Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(165)

Unified Diff: webrtc/api/rtpreceiverinterface.h

Issue 2046173002: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Moving code that needs to execute out of RTC_DCHECKs. Created 4 years, 6 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 | « webrtc/api/rtpreceiver.cc ('k') | webrtc/api/rtpsender.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/rtpreceiverinterface.h
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index 4943023111b5b4261baecdab6d23e9d9bb7d974e..53a37398cfc9222c6c6528640920fb619d728a1f 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -18,6 +18,7 @@
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
+#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/pc/mediasession.h"
@@ -26,6 +27,12 @@ namespace webrtc {
class RtpReceiverObserverInterface {
public:
+ // Note: Currently if there are multiple RtpReceivers of the same media type,
+ // they will all call OnFirstPacketReceived at once.
+ //
+ // In the future, it's likely that an RtpReceiver will only call
+ // OnFirstPacketReceived when a packet is received specifically for its
+ // SSRC/mid.
virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
protected:
@@ -36,6 +43,9 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+ // Audio or video receiver?
+ 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;
@@ -46,10 +56,10 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
virtual RtpParameters GetParameters() const = 0;
virtual bool SetParameters(const RtpParameters& parameters) = 0;
+ // Does not take ownership of observer.
+ // Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
- virtual cricket::MediaType media_type() = 0;
-
protected:
virtual ~RtpReceiverInterface() {}
};
@@ -57,11 +67,11 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
// Define proxy for RtpReceiverInterface.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
-PROXY_METHOD0(cricket::MediaType, media_type);
END_SIGNALING_PROXY()
} // namespace webrtc
« no previous file with comments | « webrtc/api/rtpreceiver.cc ('k') | webrtc/api/rtpsender.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698