Delete Timing class, timing.h, and update all users.
BUG=webrtc:6324
Review-Url: https://codereview.webrtc.org/2290203002
Cr-Commit-Position: refs/heads/master@{#14203}
diff --git a/webrtc/media/base/rtpdataengine.cc b/webrtc/media/base/rtpdataengine.cc
index 4b6647c..99aa3b1 100644
--- a/webrtc/media/base/rtpdataengine.cc
+++ b/webrtc/media/base/rtpdataengine.cc
@@ -14,7 +14,6 @@
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ratelimiter.h"
-#include "webrtc/base/timing.h"
#include "webrtc/media/base/codec.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/rtputils.h"
@@ -37,7 +36,6 @@
RtpDataEngine::RtpDataEngine() {
data_codecs_.push_back(
DataCodec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName));
- SetTiming(new rtc::Timing());
}
DataMediaChannel* RtpDataEngine::CreateChannel(
@@ -45,7 +43,7 @@
if (data_channel_type != DCT_RTP) {
return NULL;
}
- return new RtpDataMediaChannel(timing_.get());
+ return new RtpDataMediaChannel();
}
bool FindCodecByName(const std::vector<DataCodec>& codecs,
@@ -60,18 +58,13 @@
return false;
}
-RtpDataMediaChannel::RtpDataMediaChannel(rtc::Timing* timing) {
- Construct(timing);
-}
-
RtpDataMediaChannel::RtpDataMediaChannel() {
- Construct(NULL);
+ Construct();
}
-void RtpDataMediaChannel::Construct(rtc::Timing* timing) {
+void RtpDataMediaChannel::Construct() {
sending_ = false;
receiving_ = false;
- timing_ = timing;
send_limiter_.reset(new rtc::RateLimiter(kDataMaxBandwidth / 8, 1.0));
}
@@ -313,7 +306,8 @@
return false;
}
- double now = timing_->TimerNow();
+ double now =
+ rtc::TimeMicros() / static_cast<double>(rtc::kNumMicrosecsPerSec);
if (!send_limiter_->CanUse(packet_len, now)) {
LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len