Index: webrtc/media/base/rtpdataengine.cc |
diff --git a/webrtc/media/base/rtpdataengine.cc b/webrtc/media/base/rtpdataengine.cc |
index 4b6647c649acc869fd2815f754dc548f3d9d8267..99aa3b14a5690fdee1394875019ee98aeae9f0e0 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 @@ static const size_t kMaxSrtpHmacOverhead = 16; |
RtpDataEngine::RtpDataEngine() { |
data_codecs_.push_back( |
DataCodec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName)); |
- SetTiming(new rtc::Timing()); |
} |
DataMediaChannel* RtpDataEngine::CreateChannel( |
@@ -45,7 +43,7 @@ DataMediaChannel* RtpDataEngine::CreateChannel( |
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 @@ bool FindCodecByName(const std::vector<DataCodec>& codecs, |
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 @@ bool RtpDataMediaChannel::SendData( |
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 |