Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(281)

Unified Diff: talk/app/webrtc/rtpsenderinterface.h

Issue 1426443007: Revert of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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
« no previous file with comments | « talk/app/webrtc/rtpsender.cc ('k') | talk/app/webrtc/rtpsenderreceiver_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/app/webrtc/rtpsenderinterface.h
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index f54e8ca0905f3f9b2c652e73418e40f6bf2d46a3..fca98f21db5823397d789922bd21cc79b1d433b0 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,7 +35,6 @@
#include "talk/app/webrtc/proxy.h"
#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/session/media/mediasession.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -48,23 +47,9 @@
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
- // Used to set the SSRC of the sender, once a local description has been set.
- // If |ssrc| is 0, this indiates that the sender should disconnect from the
- // underlying transport (this occurs if the sender isn't seen in a local
- // description).
- virtual void SetSsrc(uint32_t ssrc) = 0;
- virtual uint32_t ssrc() const = 0;
-
- // Audio or video sender?
- virtual cricket::MediaType media_type() const = 0;
-
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
-
- // TODO(deadbeef): Support one sender having multiple stream ids.
- virtual void set_stream_id(const std::string& stream_id) = 0;
- virtual std::string stream_id() const = 0;
virtual void Stop() = 0;
@@ -76,12 +61,7 @@
BEGIN_PROXY_MAP(RtpSender)
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_METHOD1(void, SetSsrc, uint32_t)
-PROXY_CONSTMETHOD0(uint32_t, ssrc)
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD1(void, set_stream_id, const std::string&)
-PROXY_CONSTMETHOD0(std::string, stream_id)
PROXY_METHOD0(void, Stop)
END_PROXY()
« no previous file with comments | « talk/app/webrtc/rtpsender.cc ('k') | talk/app/webrtc/rtpsenderreceiver_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698