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Break out RtpClock to system_wrappers and make it more generic.

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The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-01-17 14:01:20 +00:00
parent 3b7feb2a5d
commit 20ed36dada
31 changed files with 468 additions and 379 deletions
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19
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@@ -80,7 +80,7 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
RtpRtcp::Configuration configuration_copy;
memcpy(&configuration_copy, &configuration,
sizeof(RtpRtcp::Configuration));
configuration_copy.clock = ModuleRTPUtility::GetSystemClock();
configuration_copy.clock = Clock::GetRealTimeClock();
ModuleRtpRtcpImpl* rtp_rtcp_instance =
new ModuleRtpRtcpImpl(configuration_copy);
rtp_rtcp_instance->OwnsClock();
@@ -107,9 +107,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
id_(configuration.id),
audio_(configuration.audio),
collision_detected_(false),
last_process_time_(configuration.clock->GetTimeInMS()),
last_bitrate_process_time_(configuration.clock->GetTimeInMS()),
last_packet_timeout_process_time_(configuration.clock->GetTimeInMS()),
last_process_time_(configuration.clock->TimeInMilliseconds()),
last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
last_packet_timeout_process_time_(
configuration.clock->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP.
critical_section_module_ptrs_(
CriticalSectionWrapper::CreateCriticalSection()),
@@ -216,13 +217,13 @@ 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_.GetTimeInMS();
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_.GetTimeInMS();
const WebRtc_Word64 now = clock_.TimeInMilliseconds();
last_process_time_ = now;
if (now >=
@@ -304,7 +305,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
if (dead_or_alive_active_) {
const WebRtc_Word64 now = clock_.GetTimeInMS();
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_;
@@ -339,7 +340,7 @@ 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_.GetTimeInMS();
dead_or_alive_last_timer_ = clock_.TimeInMilliseconds();
return 0;
}
@@ -1519,7 +1520,7 @@ 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_.GetTimeInMS();
const WebRtc_Word64 now = clock_.TimeInMilliseconds();
const WebRtc_Word64 time_limit = now - wait_time;
WebRtc_UWord16 nackLength = size;
WebRtc_UWord16 start_id = 0;