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-rw-r--r--media/cast/audio_receiver/audio_receiver_unittest.cc261
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