = 0; - rtp_header_.rtp_timestamp += 2 * rtp_advance_per_frame; - fake_audio_client_.AddExpectedResult( - kFirstFrameId + 2, - first_frame_capture_time + 2 * time_advance_per_frame + - target_playout_delay); - FeedOneFrameIntoReceiver(); - - // Frame 2 should not come out at this point in time. - task_runner_->RunTasks(); - EXPECT_EQ(1, fake_audio_client_.number_times_called()); - - // Enqueue a third request for an audio frame. - receiver_->GetEncodedAudioFrame(frame_encoded_callback); - task_runner_->RunTasks(); - EXPECT_EQ(1, fake_audio_client_.number_times_called()); - - // Now, advance time forward such that the receiver is convinced it should - // skip Frame 2. Frame 3 is emitted (to satisfy the second request) because a - // decision was made to skip over the no-show Frame 2. - testing_clock_->Advance(2 * time_advance_per_frame + target_playout_delay); - task_runner_->RunTasks(); - EXPECT_EQ(2, fake_audio_client_.number_times_called()); - - // Receive Frame 4 and expect it to fulfill the third request immediately. - rtp_header_.frame_id = kFirstFrameId + 3; - rtp_header_.reference_frame_id = rtp_header_.frame_id - 1; - rtp_header_.rtp_timestamp += rtp_advance_per_frame; - fake_audio_client_.AddExpectedResult( - kFirstFrameId + 3, first_frame_capture_time + 3 * time_advance_per_frame + - target_playout_delay); - FeedOneFrameIntoReceiver(); - task_runner_->RunTasks(); - EXPECT_EQ(3, fake_audio_client_.number_times_called()); - - // Move forward to the playout time of an unreceived Frame 5. Expect no - // additional frames were emitted. - testing_clock_->Advance(3 * time_advance_per_frame); - task_runner_->RunTasks(); - EXPECT_EQ(3, fake_audio_client_.number_times_called()); -} - -} // namespace cast -} // namespace media |