| Index: media/cast/audio_receiver/audio_receiver_unittest.cc
|
| 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 e53c1b93102d9e5af3670221008570d0bffef37e..0000000000000000000000000000000000000000
|
| --- 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,
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| - const base::TimeTicks& expected_playout_time) {
|
| - expected_results_.push_back(
|
| - std::make_pair(expected_frame_id, expected_playout_time));
|
| - }
|
| -
|
| - void DeliverEncodedAudioFrame(
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| - 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();
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| - task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
|
| -
|
| - cast_environment_ = new CastEnvironment(
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| - scoped_ptr<base::TickClock>(testing_clock_).Pass(),
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| - task_runner_,
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| - task_runner_,
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| - task_runner_);
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| -
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| - receiver_.reset(new AudioReceiver(cast_environment_, audio_config_,
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| - &mock_transport_));
|
| - }
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| -
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| - virtual ~AudioReceiverTest() {}
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| -
|
| - virtual void SetUp() {
|
| - payload_.assign(kMaxIpPacketSize, 0);
|
| - rtp_header_.is_key_frame = true;
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| - rtp_header_.frame_id = kFirstFrameId;
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| - rtp_header_.packet_id = 0;
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| - rtp_header_.max_packet_id = 0;
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| - rtp_header_.reference_frame_id = rtp_header_.frame_id;
|
| - rtp_header_.rtp_timestamp = 0;
|
| - }
|
| -
|
| - void FeedOneFrameIntoReceiver() {
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| - receiver_->OnReceivedPayloadData(
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| - payload_.data(), payload_.size(), rtp_header_);
|
| - }
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| -
|
| - void FeedLipSyncInfoIntoReceiver() {
|
| - const base::TimeTicks now = testing_clock_->NowTicks();
|
| - const int64 rtp_timestamp = (now - start_time_) *
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| - audio_config_.frequency / base::TimeDelta::FromSeconds(1);
|
| - CHECK_LE(0, rtp_timestamp);
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| - uint32 ntp_seconds;
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| - uint32 ntp_fraction;
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| - ConvertTimeTicksToNtp(now, &ntp_seconds, &ntp_fraction);
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| - TestRtcpPacketBuilder rtcp_packet;
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| - rtcp_packet.AddSrWithNtp(audio_config_.incoming_ssrc,
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| - ntp_seconds, ntp_fraction,
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| - static_cast<uint32>(rtp_timestamp));
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| - receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
|
| - }
|
| -
|
| - FrameReceiverConfig audio_config_;
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| - std::vector<uint8> payload_;
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| - RtpCastHeader rtp_header_;
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| - base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.
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| - base::TimeTicks start_time_;
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| - transport::MockPacedPacketSender mock_transport_;
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| - scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
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| - scoped_refptr<CastEnvironment> cast_environment_;
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| - FakeAudioClient fake_audio_client_;
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| -
|
| - // Important for the AudioReceiver to be declared last, since its dependencies
|
| - // must remain alive until after its destruction.
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| - scoped_ptr<AudioReceiver> receiver_;
|
| -};
|
| -
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| -TEST_F(AudioReceiverTest, ReceivesOneFrame) {
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| - SimpleEventSubscriber event_subscriber;
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| - cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
|
| -
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| - EXPECT_CALL(mock_transport_, SendRtcpPacket(_, _))
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| - .WillRepeatedly(testing::Return(true));
|
| -
|
| - FeedLipSyncInfoIntoReceiver();
|
| - task_runner_->RunTasks();
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| -
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| - // Enqueue a request for an audio frame.
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| - receiver_->GetEncodedAudioFrame(
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| - base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
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| - base::Unretained(&fake_audio_client_)));
|
| -
|
| - // The request should not be satisfied since no packets have been received.
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| - task_runner_->RunTasks();
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| - 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 =
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| - base::TimeDelta::FromMilliseconds(kPlayoutDelayMillis);
|
| - fake_audio_client_.AddExpectedResult(
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| - kFirstFrameId, testing_clock_->NowTicks() + target_playout_delay);
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| - FeedOneFrameIntoReceiver();
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| - task_runner_->RunTasks();
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| - EXPECT_EQ(1, fake_audio_client_.number_times_called());
|
| -
|
| - std::vector<FrameEvent> frame_events;
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| - event_subscriber.GetFrameEventsAndReset(&frame_events);
|
| -
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| - ASSERT_TRUE(!frame_events.empty());
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| - EXPECT_EQ(FRAME_ACK_SENT, frame_events.begin()->type);
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| - EXPECT_EQ(AUDIO_EVENT, frame_events.begin()->media_type);
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| - 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 =
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| - 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,
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| - 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.
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| - 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
|
|
|