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      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
    146