Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
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
diff --git a/webrtc/modules/rtp_rtcp/source/bitrate.cc b/webrtc/modules/rtp_rtcp/source/bitrate.cc
index 8f1bac8..69601d2 100644
--- a/webrtc/modules/rtp_rtcp/source/bitrate.cc
+++ b/webrtc/modules/rtp_rtcp/source/bitrate.cc
@@ -15,7 +15,7 @@
namespace webrtc {
Bitrate::Bitrate(Clock* clock)
- : clock_(*clock),
+ : clock_(clock),
packet_rate_(0),
bitrate_(0),
bitrate_next_idx_(0),
@@ -41,7 +41,7 @@
}
WebRtc_UWord32 Bitrate::BitrateNow() const {
- WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ WebRtc_Word64 now = clock_->TimeInMilliseconds();
WebRtc_Word64 diff_ms = now - time_last_rate_update_;
if (diff_ms > 10000) { // 10 seconds.
@@ -59,7 +59,7 @@
void Bitrate::Process() {
// Triggered by timer.
- WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ WebRtc_Word64 now = clock_->TimeInMilliseconds();
WebRtc_Word64 diff_ms = now - time_last_rate_update_;
if (diff_ms < 100) {
diff --git a/webrtc/modules/rtp_rtcp/source/bitrate.h b/webrtc/modules/rtp_rtcp/source/bitrate.h
index d8c8aa8..9a0217f 100644
--- a/webrtc/modules/rtp_rtcp/source/bitrate.h
+++ b/webrtc/modules/rtp_rtcp/source/bitrate.h
@@ -42,7 +42,7 @@
WebRtc_UWord32 BitrateNow() const;
protected:
- Clock& clock_;
+ Clock* clock_;
private:
WebRtc_UWord32 packet_rate_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
index 03ca64e..0438510 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
@@ -99,7 +99,6 @@
delete rtcp_receiver_;
delete dummy_rtp_rtcp_impl_;
delete test_transport_;
- delete system_clock_;
}
TEST_F(RtcpFormatRembTest, TestBasicAPI) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 7bf20ee..e300439 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -34,7 +34,7 @@
ModuleRtpRtcpImpl* owner)
: TMMBRHelp(),
_id(id),
- _clock(*clock),
+ _clock(clock),
_method(kRtcpOff),
_lastReceived(0),
_rtpRtcp(*owner),
@@ -285,7 +285,7 @@
{
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- _lastReceived = _clock.TimeInMilliseconds();
+ _lastReceived = _clock->TimeInMilliseconds();
RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
while (pktType != RTCPUtility::kRtcpNotValidCode)
@@ -406,7 +406,7 @@
_remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount;
_remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount;
- _clock.CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
+ _clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
}
else
{
@@ -466,7 +466,7 @@
return;
}
- _lastReceivedRrMs = _clock.TimeInMilliseconds();
+ _lastReceivedRrMs = _clock->TimeInMilliseconds();
const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem;
reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC;
reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
@@ -496,7 +496,7 @@
WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
- _clock.CurrentNtp(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
+ _clock->CurrentNtp(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
// time when we received this in MS
WebRtc_UWord32 receiveTimeMS = ModuleRTPUtility::ConvertNTPTimeToMS(
@@ -641,7 +641,7 @@
void RTCPReceiver::UpdateReceiveInformation(
RTCPReceiveInformation& receiveInformation) {
// Update that this remote is alive
- receiveInformation.lastTimeReceived = _clock.TimeInMilliseconds();
+ receiveInformation.lastTimeReceived = _clock->TimeInMilliseconds();
}
bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
@@ -650,7 +650,7 @@
return false;
int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
- if (_clock.TimeInMilliseconds() > _lastReceivedRrMs + time_out_ms) {
+ if (_clock->TimeInMilliseconds() > _lastReceivedRrMs + time_out_ms) {
// Reset the timer to only trigger one log.
_lastReceivedRrMs = 0;
return true;
@@ -664,7 +664,7 @@
return false;
int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
- if (_clock.TimeInMilliseconds() > _lastIncreasedSequenceNumberMs +
+ if (_clock->TimeInMilliseconds() > _lastIncreasedSequenceNumberMs +
time_out_ms) {
// Reset the timer to only trigger one log.
_lastIncreasedSequenceNumberMs = 0;
@@ -677,7 +677,7 @@
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
bool updateBoundingSet = false;
- WebRtc_Word64 timeNow = _clock.TimeInMilliseconds();
+ WebRtc_Word64 timeNow = _clock->TimeInMilliseconds();
std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
_receivedInfoMap.begin();
@@ -962,7 +962,7 @@
rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0)
{
receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem,
- _clock.TimeInMilliseconds());
+ _clock->TimeInMilliseconds());
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr;
}
}
@@ -1150,7 +1150,7 @@
// check if we have reported this FIRSequenceNumber before
if (rtcpPacket.FIRItem.CommandSequenceNumber !=
receiveInfo->lastFIRSequenceNumber) {
- WebRtc_Word64 now = _clock.TimeInMilliseconds();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
// sanity; don't go crazy with the callbacks
if ((now - receiveInfo->lastFIRRequest) > RTCP_MIN_FRAME_LENGTH_MS) {
receiveInfo->lastFIRRequest = now;
@@ -1308,7 +1308,7 @@
if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr) &&
rtcpPacketInformation.reportBlock) {
- WebRtc_Word64 now = _clock.TimeInMilliseconds();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
_cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport(
rtcpPacketInformation.remoteSSRC,
rtcpPacketInformation.fractionLost,
@@ -1378,7 +1378,7 @@
for (WebRtc_UWord32 i = 0;
(num < size) && (i < receiveInfo->TmmbrSet.lengthOfSet()); i++) {
if (receiveInfo->GetTMMBRSet(i, num, candidateSet,
- _clock.TimeInMilliseconds()) == 0) {
+ _clock->TimeInMilliseconds()) == 0) {
num++;
}
}
@@ -1425,7 +1425,7 @@
return;
}
- WebRtc_Word64 now = _clock.TimeInMilliseconds();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
if(now - _lastReceived > _packetTimeOutMS)
{
packetTimeOut = true;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 01a036f..08ff37b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -198,7 +198,7 @@
private:
WebRtc_Word32 _id;
- Clock& _clock;
+ Clock* _clock;
RTCPMethod _method;
WebRtc_Word64 _lastReceived;
ModuleRtpRtcpImpl& _rtpRtcp;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index b14db10..783b13a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -168,30 +168,29 @@
protected:
RtcpReceiverTest()
: over_use_detector_options_(),
+ system_clock_(1335900000),
remote_bitrate_observer_(),
remote_bitrate_estimator_(
RemoteBitrateEstimator::Create(
&remote_bitrate_observer_,
over_use_detector_options_,
RemoteBitrateEstimator::kMultiStreamEstimation)) {
- system_clock_ = new SimulatedClock(1335900000);
test_transport_ = new TestTransport();
RtpRtcp::Configuration configuration;
configuration.id = 0;
configuration.audio = false;
- configuration.clock = system_clock_;
+ configuration.clock = &system_clock_;
configuration.outgoing_transport = test_transport_;
configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
- rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
+ rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, rtp_rtcp_impl_);
test_transport_->SetRTCPReceiver(rtcp_receiver_);
}
~RtcpReceiverTest() {
delete rtcp_receiver_;
delete rtp_rtcp_impl_;
delete test_transport_;
- delete system_clock_;
}
// Injects an RTCP packet into the receiver.
@@ -210,7 +209,7 @@
}
OverUseDetectorOptions over_use_detector_options_;
- SimulatedClock* system_clock_;
+ SimulatedClock system_clock_;
ModuleRtpRtcpImpl* rtp_rtcp_impl_;
RTCPReceiver* rtcp_receiver_;
TestTransport* test_transport_;
@@ -246,7 +245,7 @@
rtcp_receiver_->SetSSRC(kSourceSsrc);
uint32_t sequence_number = 1234;
- system_clock_->AdvanceTimeMs(3 * kRtcpIntervalMs);
+ system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
// No RR received, shouldn't trigger a timeout.
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
@@ -256,7 +255,7 @@
PacketBuilder p1;
p1.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p1.packet(), p1.length()));
- system_clock_->AdvanceTimeMs(3 * kRtcpIntervalMs - 1);
+ system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
@@ -265,12 +264,12 @@
PacketBuilder p2;
p2.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p2.packet(), p2.length()));
- system_clock_->AdvanceTimeMs(2);
+ system_clock_.AdvanceTimeMilliseconds(2);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Advance clock enough to trigger an RR timeout too.
- system_clock_->AdvanceTimeMs(3 * kRtcpIntervalMs);
+ system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
// We should only get one timeout even though we still haven't received a new
@@ -287,14 +286,14 @@
EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
// Verify we can get a timeout again once we've received new RR.
- system_clock_->AdvanceTimeMs(2 * kRtcpIntervalMs);
+ system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
PacketBuilder p4;
p4.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
EXPECT_EQ(0, InjectRtcpPacket(p4.packet(), p4.length()));
- system_clock_->AdvanceTimeMs(kRtcpIntervalMs + 1);
+ system_clock_.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
- system_clock_->AdvanceTimeMs(2 * kRtcpIntervalMs);
+ system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
}
@@ -386,7 +385,8 @@
p.AddTmmbrBandwidth(30000, 0, 0); // 30 Kbits/sec bandwidth, no overhead.
EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
- system_clock_->AdvanceTimeMs(5000); // 5 seconds between each packet.
+ // 5 seconds between each packet.
+ system_clock_.AdvanceTimeMilliseconds(5000);
}
// It is now starttime+15.
EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
@@ -396,7 +396,7 @@
EXPECT_LT(0U, candidate_set.Tmmbr(0));
// We expect the timeout to be 25 seconds. Advance the clock by 12
// seconds, timing out the first packet.
- system_clock_->AdvanceTimeMs(12000);
+ system_clock_.AdvanceTimeMilliseconds(12000);
// Odd behaviour: Just counting them does not trigger the timeout.
EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
// Odd behaviour: There's only one left after timeout, not 2.
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 8611769..d5dd4a1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -29,7 +29,7 @@
ModuleRtpRtcpImpl* owner) :
_id(id),
_audio(audio),
- _clock(*clock),
+ _clock(clock),
_method(kRtcpOff),
_rtpRtcp(*owner),
_criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
@@ -183,11 +183,11 @@
{
if(_audio)
{
- _nextTimeToSendRTCP = _clock.TimeInMilliseconds() +
+ _nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
(RTCP_INTERVAL_AUDIO_MS/2);
} else
{
- _nextTimeToSendRTCP = _clock.TimeInMilliseconds() +
+ _nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
(RTCP_INTERVAL_VIDEO_MS/2);
}
}
@@ -304,7 +304,7 @@
last_rtp_timestamp_ = rtp_timestamp;
if (capture_time_ms < 0) {
// We don't currently get a capture time from VoiceEngine.
- last_frame_capture_time_ms_ = _clock.TimeInMilliseconds();
+ last_frame_capture_time_ms_ = _clock->TimeInMilliseconds();
} else {
last_frame_capture_time_ms_ = capture_time_ms;
}
@@ -320,7 +320,7 @@
// not first SetSSRC, probably due to a collision
// schedule a new RTCP report
// make sure that we send a RTP packet
- _nextTimeToSendRTCP = _clock.TimeInMilliseconds() + 100;
+ _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100;
}
_SSRC = ssrc;
}
@@ -451,7 +451,7 @@
a value of the RTCP bandwidth below the intended average
*/
- WebRtc_Word64 now = _clock.TimeInMilliseconds();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
CriticalSectionScoped lock(_criticalSectionRTCPSender);
@@ -590,7 +590,7 @@
// timestamp as the last frame's timestamp + the time since the last frame
// was captured.
RTPtime = start_timestamp_ + last_rtp_timestamp_ + (
- _clock.TimeInMilliseconds() - last_frame_capture_time_ms_) *
+ _clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
(freqHz / 1000);
// Add sender data
@@ -1580,7 +1580,7 @@
remoteSR);
// get our NTP as late as possible to avoid a race
- _clock.CurrentNtp(NTPsec, NTPfrac);
+ _clock->CurrentNtp(NTPsec, NTPfrac);
// Delay since last received report
WebRtc_UWord32 delaySinceLastReceivedSR = 0;
@@ -1602,7 +1602,7 @@
} else
{
// we need to send our NTP even if we dont have received any reports
- _clock.CurrentNtp(NTPsec, NTPfrac);
+ _clock->CurrentNtp(NTPsec, NTPfrac);
}
}
@@ -1697,7 +1697,7 @@
}
timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000);
}
- _nextTimeToSendRTCP = _clock.TimeInMilliseconds() + timeToNext;
+ _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + timeToNext;
}
// if the data does not fitt in the packet we fill it as much as possible
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index f76b36c..2844975 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -186,7 +186,7 @@
private:
WebRtc_Word32 _id;
const bool _audio;
- Clock& _clock;
+ Clock* _clock;
RTCPMethod _method;
ModuleRtpRtcpImpl& _rtpRtcp;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
index 7cbba88..7bd2931 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
@@ -129,7 +129,6 @@
delete rtcp_receiver_;
delete rtp_rtcp_impl_;
delete test_transport_;
- delete system_clock_;
}
// Helper function: Incoming RTCP has a specific packet type.
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index e6577a6..b5de84a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -20,7 +20,7 @@
namespace webrtc {
RTPPacketHistory::RTPPacketHistory(Clock* clock)
- : clock_(*clock),
+ : clock_(clock),
critsect_(CriticalSectionWrapper::CreateCriticalSection()),
store_(false),
prev_index_(0),
@@ -140,7 +140,7 @@
stored_seq_nums_[prev_index_] = seq_num;
stored_lengths_[prev_index_] = packet_length;
stored_times_[prev_index_] =
- (capture_time_ms > 0) ? capture_time_ms : clock_.TimeInMilliseconds();
+ (capture_time_ms > 0) ? capture_time_ms : clock_->TimeInMilliseconds();
stored_resend_times_[prev_index_] = 0; // packet not resent
stored_types_[prev_index_] = type;
@@ -244,7 +244,7 @@
}
// Verify elapsed time since last retrieve.
- int64_t now = clock_.TimeInMilliseconds();
+ int64_t now = clock_->TimeInMilliseconds();
if (min_elapsed_time_ms > 0 &&
((now - stored_resend_times_.at(index)) < min_elapsed_time_ms)) {
WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
@@ -276,7 +276,7 @@
"Failed to update resend time, seq num: %u.", sequence_number);
return;
}
- stored_resend_times_[index] = clock_.TimeInMilliseconds();
+ stored_resend_times_[index] = clock_->TimeInMilliseconds();
}
// private, lock should already be taken
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index 59e27fc..cc0a0c8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -77,7 +77,7 @@
bool FindSeqNum(uint16_t sequence_number, int32_t* index) const;
private:
- Clock& clock_;
+ Clock* clock_;
CriticalSectionWrapper* critsect_;
bool store_;
uint32_t prev_index_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index d90fff9..095606d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -12,49 +12,24 @@
#include <gtest/gtest.h>
+#include "clock.h"
#include "rtp_packet_history.h"
#include "rtp_rtcp_defines.h"
#include "typedefs.h"
namespace webrtc {
-class FakeClock : public Clock {
- public:
- FakeClock() {
- 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 WebRtc_Word64 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_;
-};
-
class RtpPacketHistoryTest : public ::testing::Test {
protected:
RtpPacketHistoryTest()
- : hist_(new RTPPacketHistory(&fake_clock_)) {
+ : fake_clock_(123456),
+ hist_(new RTPPacketHistory(&fake_clock_)) {
}
~RtpPacketHistoryTest() {
delete hist_;
}
- FakeClock fake_clock_;
+ SimulatedClock fake_clock_;
RTPPacketHistory* hist_;
enum {kPayload = 127};
enum {kSsrc = 12345678};
@@ -224,7 +199,7 @@
TEST_F(RtpPacketHistoryTest, NoCaptureTime) {
hist_->SetStorePacketsStatus(true, 10);
uint16_t len = 0;
- fake_clock_.IncrementTime(1);
+ fake_clock_.AdvanceTimeMilliseconds(1);
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
@@ -270,7 +245,7 @@
capture_time_ms, kAllowRetransmission));
hist_->UpdateResendTime(kSeqNum);
- fake_clock_.IncrementTime(100);
+ fake_clock_.AdvanceTimeMilliseconds(100);
// Time has elapsed.
len = kMaxPacketLength;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
index 8132af1..bff6def 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -171,7 +171,7 @@
return;
}
- WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ WebRtc_Word64 now = clock_->TimeInMilliseconds();
if (now - last_receive_time_ > packet_timeout_ms_) {
packet_time_out = true;
@@ -416,7 +416,7 @@
WebRtc_Word32 ret_val = rtp_media_receiver_->ParseRtpPacket(
rtp_header, specific_payload, is_red, packet, packet_length,
- clock_.TimeInMilliseconds(), is_first_packet);
+ clock_->TimeInMilliseconds(), is_first_packet);
if (ret_val < 0) {
return ret_val;
@@ -434,13 +434,13 @@
// Need to be updated after RetransmitOfOldPacket and
// RetransmitOfOldPacketUpdateStatistics.
- last_receive_time_ = clock_.TimeInMilliseconds();
+ last_receive_time_ = clock_->TimeInMilliseconds();
last_received_payload_length_ = payload_data_length;
if (!old_packet) {
if (last_received_timestamp_ != rtp_header->header.timestamp) {
last_received_timestamp_ = rtp_header->header.timestamp;
- last_received_frame_time_ms_ = clock_.TimeInMilliseconds();
+ last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
}
last_received_sequence_number_ = rtp_header->header.sequenceNumber;
last_received_transmission_time_offset_ =
@@ -466,14 +466,14 @@
received_seq_max_ = rtp_header->header.sequenceNumber;
received_inorder_packet_count_ = 1;
local_time_last_received_timestamp_ =
- GetCurrentRTP(&clock_, frequency_hz); // Time in samples.
+ GetCurrentRTP(clock_, frequency_hz); // Time in samples.
return;
}
// Count only the new packets received.
if (InOrderPacket(rtp_header->header.sequenceNumber)) {
const WebRtc_UWord32 RTPtime =
- GetCurrentRTP(&clock_, frequency_hz); // Time in samples.
+ GetCurrentRTP(clock_, frequency_hz); // Time in samples.
received_inorder_packet_count_++;
// Wrong if we use RetransmitOfOldPacket.
@@ -548,7 +548,8 @@
}
WebRtc_UWord32 frequency_khz = rtp_media_receiver_->GetFrequencyHz() / 1000;
- WebRtc_Word64 time_diff_ms = clock_.TimeInMilliseconds() - last_receive_time_;
+ WebRtc_Word64 time_diff_ms = clock_->TimeInMilliseconds() -
+ last_receive_time_;
// Diff in time stamp since last received in order.
WebRtc_Word32 rtp_time_stamp_diff_ms =
@@ -642,7 +643,7 @@
return -1;
}
// Time in samples.
- WebRtc_UWord32 diff = GetCurrentRTP(&clock_, frequency_hz) -
+ WebRtc_UWord32 diff = GetCurrentRTP(clock_, frequency_hz) -
local_time_last_received_timestamp_;
timestamp = last_received_timestamp_ + diff;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 634010e..40f2432 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -83,7 +83,6 @@
configuration_copy.clock = Clock::GetRealTimeClock();
ModuleRtpRtcpImpl* rtp_rtcp_instance =
new ModuleRtpRtcpImpl(configuration_copy);
- rtp_rtcp_instance->OwnsClock();
return rtp_rtcp_instance;
}
}
@@ -102,8 +101,7 @@
rtcp_sender_(configuration.id, configuration.audio, configuration.clock,
this),
rtcp_receiver_(configuration.id, configuration.clock, this),
- owns_clock_(false),
- clock_(*configuration.clock),
+ clock_(configuration.clock),
id_(configuration.id),
audio_(configuration.audio),
collision_detected_(false),
@@ -168,9 +166,6 @@
plot1_ = NULL;
}
#endif
- if (owns_clock_) {
- delete &clock_;
- }
}
void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
@@ -217,13 +212,13 @@
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() {
- const WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ const WebRtc_Word64 now = clock_->TimeInMilliseconds();
return kRtpRtcpMaxIdleTimeProcess - (now - last_process_time_);
}
// Process any pending tasks such as timeouts (non time critical events).
WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
- const WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ const WebRtc_Word64 now = clock_->TimeInMilliseconds();
last_process_time_ = now;
if (now >=
@@ -305,7 +300,7 @@
void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
if (dead_or_alive_active_) {
- const WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ const WebRtc_Word64 now = clock_->TimeInMilliseconds();
if (now > dead_or_alive_timeout_ms_ + dead_or_alive_last_timer_) {
// RTCP is alive if we have received a report the last 12 seconds.
dead_or_alive_last_timer_ += dead_or_alive_timeout_ms_;
@@ -340,7 +335,7 @@
dead_or_alive_active_ = enable;
dead_or_alive_timeout_ms_ = sample_time_seconds * 1000;
// Trigger the first after one period.
- dead_or_alive_last_timer_ = clock_.TimeInMilliseconds();
+ dead_or_alive_last_timer_ = clock_->TimeInMilliseconds();
return 0;
}
@@ -1304,7 +1299,7 @@
#ifdef MATLAB
if (plot1_ == NULL) {
plot1_ = eng.NewPlot(new MatlabPlot());
- plot1_->AddTimeLine(30, "b", "lost", clock_.GetTimeInMS());
+ plot1_->AddTimeLine(30, "b", "lost", clock_->TimeInMilliseconds());
}
plot1_->Append("lost", missing);
plot1_->Plot();
@@ -1520,7 +1515,7 @@
if (wait_time == 5) {
wait_time = 100; // During startup we don't have an RTT.
}
- const WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ const WebRtc_Word64 now = clock_->TimeInMilliseconds();
const WebRtc_Word64 time_limit = now - wait_time;
WebRtc_UWord16 nackLength = size;
WebRtc_UWord16 start_id = 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index cd675bf..2daba1d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -454,12 +454,6 @@
void OnRequestSendReport();
- // Following function is only called when constructing the object so no
- // need to worry about data race.
- void OwnsClock() {
- owns_clock_ = true;
- }
-
protected:
void RegisterChildModule(RtpRtcp* module);
@@ -483,8 +477,7 @@
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
- bool owns_clock_;
- Clock& clock_;
+ Clock* clock_;
private:
int64_t RtcpReportInterval();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index dd80508..bb778f7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -75,14 +75,14 @@
memset(_nackByteCount, 0, sizeof(_nackByteCount));
memset(_CSRC, 0, sizeof(_CSRC));
// We need to seed the random generator.
- srand( (WebRtc_UWord32)clock_.TimeInMilliseconds() );
+ srand(static_cast<WebRtc_UWord32>(clock_->TimeInMilliseconds()));
_ssrc = _ssrcDB.CreateSSRC(); // Can't be 0.
if (audio) {
- _audio = new RTPSenderAudio(id, &clock_, this);
+ _audio = new RTPSenderAudio(id, clock_, this);
_audio->RegisterAudioCallback(audio_feedback);
} else {
- _video = new RTPSenderVideo(id, &clock_, this);
+ _video = new RTPSenderVideo(id, clock_, this);
}
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
@@ -576,7 +576,7 @@
void RTPSender::OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
const WebRtc_UWord16* nackSequenceNumbers,
const WebRtc_UWord16 avgRTT) {
- const WebRtc_Word64 now = clock_.TimeInMilliseconds();
+ const WebRtc_Word64 now = clock_->TimeInMilliseconds();
WebRtc_UWord32 bytesReSent = 0;
// Enough bandwidth to send NACK?
@@ -700,7 +700,7 @@
WebRtcRTPHeader rtp_header;
rtpParser.Parse(rtp_header);
- int64_t diff_ms = clock_.TimeInMilliseconds() - capture_time_ms;
+ int64_t diff_ms = clock_->TimeInMilliseconds() - capture_time_ms;
if (UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms)) {
// Update stored packet in case of receiving a re-transmission request.
_packetHistory->ReplaceRTPHeader(data_buffer,
@@ -738,7 +738,7 @@
// TODO(holmer): This should be changed all over Video Engine so that negative
// time is consider invalid, while 0 is considered a valid time.
if (capture_time_ms > 0) {
- int64_t time_now = clock_.TimeInMilliseconds();
+ int64_t time_now = clock_->TimeInMilliseconds();
UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
rtp_header, time_now - capture_time_ms);
}
@@ -1027,7 +1027,7 @@
} else {
frequency_hz = kDefaultVideoFrequency;
}
- WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(&clock_,
+ WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(clock_,
frequency_hz);
// will be ignored if it's already configured via API
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 2a28750..34f650f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -17,7 +17,7 @@
RTPSenderAudio::RTPSenderAudio(const WebRtc_Word32 id, Clock* clock,
RTPSenderInterface* rtpSender) :
_id(id),
- _clock(*clock),
+ _clock(clock),
_rtpSender(rtpSender),
_audioFeedbackCritsect(CriticalSectionWrapper::CreateCriticalSection()),
_audioFeedback(NULL),
@@ -217,7 +217,7 @@
telephoneEvent = _dtmfKey;
return true;
}
- WebRtc_Word64 delaySinceLastDTMF = _clock.TimeInMilliseconds() -
+ WebRtc_Word64 delaySinceLastDTMF = _clock->TimeInMilliseconds() -
_dtmfTimeLastSent;
if(delaySinceLastDTMF < 100)
{
@@ -246,7 +246,7 @@
if (!_dtmfEventIsOn && PendingDTMF()) {
CriticalSectionScoped cs(_sendAudioCritsect);
- WebRtc_Word64 delaySinceLastDTMF = _clock.TimeInMilliseconds() -
+ WebRtc_Word64 delaySinceLastDTMF = _clock->TimeInMilliseconds() -
_dtmfTimeLastSent;
if (delaySinceLastDTMF > 100) {
@@ -297,7 +297,7 @@
} else {
ended = true;
_dtmfEventIsOn = false;
- _dtmfTimeLastSent = _clock.TimeInMilliseconds();
+ _dtmfTimeLastSent = _clock->TimeInMilliseconds();
}
// don't hold the critsect while calling SendTelephoneEventPacket
_sendAudioCritsect->Leave();
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 2622812..535c0d0 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -89,7 +89,7 @@
private:
WebRtc_Word32 _id;
- Clock& _clock;
+ Clock* _clock;
RTPSenderInterface* _rtpSender;
CriticalSectionWrapper* _audioFeedbackCritsect;
RtpAudioFeedback* _audioFeedback;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 055be57..cc00d40 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -33,33 +33,6 @@
const int kMaxPacketLength = 1500;
} // namespace
-class FakeClockTest : public Clock {
- public:
- FakeClockTest() {
- 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 WebRtc_Word64 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_;
-};
-
class LoopbackTransportTest : public webrtc::Transport {
public:
LoopbackTransportTest()
@@ -83,14 +56,14 @@
class RtpSenderTest : public ::testing::Test {
protected:
RtpSenderTest()
- : fake_clock_(),
+ : fake_clock_(123456),
rtp_sender_(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
NULL)),
kMarkerBit(true),
kType(kRtpExtensionTransmissionTimeOffset) {
rtp_sender_->SetSequenceNumber(kSeqNum);
}
- FakeClockTest fake_clock_;
+ SimulatedClock fake_clock_;
scoped_ptr<RTPSender> rtp_sender_;
LoopbackTransportTest transport_;
const bool kMarkerBit;
@@ -232,7 +205,7 @@
kAllowRetransmission));
EXPECT_EQ(0, transport_.packets_sent_);
const int kStoredTimeInMs = 100;
- fake_clock_.IncrementTime(kStoredTimeInMs);
+ fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
// Process send bucket. Packet should now be sent.
EXPECT_EQ(1, transport_.packets_sent_);
EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index 3503af2..4c9f36e 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -22,7 +22,7 @@
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 @@
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) {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index d4bc20c..b3a30a2 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -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 {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index ce899d7..2c0c2c2 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -119,7 +119,7 @@
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 @@
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 @@
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 @@
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();
}
}
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
index 6fa81e71..b72d0b3 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
@@ -94,7 +94,12 @@
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 @@
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 @@
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 @@
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 @@
nack_list[n++] = (*it);
}
video_module_->SendNACK(nack_list, n);
- fake_clock.IncrementTime(33);
+ fake_clock.AdvanceTimeMilliseconds(33);
video_module_->Process();
// Prepare next frame.
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index d25d4cb..d2ea532 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -78,7 +78,7 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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));
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 7a9d3db..b87904a 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -28,7 +28,8 @@
: 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 @@
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 @@
EXPECT_EQ(packet_size - 12, receiver_->rtp_header().header.paddingLength);
}
timestamp += 3000;
- fake_clock.IncrementTime(33);
+ fake_clock.AdvanceTimeMilliseconds(33);
}
}
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
index ad7edcd..b5adf35 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding.h
@@ -19,7 +19,7 @@
namespace webrtc
{
-class TickTimeBase;
+class Clock;
class VideoEncoder;
class VideoDecoder;
struct CodecSpecificInfo;
@@ -49,7 +49,7 @@
static VideoCodingModule* Create(const WebRtc_Word32 id);
static VideoCodingModule* Create(const WebRtc_Word32 id,
- TickTimeBase* clock);
+ Clock* clock);
static void Destroy(VideoCodingModule* module);
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.cc b/webrtc/modules/video_coding/main/source/generic_decoder.cc
index 3139ba8..a9bb763 100644
--- a/webrtc/modules/video_coding/main/source/generic_decoder.cc
+++ b/webrtc/modules/video_coding/main/source/generic_decoder.cc
@@ -12,12 +12,12 @@
#include "trace.h"
#include "generic_decoder.h"
#include "internal_defines.h"
-#include "tick_time_base.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
- TickTimeBase* clock)
+ Clock* clock)
:
_critSect(CriticalSectionWrapper::CreateCriticalSection()),
_clock(clock),
@@ -56,7 +56,7 @@
_timing.StopDecodeTimer(
decodedImage.timestamp(),
frameInfo->decodeStartTimeMs,
- _clock->MillisecondTimestamp());
+ _clock->TimeInMilliseconds());
if (_receiveCallback != NULL)
{
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.h b/webrtc/modules/video_coding/main/source/generic_decoder.h
index 016142a..084bd76 100644
--- a/webrtc/modules/video_coding/main/source/generic_decoder.h
+++ b/webrtc/modules/video_coding/main/source/generic_decoder.h
@@ -34,7 +34,7 @@
class VCMDecodedFrameCallback : public DecodedImageCallback
{
public:
- VCMDecodedFrameCallback(VCMTiming& timing, TickTimeBase* clock);
+ VCMDecodedFrameCallback(VCMTiming& timing, Clock* clock);
virtual ~VCMDecodedFrameCallback();
void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
@@ -49,7 +49,7 @@
private:
CriticalSectionWrapper* _critSect;
- TickTimeBase* _clock;
+ Clock* _clock;
I420VideoFrame _frame;
VCMReceiveCallback* _receiveCallback;
VCMTiming& _timing;
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index c80b84a..8121497 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -12,16 +12,16 @@
#include <algorithm>
#include <cassert>
-#include "modules/video_coding/main/source/event.h"
-#include "modules/video_coding/main/source/frame_buffer.h"
-#include "modules/video_coding/main/source/inter_frame_delay.h"
-#include "modules/video_coding/main/source/internal_defines.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/video_coding/main/source/jitter_estimator.h"
-#include "modules/video_coding/main/source/packet.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/trace.h"
+#include "webrtc/modules/video_coding/main/source/event.h"
+#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
+#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
+#include "webrtc/modules/video_coding/main/source/internal_defines.h"
+#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
+#include "webrtc/modules/video_coding/main/source/packet.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
@@ -61,7 +61,7 @@
}
};
-VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
+VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
int vcm_id,
int receiver_id,
bool master)
@@ -90,7 +90,7 @@
num_consecutive_old_packets_(0),
num_discarded_packets_(0),
jitter_estimate_(vcm_id, receiver_id),
- inter_frame_delay_(clock_->MillisecondTimestamp()),
+ inter_frame_delay_(clock_->TimeInMilliseconds()),
rtt_ms_(kDefaultRtt),
nack_mode_(kNoNack),
low_rtt_nack_threshold_ms_(-1),
@@ -177,7 +177,7 @@
incoming_frame_rate_ = 0;
incoming_bit_count_ = 0;
incoming_bit_rate_ = 0;
- time_last_incoming_frame_count_ = clock_->MillisecondTimestamp();
+ time_last_incoming_frame_count_ = clock_->TimeInMilliseconds();
memset(receive_statistics_, 0, sizeof(receive_statistics_));
num_consecutive_old_frames_ = 0;
@@ -241,7 +241,7 @@
num_consecutive_old_packets_ = 0;
// Also reset the jitter and delay estimates
jitter_estimate_.Reset();
- inter_frame_delay_.Reset(clock_->MillisecondTimestamp());
+ inter_frame_delay_.Reset(clock_->TimeInMilliseconds());
waiting_for_completion_.frame_size = 0;
waiting_for_completion_.timestamp = 0;
waiting_for_completion_.latest_packet_time = -1;
@@ -278,7 +278,7 @@
assert(framerate);
assert(bitrate);
CriticalSectionScoped cs(crit_sect_);
- const int64_t now = clock_->MillisecondTimestamp();
+ const int64_t now = clock_->TimeInMilliseconds();
int64_t diff = now - time_last_incoming_frame_count_;
if (diff < 1000 && incoming_frame_rate_ > 0 && incoming_bit_rate_ > 0) {
// Make sure we report something even though less than
@@ -323,7 +323,7 @@
} else {
// No frames since last call
- time_last_incoming_frame_count_ = clock_->MillisecondTimestamp();
+ time_last_incoming_frame_count_ = clock_->TimeInMilliseconds();
*framerate = 0;
bitrate = 0;
incoming_bit_rate_ = 0;
@@ -437,8 +437,8 @@
crit_sect_->Leave();
return NULL;
}
- const int64_t end_wait_time_ms = clock_->MillisecondTimestamp()
- + max_wait_time_ms;
+ const int64_t end_wait_time_ms = clock_->TimeInMilliseconds() +
+ max_wait_time_ms;
int64_t wait_time_ms = max_wait_time_ms;
while (wait_time_ms > 0) {
crit_sect_->Leave();
@@ -457,8 +457,7 @@
CleanUpOldFrames();
it = FindOldestCompleteContinuousFrame(false);
if (it == frame_list_.end()) {
- wait_time_ms = end_wait_time_ms -
- clock_->MillisecondTimestamp();
+ wait_time_ms = end_wait_time_ms - clock_->TimeInMilliseconds();
} else {
break;
}
@@ -663,7 +662,7 @@
const VCMPacket& packet) {
assert(encoded_frame);
CriticalSectionScoped cs(crit_sect_);
- int64_t now_ms = clock_->MillisecondTimestamp();
+ int64_t now_ms = clock_->TimeInMilliseconds();
VCMFrameBufferEnum buffer_return = kSizeError;
VCMFrameBufferEnum ret = kSizeError;
VCMFrameBuffer* frame = static_cast<VCMFrameBuffer*>(encoded_frame);
@@ -673,7 +672,7 @@
if (first_packet_) {
// Now it's time to start estimating jitter
// reset the delay estimate.
- inter_frame_delay_.Reset(clock_->MillisecondTimestamp());
+ inter_frame_delay_.Reset(clock_->TimeInMilliseconds());
first_packet_ = false;
}
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index 5b30fb8..d0a2569 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -35,7 +35,7 @@
typedef std::list<VCMFrameBuffer*> FrameList;
// forward declarations
-class TickTimeBase;
+class Clock;
class VCMFrameBuffer;
class VCMPacket;
class VCMEncodedFrame;
@@ -49,7 +49,7 @@
class VCMJitterBuffer {
public:
- VCMJitterBuffer(TickTimeBase* clock, int vcm_id = -1, int receiver_id = -1,
+ VCMJitterBuffer(Clock* clock, int vcm_id = -1, int receiver_id = -1,
bool master = true);
virtual ~VCMJitterBuffer();
@@ -206,7 +206,7 @@
int vcm_id_;
int receiver_id_;
- TickTimeBase* clock_;
+ Clock* clock_;
// If we are running (have started) or not.
bool running_;
CriticalSectionWrapper* crit_sect_;
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
index 88ca597..af79442 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -15,8 +15,8 @@
#include "gtest/gtest.h"
#include "modules/video_coding/main/source/jitter_buffer.h"
#include "modules/video_coding/main/source/media_opt_util.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
#include "modules/video_coding/main/source/packet.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
@@ -145,10 +145,10 @@
enum { kDefaultFramePeriodMs = 1000 / kDefaultFrameRate };
virtual void SetUp() {
- clock_ = new FakeTickTime(0);
- jitter_buffer_ = new VCMJitterBuffer(clock_);
+ clock_.reset(new SimulatedClock(0));
+ jitter_buffer_ = new VCMJitterBuffer(clock_.get());
stream_generator = new StreamGenerator(0, 0,
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
jitter_buffer_->Start();
memset(data_buffer_, 0, kDataBufferSize);
}
@@ -157,7 +157,6 @@
jitter_buffer_->Stop();
delete stream_generator;
delete jitter_buffer_;
- delete clock_;
}
VCMFrameBufferEnum InsertPacketAndPop(int index) {
@@ -190,9 +189,9 @@
stream_generator->GenerateFrame(frame_type,
(frame_type != kFrameEmpty) ? 1 : 0,
(frame_type == kFrameEmpty) ? 1 : 0,
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
}
void InsertFrames(int num_frames, FrameType frame_type) {
@@ -203,8 +202,8 @@
void DropFrame(int num_packets) {
stream_generator->GenerateFrame(kVideoFrameDelta, num_packets, 0,
- clock_->MillisecondTimestamp());
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ clock_->TimeInMilliseconds());
+ clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
}
bool DecodeCompleteFrame() {
@@ -223,7 +222,7 @@
VCMJitterBuffer* jitter_buffer_;
StreamGenerator* stream_generator;
- FakeTickTime* clock_;
+ scoped_ptr<SimulatedClock> clock_;
uint8_t data_buffer_[kDataBufferSize];
};
@@ -258,7 +257,7 @@
TEST_F(TestRunningJitterBuffer, TestEmptyPackets) {
// Make sure a frame can get complete even though empty packets are missing.
stream_generator->GenerateFrame(kVideoFrameKey, 3, 3,
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
EXPECT_EQ(kFirstPacket, InsertPacketAndPop(4));
EXPECT_EQ(kIncomplete, InsertPacketAndPop(4));
EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
@@ -319,8 +318,8 @@
// | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
// ----------------------------------------------------------------
stream_generator->GenerateFrame(kVideoFrameKey, 100, 0,
- clock_->MillisecondTimestamp());
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
+ clock_->TimeInMilliseconds());
+ clock_->AdvanceTimeMilliseconds(kDefaultFramePeriodMs);
EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
// Verify that the frame is incomplete.
EXPECT_FALSE(DecodeCompleteFrame());
@@ -348,11 +347,11 @@
// ------- ------------------------------------------------------------
// | 65532 | | 65533 | 65534 | 65535 | x | 1 | .. | 9 | x | 11 |.....| 96 |
// ------- ------------------------------------------------------------
- stream_generator->Init(65532, 0, clock_->MillisecondTimestamp());
+ stream_generator->Init(65532, 0, clock_->TimeInMilliseconds());
InsertFrame(kVideoFrameKey);
EXPECT_TRUE(DecodeCompleteFrame());
stream_generator->GenerateFrame(kVideoFrameDelta, 100, 0,
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
while (stream_generator->PacketsRemaining() > 1) {
if (stream_generator->NextSequenceNumber() % 10 != 0)
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.cc b/webrtc/modules/video_coding/main/source/media_optimization.cc
index 8dd47fc..dd428b4 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.cc
+++ b/webrtc/modules/video_coding/main/source/media_optimization.cc
@@ -13,12 +13,12 @@
#include "content_metrics_processing.h"
#include "frame_dropper.h"
#include "qm_select.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id,
- TickTimeBase* clock):
+ Clock* clock):
_id(id),
_clock(clock),
_maxBitRate(0),
@@ -46,7 +46,7 @@
memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
_frameDropper = new VCMFrameDropper(_id);
- _lossProtLogic = new VCMLossProtectionLogic(_clock->MillisecondTimestamp());
+ _lossProtLogic = new VCMLossProtectionLogic(_clock->TimeInMilliseconds());
_content = new VCMContentMetricsProcessing();
_qmResolution = new VCMQmResolution();
}
@@ -66,12 +66,12 @@
memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
_incomingFrameRate = 0.0;
_frameDropper->Reset();
- _lossProtLogic->Reset(_clock->MillisecondTimestamp());
+ _lossProtLogic->Reset(_clock->TimeInMilliseconds());
_frameDropper->SetRates(0, 0);
_content->Reset();
_qmResolution->Reset();
_lossProtLogic->UpdateFrameRate(_incomingFrameRate);
- _lossProtLogic->Reset(_clock->MillisecondTimestamp());
+ _lossProtLogic->Reset(_clock->TimeInMilliseconds());
_sendStatisticsZeroEncode = 0;
_targetBitRate = 0;
_codecWidth = 0;
@@ -122,7 +122,7 @@
// Use max window filter for now.
FilterPacketLossMode filter_mode = kMaxFilter;
WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss(
- _clock->MillisecondTimestamp(), filter_mode, fractionLost);
+ _clock->TimeInMilliseconds(), filter_mode, fractionLost);
// For now use the filtered loss for computing the robustness settings
_lossProtLogic->UpdateFilteredLossPr(packetLossEnc);
@@ -274,7 +274,7 @@
// has changed. If native dimension values have changed, then either user
// initiated change, or QM initiated change. Will be able to determine only
// after the processing of the first frame.
- _lastChangeTime = _clock->MillisecondTimestamp();
+ _lastChangeTime = _clock->TimeInMilliseconds();
_content->Reset();
_content->UpdateFrameRate(frameRate);
@@ -359,7 +359,7 @@
float
VCMMediaOptimization::SentBitRate()
{
- UpdateBitRateEstimate(-1, _clock->MillisecondTimestamp());
+ UpdateBitRateEstimate(-1, _clock->TimeInMilliseconds());
return _avgSentBitRateBps / 1000.0f;
}
@@ -374,7 +374,7 @@
FrameType encodedFrameType)
{
// look into the ViE version - debug mode - needs also number of layers.
- UpdateBitRateEstimate(encodedLength, _clock->MillisecondTimestamp());
+ UpdateBitRateEstimate(encodedLength, _clock->TimeInMilliseconds());
if(encodedLength > 0)
{
const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
@@ -388,12 +388,12 @@
if (deltaFrame)
{
_lossProtLogic->UpdatePacketsPerFrame(
- minPacketsPerFrame, _clock->MillisecondTimestamp());
+ minPacketsPerFrame, _clock->TimeInMilliseconds());
}
else
{
_lossProtLogic->UpdatePacketsPerFrameKey(
- minPacketsPerFrame, _clock->MillisecondTimestamp());
+ minPacketsPerFrame, _clock->TimeInMilliseconds());
}
if (_enableQm)
@@ -544,7 +544,7 @@
_qmResolution->ResetRates();
// Reset counters
- _lastQMUpdateTime = _clock->MillisecondTimestamp();
+ _lastQMUpdateTime = _clock->TimeInMilliseconds();
// Reset content metrics
_content->Reset();
@@ -567,7 +567,7 @@
// (to sample the metrics) from the event lastChangeTime
// lastChangeTime is the time where user changed the size/rate/frame rate
// (via SetEncodingData)
- WebRtc_Word64 now = _clock->MillisecondTimestamp();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
if ((now - _lastQMUpdateTime) < kQmMinIntervalMs ||
(now - _lastChangeTime) < kQmMinIntervalMs)
{
@@ -619,7 +619,7 @@
void
VCMMediaOptimization::UpdateIncomingFrameRate()
{
- WebRtc_Word64 now = _clock->MillisecondTimestamp();
+ WebRtc_Word64 now = _clock->TimeInMilliseconds();
if (_incomingFrameTimes[0] == 0)
{
// first no shift
@@ -667,7 +667,7 @@
WebRtc_UWord32
VCMMediaOptimization::InputFrameRate()
{
- ProcessIncomingFrameRate(_clock->MillisecondTimestamp());
+ ProcessIncomingFrameRate(_clock->TimeInMilliseconds());
return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
}
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/main/source/media_optimization.h
index 3b6c978..a642c5d 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/webrtc/modules/video_coding/main/source/media_optimization.h
@@ -23,7 +23,7 @@
enum { kBitrateMaxFrameSamples = 60 };
enum { kBitrateAverageWinMs = 1000 };
-class TickTimeBase;
+class Clock;
class VCMContentMetricsProcessing;
class VCMFrameDropper;
@@ -38,7 +38,7 @@
class VCMMediaOptimization
{
public:
- VCMMediaOptimization(WebRtc_Word32 id, TickTimeBase* clock);
+ VCMMediaOptimization(WebRtc_Word32 id, Clock* clock);
~VCMMediaOptimization(void);
/*
* Reset the Media Optimization module
@@ -163,7 +163,7 @@
enum { kFrameHistoryWinMs = 2000};
WebRtc_Word32 _id;
- TickTimeBase* _clock;
+ Clock* _clock;
WebRtc_Word32 _maxBitRate;
VideoCodecType _sendCodecType;
WebRtc_UWord16 _codecWidth;
diff --git a/webrtc/modules/video_coding/main/source/mock/fake_tick_time.h b/webrtc/modules/video_coding/main/source/mock/fake_tick_time.h
deleted file mode 100644
index c6da348..0000000
--- a/webrtc/modules/video_coding/main/source/mock/fake_tick_time.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
-
-#include <assert.h>
-
-#include <limits>
-
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-namespace webrtc {
-
-// Provides a fake implementation of TickTimeBase, intended for offline
-// testing. This implementation does not query the system clock, but returns a
-// time value set by the user when creating the object, and incremented with
-// the method IncrementDebugClock.
-class FakeTickTime : public TickTimeBase {
- public:
- explicit FakeTickTime(int64_t start_time_ms) : fake_now_ms_(start_time_ms) {}
- virtual ~FakeTickTime() {}
- virtual int64_t MillisecondTimestamp() const {
- return fake_now_ms_;
- }
- virtual int64_t MicrosecondTimestamp() const {
- return 1000 * fake_now_ms_;
- }
- virtual void IncrementDebugClock(int64_t increase_ms) {
- assert(increase_ms <= std::numeric_limits<int64_t>::max() - fake_now_ms_);
- fake_now_ms_ += increase_ms;
- }
-
- private:
- int64_t fake_now_ms_;
-};
-
-} // namespace
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
diff --git a/webrtc/modules/video_coding/main/source/receiver.cc b/webrtc/modules/video_coding/main/source/receiver.cc
index ce209e1..efb3ecd 100644
--- a/webrtc/modules/video_coding/main/source/receiver.cc
+++ b/webrtc/modules/video_coding/main/source/receiver.cc
@@ -16,13 +16,13 @@
#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
-#include "webrtc/modules/video_coding/main/source/tick_time_base.h"
+#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
VCMReceiver::VCMReceiver(VCMTiming* timing,
- TickTimeBase* clock,
+ Clock* clock,
int32_t vcm_id,
int32_t receiver_id,
bool master)
@@ -95,10 +95,10 @@
VCMId(vcm_id_, receiver_id_),
"Packet seq_no %u of frame %u at %u",
packet.seqNum, packet.timestamp,
- MaskWord64ToUWord32(clock_->MillisecondTimestamp()));
+ MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
}
- const int64_t now_ms = clock_->MillisecondTimestamp();
+ const int64_t now_ms = clock_->TimeInMilliseconds();
int64_t render_time_ms = timing_->RenderTimeMs(packet.timestamp, now_ms);
@@ -106,7 +106,7 @@
// Render time error. Assume that this is due to some change in the
// incoming video stream and reset the JB and the timing.
jitter_buffer_.Flush();
- timing_->Reset(clock_->MillisecondTimestamp());
+ timing_->Reset(clock_->TimeInMilliseconds());
return VCM_FLUSH_INDICATOR;
} else if (render_time_ms < now_ms - kMaxVideoDelayMs) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
@@ -115,7 +115,7 @@
"Flushing jitter buffer and resetting timing.",
kMaxVideoDelayMs);
jitter_buffer_.Flush();
- timing_->Reset(clock_->MillisecondTimestamp());
+ timing_->Reset(clock_->TimeInMilliseconds());
return VCM_FLUSH_INDICATOR;
} else if (timing_->TargetVideoDelay() > kMaxVideoDelayMs) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
@@ -123,13 +123,13 @@
"More than %u ms target delay. Flushing jitter buffer and"
"resetting timing.", kMaxVideoDelayMs);
jitter_buffer_.Flush();
- timing_->Reset(clock_->MillisecondTimestamp());
+ timing_->Reset(clock_->TimeInMilliseconds());
return VCM_FLUSH_INDICATOR;
}
// First packet received belonging to this frame.
if (buffer->Length() == 0) {
- const int64_t now_ms = clock_->MillisecondTimestamp();
+ const int64_t now_ms = clock_->TimeInMilliseconds();
if (master_) {
// Only trace the primary receiver to make it possible to parse and plot
// the trace file.
@@ -171,7 +171,7 @@
// is thread-safe.
FrameType incoming_frame_type = kVideoFrameDelta;
next_render_time_ms = -1;
- const int64_t start_time_ms = clock_->MillisecondTimestamp();
+ const int64_t start_time_ms = clock_->TimeInMilliseconds();
int64_t ret = jitter_buffer_.NextTimestamp(max_wait_time_ms,
&incoming_frame_type,
&next_render_time_ms);
@@ -186,7 +186,7 @@
timing_->UpdateCurrentDelay(time_stamp);
const int32_t temp_wait_time = max_wait_time_ms -
- static_cast<int32_t>(clock_->MillisecondTimestamp() - start_time_ms);
+ static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
uint16_t new_max_wait_time = static_cast<uint16_t>(VCM_MAX(temp_wait_time,
0));
@@ -223,7 +223,7 @@
VCMReceiver* dual_receiver) {
// How long can we wait until we must decode the next frame.
uint32_t wait_time_ms = timing_->MaxWaitingTime(
- next_render_time_ms, clock_->MillisecondTimestamp());
+ next_render_time_ms, clock_->TimeInMilliseconds());
// Try to get a complete frame from the jitter buffer.
VCMEncodedFrame* frame = jitter_buffer_.GetCompleteFrameForDecoding(0);
@@ -257,7 +257,7 @@
if (frame == NULL) {
// Get an incomplete frame.
if (timing_->MaxWaitingTime(next_render_time_ms,
- clock_->MillisecondTimestamp()) > 0) {
+ clock_->TimeInMilliseconds()) > 0) {
// Still time to wait for a complete frame.
return NULL;
}
@@ -286,7 +286,7 @@
// frame to the decoder, which will render the frame as soon as it has been
// decoded.
uint32_t wait_time_ms = timing_->MaxWaitingTime(
- next_render_time_ms, clock_->MillisecondTimestamp());
+ next_render_time_ms, clock_->TimeInMilliseconds());
if (max_wait_time_ms < wait_time_ms) {
// If we're not allowed to wait until the frame is supposed to be rendered
// we will have to return NULL for now.
diff --git a/webrtc/modules/video_coding/main/source/receiver.h b/webrtc/modules/video_coding/main/source/receiver.h
index 9546f18..9d18e24 100644
--- a/webrtc/modules/video_coding/main/source/receiver.h
+++ b/webrtc/modules/video_coding/main/source/receiver.h
@@ -14,11 +14,11 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
#include "webrtc/modules/video_coding/main/source/packet.h"
-#include "webrtc/modules/video_coding/main/source/tick_time_base.h"
#include "webrtc/modules/video_coding/main/source/timing.h"
namespace webrtc {
+class Clock;
class VCMEncodedFrame;
enum VCMNackStatus {
@@ -36,7 +36,7 @@
class VCMReceiver {
public:
VCMReceiver(VCMTiming* timing,
- TickTimeBase* clock,
+ Clock* clock,
int32_t vcm_id = -1,
int32_t receiver_id = -1,
bool master = true);
@@ -81,7 +81,7 @@
CriticalSectionWrapper* crit_sect_;
int32_t vcm_id_;
- TickTimeBase* clock_;
+ Clock* clock_;
int32_t receiver_id_;
bool master_;
VCMJitterBuffer jitter_buffer_;
diff --git a/webrtc/modules/video_coding/main/source/tick_time_base.h b/webrtc/modules/video_coding/main/source/tick_time_base.h
deleted file mode 100644
index a212591..0000000
--- a/webrtc/modules/video_coding/main/source/tick_time_base.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
-
-#include "system_wrappers/interface/tick_util.h"
-
-namespace webrtc {
-
-// This class provides a mockable wrapper to TickTime.
-class TickTimeBase {
- public:
- virtual ~TickTimeBase() {}
-
- // "Now" in milliseconds.
- virtual int64_t MillisecondTimestamp() const {
- return TickTime::MillisecondTimestamp();
- }
-
- // "Now" in microseconds.
- virtual int64_t MicrosecondTimestamp() const {
- return TickTime::MicrosecondTimestamp();
- }
-};
-
-} // namespace
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
index e272eb9..37ebbdd 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
+++ b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
@@ -9,13 +9,13 @@
*/
#include "internal_defines.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "timestamp_extrapolator.h"
#include "trace.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
-VCMTimestampExtrapolator::VCMTimestampExtrapolator(TickTimeBase* clock,
+VCMTimestampExtrapolator::VCMTimestampExtrapolator(Clock* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 id)
:
@@ -38,7 +38,7 @@
_accMaxError(7000),
_P11(1e10)
{
- Reset(_clock->MillisecondTimestamp());
+ Reset(_clock->TimeInMilliseconds());
}
VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
@@ -56,7 +56,7 @@
}
else
{
- _startMs = _clock->MillisecondTimestamp();
+ _startMs = _clock->TimeInMilliseconds();
}
_prevMs = _startMs;
_firstTimestamp = 0;
diff --git a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
index 901d8d4..1661f54 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
+++ b/webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
@@ -17,12 +17,12 @@
namespace webrtc
{
-class TickTimeBase;
+class Clock;
class VCMTimestampExtrapolator
{
public:
- VCMTimestampExtrapolator(TickTimeBase* clock,
+ VCMTimestampExtrapolator(Clock* clock,
WebRtc_Word32 vcmId = 0,
WebRtc_Word32 receiverId = 0);
~VCMTimestampExtrapolator();
@@ -37,16 +37,16 @@
RWLockWrapper* _rwLock;
WebRtc_Word32 _vcmId;
WebRtc_Word32 _id;
- TickTimeBase* _clock;
- double _w[2];
- double _P[2][2];
+ Clock* _clock;
+ double _w[2];
+ double _P[2][2];
WebRtc_Word64 _startMs;
WebRtc_Word64 _prevMs;
WebRtc_UWord32 _firstTimestamp;
WebRtc_Word32 _wrapArounds;
WebRtc_UWord32 _prevTs90khz;
- const double _lambda;
- bool _firstAfterReset;
+ const double _lambda;
+ bool _firstAfterReset;
WebRtc_UWord32 _packetCount;
const WebRtc_UWord32 _startUpFilterDelayInPackets;
diff --git a/webrtc/modules/video_coding/main/source/timing.cc b/webrtc/modules/video_coding/main/source/timing.cc
index aca05fa..36131b1b 100644
--- a/webrtc/modules/video_coding/main/source/timing.cc
+++ b/webrtc/modules/video_coding/main/source/timing.cc
@@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "trace.h"
-#include "internal_defines.h"
-#include "jitter_buffer_common.h"
-#include "timing.h"
-#include "timestamp_extrapolator.h"
+#include "webrtc/modules/video_coding/main/source/timing.h"
+
+#include "webrtc/modules/video_coding/main/source/internal_defines.h"
+#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
+#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
-VCMTiming::VCMTiming(TickTimeBase* clock,
+VCMTiming::VCMTiming(Clock* clock,
WebRtc_Word32 vcmId,
WebRtc_Word32 timingId,
VCMTiming* masterTiming)
diff --git a/webrtc/modules/video_coding/main/source/timing.h b/webrtc/modules/video_coding/main/source/timing.h
index 41a4945..ac650ec 100644
--- a/webrtc/modules/video_coding/main/source/timing.h
+++ b/webrtc/modules/video_coding/main/source/timing.h
@@ -18,7 +18,7 @@
namespace webrtc
{
-class TickTimeBase;
+class Clock;
class VCMTimestampExtrapolator;
class VCMTiming
@@ -26,7 +26,7 @@
public:
// The primary timing component should be passed
// if this is the dual timing component.
- VCMTiming(TickTimeBase* clock,
+ VCMTiming(Clock* clock,
WebRtc_Word32 vcmId = 0,
WebRtc_Word32 timingId = 0,
VCMTiming* masterTiming = NULL);
@@ -94,7 +94,7 @@
private:
CriticalSectionWrapper* _critSect;
WebRtc_Word32 _vcmId;
- TickTimeBase* _clock;
+ Clock* _clock;
WebRtc_Word32 _timingId;
bool _master;
VCMTimestampExtrapolator* _tsExtrapolator;
diff --git a/webrtc/modules/video_coding/main/source/video_coding.gypi b/webrtc/modules/video_coding/main/source/video_coding.gypi
index 2b9d4bd..b795f90 100644
--- a/webrtc/modules/video_coding/main/source/video_coding.gypi
+++ b/webrtc/modules/video_coding/main/source/video_coding.gypi
@@ -62,7 +62,6 @@
'receiver.h',
'rtt_filter.h',
'session_info.h',
- 'tick_time_base.h',
'timestamp_extrapolator.h',
'timestamp_map.h',
'timing.h',
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
index 8fba7de..11725ad 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.cc
@@ -16,7 +16,7 @@
#include "packet.h"
#include "trace.h"
#include "video_codec_interface.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc
{
@@ -34,22 +34,20 @@
{
return static_cast<WebRtc_UWord32>(
VCM_MAX(static_cast<WebRtc_Word64>(_periodMs) -
- (_clock->MillisecondTimestamp() - _latestMs), 0));
+ (_clock->TimeInMilliseconds() - _latestMs), 0));
}
void
VCMProcessTimer::Processed()
{
- _latestMs = _clock->MillisecondTimestamp();
+ _latestMs = _clock->TimeInMilliseconds();
}
VideoCodingModuleImpl::VideoCodingModuleImpl(const WebRtc_Word32 id,
- TickTimeBase* clock,
- bool delete_clock_on_destroy)
+ Clock* clock)
:
_id(id),
clock_(clock),
-delete_clock_on_destroy_(delete_clock_on_destroy),
_receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
_receiverInited(false),
_timing(clock_, id, 1),
@@ -99,7 +97,6 @@
}
delete _receiveCritSect;
delete _sendCritSect;
- if (delete_clock_on_destroy_) delete clock_;
#ifdef DEBUG_DECODER_BIT_STREAM
fclose(_bitStreamBeforeDecoder);
#endif
@@ -112,14 +109,14 @@
VideoCodingModule*
VideoCodingModule::Create(const WebRtc_Word32 id)
{
- return new VideoCodingModuleImpl(id, new TickTimeBase(), true);
+ return new VideoCodingModuleImpl(id, Clock::GetRealTimeClock());
}
VideoCodingModule*
-VideoCodingModule::Create(const WebRtc_Word32 id, TickTimeBase* clock)
+VideoCodingModule::Create(const WebRtc_Word32 id, Clock* clock)
{
assert(clock);
- return new VideoCodingModuleImpl(id, clock, false);
+ return new VideoCodingModuleImpl(id, clock);
}
void
@@ -890,7 +887,7 @@
// If this frame was too late, we should adjust the delay accordingly
_timing.UpdateCurrentDelay(frame->RenderTimeMs(),
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
#ifdef DEBUG_DECODER_BIT_STREAM
if (_bitStreamBeforeDecoder != NULL)
@@ -1001,7 +998,7 @@
dualFrame->TimeStamp());
// Decode dualFrame and try to catch up
WebRtc_Word32 ret = _dualDecoder->Decode(*dualFrame,
- clock_->MillisecondTimestamp());
+ clock_->TimeInMilliseconds());
if (ret != WEBRTC_VIDEO_CODEC_OK)
{
WEBRTC_TRACE(webrtc::kTraceWarning,
@@ -1049,7 +1046,7 @@
return VCM_NO_CODEC_REGISTERED;
}
// Decode a frame
- WebRtc_Word32 ret = _decoder->Decode(frame, clock_->MillisecondTimestamp());
+ WebRtc_Word32 ret = _decoder->Decode(frame, clock_->TimeInMilliseconds());
// Check for failed decoding, run frame type request callback if needed.
if (ret < 0)
diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.h b/webrtc/modules/video_coding/main/source/video_coding_impl.h
index 80e3c81..e09872e 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_impl.h
+++ b/webrtc/modules/video_coding/main/source/video_coding_impl.h
@@ -15,16 +15,16 @@
#include <vector>
-#include "modules/video_coding/main/source/codec_database.h"
-#include "modules/video_coding/main/source/frame_buffer.h"
-#include "modules/video_coding/main/source/generic_decoder.h"
-#include "modules/video_coding/main/source/generic_encoder.h"
-#include "modules/video_coding/main/source/jitter_buffer.h"
-#include "modules/video_coding/main/source/media_optimization.h"
-#include "modules/video_coding/main/source/receiver.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "modules/video_coding/main/source/timing.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/modules/video_coding/main/source/codec_database.h"
+#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
+#include "webrtc/modules/video_coding/main/source/generic_decoder.h"
+#include "webrtc/modules/video_coding/main/source/generic_encoder.h"
+#include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
+#include "webrtc/modules/video_coding/main/source/media_optimization.h"
+#include "webrtc/modules/video_coding/main/source/receiver.h"
+#include "webrtc/modules/video_coding/main/source/timing.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc
{
@@ -32,16 +32,16 @@
class VCMProcessTimer
{
public:
- VCMProcessTimer(WebRtc_UWord32 periodMs, TickTimeBase* clock)
+ VCMProcessTimer(WebRtc_UWord32 periodMs, Clock* clock)
: _clock(clock),
_periodMs(periodMs),
- _latestMs(_clock->MillisecondTimestamp()) {}
+ _latestMs(_clock->TimeInMilliseconds()) {}
WebRtc_UWord32 Period() const;
WebRtc_UWord32 TimeUntilProcess() const;
void Processed();
private:
- TickTimeBase* _clock;
+ Clock* _clock;
WebRtc_UWord32 _periodMs;
WebRtc_Word64 _latestMs;
};
@@ -59,8 +59,7 @@
{
public:
VideoCodingModuleImpl(const WebRtc_Word32 id,
- TickTimeBase* clock,
- bool delete_clock_on_destroy);
+ Clock* clock);
virtual ~VideoCodingModuleImpl();
@@ -275,8 +274,7 @@
private:
WebRtc_Word32 _id;
- TickTimeBase* clock_;
- bool delete_clock_on_destroy_;
+ Clock* clock_;
CriticalSectionWrapper* _receiveCritSect;
bool _receiverInited;
VCMTiming _timing;
diff --git a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
index 0ee9657..ccf37eb 100644
--- a/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
@@ -13,7 +13,7 @@
#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
#include "modules/video_coding/main/interface/video_coding.h"
#include "modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
@@ -32,9 +32,9 @@
static const size_t kPayloadLen = 10;
virtual void SetUp() {
- clock_ = new FakeTickTime(0);
- ASSERT_TRUE(clock_ != NULL);
- vcm_ = VideoCodingModule::Create(0, clock_);
+ clock_.reset(new SimulatedClock(0));
+ ASSERT_TRUE(clock_.get() != NULL);
+ vcm_ = VideoCodingModule::Create(0, clock_.get());
ASSERT_TRUE(vcm_ != NULL);
ASSERT_EQ(0, vcm_->InitializeReceiver());
ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
@@ -48,7 +48,6 @@
virtual void TearDown() {
VideoCodingModule::Destroy(vcm_);
- delete clock_;
}
void InsertPacket(uint32_t timestamp,
@@ -77,7 +76,7 @@
MockPacketRequestCallback request_callback_;
NiceMock<MockVideoDecoder> decoder_;
NiceMock<MockVideoDecoder> decoderCopy_;
- FakeTickTime* clock_;
+ scoped_ptr<SimulatedClock> clock_;
};
TEST_F(VCMRobustnessTest, TestHardNack) {
@@ -112,21 +111,21 @@
ASSERT_EQ(VCM_OK, vcm_->Decode(0));
ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
ASSERT_EQ(VCM_OK, vcm_->Process());
ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
InsertPacket(6000, 8, false, true, kVideoFrameDelta);
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
ASSERT_EQ(VCM_OK, vcm_->Process());
ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
InsertPacket(6000, 6, true, false, kVideoFrameDelta);
InsertPacket(6000, 7, false, false, kVideoFrameDelta);
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
ASSERT_EQ(VCM_OK, vcm_->Process());
ASSERT_EQ(VCM_OK, vcm_->Decode(0));
@@ -149,7 +148,7 @@
EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
ASSERT_EQ(VCM_OK, vcm_->Process());
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
ASSERT_EQ(VCM_OK, vcm_->Process());
@@ -217,13 +216,13 @@
InsertPacket(0, 2, false, true, kVideoFrameKey);
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(3000, 3, true, false, kVideoFrameDelta);
// Packet 4 missing
InsertPacket(3000, 5, false, true, kVideoFrameDelta);
EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(6000, 6, true, false, kVideoFrameDelta);
InsertPacket(6000, 7, false, false, kVideoFrameDelta);
InsertPacket(6000, 8, false, true, kVideoFrameDelta);
@@ -232,7 +231,7 @@
// Spawn a decoder copy.
EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
EXPECT_EQ(VCM_OK, vcm_->Process()); // Generate NACK list.
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
@@ -299,25 +298,25 @@
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(3000, 3, true, false, kVideoFrameDelta);
// Packet 4 missing
InsertPacket(3000, 5, false, true, kVideoFrameDelta);
EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(6000, 6, true, false, kVideoFrameDelta);
InsertPacket(6000, 7, false, false, kVideoFrameDelta);
InsertPacket(6000, 8, false, true, kVideoFrameDelta);
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(23);
+ clock_->AdvanceTimeMilliseconds(23);
InsertPacket(3000, 4, false, false, kVideoFrameDelta);
InsertPacket(9000, 9, true, false, kVideoFrameDelta);
@@ -371,14 +370,14 @@
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(3000, 3, true, false, kVideoFrameDelta);
// Packet 4 missing
InsertPacket(3000, 5, false, true, kVideoFrameDelta);
EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(33);
+ clock_->AdvanceTimeMilliseconds(33);
InsertPacket(6000, 6, true, false, kVideoFrameDelta);
InsertPacket(6000, 7, false, false, kVideoFrameDelta);
InsertPacket(6000, 8, false, true, kVideoFrameDelta);
@@ -386,11 +385,12 @@
// Schedule key frame request.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(10);
+ clock_->AdvanceTimeMilliseconds(10);
EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
- clock_->IncrementDebugClock(500); // Wait for the key request timer to set.
+ // Wait for the key request timer to set.
+ clock_->AdvanceTimeMilliseconds(500);
EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect key frame request.
}
} // namespace webrtc
diff --git a/webrtc/modules/video_coding/main/test/decode_from_storage_test.cc b/webrtc/modules/video_coding/main/test/decode_from_storage_test.cc
index 628d509..5d1c916 100644
--- a/webrtc/modules/video_coding/main/test/decode_from_storage_test.cc
+++ b/webrtc/modules/video_coding/main/test/decode_from_storage_test.cc
@@ -14,7 +14,7 @@
#include "trace.h"
#include "../source/event.h"
#include "rtp_player.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
@@ -64,7 +64,7 @@
Trace::SetLevelFilter(webrtc::kTraceAll);
- FakeTickTime clock(0);
+ SimulatedClock clock(0);
// TODO(hlundin): This test was not verified after changing to FakeTickTime.
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
VideoCodingModule* vcmPlayback = VideoCodingModule::Create(2, &clock);
@@ -125,9 +125,9 @@
ret = 0;
// RTP stream main loop
- while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
+ while ((ret = rtpStream.NextPacket(clock.TimeInMilliseconds())) == 0)
{
- if (clock.MillisecondTimestamp() % 5 == 0)
+ if (clock.TimeInMilliseconds() % 5 == 0)
{
ret = vcm->Decode();
if (ret < 0)
@@ -139,11 +139,11 @@
{
vcm->Process();
}
- if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
+ if (MAX_RUNTIME_MS > -1 && clock.TimeInMilliseconds() >= MAX_RUNTIME_MS)
{
break;
}
- clock.IncrementDebugClock(1);
+ clock.AdvanceTimeMilliseconds(1);
}
switch (ret)
diff --git a/webrtc/modules/video_coding/main/test/generic_codec_test.cc b/webrtc/modules/video_coding/main/test/generic_codec_test.cc
index 2220d3e..3f21a65 100644
--- a/webrtc/modules/video_coding/main/test/generic_codec_test.cc
+++ b/webrtc/modules/video_coding/main/test/generic_codec_test.cc
@@ -15,7 +15,7 @@
#include "rtp_rtcp.h"
#include "common_video/interface/i420_video_frame.h"
#include "test_macros.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
@@ -27,7 +27,7 @@
printf("\n\nEnable debug events to run this test!\n\n");
return -1;
#endif
- FakeTickTime clock(0);
+ SimulatedClock clock(0);
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
GenericCodecTest* get = new GenericCodecTest(vcm, &clock);
Trace::CreateTrace();
@@ -41,7 +41,8 @@
return 0;
}
-GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm, FakeTickTime* clock):
+GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm,
+ SimulatedClock* clock):
_clock(clock),
_vcm(vcm),
_width(0),
@@ -332,10 +333,6 @@
IncrementDebugClock(_frameRate);
// The following should be uncommneted for timing tests. Release tests only include
// compliance with full sequence bit rate.
-
-
- //totalBytes = WaitForEncodedFrame();
- //currentTime = VCMTickTime::MillisecondTimestamp();//clock()/(double)CLOCKS_PER_SEC;
if (_frameCnt == _frameRate)// @ 1sec
{
totalBytesOneSec = _encodeCompleteCallback->EncodedBytes();//totalBytes;
@@ -482,8 +479,8 @@
float
GenericCodecTest::WaitForEncodedFrame() const
{
- WebRtc_Word64 startTime = _clock->MillisecondTimestamp();
- while (_clock->MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs*10)
+ WebRtc_Word64 startTime = _clock->TimeInMilliseconds();
+ while (_clock->TimeInMilliseconds() - startTime < kMaxWaitEncTimeMs*10)
{
if (_encodeCompleteCallback->EncodeComplete())
{
@@ -496,7 +493,7 @@
void
GenericCodecTest::IncrementDebugClock(float frameRate)
{
- _clock->IncrementDebugClock(1000/frameRate);
+ _clock->AdvanceTimeMilliseconds(1000/frameRate);
}
int
diff --git a/webrtc/modules/video_coding/main/test/generic_codec_test.h b/webrtc/modules/video_coding/main/test/generic_codec_test.h
index f60893b..f0f46dc 100644
--- a/webrtc/modules/video_coding/main/test/generic_codec_test.h
+++ b/webrtc/modules/video_coding/main/test/generic_codec_test.h
@@ -31,13 +31,13 @@
int VCMGenericCodecTest(CmdArgs& args);
-class FakeTickTime;
+class SimulatedClock;
class GenericCodecTest
{
public:
GenericCodecTest(webrtc::VideoCodingModule* vcm,
- webrtc::FakeTickTime* clock);
+ webrtc::SimulatedClock* clock);
~GenericCodecTest();
static int RunTest(CmdArgs& args);
WebRtc_Word32 Perform(CmdArgs& args);
@@ -49,7 +49,7 @@
WebRtc_Word32 TearDown();
void IncrementDebugClock(float frameRate);
- webrtc::FakeTickTime* _clock;
+ webrtc::SimulatedClock* _clock;
webrtc::VideoCodingModule* _vcm;
webrtc::VideoCodec _sendCodec;
webrtc::VideoCodec _receiveCodec;
diff --git a/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc b/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
index 99d8ac5..69c4c20 100644
--- a/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
+++ b/webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
@@ -19,10 +19,10 @@
#include "jitter_estimate_test.h"
#include "jitter_estimator.h"
#include "media_opt_util.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "packet.h"
#include "test_util.h"
#include "test_macros.h"
+#include "webrtc/system_wrappers/interface/clock.h"
// TODO(holmer): Get rid of this to conform with style guide.
using namespace webrtc;
@@ -97,7 +97,7 @@
#if defined(EVENT_DEBUG)
return -1;
#endif
- TickTimeBase clock;
+ Clock* clock = Clock::GetRealTimeClock();
// Start test
WebRtc_UWord16 seqNum = 1234;
@@ -106,7 +106,7 @@
WebRtc_UWord8 data[1500];
VCMPacket packet(data, size, seqNum, timeStamp, true);
- VCMJitterBuffer jb(&clock);
+ VCMJitterBuffer jb(clock);
seqNum = 1234;
timeStamp = 123*90;
diff --git a/webrtc/modules/video_coding/main/test/media_opt_test.cc b/webrtc/modules/video_coding/main/test/media_opt_test.cc
index 8d398d2..12697cf 100644
--- a/webrtc/modules/video_coding/main/test/media_opt_test.cc
+++ b/webrtc/modules/video_coding/main/test/media_opt_test.cc
@@ -32,9 +32,9 @@
Trace::CreateTrace();
Trace::SetTraceFile((test::OutputPath() + "mediaOptTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
- TickTimeBase clock;
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- MediaOptTest* mot = new MediaOptTest(vcm, &clock);
+ Clock* clock = Clock::GetRealTimeClock();
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
+ MediaOptTest* mot = new MediaOptTest(vcm, clock);
if (testNum == 0)
{ // regular
mot->Setup(0, args);
@@ -65,7 +65,7 @@
}
-MediaOptTest::MediaOptTest(VideoCodingModule* vcm, TickTimeBase* clock)
+MediaOptTest::MediaOptTest(VideoCodingModule* vcm, Clock* clock)
: _vcm(vcm),
_rtp(NULL),
_outgoingTransport(NULL),
diff --git a/webrtc/modules/video_coding/main/test/media_opt_test.h b/webrtc/modules/video_coding/main/test/media_opt_test.h
index 5f210e6..24f53ab 100644
--- a/webrtc/modules/video_coding/main/test/media_opt_test.h
+++ b/webrtc/modules/video_coding/main/test/media_opt_test.h
@@ -34,7 +34,7 @@
{
public:
MediaOptTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
+ webrtc::Clock* clock);
~MediaOptTest();
static int RunTest(int testNum, CmdArgs& args);
@@ -57,7 +57,7 @@
webrtc::RTPSendCompleteCallback* _outgoingTransport;
RtpDataCallback* _dataCallback;
- webrtc::TickTimeBase* _clock;
+ webrtc::Clock* _clock;
std::string _inname;
std::string _outname;
std::string _actualSourcename;
diff --git a/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc b/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
index 6ec9389..b99c31a 100644
--- a/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
+++ b/webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
@@ -143,12 +143,12 @@
printf("Cannot read file %s.\n", outname.c_str());
return -1;
}
- TickTimeBase clock;
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
+ Clock* clock = Clock::GetRealTimeClock();
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
RtpDataCallback dataCallback(vcm);
RTPSendCompleteCallback* outgoingTransport =
- new RTPSendCompleteCallback(&clock, "dump.rtp");
+ new RTPSendCompleteCallback(clock, "dump.rtp");
RtpRtcp::Configuration configuration;
configuration.id = 1;
diff --git a/webrtc/modules/video_coding/main/test/mt_test_common.cc b/webrtc/modules/video_coding/main/test/mt_test_common.cc
index ece0d9c..e27f166 100644
--- a/webrtc/modules/video_coding/main/test/mt_test_common.cc
+++ b/webrtc/modules/video_coding/main/test/mt_test_common.cc
@@ -12,12 +12,12 @@
#include <cmath>
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "rtp_dump.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
-TransportCallback::TransportCallback(TickTimeBase* clock, const char* filename)
+TransportCallback::TransportCallback(Clock* clock, const char* filename)
: RTPSendCompleteCallback(clock, filename) {
}
@@ -47,8 +47,8 @@
transmitPacket = PacketLoss();
}
- TickTimeBase clock;
- int64_t now = clock.MillisecondTimestamp();
+ Clock* clock = Clock::GetRealTimeClock();
+ int64_t now = clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{
@@ -72,8 +72,8 @@
{
// Are we ready to send packets to the receiver?
RtpPacket* packet = NULL;
- TickTimeBase clock;
- int64_t now = clock.MillisecondTimestamp();
+ Clock* clock = Clock::GetRealTimeClock();
+ int64_t now = clock->TimeInMilliseconds();
while (!_rtpPackets.empty())
{
diff --git a/webrtc/modules/video_coding/main/test/mt_test_common.h b/webrtc/modules/video_coding/main/test/mt_test_common.h
index c17d269..e1ae0c6 100644
--- a/webrtc/modules/video_coding/main/test/mt_test_common.h
+++ b/webrtc/modules/video_coding/main/test/mt_test_common.h
@@ -47,7 +47,7 @@
{
public:
// constructor input: (receive side) rtp module to send encoded data to
- TransportCallback(TickTimeBase* clock, const char* filename = NULL);
+ TransportCallback(Clock* clock, const char* filename = NULL);
virtual ~TransportCallback();
// Add packets to list
// Incorporate network conditions - delay and packet loss
diff --git a/webrtc/modules/video_coding/main/test/normal_test.cc b/webrtc/modules/video_coding/main/test/normal_test.cc
index 96e5008..e88ff87 100644
--- a/webrtc/modules/video_coding/main/test/normal_test.cc
+++ b/webrtc/modules/video_coding/main/test/normal_test.cc
@@ -18,12 +18,12 @@
#include "../source/event.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "common_types.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
#include "test_callbacks.h"
#include "test_macros.h"
#include "test_util.h"
#include "trace.h"
#include "testsupport/metrics/video_metrics.h"
+#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
@@ -31,17 +31,18 @@
{
#if defined(EVENT_DEBUG)
printf("SIMULATION TIME\n");
- FakeTickTime clock(0);
+ SimulatedClock sim_clock;
+ SimulatedClock* clock = &sim_clock;
#else
printf("REAL-TIME\n");
- TickTimeBase clock;
+ Clock* clock = Clock::GetRealTimeClock();
#endif
Trace::CreateTrace();
Trace::SetTraceFile(
(test::OutputPath() + "VCMNormalTestTrace.txt").c_str());
Trace::SetLevelFilter(webrtc::kTraceAll);
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- NormalTest VCMNTest(vcm, &clock);
+ VideoCodingModule* vcm = VideoCodingModule::Create(1, clock);
+ NormalTest VCMNTest(vcm, clock);
VCMNTest.Perform(args);
VideoCodingModule::Destroy(vcm);
Trace::ReturnTrace();
@@ -183,7 +184,7 @@
//VCM Normal Test Class implementation
-NormalTest::NormalTest(VideoCodingModule* vcm, TickTimeBase* clock)
+NormalTest::NormalTest(VideoCodingModule* vcm, Clock* clock)
:
_clock(clock),
_vcm(vcm),
@@ -289,7 +290,7 @@
while (feof(_sourceFile) == 0) {
#if !defined(EVENT_DEBUG)
- WebRtc_Word64 processStartTime = _clock->MillisecondTimestamp();
+ WebRtc_Word64 processStartTime = _clock->TimeInMilliseconds();
#endif
TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0 ||
feof(_sourceFile));
@@ -332,10 +333,10 @@
1000.0f / static_cast<float>(_sendCodec.maxFramerate) + 0.5f);
#if defined(EVENT_DEBUG)
- static_cast<FakeTickTime*>(_clock)->IncrementDebugClock(framePeriod);
+ static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(framePeriod);
#else
WebRtc_Word64 timeSpent =
- _clock->MillisecondTimestamp() - processStartTime;
+ _clock->TimeInMilliseconds() - processStartTime;
if (timeSpent < framePeriod)
{
waitEvent->Wait(framePeriod - timeSpent);
diff --git a/webrtc/modules/video_coding/main/test/normal_test.h b/webrtc/modules/video_coding/main/test/normal_test.h
index 0394d6e..14d55a1 100644
--- a/webrtc/modules/video_coding/main/test/normal_test.h
+++ b/webrtc/modules/video_coding/main/test/normal_test.h
@@ -86,7 +86,7 @@
{
public:
NormalTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
+ webrtc::Clock* clock);
~NormalTest();
static int RunTest(const CmdArgs& args);
WebRtc_Word32 Perform(const CmdArgs& args);
@@ -108,7 +108,7 @@
// calculating pipeline delay, and decoding time
void FrameDecoded(WebRtc_UWord32 timeStamp);
- webrtc::TickTimeBase* _clock;
+ webrtc::Clock* _clock;
webrtc::VideoCodingModule* _vcm;
webrtc::VideoCodec _sendCodec;
webrtc::VideoCodec _receiveCodec;
diff --git a/webrtc/modules/video_coding/main/test/quality_modes_test.cc b/webrtc/modules/video_coding/main/test/quality_modes_test.cc
index 57632cc..81f3d9d 100644
--- a/webrtc/modules/video_coding/main/test/quality_modes_test.cc
+++ b/webrtc/modules/video_coding/main/test/quality_modes_test.cc
@@ -17,20 +17,19 @@
#include "common_video/libyuv/include/webrtc_libyuv.h"
#include "modules/video_coding/main/source/event.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "modules/video_coding/main/test/test_callbacks.h"
#include "modules/video_coding/main/test/test_macros.h"
#include "modules/video_coding/main/test/test_util.h"
#include "system_wrappers/interface/data_log.h"
#include "system_wrappers/interface/data_log.h"
#include "testsupport/metrics/video_metrics.h"
+#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
int qualityModeTest(const CmdArgs& args)
{
- FakeTickTime clock(0);
+ SimulatedClock clock(0);
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
QualityModesTest QMTest(vcm, &clock);
QMTest.Perform(args);
@@ -39,7 +38,7 @@
}
QualityModesTest::QualityModesTest(VideoCodingModule* vcm,
- TickTimeBase* clock):
+ Clock* clock):
NormalTest(vcm, clock),
_vpm()
{
@@ -367,8 +366,8 @@
DataLog::InsertCell(feature_table_name_, "frame rate", _nativeFrameRate);
DataLog::NextRow(feature_table_name_);
- static_cast<FakeTickTime*>(
- _clock)->IncrementDebugClock(1000 / _nativeFrameRate);
+ static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(
+ 1000 / _nativeFrameRate);
}
} while (feof(_sourceFile) == 0);
diff --git a/webrtc/modules/video_coding/main/test/quality_modes_test.h b/webrtc/modules/video_coding/main/test/quality_modes_test.h
index 10f023b..0bb7e09 100644
--- a/webrtc/modules/video_coding/main/test/quality_modes_test.h
+++ b/webrtc/modules/video_coding/main/test/quality_modes_test.h
@@ -22,7 +22,7 @@
{
public:
QualityModesTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
+ webrtc::Clock* clock);
virtual ~QualityModesTest();
WebRtc_Word32 Perform(const CmdArgs& args);
diff --git a/webrtc/modules/video_coding/main/test/receiver_timing_tests.cc b/webrtc/modules/video_coding/main/test/receiver_timing_tests.cc
index 0b09256..b273b0e 100644
--- a/webrtc/modules/video_coding/main/test/receiver_timing_tests.cc
+++ b/webrtc/modules/video_coding/main/test/receiver_timing_tests.cc
@@ -61,8 +61,8 @@
// A static random seed
srand(0);
- TickTimeBase clock;
- VCMTiming timing(&clock);
+ Clock* clock = Clock::GetRealTimeClock();
+ VCMTiming timing(clock);
float clockInMs = 0.0;
WebRtc_UWord32 waitTime = 0;
WebRtc_UWord32 jitterDelayMs = 0;
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index 9eacf34..de06224 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -20,8 +20,8 @@
#include "../source/internal_defines.h"
#include "gtest/gtest.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "rtp_rtcp.h"
+#include "webrtc/system_wrappers/interface/clock.h"
using namespace webrtc;
@@ -137,7 +137,7 @@
RTPPlayer::RTPPlayer(const char* filename,
RtpData* callback,
- TickTimeBase* clock)
+ Clock* clock)
:
_clock(clock),
_rtpModule(NULL),
@@ -273,7 +273,8 @@
WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
{
- WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (_clock->MillisecondTimestamp() - _firstPacketTimeMs);
+ WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) -
+ (_clock->TimeInMilliseconds() - _firstPacketTimeMs);
if (timeLeft < 0)
{
return 0;
@@ -293,7 +294,7 @@
delete resend_packet;
_resendPacketCount++;
if (ret > 0) {
- _lostPackets.SetPacketResent(seqNo, _clock->MillisecondTimestamp());
+ _lostPackets.SetPacketResent(seqNo, _clock->TimeInMilliseconds());
} else if (ret < 0) {
return ret;
}
@@ -307,7 +308,7 @@
if (_firstPacket)
{
_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
- _firstPacketTimeMs = _clock->MillisecondTimestamp();
+ _firstPacketTimeMs = _clock->TimeInMilliseconds();
}
if (_reordering && _reorderBuffer == NULL)
{
@@ -428,8 +429,8 @@
for (int i=0; i < length; i++)
{
_lostPackets.SetResendTime(sequenceNumbers[i],
- _clock->MillisecondTimestamp() + _rttMs,
- _clock->MillisecondTimestamp());
+ _clock->TimeInMilliseconds() + _rttMs,
+ _clock->TimeInMilliseconds());
}
return 0;
}
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h
index 9920d0c..d6b0734 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/webrtc/modules/video_coding/main/test/rtp_player.h
@@ -15,7 +15,7 @@
#include "rtp_rtcp.h"
#include "critical_section_wrapper.h"
#include "video_coding_defines.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
+#include "webrtc/system_wrappers/interface/clock.h"
#include <stdio.h>
#include <list>
@@ -78,7 +78,7 @@
public:
RTPPlayer(const char* filename,
webrtc::RtpData* callback,
- webrtc::TickTimeBase* clock);
+ webrtc::Clock* clock);
virtual ~RTPPlayer();
WebRtc_Word32 Initialize(const PayloadTypeList* payloadList);
@@ -93,7 +93,7 @@
WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
WebRtc_Word32 ReadHeader();
- webrtc::TickTimeBase* _clock;
+ webrtc::Clock* _clock;
FILE* _rtpFile;
webrtc::RtpRtcp* _rtpModule;
WebRtc_UWord32 _nextRtpTime;
diff --git a/webrtc/modules/video_coding/main/test/test_callbacks.cc b/webrtc/modules/video_coding/main/test/test_callbacks.cc
index 12824ba..4d7d3d3 100644
--- a/webrtc/modules/video_coding/main/test/test_callbacks.cc
+++ b/webrtc/modules/video_coding/main/test/test_callbacks.cc
@@ -13,9 +13,9 @@
#include <cmath>
#include "common_video/libyuv/include/webrtc_libyuv.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
#include "rtp_dump.h"
#include "test_macros.h"
+#include "webrtc/system_wrappers/interface/clock.h"
namespace webrtc {
@@ -204,7 +204,7 @@
return _decodedBytes;
}
-RTPSendCompleteCallback::RTPSendCompleteCallback(TickTimeBase* clock,
+RTPSendCompleteCallback::RTPSendCompleteCallback(Clock* clock,
const char* filename):
_clock(clock),
_sendCount(0),
@@ -258,7 +258,7 @@
bool transmitPacket = true;
transmitPacket = PacketLoss();
- WebRtc_UWord64 now = _clock->MillisecondTimestamp();
+ int64_t now = _clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{
diff --git a/webrtc/modules/video_coding/main/test/test_callbacks.h b/webrtc/modules/video_coding/main/test/test_callbacks.h
index f2c419b..01c1b7c 100644
--- a/webrtc/modules/video_coding/main/test/test_callbacks.h
+++ b/webrtc/modules/video_coding/main/test/test_callbacks.h
@@ -157,7 +157,7 @@
{
public:
// Constructor input: (receive side) rtp module to send encoded data to
- RTPSendCompleteCallback(TickTimeBase* clock,
+ RTPSendCompleteCallback(Clock* clock,
const char* filename = NULL);
virtual ~RTPSendCompleteCallback();
@@ -186,7 +186,7 @@
// Random uniform loss model
bool UnifomLoss(double lossPct);
- TickTimeBase* _clock;
+ Clock* _clock;
WebRtc_UWord32 _sendCount;
RtpRtcp* _rtp;
double _lossPct;
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play.cc b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
index 49b7f73..bd22168 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play.cc
+++ b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
@@ -17,7 +17,7 @@
#include "../source/internal_defines.h"
#include "test_macros.h"
#include "rtp_player.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
+#include "webrtc/system_wrappers/interface/clock.h"
#include <stdio.h>
#include <string.h>
@@ -130,7 +130,7 @@
if (outFile == "")
outFile = test::OutputPath() + "RtpPlay_decoded.yuv";
FrameReceiveCallback receiveCallback(outFile);
- FakeTickTime clock(0);
+ SimulatedClock clock(0);
VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
RtpDataCallback dataCallback(vcm);
RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback, &clock);
@@ -198,9 +198,9 @@
ret = 0;
// RTP stream main loop
- while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
+ while ((ret = rtpStream.NextPacket(clock.TimeInMilliseconds())) == 0)
{
- if (clock.MillisecondTimestamp() % 5 == 0)
+ if (clock.TimeInMilliseconds() % 5 == 0)
{
ret = vcm->Decode();
if (ret < 0)
@@ -214,12 +214,12 @@
{
vcm->Process();
}
- if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >=
+ if (MAX_RUNTIME_MS > -1 && clock.TimeInMilliseconds() >=
MAX_RUNTIME_MS)
{
break;
}
- clock.IncrementDebugClock(1);
+ clock.AdvanceTimeMilliseconds(1);
}
// Tear down
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc b/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
index ecb7ab6..911eb5c 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
+++ b/webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
@@ -8,17 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "receiver_tests.h"
-#include "video_coding.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "../source/event.h"
-#include "test_macros.h"
-#include "rtp_player.h"
-
#include <string.h>
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/main/source/event.h"
+#include "webrtc/modules/video_coding/main/test/receiver_tests.h"
+#include "webrtc/modules/video_coding/main/test/rtp_player.h"
+#include "webrtc/modules/video_coding/main/test/test_macros.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+
using namespace webrtc;
bool ProcessingThread(void* obj)
@@ -39,8 +40,8 @@
SharedState* state = static_cast<SharedState*>(obj);
EventWrapper& waitEvent = *EventWrapper::Create();
// RTP stream main loop
- TickTimeBase clock;
- if (state->_rtpPlayer.NextPacket(clock.MillisecondTimestamp()) < 0)
+ Clock* clock = Clock::GetRealTimeClock();
+ if (state->_rtpPlayer.NextPacket(clock->TimeInMilliseconds()) < 0)
{
return false;
}
@@ -82,9 +83,9 @@
(protection == kProtectionDualDecoder ||
protection == kProtectionNack ||
kProtectionNackFEC));
- TickTimeBase clock;
+ Clock* clock = Clock::GetRealTimeClock();
VideoCodingModule* vcm =
- VideoCodingModule::Create(1, &clock);
+ VideoCodingModule::Create(1, clock);
RtpDataCallback dataCallback(vcm);
std::string rtpFilename;
rtpFilename = args.inputFile;
@@ -137,7 +138,7 @@
}
printf("Watch %s to verify that the output is reasonable\n", outFilename.c_str());
}
- RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
+ RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, clock);
PayloadTypeList payloadTypes;
payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
kVideoCodecVP8));
@@ -164,10 +165,10 @@
}
// Create and start all threads
- ThreadWrapper* processingThread = ThreadWrapper::CreateThread(ProcessingThread,
- &mtState, kNormalPriority, "ProcessingThread");
- ThreadWrapper* rtpReaderThread = ThreadWrapper::CreateThread(RtpReaderThread,
- &mtState, kNormalPriority, "RtpReaderThread");
+ ThreadWrapper* processingThread = ThreadWrapper::CreateThread(
+ ProcessingThread, &mtState, kNormalPriority, "ProcessingThread");
+ ThreadWrapper* rtpReaderThread = ThreadWrapper::CreateThread(
+ RtpReaderThread, &mtState, kNormalPriority, "RtpReaderThread");
ThreadWrapper* decodeThread = ThreadWrapper::CreateThread(DecodeThread,
&mtState, kNormalPriority, "DecodeThread");
