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// 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 "base/bind.h"
#include "base/memory/ref_counted.h"
#include "base/memory/scoped_ptr.h"
#include "base/test/simple_test_tick_clock.h"
#include "media/cast/audio_receiver/audio_receiver.h"
#include "media/cast/cast_defines.h"
#include "media/cast/cast_environment.h"
#include "media/cast/logging/simple_event_subscriber.h"
#include "media/cast/rtcp/test_rtcp_packet_builder.h"
#include "media/cast/test/fake_single_thread_task_runner.h"
#include "media/cast/transport/pacing/mock_paced_packet_sender.h"
#include "testing/gmock/include/gmock/gmock.h"
namespace media {
namespace cast {
static const int64 kStartMillisecond = GG_INT64_C(12345678900000);
namespace {
class TestAudioEncoderCallback
: public base::RefCountedThreadSafe<TestAudioEncoderCallback> {
public:
TestAudioEncoderCallback() : num_called_(0) {}
void SetExpectedResult(uint8 expected_frame_id,
const base::TimeTicks& expected_playout_time) {
expected_frame_id_ = expected_frame_id;
expected_playout_time_ = expected_playout_time;
}
void DeliverEncodedAudioFrame(
scoped_ptr<transport::EncodedAudioFrame> audio_frame,
const base::TimeTicks& playout_time) {
EXPECT_EQ(expected_frame_id_, audio_frame->frame_id);
EXPECT_EQ(transport::kPcm16, audio_frame->codec);
EXPECT_EQ(expected_playout_time_, playout_time);
num_called_++;
}
int number_times_called() const { return num_called_; }
protected:
virtual ~TestAudioEncoderCallback() {}
private:
friend class base::RefCountedThreadSafe<TestAudioEncoderCallback>;
int num_called_;
uint8 expected_frame_id_;
base::TimeTicks expected_playout_time_;
DISALLOW_COPY_AND_ASSIGN(TestAudioEncoderCallback);
};
} // namespace
class PeerAudioReceiver : public AudioReceiver {
public:
PeerAudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
const AudioReceiverConfig& audio_config,
transport::PacedPacketSender* const packet_sender)
: AudioReceiver(cast_environment, audio_config, packet_sender) {}
using AudioReceiver::IncomingParsedRtpPacket;
};
class AudioReceiverTest : public ::testing::Test {
protected:
AudioReceiverTest() {
// Configure the audio receiver to use PCM16.
audio_config_.rtp_payload_type = 127;
audio_config_.frequency = 16000;
audio_config_.channels = 1;
audio_config_.codec = transport::kPcm16;
audio_config_.use_external_decoder = false;
audio_config_.feedback_ssrc = 1234;
testing_clock_ = new base::SimpleTestTickClock();
testing_clock_->Advance(
base::TimeDelta::FromMilliseconds(kStartMillisecond));
task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
CastLoggingConfig logging_config(GetDefaultCastReceiverLoggingConfig());
logging_config.enable_raw_data_collection = true;
cast_environment_ = new CastEnvironment(
scoped_ptr<base::TickClock>(testing_clock_).Pass(), task_runner_,
task_runner_, task_runner_, task_runner_, task_runner_, task_runner_,
logging_config);
test_audio_encoder_callback_ = new TestAudioEncoderCallback();
}
void Configure(bool use_external_decoder) {
audio_config_.use_external_decoder = use_external_decoder;
receiver_.reset(new PeerAudioReceiver(cast_environment_, audio_config_,
&mock_transport_));
}
virtual ~AudioReceiverTest() {}
static void DummyDeletePacket(const uint8* packet) {};
virtual void SetUp() {
payload_.assign(kMaxIpPacketSize, 0);
rtp_header_.is_key_frame = true;
rtp_header_.frame_id = 0;
rtp_header_.packet_id = 0;
rtp_header_.max_packet_id = 0;
rtp_header_.is_reference = false;
rtp_header_.reference_frame_id = 0;
rtp_header_.webrtc.header.timestamp = 0;
}
AudioReceiverConfig audio_config_;
std::vector<uint8> payload_;
RtpCastHeader rtp_header_;
base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment.
transport::MockPacedPacketSender mock_transport_;
scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
scoped_ptr<PeerAudioReceiver> receiver_;
scoped_refptr<CastEnvironment> cast_environment_;
scoped_refptr<TestAudioEncoderCallback> test_audio_encoder_callback_;
};
TEST_F(AudioReceiverTest, GetOnePacketEncodedframe) {
SimpleEventSubscriber event_subscriber;
cast_environment_->Logging()->AddRawEventSubscriber(&event_subscriber);
Configure(true);
EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_)).Times(1);
receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
rtp_header_);
transport::EncodedAudioFrame audio_frame;
base::TimeTicks playout_time;
test_audio_encoder_callback_->SetExpectedResult(0,
testing_clock_->NowTicks());
AudioFrameEncodedCallback frame_encoded_callback =
base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
test_audio_encoder_callback_.get());
receiver_->GetEncodedAudioFrame(frame_encoded_callback);
task_runner_->RunTasks();
EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
std::vector<FrameEvent> frame_events;
event_subscriber.GetFrameEventsAndReset(&frame_events);
ASSERT_TRUE(!frame_events.empty());
EXPECT_EQ(kAudioAckSent, frame_events.begin()->type);
EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
EXPECT_EQ(rtp_header_.webrtc.header.timestamp,
frame_events.begin()->rtp_timestamp);
cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
}
TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
Configure(true);
EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_))
.WillRepeatedly(testing::Return(true));
AudioFrameEncodedCallback frame_encoded_callback =
base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
test_audio_encoder_callback_.get());
receiver_->GetEncodedAudioFrame(frame_encoded_callback);
receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
rtp_header_);
transport::EncodedAudioFrame audio_frame;
base::TimeTicks playout_time;
test_audio_encoder_callback_->SetExpectedResult(0,
testing_clock_->NowTicks());
task_runner_->RunTasks();
EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
TestRtcpPacketBuilder rtcp_packet;
uint32 ntp_high;
uint32 ntp_low;
ConvertTimeTicksToNtp(testing_clock_->NowTicks(), &ntp_high, &ntp_low);
rtcp_packet.AddSrWithNtp(audio_config_.feedback_ssrc, ntp_high, ntp_low,
rtp_header_.webrtc.header.timestamp);
testing_clock_->Advance(base::TimeDelta::FromMilliseconds(20));
receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
// Make sure that we are not continuous and that the RTP timestamp represent a
// time in the future.
rtp_header_.is_key_frame = false;
rtp_header_.frame_id = 2;
rtp_header_.is_reference = true;
rtp_header_.reference_frame_id = 0;
rtp_header_.webrtc.header.timestamp = 960;
test_audio_encoder_callback_->SetExpectedResult(
2, testing_clock_->NowTicks() + base::TimeDelta::FromMilliseconds(100));
receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
rtp_header_);
receiver_->GetEncodedAudioFrame(frame_encoded_callback);
task_runner_->RunTasks();
// Frame 2 should not come out at this point in time.
EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
// Through on one more pending callback.
receiver_->GetEncodedAudioFrame(frame_encoded_callback);
testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
task_runner_->RunTasks();
EXPECT_EQ(2, test_audio_encoder_callback_->number_times_called());
test_audio_encoder_callback_->SetExpectedResult(3,
testing_clock_->NowTicks());
// Through on one more pending audio frame.
rtp_header_.frame_id = 3;
rtp_header_.is_reference = false;
rtp_header_.reference_frame_id = 0;
rtp_header_.webrtc.header.timestamp = 1280;
receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
rtp_header_);
receiver_->GetEncodedAudioFrame(frame_encoded_callback);
task_runner_->RunTasks();
EXPECT_EQ(3, test_audio_encoder_callback_->number_times_called());
}
// TODO(mikhal): Add encoded frames.
TEST_F(AudioReceiverTest, GetRawFrame) {}
} // namespace cast
} // namespace media
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