commit dc1c62cd30e42d5bc31db347126476ca8e1b6c58
author skvlad <skvlad@webrtc.org> Wed Mar 16 19:07:43 2016 -0700
committer Commit bot <commit-bot@chromium.org> Thu Mar 17 02:07:49 2016 +0000
tree 2b52872047633fa01a266312fcc096fd7195ee3b
parent df6416aa502d5a9694875a22fcdf286c10f836ea

Enable setting the maximum bitrate limit in RtpSender.

This change allows the application to limit the bitrate of the outgoing
audio and video streams at runtime. The API roughly follows the WebRTC
API draft, defining the RTCRtpParameters structure witn exactly one
encoding (simulcast streams are not exposed in the API for now).
(https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters)

BUG=

Review URL: https://codereview.webrtc.org/1788583004

Cr-Commit-Position: refs/heads/master@{#12025}

webrtc/api/rtpsenderinterface.h

diff --git a/webrtc/api/rtpsenderinterface.h b/webrtc/api/rtpsenderinterface.h
index c3dd52f..776d01a 100644
--- a/webrtc/api/rtpsenderinterface.h
+++ b/webrtc/api/rtpsenderinterface.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"
@@ -51,6 +52,9 @@
 
   virtual void Stop() = 0;
 
+  virtual RtpParameters GetParameters() const = 0;
+  virtual bool SetParameters(const RtpParameters& parameters) = 0;
+
  protected:
   virtual ~RtpSenderInterface() {}
 };
@@ -66,6 +70,8 @@
 PROXY_METHOD1(void, set_stream_id, const std::string&)
 PROXY_CONSTMETHOD0(std::string, stream_id)
 PROXY_METHOD0(void, Stop)
+PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
+PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
 END_PROXY()
 
 }  // namespace webrtc