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Update talk to 61699344.

Browse Source 
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
This commit is contained in:
mallinath@webrtc.org 2014-02-17 18:49:41 +00:00
parent e3842897e2
commit 92fdfebedd
5 changed files with 33 additions and 5 deletions
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10
talk/base/asyncpacketsocket.h
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@ -45,10 +45,12 @@ struct PacketTimeUpdateParams {
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

4
talk/media/base/mediachannel.h
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@ -539,6 +539,10 @@ class MediaChannel : public sigslot::has_slots<> {
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.

11
talk/media/webrtc/webrtcvideoengine.cc
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@ -2467,7 +2467,7 @@ bool WebRtcVideoMediaChannel::GetStats(const StatsOptions& options,
// 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 @@ bool WebRtcVideoMediaChannel::SetSendRtpHeaderExtensions(
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";

1
talk/media/webrtc/webrtcvideoengine.h
View File

@ -273,6 +273,7 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
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);

12
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@ -1886,6 +1886,18 @@ TEST_F(WebRtcVideoMediaChannelTest, SetRecvCodecsUnsupportedCodec) {
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();
}