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(Auto)update libjingle 68379861-> 68445177

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-03 09:42:15 +00:00
parent 044bdacfef
commit b525a9d790
4 changed files with 22 additions and 2 deletions
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2
talk/app/webrtc/statscollector.cc
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@ -297,6 +297,8 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
info.decoding_cng);
report->AddValue(StatsReport::kStatsValueNameDecodingPLCCNG,
info.decoding_plc_cng);
report->AddValue(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
info.capture_start_ntp_time_ms);
}
void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {

7
talk/media/base/mediachannel.h
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@ -829,7 +829,8 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
decoding_normal(0),
decoding_plc(0),
decoding_cng(0),
decoding_plc_cng(0) {
decoding_plc_cng(0),
capture_start_ntp_time_ms(-1) {
}
int ext_seqnum;
@ -846,6 +847,8 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
int decoding_plc;
int decoding_cng;
int decoding_plc_cng;
// Estimated capture start time in NTP time in ms.
int64 capture_start_ntp_time_ms;
};
struct VideoSenderInfo : public MediaSenderInfo {
@ -912,7 +915,7 @@ struct VideoReceiverInfo : public MediaReceiverInfo {
render_delay_ms(0),
target_delay_ms(0),
current_delay_ms(0),
capture_start_ntp_time_ms(0) {
capture_start_ntp_time_ms(-1) {
}
std::vector<SsrcGroup> ssrc_groups;

12
talk/media/webrtc/webrtcvideoengine.cc
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@ -62,7 +62,9 @@
#include "talk/media/webrtc/webrtcvoiceengine.h"
#include "webrtc/experiments.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#ifdef WEBRTC_CHROMIUM_BUILD
#include "webrtc/system_wrappers/interface/field_trial.h"
#endif
#if !defined(LIBPEERCONNECTION_LIB)
#include "talk/media/webrtc/webrtcmediaengine.h"
@ -2514,6 +2516,15 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options,
send_codec_->maxBitrate, kMaxVideoBitrate);
}
sinfo.adapt_reason = send_channel->CurrentAdaptReason();
#ifdef USE_WEBRTC_DEV_BRANCH
webrtc::CpuOveruseMetrics metrics;
engine()->vie()->base()->GetCpuOveruseMetrics(channel_id, &metrics);
sinfo.capture_jitter_ms = metrics.capture_jitter_ms;
sinfo.avg_encode_ms = metrics.avg_encode_time_ms;
sinfo.encode_usage_percent = metrics.encode_usage_percent;
sinfo.capture_queue_delay_ms_per_s = metrics.capture_queue_delay_ms_per_s;
#else
sinfo.capture_jitter_ms = -1;
sinfo.avg_encode_ms = -1;
sinfo.encode_usage_percent = -1;
@ -2534,6 +2545,7 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options,
sinfo.encode_usage_percent = encode_usage_percent;
sinfo.capture_queue_delay_ms_per_s = capture_queue_delay_ms_per_s;
}
#endif
webrtc::RtcpPacketTypeCounter rtcp_sent;
webrtc::RtcpPacketTypeCounter rtcp_received;

3
talk/media/webrtc/webrtcvoiceengine.cc
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@ -3267,6 +3267,9 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
rinfo.packets_lost = cs.cumulativeLost;
rinfo.ext_seqnum = cs.extendedMax;
#ifdef USE_WEBRTC_DEV_BRANCH
rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
#endif
// Convert samples to milliseconds.
if (codec.plfreq / 1000 > 0) {
rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);