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Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.

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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-01-21 07:42:11 +00:00
parent a3c82bf667
commit a678a3baee
67 changed files with 367 additions and 544 deletions
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4
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
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@ -22,7 +22,7 @@ using namespace webrtc;
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest() {
RtpRtcpAPITest() : module(NULL), fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[1] = 2345;
test_id = 123;
@ -50,7 +50,7 @@ class RtpRtcpAPITest : public ::testing::Test {
WebRtc_UWord32 test_timestamp;
WebRtc_UWord16 test_sequence_number;
WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpAPITest, Basic) {

25
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
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@ -14,31 +14,6 @@
namespace webrtc {
class FakeRtpRtcpClock : public Clock {
public:
FakeRtpRtcpClock() {
time_in_ms_ = 123456;
}
// Return a timestamp in milliseconds relative to some arbitrary
// source; the source is fixed for this clock.
virtual WebRtc_Word64 TimeInMilliseconds() {
return time_in_ms_;
}
virtual int64_t TimeInMicroseconds() {
return time_in_ms_ * 1000;
}
// Retrieve an NTP absolute timestamp.
virtual void CurrentNtp(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
secs = time_in_ms_ / 1000;
frac = (time_in_ms_ % 1000) * 4294967;
}
void IncrementTime(WebRtc_UWord32 time_increment_ms) {
time_in_ms_ += time_increment_ms;
}
private:
WebRtc_Word64 time_in_ms_;
};
// This class sends all its packet straight to the provided RtpRtcp module.
// with optional packet loss.
class LoopBackTransport : public webrtc::Transport {

8
webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
View File

@ -119,7 +119,7 @@ class AudioFeedback : public RtpAudioFeedback {
class RtpRtcpAudioTest : public ::testing::Test {
protected:
RtpRtcpAudioTest() {
RtpRtcpAudioTest() : fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[2] = 2345;
test_id = 123;
@ -183,7 +183,7 @@ class RtpRtcpAudioTest : public ::testing::Test {
WebRtc_UWord32 test_timestamp;
WebRtc_UWord16 test_sequence_number;
WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpAudioTest, Basic) {
@ -317,7 +317,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
for (;timeStamp <= 250 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
timeStamp, -1, test, 4));
fake_clock.IncrementTime(20);
fake_clock.AdvanceTimeMilliseconds(20);
module1->Process();
}
EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
@ -325,7 +325,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
for (;timeStamp <= 740 * 160; timeStamp += 160) {
EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
timeStamp, -1, test, 4));
fake_clock.IncrementTime(20);
fake_clock.AdvanceTimeMilliseconds(20);
module1->Process();
}
}

15
webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
View File

@ -94,7 +94,12 @@ class NackLoopBackTransport : public webrtc::Transport {
class RtpRtcpNackTest : public ::testing::Test {
protected:
RtpRtcpNackTest() {}
RtpRtcpNackTest()
: video_module_(NULL),
transport_(NULL),
nack_receiver_(NULL),
payload_data_length(sizeof(payload_data)),
fake_clock(123456) {}
~RtpRtcpNackTest() {}
virtual void SetUp() {
@ -127,8 +132,6 @@ class RtpRtcpNackTest : public ::testing::Test {
EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
EXPECT_EQ(0, video_module_->RegisterReceivePayload(video_codec));
payload_data_length = sizeof(payload_data);
for (int n = 0; n < payload_data_length; n++) {
payload_data[n] = n % 10;
}
@ -145,7 +148,7 @@ class RtpRtcpNackTest : public ::testing::Test {
VerifyingNackReceiver* nack_receiver_;
WebRtc_UWord8 payload_data[65000];
int payload_data_length;
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpNackTest, RTCP) {
@ -185,7 +188,7 @@ TEST_F(RtpRtcpNackTest, RTCP) {
nack_list[n++] = (*it);
}
video_module_->SendNACK(nack_list, n);
fake_clock.IncrementTime(33);
fake_clock.AdvanceTimeMilliseconds(33);
video_module_->Process();
// Prepare next frame.
@ -242,7 +245,7 @@ TEST_F(RtpRtcpNackTest, RTX) {
nack_list[n++] = (*it);
}
video_module_->SendNACK(nack_list, n);
fake_clock.IncrementTime(33);
fake_clock.AdvanceTimeMilliseconds(33);
video_module_->Process();
// Prepare next frame.

18
webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
View File

@ -78,7 +78,7 @@ class RtcpCallback : public RtcpFeedback, public RtcpIntraFrameObserver {
class RtpRtcpRtcpTest : public ::testing::Test {
protected:
RtpRtcpRtcpTest() {
RtpRtcpRtcpTest() : fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[1] = 2345;
test_id = 123;
@ -171,7 +171,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
WebRtc_UWord32 test_timestamp;
WebRtc_UWord16 test_sequence_number;
WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
@ -194,9 +194,9 @@ TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[1], "jane@192.168.0.2"));
// send RTCP packet, triggered by timer
fake_clock.IncrementTime(7500);
fake_clock.AdvanceTimeMilliseconds(7500);
module1->Process();
fake_clock.IncrementTime(100);
fake_clock.AdvanceTimeMilliseconds(100);
module2->Process();
char cName[RTCP_CNAME_SIZE];
@ -251,9 +251,9 @@ TEST_F(RtpRtcpRtcpTest, RTCP) {
300));
// send RTCP packet, triggered by timer
fake_clock.IncrementTime(7500);
fake_clock.AdvanceTimeMilliseconds(7500);
module1->Process();
fake_clock.IncrementTime(100);
fake_clock.AdvanceTimeMilliseconds(100);
module2->Process();
WebRtc_UWord32 receivedNTPsecs = 0;
@ -318,7 +318,7 @@ TEST_F(RtpRtcpRtcpTest, RTCP) {
EXPECT_EQ(0, module1->SetSendingStatus(false));
// Send RTCP packet, triggered by timer.
fake_clock.IncrementTime(5000);
fake_clock.AdvanceTimeMilliseconds(5000);
module1->Process();
module2->Process();
}
@ -330,9 +330,9 @@ TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
EXPECT_EQ(0u, report_blocks.size());
// send RTCP packet, triggered by timer
fake_clock.IncrementTime(7500);
fake_clock.AdvanceTimeMilliseconds(7500);
module1->Process();
fake_clock.IncrementTime(100);
fake_clock.AdvanceTimeMilliseconds(100);
module2->Process();
EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));

7
webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
View File

@ -28,7 +28,8 @@ class RtpRtcpVideoTest : public ::testing::Test {
: test_id_(123),
test_ssrc_(3456),
test_timestamp_(4567),
test_sequence_number_(2345) {
test_sequence_number_(2345),
fake_clock(123456) {
}
~RtpRtcpVideoTest() {}
@ -124,7 +125,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
WebRtc_UWord16 test_sequence_number_;
WebRtc_UWord8 video_frame_[65000];
int payload_data_length_;
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
enum { kPayloadType = 100 };
};
@ -157,7 +158,7 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
EXPECT_EQ(packet_size - 12, receiver_->rtp_header().header.paddingLength);
}
timestamp += 3000;
fake_clock.IncrementTime(33);
fake_clock.AdvanceTimeMilliseconds(33);
}
}