src/media/cast/sender/audio_sender_unittest.cc

   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
   3 // found in the LICENSE file.
   4
   5 #include <stdint.h>
   6
   7 #include "base/bind.h"
   8 #include "base/bind_helpers.h"
   9 #include "base/memory/scoped_ptr.h"
  10 #include "base/test/simple_test_tick_clock.h"
  11 #include "media/base/media.h"
  12 #include "media/cast/cast_config.h"
  13 #include "media/cast/cast_environment.h"
  14 #include "media/cast/net/cast_transport_config.h"
  15 #include "media/cast/net/cast_transport_sender_impl.h"
  16 #include "media/cast/sender/audio_sender.h"
  17 #include "media/cast/test/fake_single_thread_task_runner.h"
  18 #include "media/cast/test/utility/audio_utility.h"
  19 #include "testing/gtest/include/gtest/gtest.h"
  20
  21 namespace media {
  22 namespace cast {
  23
  24 class TestPacketSender : public PacketSender {
  25  public:
  26   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
  27
  28   virtual bool SendPacket(PacketRef packet,
  29                           const base::Closure& cb) OVERRIDE {
  30     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
  31       ++number_of_rtcp_packets_;
  32     } else {
  33       // Check that at least one RTCP packet was sent before the first RTP
  34       // packet.  This confirms that the receiver will have the necessary lip
  35       // sync info before it has to calculate the playout time of the first
  36       // frame.
  37       if (number_of_rtp_packets_ == 0)
  38         EXPECT_LE(1, number_of_rtcp_packets_);
  39       ++number_of_rtp_packets_;
  40     }
  41     return true;
  42   }
  43
  44   virtual int64 GetBytesSent() OVERRIDE {
  45     return 0;
  46   }
  47
  48   int number_of_rtp_packets() const { return number_of_rtp_packets_; }
  49
  50   int number_of_rtcp_packets() const { return number_of_rtcp_packets_; }
  51
  52  private:
  53   int number_of_rtp_packets_;
  54   int number_of_rtcp_packets_;
  55
  56   DISALLOW_COPY_AND_ASSIGN(TestPacketSender);
  57 };
  58
  59 class AudioSenderTest : public ::testing::Test {
  60  protected:
  61   AudioSenderTest() {
  62     InitializeMediaLibraryForTesting();
  63     testing_clock_ = new base::SimpleTestTickClock();
  64     testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
  65     task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
  66     cast_environment_ =
  67         new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(),
  68                             task_runner_,
  69                             task_runner_,
  70                             task_runner_);
  71     audio_config_.codec = CODEC_AUDIO_OPUS;
  72     audio_config_.use_external_encoder = false;
  73     audio_config_.frequency = kDefaultAudioSamplingRate;
  74     audio_config_.channels = 2;
  75     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
  76     audio_config_.rtp_payload_type = 127;
  77
  78     net::IPEndPoint dummy_endpoint;
  79
  80     transport_sender_.reset(new CastTransportSenderImpl(
  81         NULL,
  82         testing_clock_,
  83         dummy_endpoint,
  84         make_scoped_ptr(new base::DictionaryValue),
  85         base::Bind(&UpdateCastTransportStatus),
  86         BulkRawEventsCallback(),
  87         base::TimeDelta(),
  88         task_runner_,
  89         &transport_));
  90     audio_sender_.reset(new AudioSender(
  91         cast_environment_, audio_config_, transport_sender_.get()));
  92     task_runner_->RunTasks();
  93   }
  94
  95   virtual ~AudioSenderTest() {}
  96
  97   static void UpdateCastTransportStatus(CastTransportStatus status) {
  98     EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
  99   }
 100
 101   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
 102   TestPacketSender transport_;
 103   scoped_ptr<CastTransportSenderImpl> transport_sender_;
 104   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
 105   scoped_ptr<AudioSender> audio_sender_;
 106   scoped_refptr<CastEnvironment> cast_environment_;
 107   AudioSenderConfig audio_config_;
 108 };
 109
 110 TEST_F(AudioSenderTest, Encode20ms) {
 111   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
 112   scoped_ptr<AudioBus> bus(
 113       TestAudioBusFactory(audio_config_.channels,
 114                           audio_config_.frequency,
 115                           TestAudioBusFactory::kMiddleANoteFreq,
 116                           0.5f).NextAudioBus(kDuration));
 117
 118   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
 119   task_runner_->RunTasks();
 120   EXPECT_LE(1, transport_.number_of_rtp_packets());
 121   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 122 }
 123
 124 TEST_F(AudioSenderTest, RtcpTimer) {
 125   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
 126   scoped_ptr<AudioBus> bus(
 127       TestAudioBusFactory(audio_config_.channels,
 128                           audio_config_.frequency,
 129                           TestAudioBusFactory::kMiddleANoteFreq,
 130                           0.5f).NextAudioBus(kDuration));
 131
 132   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
 133   task_runner_->RunTasks();
 134
 135   // Make sure that we send at least one RTCP packet.
 136   base::TimeDelta max_rtcp_timeout =
 137       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
 138   testing_clock_->Advance(max_rtcp_timeout);
 139   task_runner_->RunTasks();
 140   EXPECT_LE(1, transport_.number_of_rtp_packets());
 141   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 142 }
 143
 144 }  // namespace cast
 145 }  // namespace media