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   bool SendPacket(PacketRef packet, const base::Closure& cb) override {
  29     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
  30       ++number_of_rtcp_packets_;
  31     } else {
  32       // Check that at least one RTCP packet was sent before the first RTP
  33       // packet.  This confirms that the receiver will have the necessary lip
  34       // sync info before it has to calculate the playout time of the first
  35       // frame.
  36       if (number_of_rtp_packets_ == 0)
  37         EXPECT_LE(1, number_of_rtcp_packets_);
  38       ++number_of_rtp_packets_;
  39     }
  40     return true;
  41   }
  42
  43   int64 GetBytesSent() override { return 0; }
  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         make_scoped_ptr(new base::DictionaryValue),
  82         base::Bind(&UpdateCastTransportStatus),
  83         BulkRawEventsCallback(),
  84         base::TimeDelta(),
  85         task_runner_,
  86         &transport_));
  87     audio_sender_.reset(new AudioSender(
  88         cast_environment_, audio_config_, transport_sender_.get()));
  89     task_runner_->RunTasks();
  90   }
  91
  92   ~AudioSenderTest() override {}
  93
  94   static void UpdateCastTransportStatus(CastTransportStatus status) {
  95     EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
  96   }
  97
  98   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
  99   TestPacketSender transport_;
 100   scoped_ptr<CastTransportSenderImpl> transport_sender_;
 101   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
 102   scoped_ptr<AudioSender> audio_sender_;
 103   scoped_refptr<CastEnvironment> cast_environment_;
 104   AudioSenderConfig audio_config_;
 105 };
 106
 107 TEST_F(AudioSenderTest, Encode20ms) {
 108   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
 109   scoped_ptr<AudioBus> bus(
 110       TestAudioBusFactory(audio_config_.channels,
 111                           audio_config_.frequency,
 112                           TestAudioBusFactory::kMiddleANoteFreq,
 113                           0.5f).NextAudioBus(kDuration));
 114
 115   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
 116   task_runner_->RunTasks();
 117   EXPECT_LE(1, transport_.number_of_rtp_packets());
 118   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 119 }
 120
 121 TEST_F(AudioSenderTest, RtcpTimer) {
 122   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
 123   scoped_ptr<AudioBus> bus(
 124       TestAudioBusFactory(audio_config_.channels,
 125                           audio_config_.frequency,
 126                           TestAudioBusFactory::kMiddleANoteFreq,
 127                           0.5f).NextAudioBus(kDuration));
 128
 129   audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
 130   task_runner_->RunTasks();
 131
 132   // Make sure that we send at least one RTCP packet.
 133   base::TimeDelta max_rtcp_timeout =
 134       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
 135   testing_clock_->Advance(max_rtcp_timeout);
 136   task_runner_->RunTasks();
 137   EXPECT_LE(1, transport_.number_of_rtp_packets());
 138   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 139 }
 140
 141 }  // namespace cast
 142 }  // namespace media