Index: media/cast/audio_sender/audio_sender_unittest.cc |
diff --git a/media/cast/audio_sender/audio_sender_unittest.cc b/media/cast/audio_sender/audio_sender_unittest.cc |
deleted file mode 100644 |
index 69b6d85b83dfbe11f605fe4c304751e32f2d2634..0000000000000000000000000000000000000000 |
--- a/media/cast/audio_sender/audio_sender_unittest.cc |
+++ /dev/null |
@@ -1,141 +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 <stdint.h> |
- |
-#include "base/bind.h" |
-#include "base/bind_helpers.h" |
-#include "base/memory/scoped_ptr.h" |
-#include "base/test/simple_test_tick_clock.h" |
-#include "media/base/media.h" |
-#include "media/cast/audio_sender/audio_sender.h" |
-#include "media/cast/cast_config.h" |
-#include "media/cast/cast_environment.h" |
-#include "media/cast/rtcp/rtcp.h" |
-#include "media/cast/test/fake_single_thread_task_runner.h" |
-#include "media/cast/test/utility/audio_utility.h" |
-#include "media/cast/transport/cast_transport_config.h" |
-#include "media/cast/transport/cast_transport_sender_impl.h" |
-#include "testing/gtest/include/gtest/gtest.h" |
- |
-namespace media { |
-namespace cast { |
- |
-class TestPacketSender : public transport::PacketSender { |
- public: |
- TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {} |
- |
- virtual bool SendPacket(transport::PacketRef packet, |
- const base::Closure& cb) OVERRIDE { |
- if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { |
- ++number_of_rtcp_packets_; |
- } else { |
- // Check that at least one RTCP packet was sent before the first RTP |
- // packet. This confirms that the receiver will have the necessary lip |
- // sync info before it has to calculate the playout time of the first |
- // frame. |
- if (number_of_rtp_packets_ == 0) |
- EXPECT_LE(1, number_of_rtcp_packets_); |
- ++number_of_rtp_packets_; |
- } |
- return true; |
- } |
- |
- int number_of_rtp_packets() const { return number_of_rtp_packets_; } |
- |
- int number_of_rtcp_packets() const { return number_of_rtcp_packets_; } |
- |
- private: |
- int number_of_rtp_packets_; |
- int number_of_rtcp_packets_; |
- |
- DISALLOW_COPY_AND_ASSIGN(TestPacketSender); |
-}; |
- |
-class AudioSenderTest : public ::testing::Test { |
- protected: |
- AudioSenderTest() { |
- InitializeMediaLibraryForTesting(); |
- testing_clock_ = new base::SimpleTestTickClock(); |
- testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks()); |
- task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_); |
- cast_environment_ = |
- new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(), |
- task_runner_, |
- task_runner_, |
- task_runner_); |
- audio_config_.codec = transport::CODEC_AUDIO_OPUS; |
- audio_config_.use_external_encoder = false; |
- audio_config_.frequency = kDefaultAudioSamplingRate; |
- audio_config_.channels = 2; |
- audio_config_.bitrate = kDefaultAudioEncoderBitrate; |
- audio_config_.rtp_payload_type = 127; |
- |
- net::IPEndPoint dummy_endpoint; |
- |
- transport_sender_.reset(new transport::CastTransportSenderImpl( |
- NULL, |
- testing_clock_, |
- dummy_endpoint, |
- base::Bind(&UpdateCastTransportStatus), |
- transport::BulkRawEventsCallback(), |
- base::TimeDelta(), |
- task_runner_, |
- &transport_)); |
- audio_sender_.reset(new AudioSender( |
- cast_environment_, audio_config_, transport_sender_.get())); |
- task_runner_->RunTasks(); |
- } |
- |
- virtual ~AudioSenderTest() {} |
- |
- static void UpdateCastTransportStatus(transport::CastTransportStatus status) { |
- EXPECT_EQ(transport::TRANSPORT_AUDIO_INITIALIZED, status); |
- } |
- |
- base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment. |
- TestPacketSender transport_; |
- scoped_ptr<transport::CastTransportSenderImpl> transport_sender_; |
- scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_; |
- scoped_ptr<AudioSender> audio_sender_; |
- scoped_refptr<CastEnvironment> cast_environment_; |
- AudioSenderConfig audio_config_; |
-}; |
- |
-TEST_F(AudioSenderTest, Encode20ms) { |
- const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20); |
- scoped_ptr<AudioBus> bus( |
- TestAudioBusFactory(audio_config_.channels, |
- audio_config_.frequency, |
- TestAudioBusFactory::kMiddleANoteFreq, |
- 0.5f).NextAudioBus(kDuration)); |
- |
- audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks()); |
- task_runner_->RunTasks(); |
- EXPECT_LE(1, transport_.number_of_rtp_packets()); |
- EXPECT_LE(1, transport_.number_of_rtcp_packets()); |
-} |
- |
-TEST_F(AudioSenderTest, RtcpTimer) { |
- const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20); |
- scoped_ptr<AudioBus> bus( |
- TestAudioBusFactory(audio_config_.channels, |
- audio_config_.frequency, |
- TestAudioBusFactory::kMiddleANoteFreq, |
- 0.5f).NextAudioBus(kDuration)); |
- |
- audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks()); |
- task_runner_->RunTasks(); |
- |
- // Make sure that we send at least one RTCP packet. |
- base::TimeDelta max_rtcp_timeout = |
- base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2); |
- testing_clock_->Advance(max_rtcp_timeout); |
- task_runner_->RunTasks(); |
- EXPECT_LE(1, transport_.number_of_rtp_packets()); |
- EXPECT_LE(1, transport_.number_of_rtcp_packets()); |
-} |
- |
-} // namespace cast |
-} // namespace media |