Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1044004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
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@@ -83,7 +83,6 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
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configuration_copy.clock = Clock::GetRealTimeClock();
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ModuleRtpRtcpImpl* rtp_rtcp_instance =
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new ModuleRtpRtcpImpl(configuration_copy);
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rtp_rtcp_instance->OwnsClock();
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return rtp_rtcp_instance;
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}
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}
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@@ -102,8 +101,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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rtcp_sender_(configuration.id, configuration.audio, configuration.clock,
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this),
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rtcp_receiver_(configuration.id, configuration.clock, this),
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owns_clock_(false),
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clock_(*configuration.clock),
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clock_(configuration.clock),
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id_(configuration.id),
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audio_(configuration.audio),
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collision_detected_(false),
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@@ -168,9 +166,6 @@ ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
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plot1_ = NULL;
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}
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#endif
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if (owns_clock_) {
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delete &clock_;
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}
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}
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void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
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@@ -217,13 +212,13 @@ void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* remove_module) {
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// Returns the number of milliseconds until the module want a worker thread
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// to call Process.
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WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() {
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const WebRtc_Word64 now = clock_.TimeInMilliseconds();
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const WebRtc_Word64 now = clock_->TimeInMilliseconds();
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return kRtpRtcpMaxIdleTimeProcess - (now - last_process_time_);
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}
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// Process any pending tasks such as timeouts (non time critical events).
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WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
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const WebRtc_Word64 now = clock_.TimeInMilliseconds();
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const WebRtc_Word64 now = clock_->TimeInMilliseconds();
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last_process_time_ = now;
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if (now >=
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@@ -305,7 +300,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
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void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
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if (dead_or_alive_active_) {
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const WebRtc_Word64 now = clock_.TimeInMilliseconds();
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const WebRtc_Word64 now = clock_->TimeInMilliseconds();
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if (now > dead_or_alive_timeout_ms_ + dead_or_alive_last_timer_) {
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// RTCP is alive if we have received a report the last 12 seconds.
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dead_or_alive_last_timer_ += dead_or_alive_timeout_ms_;
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@@ -340,7 +335,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
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dead_or_alive_active_ = enable;
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dead_or_alive_timeout_ms_ = sample_time_seconds * 1000;
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// Trigger the first after one period.
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dead_or_alive_last_timer_ = clock_.TimeInMilliseconds();
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dead_or_alive_last_timer_ = clock_->TimeInMilliseconds();
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return 0;
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}
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@@ -1304,7 +1299,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
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#ifdef MATLAB
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if (plot1_ == NULL) {
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plot1_ = eng.NewPlot(new MatlabPlot());
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plot1_->AddTimeLine(30, "b", "lost", clock_.GetTimeInMS());
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plot1_->AddTimeLine(30, "b", "lost", clock_->TimeInMilliseconds());
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}
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plot1_->Append("lost", missing);
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plot1_->Plot();
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@@ -1520,7 +1515,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nack_list,
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if (wait_time == 5) {
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wait_time = 100; // During startup we don't have an RTT.
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}
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const WebRtc_Word64 now = clock_.TimeInMilliseconds();
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const WebRtc_Word64 now = clock_->TimeInMilliseconds();
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const WebRtc_Word64 time_limit = now - wait_time;
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WebRtc_UWord16 nackLength = size;
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WebRtc_UWord16 start_id = 0;
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