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// Copyright 2014 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/cast_config.h"
#include "media/cast/cast_environment.h"
#include "media/cast/net/cast_transport_config.h"
#include "media/cast/net/cast_transport_sender_impl.h"
#include "media/cast/sender/audio_sender.h"
#include "media/cast/test/fake_single_thread_task_runner.h"
#include "media/cast/test/utility/audio_utility.h"
#include "testing/gtest/include/gtest/gtest.h"
namespace media {
namespace cast {
namespace {
void SaveOperationalStatus(OperationalStatus* out_status,
OperationalStatus in_status) {
DVLOG(1) << "OperationalStatus transitioning from " << *out_status << " to "
<< in_status;
*out_status = in_status;
}
} // namespace
class TestPacketSender : public PacketSender {
public:
TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
bool SendPacket(PacketRef packet, const base::Closure& cb) final {
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;
}
int64 GetBytesSent() final { return 0; }
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 = 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 CastTransportSenderImpl(
NULL,
testing_clock_,
net::IPEndPoint(),
dummy_endpoint,
make_scoped_ptr(new base::DictionaryValue),
base::Bind(&UpdateCastTransportStatus),
BulkRawEventsCallback(),
base::TimeDelta(),
task_runner_,
PacketReceiverCallback(),
&transport_));
OperationalStatus operational_status = STATUS_UNINITIALIZED;
audio_sender_.reset(new AudioSender(
cast_environment_,
audio_config_,
base::Bind(&SaveOperationalStatus, &operational_status),
transport_sender_.get()));
task_runner_->RunTasks();
CHECK_EQ(STATUS_INITIALIZED, operational_status);
}
~AudioSenderTest() override {}
static void UpdateCastTransportStatus(CastTransportStatus status) {
EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
}
base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.
TestPacketSender transport_;
scoped_ptr<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
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