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Revert r4328

Browse Source 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
elham@webrtc.org 2013-07-15 20:59:52 +00:00
parent 8543c1c77c
commit 6f5707e184
5 changed files with 9 additions and 26 deletions
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13
webrtc/modules/rtp_rtcp/interface/receive_statistics.h
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@ -69,18 +69,5 @@ class ReceiveStatistics : public Module {
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
};
class NullReceiveStatistics : public ReceiveStatistics {
public:
virtual void IncomingPacket(const RTPHeader& rtp_header, size_t bytes,
bool retransmitted, bool in_order) {}
virtual void GetActiveStatisticians(
StatisticianMap* statisticians) const { statisticians->clear(); }
virtual StreamStatistician* GetStatistician(uint32_t ssrc) const {
return NULL;
}
virtual int32_t TimeUntilNextProcess() { return 0; }
virtual int32_t Process() { return 0; }
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_

13
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -1689,12 +1689,14 @@ int RTCPSender::PrepareRTCP(
// If the data does not fit in the packet we fill it as much as possible.
int32_t buildVal = 0;
// We need to send our NTP even if we haven't received any reports.
_clock->CurrentNtp(NTPsec, NTPfrac);
if (ShouldSendReportBlocks(rtcpPacketTypeFlags)) {
ReceiveStatistics::StatisticianMap statisticians;
receive_statistics_->GetActiveStatisticians(&statisticians);
if (!statisticians.empty()) {
if (statisticians.empty()) {
// We need to send our NTP even if we dont have received any
// reports.
_clock->CurrentNtp(NTPsec, NTPfrac);
} else {
ReceiveStatistics::StatisticianMap::const_iterator it;
int i;
for (it = statisticians.begin(), i = 0; it != statisticians.end();
@ -1703,8 +1705,9 @@ int RTCPSender::PrepareRTCP(
if (PrepareReport(it->second, &report_block, &NTPsec, &NTPfrac))
AddReportBlock(it->first, &internal_report_blocks_, &report_block);
}
if (_IJ && !statisticians.empty()) {
rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
if (_IJ && !statisticians.empty())
{
rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
}
_lastRTCPTime[0] = Clock::NtpToMs(NTPsec, NTPfrac);
}

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -41,7 +41,7 @@ RtpRtcp::Configuration::Configuration()
audio(false),
clock(NULL),
default_module(NULL),
receive_statistics(NullObjectReceiveStatistics()),
receive_statistics(),
outgoing_transport(NULL),
rtcp_feedback(NULL),
intra_frame_callback(NULL),

5
webrtc/modules/rtp_rtcp/source/rtp_utility.cc
View File

@ -61,11 +61,6 @@ RtpAudioFeedback* NullObjectRtpAudioFeedback() {
return &null_rtp_audio_feedback;
}
ReceiveStatistics* NullObjectReceiveStatistics() {
static NullReceiveStatistics null_receive_statistics;
return &null_receive_statistics;
}
namespace ModuleRTPUtility {
enum {

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

@ -14,7 +14,6 @@
#include <cstddef> // size_t, ptrdiff_t
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/typedefs.h"
@ -26,7 +25,6 @@ const uint8_t kRtpMarkerBitMask = 0x80;
RtpData* NullObjectRtpData();
RtpFeedback* NullObjectRtpFeedback();
RtpAudioFeedback* NullObjectRtpAudioFeedback();
ReceiveStatistics* NullObjectReceiveStatistics();
namespace ModuleRTPUtility
{