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

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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2013-01-21 07:42:11 +00:00
parent a3c82bf667
commit a678a3baee
67 changed files with 367 additions and 544 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/rtp_rtcp/source/bitrate.cc
View File

@ -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::BitrateLast() const {
}
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 @@ WebRtc_UWord32 Bitrate::BitrateNow() const {
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) {

2
webrtc/modules/rtp_rtcp/source/bitrate.h
View File

@ -42,7 +42,7 @@ class Bitrate {
WebRtc_UWord32 BitrateNow() const;
protected:
Clock& clock_;
Clock* clock_;
private:
WebRtc_UWord32 packet_rate_;

1
webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
View File

@ -99,7 +99,6 @@ void RtcpFormatRembTest::TearDown() {
delete rtcp_receiver_;
delete dummy_rtp_rtcp_impl_;
delete test_transport_;
delete system_clock_;
}
TEST_F(RtcpFormatRembTest, TestBasicAPI) {

28
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -34,7 +34,7 @@ RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id, Clock* clock,
ModuleRtpRtcpImpl* owner)
: TMMBRHelp(),
_id(id),
_clock(*clock),
_clock(clock),
_method(kRtcpOff),
_lastReceived(0),
_rtpRtcp(*owner),
@ -285,7 +285,7 @@ RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
{
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
_lastReceived = _clock.TimeInMilliseconds();
_lastReceived = _clock->TimeInMilliseconds();
RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
while (pktType != RTCPUtility::kRtcpNotValidCode)
@ -406,7 +406,7 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
_remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount;
_remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount;
_clock.CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
_clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
}
else
{
@ -466,7 +466,7 @@ RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
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 @@ RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
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 @@ RTCPReceiver::GetReceiveInformation(WebRtc_UWord32 remoteSSRC) {
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 @@ bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
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 @@ bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
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 @@ bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
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 @@ RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0)
{
receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem,
_clock.TimeInMilliseconds());
_clock->TimeInMilliseconds());
rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr;
}
}
@ -1150,7 +1150,7 @@ void RTCPReceiver::HandleFIRItem(RTCPReceiveInformation* receiveInfo,
// 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 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
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 @@ WebRtc_Word32 RTCPReceiver::TMMBRReceived(const WebRtc_UWord32 size,
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 @@ void RTCPReceiver::PacketTimeout()
return;
}
WebRtc_Word64 now = _clock.TimeInMilliseconds();
WebRtc_Word64 now = _clock->TimeInMilliseconds();
if(now - _lastReceived > _packetTimeOutMS)
{
packetTimeOut = true;

2
webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
View File

@ -198,7 +198,7 @@ protected:
private:
WebRtc_Word32 _id;
Clock& _clock;
Clock* _clock;
RTCPMethod _method;
WebRtc_Word64 _lastReceived;
ModuleRtpRtcpImpl& _rtpRtcp;

28
webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
View File

@ -168,30 +168,29 @@ class RtcpReceiverTest : public ::testing::Test {
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 @@ class RtcpReceiverTest : public ::testing::Test {
}
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 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
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 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
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 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
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 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
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 @@ TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
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 @@ TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
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.

20
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -29,7 +29,7 @@ RTCPSender::RTCPSender(const WebRtc_Word32 id,
ModuleRtpRtcpImpl* owner) :
_id(id),
_audio(audio),
_clock(*clock),
_clock(clock),
_method(kRtcpOff),
_rtpRtcp(*owner),
_criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
@ -183,11 +183,11 @@ RTCPSender::SetRTCPStatus(const RTCPMethod method)
{
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 @@ void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
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 @@ RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc)
// 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 @@ From RFC 3550
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 @@ RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer,
// 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 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
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 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
} 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 @@ RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
}
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

2
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -186,7 +186,7 @@ private:
private:
WebRtc_Word32 _id;
const bool _audio;
Clock& _clock;
Clock* _clock;
RTCPMethod _method;
ModuleRtpRtcpImpl& _rtpRtcp;

1
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -129,7 +129,6 @@ class RtcpSenderTest : public ::testing::Test {
delete rtcp_receiver_;
delete rtp_rtcp_impl_;
delete test_transport_;
delete system_clock_;
}
// Helper function: Incoming RTCP has a specific packet type.

8
webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
View File

@ -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 @@ int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
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 @@ bool RTPPacketHistory::GetRTPPacket(uint16_t sequence_number,
}
// 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 @@ void RTPPacketHistory::UpdateResendTime(uint16_t sequence_number) {
"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

2
webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
View File

@ -77,7 +77,7 @@ class RTPPacketHistory {
bool FindSeqNum(uint16_t sequence_number, int32_t* index) const;
private:
Clock& clock_;
Clock* clock_;
CriticalSectionWrapper* critsect_;
bool store_;
uint32_t prev_index_;

37
webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
View File

@ -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, ReplaceRtpHeader) {
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 @@ TEST_F(RtpPacketHistoryTest, MinResendTime) {
capture_time_ms, kAllowRetransmission));
hist_->UpdateResendTime(kSeqNum);
fake_clock_.IncrementTime(100);
fake_clock_.AdvanceTimeMilliseconds(100);
// Time has elapsed.
len = kMaxPacketLength;

17
webrtc/modules/rtp_rtcp/source/rtp_receiver.cc
View File

@ -171,7 +171,7 @@ void RTPReceiver::PacketTimeout() {
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 RTPReceiver::IncomingRTPPacket(
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 @@ WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
// 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 @@ void RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtp_header,
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 @@ bool RTPReceiver::RetransmitOfOldPacket(
}
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 @@ WebRtc_Word32 RTPReceiver::EstimatedRemoteTimeStamp(
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;

19
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -83,7 +83,6 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
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 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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 @@ ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
plot1_ = NULL;
}
#endif
if (owns_clock_) {
delete &clock_;
}
}
void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
@ -217,13 +212,13 @@ void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* remove_module) {
// 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 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
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 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
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 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
#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 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nack_list,
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;

9
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -454,12 +454,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
bool owns_clock_;
Clock& clock_;
Clock* clock_;
private:
int64_t RtcpReportInterval();

14
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -75,14 +75,14 @@ RTPSender::RTPSender(const WebRtc_Word32 id,
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 @@ int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
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 @@ void RTPSender::TimeToSendPacket(uint16_t sequence_number,
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 @@ WebRtc_Word32 RTPSender::SendToNetwork(uint8_t* buffer,
// 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 @@ void RTPSender::SetSendingStatus(const bool enabled) {
} 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

8
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -17,7 +17,7 @@ namespace webrtc {
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 @@ RTPSenderAudio::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
telephoneEvent = _dtmfKey;
return true;
}
WebRtc_Word64 delaySinceLastDTMF = _clock.TimeInMilliseconds() -
WebRtc_Word64 delaySinceLastDTMF = _clock->TimeInMilliseconds() -
_dtmfTimeLastSent;
if(delaySinceLastDTMF < 100)
{
@ -246,7 +246,7 @@ WebRtc_Word32 RTPSenderAudio::SendAudio(
if (!_dtmfEventIsOn && PendingDTMF()) {
CriticalSectionScoped cs(_sendAudioCritsect);
WebRtc_Word64 delaySinceLastDTMF = _clock.TimeInMilliseconds() -
WebRtc_Word64 delaySinceLastDTMF = _clock->TimeInMilliseconds() -
_dtmfTimeLastSent;
if (delaySinceLastDTMF > 100) {
@ -297,7 +297,7 @@ WebRtc_Word32 RTPSenderAudio::SendAudio(
} else {
ended = true;
_dtmfEventIsOn = false;
_dtmfTimeLastSent = _clock.TimeInMilliseconds();
_dtmfTimeLastSent = _clock->TimeInMilliseconds();
}
// don't hold the critsect while calling SendTelephoneEventPacket
_sendAudioCritsect->Leave();

2
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
View File

@ -89,7 +89,7 @@ protected:
private:
WebRtc_Word32 _id;
Clock& _clock;
Clock* _clock;
RTPSenderInterface* _rtpSender;
CriticalSectionWrapper* _audioFeedbackCritsect;
RtpAudioFeedback* _audioFeedback;

33
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -33,33 +33,6 @@ const int kTimeOffset = 22222;
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 LoopbackTransportTest : public webrtc::Transport {
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 @@ TEST_F(RtpSenderTest, DISABLED_TrafficSmoothing) {
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_);

4
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -22,7 +22,7 @@ using namespace webrtc;
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest() {
RtpRtcpAPITest() : module(NULL), fake_clock(123456) {
test_CSRC[0] = 1234;
test_CSRC[1] = 2345;
test_id = 123;
@ -50,7 +50,7 @@ class RtpRtcpAPITest : public ::testing::Test {
WebRtc_UWord32 test_timestamp;
WebRtc_UWord16 test_sequence_number;
WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
FakeRtpRtcpClock fake_clock;
SimulatedClock fake_clock;
};
TEST_F(RtpRtcpAPITest, Basic) {

25
webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
View File

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

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

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

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

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

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

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

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

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

4
webrtc/modules/video_coding/main/interface/video_coding.h
View File

@ -19,7 +19,7 @@
namespace webrtc
{
class TickTimeBase;
class Clock;
class VideoEncoder;
class VideoDecoder;
struct CodecSpecificInfo;
@ -49,7 +49,7 @@ public:
static VideoCodingModule* Create(const WebRtc_Word32 id);
static VideoCodingModule* Create(const WebRtc_Word32 id,
TickTimeBase* clock);
Clock* clock);
static void Destroy(VideoCodingModule* module);

6
webrtc/modules/video_coding/main/source/generic_decoder.cc
View File

@ -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 @@ WebRtc_Word32 VCMDecodedFrameCallback::Decoded(I420VideoFrame& decodedImage)
_timing.StopDecodeTimer(
decodedImage.timestamp(),
frameInfo->decodeStartTimeMs,
_clock->MillisecondTimestamp());
_clock->TimeInMilliseconds());
if (_receiveCallback != NULL)
{

4
webrtc/modules/video_coding/main/source/generic_decoder.h
View File

@ -34,7 +34,7 @@ struct VCMFrameInformation
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 @@ public:
private:
CriticalSectionWrapper* _critSect;
TickTimeBase* _clock;
Clock* _clock;
I420VideoFrame _frame;
VCMReceiveCallback* _receiveCallback;
VCMTiming& _timing;

43
webrtc/modules/video_coding/main/source/jitter_buffer.cc
View File

@ -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 @@ class CompleteDecodableKeyFrameCriteria {
}
};
VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
int vcm_id,
int receiver_id,
bool master)
@ -90,7 +90,7 @@ VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
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 @@ void VCMJitterBuffer::Start() {
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 @@ void VCMJitterBuffer::Flush() {
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 @@ void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
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 @@ void VCMJitterBuffer::IncomingRateStatistics(unsigned int* framerate,
} 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 @@ VCMEncodedFrame* VCMJitterBuffer::GetCompleteFrameForDecoding(
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 @@ VCMEncodedFrame* VCMJitterBuffer::GetCompleteFrameForDecoding(
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 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(VCMEncodedFrame* encoded_frame,
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 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(VCMEncodedFrame* encoded_frame,
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;
}

6
webrtc/modules/video_coding/main/source/jitter_buffer.h
View File

@ -35,7 +35,7 @@ enum VCMNackMode {
typedef std::list<VCMFrameBuffer*> FrameList;
// forward declarations
class TickTimeBase;
class Clock;
class VCMFrameBuffer;
class VCMPacket;
class VCMEncodedFrame;
@ -49,7 +49,7 @@ struct VCMJitterSample {
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 @@ class VCMJitterBuffer {
int vcm_id_;
int receiver_id_;
TickTimeBase* clock_;
Clock* clock_;
// If we are running (have started) or not.
bool running_;
CriticalSectionWrapper* crit_sect_;

29
webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
View File

@ -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 @@ class TestRunningJitterBuffer : public ::testing::Test {
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 @@ class TestRunningJitterBuffer : public ::testing::Test {
jitter_buffer_->Stop();
delete stream_generator;
delete jitter_buffer_;
delete clock_;
}
VCMFrameBufferEnum InsertPacketAndPop(int index) {
@ -190,9 +189,9 @@ class TestRunningJitterBuffer : public ::testing::Test {
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 @@ class TestRunningJitterBuffer : public ::testing::Test {
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 @@ class TestRunningJitterBuffer : public ::testing::Test {
VCMJitterBuffer* jitter_buffer_;
StreamGenerator* stream_generator;
FakeTickTime* clock_;
scoped_ptr<SimulatedClock> clock_;
uint8_t data_buffer_[kDataBufferSize];
};
@ -258,7 +257,7 @@ TEST_F(TestRunningJitterBuffer, TestFull) {
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 @@ TEST_F(TestJitterBufferNack, TestNormalOperation) {
// | 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 @@ TEST_F(TestJitterBufferNack, TestNormalOperationWrap) {
// ------- ------------------------------------------------------------
// | 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)

30
webrtc/modules/video_coding/main/source/media_optimization.cc
View File

@ -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 @@ _numLayers(0)
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 @@ VCMMediaOptimization::Reset()
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 @@ VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
// 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 @@ VCMMediaOptimization::SetEncodingData(VideoCodecType sendCodecType,
// 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 @@ VCMMediaOptimization::SentFrameRate()
float
VCMMediaOptimization::SentBitRate()
{
UpdateBitRateEstimate(-1, _clock->MillisecondTimestamp());
UpdateBitRateEstimate(-1, _clock->TimeInMilliseconds());
return _avgSentBitRateBps / 1000.0f;
}
@ -374,7 +374,7 @@ VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
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 @@ VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
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 @@ VCMMediaOptimization::SelectQuality()
_qmResolution->ResetRates();
// Reset counters
_lastQMUpdateTime = _clock->MillisecondTimestamp();
_lastQMUpdateTime = _clock->TimeInMilliseconds();
// Reset content metrics
_content->Reset();
@ -567,7 +567,7 @@ VCMMediaOptimization::checkStatusForQMchange()
// (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 @@ bool VCMMediaOptimization::QMUpdate(VCMResolutionScale* qm) {
void
VCMMediaOptimization::UpdateIncomingFrameRate()
{
WebRtc_Word64 now = _clock->MillisecondTimestamp();
WebRtc_Word64 now = _clock->TimeInMilliseconds();
if (_incomingFrameTimes[0] == 0)
{
// first no shift
@ -667,7 +667,7 @@ VCMMediaOptimization::ProcessIncomingFrameRate(WebRtc_Word64 now)
WebRtc_UWord32
VCMMediaOptimization::InputFrameRate()
{
ProcessIncomingFrameRate(_clock->MillisecondTimestamp());
ProcessIncomingFrameRate(_clock->TimeInMilliseconds());
return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
}

6
webrtc/modules/video_coding/main/source/media_optimization.h
View File

@ -23,7 +23,7 @@ namespace webrtc
enum { kBitrateMaxFrameSamples = 60 };
enum { kBitrateAverageWinMs = 1000 };
class TickTimeBase;
class Clock;
class VCMContentMetricsProcessing;
class VCMFrameDropper;
@ -38,7 +38,7 @@ struct VCMEncodedFrameSample
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 @@ private:
enum { kFrameHistoryWinMs = 2000};
WebRtc_Word32 _id;
TickTimeBase* _clock;
Clock* _clock;
WebRtc_Word32 _maxBitRate;
VideoCodecType _sendCodecType;
WebRtc_UWord16 _codecWidth;

47
webrtc/modules/video_coding/main/source/mock/fake_tick_time.h
View File

@ -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_

26
webrtc/modules/video_coding/main/source/receiver.cc
View File

@ -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 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, uint16_t frame_width,
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 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, uint16_t frame_width,
// 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 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, uint16_t frame_width,
"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 @@ int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, uint16_t frame_width,
"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 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
// 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 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
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 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
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 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
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 @@ VCMEncodedFrame* VCMReceiver::FrameForRendering(uint16_t max_wait_time_ms,
// 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.

6
webrtc/modules/video_coding/main/source/receiver.h
View File

@ -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 @@ enum VCMReceiverState {
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 @@ class VCMReceiver {
CriticalSectionWrapper* crit_sect_;
int32_t vcm_id_;
TickTimeBase* clock_;
Clock* clock_;
int32_t receiver_id_;
bool master_;
VCMJitterBuffer jitter_buffer_;

36
webrtc/modules/video_coding/main/source/tick_time_base.h
View File

@ -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_

8
webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc
View File

@ -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 @@ _accDrift(6600), // in timestamp ticks, i.e. 15 ms
_accMaxError(7000),
_P11(1e10)
{
Reset(_clock->MillisecondTimestamp());
Reset(_clock->TimeInMilliseconds());
}
VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
@ -56,7 +56,7 @@ VCMTimestampExtrapolator::Reset(const WebRtc_Word64 nowMs /* = -1 */)
}
else
{
_startMs = _clock->MillisecondTimestamp();
_startMs = _clock->TimeInMilliseconds();
}
_prevMs = _startMs;
_firstTimestamp = 0;

14
webrtc/modules/video_coding/main/source/timestamp_extrapolator.h
View File

@ -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 @@ private:
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;

14
webrtc/modules/video_coding/main/source/timing.cc
View File

@ -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)

6
webrtc/modules/video_coding/main/source/timing.h
View File

@ -18,7 +18,7 @@
namespace webrtc
{
class TickTimeBase;
class Clock;
class VCMTimestampExtrapolator;
class VCMTiming
@ -26,7 +26,7 @@ class VCMTiming
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 @@ protected:
private:
CriticalSectionWrapper* _critSect;
WebRtc_Word32 _vcmId;
TickTimeBase* _clock;
Clock* _clock;
WebRtc_Word32 _timingId;
bool _master;
VCMTimestampExtrapolator* _tsExtrapolator;

1
webrtc/modules/video_coding/main/source/video_coding.gypi
View File

@ -62,7 +62,6 @@
'receiver.h',
'rtt_filter.h',
'session_info.h',
'tick_time_base.h',
'timestamp_extrapolator.h',
'timestamp_map.h',
'timing.h',

23
webrtc/modules/video_coding/main/source/video_coding_impl.cc
View File

@ -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 @@ VCMProcessTimer::TimeUntilProcess() const
{
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 @@ VideoCodingModuleImpl::~VideoCodingModuleImpl()
}
delete _receiveCritSect;
delete _sendCritSect;
if (delete_clock_on_destroy_) delete clock_;
#ifdef DEBUG_DECODER_BIT_STREAM
fclose(_bitStreamBeforeDecoder);
#endif
@ -112,14 +109,14 @@ VideoCodingModuleImpl::~VideoCodingModuleImpl()
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 @@ VideoCodingModuleImpl::Decode(WebRtc_UWord16 maxWaitTimeMs)
// 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 @@ VideoCodingModuleImpl::DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs)
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 @@ VideoCodingModuleImpl::Decode(const VCMEncodedFrame& frame)
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)

32
webrtc/modules/video_coding/main/source/video_coding_impl.h
View File

@ -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 @@ namespace webrtc
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 @@ class VideoCodingModuleImpl : public VideoCodingModule
{
public:
VideoCodingModuleImpl(const WebRtc_Word32 id,
TickTimeBase* clock,
bool delete_clock_on_destroy);
Clock* clock);
virtual ~VideoCodingModuleImpl();
@ -275,8 +274,7 @@ protected:
private:
WebRtc_Word32 _id;
TickTimeBase* clock_;
bool delete_clock_on_destroy_;
Clock* clock_;
CriticalSectionWrapper* _receiveCritSect;
bool _receiverInited;
VCMTiming _timing;

42
webrtc/modules/video_coding/main/source/video_coding_robustness_unittest.cc
View File

@ -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 @@ class VCMRobustnessTest : public ::testing::Test {
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 @@ class VCMRobustnessTest : public ::testing::Test {
virtual void TearDown() {
VideoCodingModule::Destroy(vcm_);
delete clock_;
}
void InsertPacket(uint32_t timestamp,
@ -77,7 +76,7 @@ class VCMRobustnessTest : public ::testing::Test {
MockPacketRequestCallback request_callback_;
NiceMock<MockVideoDecoder> decoder_;
NiceMock<MockVideoDecoder> decoderCopy_;
FakeTickTime* clock_;
scoped_ptr<SimulatedClock> clock_;
};
TEST_F(VCMRobustnessTest, TestHardNack) {
@ -112,21 +111,21 @@ TEST_F(VCMRobustnessTest, TestHardNack) {
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 @@ TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
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 @@ TEST_F(VCMRobustnessTest, TestDualDecoder) {
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 @@ TEST_F(VCMRobustnessTest, TestDualDecoder) {
// 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 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
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 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithoutErrors) {
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 @@ TEST_F(VCMRobustnessTest, TestModeNoneWithoutErrors) {
// 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

12
webrtc/modules/video_coding/main/test/decode_from_storage_test.cc
View File

@ -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 @@ int DecodeFromStorageTest(CmdArgs& args)
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 @@ int DecodeFromStorageTest(CmdArgs& args)
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 @@ int DecodeFromStorageTest(CmdArgs& args)
{
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)

17
webrtc/modules/video_coding/main/test/generic_codec_test.cc
View File

@ -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 @@ int GenericCodecTest::RunTest(CmdArgs& args)
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 @@ int GenericCodecTest::RunTest(CmdArgs& args)
return 0;
}
GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm, FakeTickTime* clock):
GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm,
SimulatedClock* clock):
_clock(clock),
_vcm(vcm),
_width(0),
@ -332,10 +333,6 @@ GenericCodecTest::Perform(CmdArgs& args)
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 @@ GenericCodecTest::Print()
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 @@ GenericCodecTest::WaitForEncodedFrame() const
void
GenericCodecTest::IncrementDebugClock(float frameRate)
{
_clock->IncrementDebugClock(1000/frameRate);
_clock->AdvanceTimeMilliseconds(1000/frameRate);
}
int

6
webrtc/modules/video_coding/main/test/generic_codec_test.h
View File

@ -31,13 +31,13 @@ namespace webrtc {
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 @@ private:
WebRtc_Word32 TearDown();
void IncrementDebugClock(float frameRate);
webrtc::FakeTickTime* _clock;
webrtc::SimulatedClock* _clock;
webrtc::VideoCodingModule* _vcm;
webrtc::VideoCodec _sendCodec;
webrtc::VideoCodec _receiveCodec;

6
webrtc/modules/video_coding/main/test/jitter_buffer_test.cc
View File

@ -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 @@ int JitterBufferTest(CmdArgs& args)
#if defined(EVENT_DEBUG)
return -1;
#endif
TickTimeBase clock;
Clock* clock = Clock::GetRealTimeClock();
// Start test
WebRtc_UWord16 seqNum = 1234;
@ -106,7 +106,7 @@ int JitterBufferTest(CmdArgs& args)
WebRtc_UWord8 data[1500];
VCMPacket packet(data, size, seqNum, timeStamp, true);
VCMJitterBuffer jb(&clock);
VCMJitterBuffer jb(clock);
seqNum = 1234;
timeStamp = 123*90;

8
webrtc/modules/video_coding/main/test/media_opt_test.cc
View File

@ -32,9 +32,9 @@ int MediaOptTest::RunTest(int testNum, CmdArgs& args)
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 @@ int MediaOptTest::RunTest(int testNum, CmdArgs& args)
}
MediaOptTest::MediaOptTest(VideoCodingModule* vcm, TickTimeBase* clock)
MediaOptTest::MediaOptTest(VideoCodingModule* vcm, Clock* clock)
: _vcm(vcm),
_rtp(NULL),
_outgoingTransport(NULL),

4
webrtc/modules/video_coding/main/test/media_opt_test.h
View File

@ -34,7 +34,7 @@ class MediaOptTest
{
public:
MediaOptTest(webrtc::VideoCodingModule* vcm,
webrtc::TickTimeBase* clock);
webrtc::Clock* clock);
~MediaOptTest();
static int RunTest(int testNum, CmdArgs& args);
@ -57,7 +57,7 @@ private:
webrtc::RTPSendCompleteCallback* _outgoingTransport;
RtpDataCallback* _dataCallback;
webrtc::TickTimeBase* _clock;
webrtc::Clock* _clock;
std::string _inname;
std::string _outname;
std::string _actualSourcename;

6
webrtc/modules/video_coding/main/test/mt_rx_tx_test.cc
View File

@ -143,12 +143,12 @@ int MTRxTxTest(CmdArgs& args)
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;

12
webrtc/modules/video_coding/main/test/mt_test_common.cc
View File

@ -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 @@ TransportCallback::SendPacket(int channel, const void *data, int len)
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 @@ TransportCallback::TransportPackets()
{
// 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())
{

2
webrtc/modules/video_coding/main/test/mt_test_common.h
View File

@ -47,7 +47,7 @@ class TransportCallback:public RTPSendCompleteCallback
{
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

19
webrtc/modules/video_coding/main/test/normal_test.cc
View File

@ -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 @@ int NormalTest::RunTest(const CmdArgs& args)
{
#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 @@ VCMNTDecodeCompleCallback::DecodedBytes()
//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 @@ NormalTest::Perform(const CmdArgs& args)
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 @@ NormalTest::Perform(const CmdArgs& args)
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);

4
webrtc/modules/video_coding/main/test/normal_test.h
View File

@ -86,7 +86,7 @@ class NormalTest
{
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 @@ protected:
// 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;

11
webrtc/modules/video_coding/main/test/quality_modes_test.cc
View File

@ -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 @@ int qualityModeTest(const CmdArgs& args)
}
QualityModesTest::QualityModesTest(VideoCodingModule* vcm,
TickTimeBase* clock):
Clock* clock):
NormalTest(vcm, clock),
_vpm()
{
@ -367,8 +366,8 @@ QualityModesTest::Perform(const CmdArgs& args)
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);

2
webrtc/modules/video_coding/main/test/quality_modes_test.h
View File

@ -22,7 +22,7 @@ class QualityModesTest : public NormalTest
{
public:
QualityModesTest(webrtc::VideoCodingModule* vcm,
webrtc::TickTimeBase* clock);
webrtc::Clock* clock);
virtual ~QualityModesTest();
WebRtc_Word32 Perform(const CmdArgs& args);

4
webrtc/modules/video_coding/main/test/receiver_timing_tests.cc
View File

@ -61,8 +61,8 @@ int ReceiverTimingTests(CmdArgs& args)
// 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;

15
webrtc/modules/video_coding/main/test/rtp_player.cc
View File

@ -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 @@ void LostPackets::Print() const {
RTPPlayer::RTPPlayer(const char* filename,
RtpData* callback,
TickTimeBase* clock)
Clock* clock)
:
_clock(clock),
_rtpModule(NULL),
@ -273,7 +273,8 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
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 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
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 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
if (_firstPacket)
{
_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
_firstPacketTimeMs = _clock->MillisecondTimestamp();
_firstPacketTimeMs = _clock->TimeInMilliseconds();
}
if (_reordering && _reorderBuffer == NULL)
{
@ -428,8 +429,8 @@ WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, We
for (int i=0; i < length; i++)
{
_lostPackets.SetResendTime(sequenceNumbers[i],
_clock->MillisecondTimestamp() + _rttMs,
_clock->MillisecondTimestamp());
_clock->TimeInMilliseconds() + _rttMs,
_clock->TimeInMilliseconds());
}
return 0;
}

6
webrtc/modules/video_coding/main/test/rtp_player.h
View File

@ -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 @@ class RTPPlayer : public webrtc::VCMPacketRequestCallback
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 @@ private:
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;

6
webrtc/modules/video_coding/main/test/test_callbacks.cc
View File

@ -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 @@ VCMDecodeCompleteCallback::DecodedBytes()
return _decodedBytes;
}
RTPSendCompleteCallback::RTPSendCompleteCallback(TickTimeBase* clock,
RTPSendCompleteCallback::RTPSendCompleteCallback(Clock* clock,
const char* filename):
_clock(clock),
_sendCount(0),
@ -258,7 +258,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
bool transmitPacket = true;
transmitPacket = PacketLoss();
WebRtc_UWord64 now = _clock->MillisecondTimestamp();
int64_t now = _clock->TimeInMilliseconds();
// Insert outgoing packet into list
if (transmitPacket)
{

4
webrtc/modules/video_coding/main/test/test_callbacks.h
View File

@ -157,7 +157,7 @@ class RTPSendCompleteCallback: public Transport
{
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 @@ protected:
// Random uniform loss model
bool UnifomLoss(double lossPct);
TickTimeBase* _clock;
Clock* _clock;
WebRtc_UWord32 _sendCount;
RtpRtcp* _rtp;
double _lossPct;

12
webrtc/modules/video_coding/main/test/video_rtp_play.cc
View File

@ -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 @@ int RtpPlay(CmdArgs& args)
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 @@ int RtpPlay(CmdArgs& args)
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 @@ int RtpPlay(CmdArgs& args)
{
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

37
webrtc/modules/video_coding/main/test/video_rtp_play_mt.cc
View File

@ -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 @@ bool RtpReaderThread(void* obj)
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 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
(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 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
}
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 @@ int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTe
}
// 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");

5
webrtc/system_wrappers/interface/clock.h
View File

@ -43,7 +43,6 @@ class Clock {
class SimulatedClock : public Clock {
public:
SimulatedClock();
explicit SimulatedClock(int64_t initial_time_us);
virtual ~SimulatedClock() {}
@ -61,8 +60,8 @@ class SimulatedClock : public Clock {
// Advance the simulated clock with a given number of milliseconds or
// microseconds.
void AdvanceTimeMs(int64_t milliseconds);
void AdvanceTimeUs(int64_t microseconds);
void AdvanceTimeMilliseconds(int64_t milliseconds);
void AdvanceTimeMicroseconds(int64_t microseconds);
private:
int64_t time_us_;

14
webrtc/system_wrappers/source/clock.cc
View File

@ -220,16 +220,16 @@ static WindowsHelpTimer global_help_timer = {0, 0, {{ 0, 0}, 0}, 0};
Clock* Clock::GetRealTimeClock() {
#if defined(_WIN32)
return new WindowsRealTimeClock(&global_help_timer);
static WindowsRealTimeClock clock(&global_help_timer);
return &clock;
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
return new UnixRealTimeClock();
static UnixRealTimeClock clock;
return &clock;
#else
return NULL;
#endif
}
SimulatedClock::SimulatedClock() : time_us_(0) {}
SimulatedClock::SimulatedClock(int64_t initial_time_us)
: time_us_(initial_time_us) {}
@ -247,11 +247,11 @@ void SimulatedClock::CurrentNtp(uint32_t& seconds, uint32_t& fractions) {
kMagicNtpFractionalUnit / 1000);
}
void SimulatedClock::AdvanceTimeMs(int64_t milliseconds) {
AdvanceTimeUs(1000 * milliseconds);
void SimulatedClock::AdvanceTimeMilliseconds(int64_t milliseconds) {
AdvanceTimeMicroseconds(1000 * milliseconds);
}
void SimulatedClock::AdvanceTimeUs(int64_t microseconds) {
void SimulatedClock::AdvanceTimeMicroseconds(int64_t microseconds) {
time_us_ += microseconds;
}