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Side by Side Diff: webrtc/api/rtpsender.h

Issue 1788583004: Enable setting the maximum bitrate limit in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased on top of the latest master branch Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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90 90
91 std::string id() const override { return id_; } 91 std::string id() const override { return id_; }
92 92
93 void set_stream_id(const std::string& stream_id) override { 93 void set_stream_id(const std::string& stream_id) override {
94 stream_id_ = stream_id; 94 stream_id_ = stream_id;
95 } 95 }
96 std::string stream_id() const override { return stream_id_; } 96 std::string stream_id() const override { return stream_id_; }
97 97
98 void Stop() override; 98 void Stop() override;
99 99
100 RtpParameters GetParameters() const;
101 bool SetParameters(const RtpParameters& parameters);
102
100 private: 103 private:
101 bool can_send_track() const { return track_ && ssrc_; } 104 bool can_send_track() const { return track_ && ssrc_; }
102 // Helper function to construct options for 105 // Helper function to construct options for
103 // AudioProviderInterface::SetAudioSend. 106 // AudioProviderInterface::SetAudioSend.
104 void SetAudioSend(); 107 void SetAudioSend();
105 108
106 std::string id_; 109 std::string id_;
107 std::string stream_id_; 110 std::string stream_id_;
108 AudioProviderInterface* provider_; 111 AudioProviderInterface* provider_;
109 StatsCollector* stats_; 112 StatsCollector* stats_;
(...skipping 41 matching lines...) Expand 10 before | Expand all | Expand 10 after
151 154
152 std::string id() const override { return id_; } 155 std::string id() const override { return id_; }
153 156
154 void set_stream_id(const std::string& stream_id) override { 157 void set_stream_id(const std::string& stream_id) override {
155 stream_id_ = stream_id; 158 stream_id_ = stream_id;
156 } 159 }
157 std::string stream_id() const override { return stream_id_; } 160 std::string stream_id() const override { return stream_id_; }
158 161
159 void Stop() override; 162 void Stop() override;
160 163
164 RtpParameters GetParameters() const;
165 bool SetParameters(const RtpParameters& parameters);
166
161 private: 167 private:
162 bool can_send_track() const { return track_ && ssrc_; } 168 bool can_send_track() const { return track_ && ssrc_; }
163 // Helper function to construct options for 169 // Helper function to construct options for
164 // VideoProviderInterface::SetVideoSend. 170 // VideoProviderInterface::SetVideoSend.
165 void SetVideoSend(); 171 void SetVideoSend();
166 172
167 std::string id_; 173 std::string id_;
168 std::string stream_id_; 174 std::string stream_id_;
169 VideoProviderInterface* provider_; 175 VideoProviderInterface* provider_;
170 rtc::scoped_refptr<VideoTrackInterface> track_; 176 rtc::scoped_refptr<VideoTrackInterface> track_;
171 uint32_t ssrc_ = 0; 177 uint32_t ssrc_ = 0;
172 bool cached_track_enabled_ = false; 178 bool cached_track_enabled_ = false;
173 bool stopped_ = false; 179 bool stopped_ = false;
174 }; 180 };
175 181
176 } // namespace webrtc 182 } // namespace webrtc
177 183
178 #endif // WEBRTC_API_RTPSENDER_H_ 184 #endif // WEBRTC_API_RTPSENDER_H_
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