Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(448)

Unified Diff: talk/session/media/channel.h

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add missing updated_options Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: talk/session/media/channel.h
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index d7f93c73b4e7f82c2602726c7fa271d4fae05888..27088c921c5ccabb2d586c3d684d2d4bad13ddc7 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -199,9 +199,8 @@ class BaseChannel
// 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 @@ class BaseChannel
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 @@ class BaseChannel
// 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 @@ class VideoChannel : public BaseChannel {
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:

Powered by Google App Engine
This is Rietveld 408576698