Update talk to 61699344.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/base/asyncpacketsocket.h b/talk/base/asyncpacketsocket.h
index 4fcc271..b1d5a4a 100644
--- a/talk/base/asyncpacketsocket.h
+++ b/talk/base/asyncpacketsocket.h
@@ -45,10 +45,12 @@
srtp_packet_index(-1) {
}
- int rtp_sendtime_extension_id; // extension header id present in packet.
- Buffer srtp_auth_key; // Authentication key.
- int srtp_auth_tag_len; // Authentication tag length.
- int64 srtp_packet_index; // Required for Rtp Packet authentication.
+ int rtp_sendtime_extension_id; // extension header id present in packet.
+ std::vector<char> srtp_auth_key; // Authentication key.
+ int srtp_auth_tag_len; // Authentication tag length.
+ int64 srtp_packet_index; // Required for Rtp Packet authentication.
+ int payload_len; // Raw payload length, before any wrapping
+ // like TURN/GTURN.
};
// This structure holds meta information for the packet which is about to send
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 1a84704..7c42b70 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -539,6 +539,10 @@
const std::vector<RtpHeaderExtension>& extensions) = 0;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
+ // Returns the absoulte sendtime extension id value from media channel.
+ virtual int GetRtpSendTimeExtnId() const {
+ return -1;
+ }
// Sets the initial bandwidth to use when sending starts.
virtual bool SetStartSendBandwidth(int bps) = 0;
// Sets the maximum allowed bandwidth to use when sending data.
diff --git a/talk/media/webrtc/webrtcvideoengine.cc b/talk/media/webrtc/webrtcvideoengine.cc
index 9a94b88..3e6f928 100644
--- a/talk/media/webrtc/webrtcvideoengine.cc
+++ b/talk/media/webrtc/webrtcvideoengine.cc
@@ -2467,7 +2467,7 @@
// Only call for the default channel because the returned stats are
// collected for all the channels using the same estimator.
if (engine_->vie()->rtp()->GetReceiveBandwidthEstimatorStats(
- recv_channels_[0]->channel_id(), &additional_stats)) {
+ recv_channels_[0]->channel_id(), &additional_stats) == 0) {
bwe.total_received_propagation_delta_ms =
additional_stats.total_propagation_time_delta_ms;
bwe.recent_received_propagation_delta_ms.swap(
@@ -2664,6 +2664,15 @@
return true;
}
+int WebRtcVideoMediaChannel::GetRtpSendTimeExtnId() const {
+ const RtpHeaderExtension* send_time_extension = FindHeaderExtension(
+ send_extensions_, kRtpAbsoluteSendTimeHeaderExtension);
+ if (send_time_extension) {
+ return send_time_extension->id;
+ }
+ return -1;
+}
+
bool WebRtcVideoMediaChannel::SetStartSendBandwidth(int bps) {
LOG(LS_INFO) << "WebRtcVideoMediaChannel::SetStartSendBandwidth";
diff --git a/talk/media/webrtc/webrtcvideoengine.h b/talk/media/webrtc/webrtcvideoengine.h
index cb19142..668d760 100644
--- a/talk/media/webrtc/webrtcvideoengine.h
+++ b/talk/media/webrtc/webrtcvideoengine.h
@@ -273,6 +273,7 @@
const std::vector<RtpHeaderExtension>& extensions);
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions);
+ virtual int GetRtpSendTimeExtnId() const;
virtual bool SetStartSendBandwidth(int bps);
virtual bool SetMaxSendBandwidth(int bps);
virtual bool SetOptions(const VideoOptions &options);
diff --git a/talk/media/webrtc/webrtcvideoengine_unittest.cc b/talk/media/webrtc/webrtcvideoengine_unittest.cc
index f2af10a..f5622ef 100644
--- a/talk/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine_unittest.cc
@@ -1886,6 +1886,18 @@
EXPECT_FALSE(channel_->SetRecvCodecs(codecs));
}
+TEST_F(WebRtcVideoMediaChannelTest, GetRtpSendTimeExtension) {
+ // Enable RTP timestamp extension.
+ const int id = 12;
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ extensions.push_back(cricket::RtpHeaderExtension(
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", id));
+
+ // Verify the send extension id.
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ EXPECT_EQ(id, channel_->GetRtpSendTimeExtnId());
+}
+
TEST_F(WebRtcVideoMediaChannelTest, SetSend) {
Base::SetSend();
}
