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}
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@ -18,6 +18,7 @@
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/proxy.h"
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#include "webrtc/api/rtpparameters.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/pc/mediasession.h"
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@ -51,6 +52,9 @@ class RtpSenderInterface : public rtc::RefCountInterface {
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virtual void Stop() = 0;
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virtual RtpParameters GetParameters() const = 0;
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virtual bool SetParameters(const RtpParameters& parameters) = 0;
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protected:
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virtual ~RtpSenderInterface() {}
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};
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@ -66,6 +70,8 @@ PROXY_CONSTMETHOD0(std::string, id)
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PROXY_METHOD1(void, set_stream_id, const std::string&)
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PROXY_CONSTMETHOD0(std::string, stream_id)
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PROXY_METHOD0(void, Stop)
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PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
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END_PROXY()
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} // namespace webrtc
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