dc1c62cd30
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}
31 lines
866 B
C++
31 lines
866 B
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_API_RTPPARAMETERS_H_
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#define WEBRTC_API_RTPPARAMETERS_H_
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#include <vector>
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namespace webrtc {
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// These structures are defined as part of the RtpSender interface.
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// See http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface for details.
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struct RtpEncodingParameters {
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int max_bitrate_bps = -1;
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};
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struct RtpParameters {
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std::vector<RtpEncodingParameters> encodings;
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};
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} // namespace webrtc
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#endif // WEBRTC_API_RTPPARAMETERS_H_
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