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