Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index d7f93c7..27088c9 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -199,9 +199,8 @@
// NetworkInterface implementation, called by MediaEngine
virtual bool SendPacket(rtc::Buffer* packet,
- rtc::DiffServCodePoint dscp);
- virtual bool SendRtcp(rtc::Buffer* packet,
- rtc::DiffServCodePoint dscp);
+ const rtc::PacketOptions& options);
+ virtual bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options);
// From TransportChannel
void OnWritableState(TransportChannel* channel);
@@ -214,8 +213,9 @@
bool PacketIsRtcp(const TransportChannel* channel, const char* data,
size_t len);
- bool SendPacket(bool rtcp, rtc::Buffer* packet,
- rtc::DiffServCodePoint dscp);
+ bool SendPacket(bool rtcp,
+ rtc::Buffer* packet,
+ const rtc::PacketOptions& options);
virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
void HandlePacket(bool rtcp, rtc::Buffer* packet,
const rtc::PacketTime& packet_time);
@@ -261,7 +261,7 @@
// Helper method to get RTP Absoulute SendTime extension header id if
// present in remote supported extensions list.
void MaybeCacheRtpAbsSendTimeHeaderExtension(
- const std::vector<RtpHeaderExtension>& extensions);
+ const std::vector<RtpHeaderExtension>& extensions);
bool CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
bool* dtls,
@@ -470,8 +470,6 @@
bool SendIntraFrame();
bool RequestIntraFrame();
- // Configure sending media on the stream with SSRC |ssrc|
- // If there is only one sending stream SSRC 0 can be used.
bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
private:
