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Side by Side Diff: webrtc/api/webrtcsession.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 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 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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238 // AudioMediaProviderInterface implementation. 238 // AudioMediaProviderInterface implementation.
239 void SetAudioPlayout(uint32_t ssrc, bool enable) override; 239 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
240 void SetAudioSend(uint32_t ssrc, 240 void SetAudioSend(uint32_t ssrc,
241 bool enable, 241 bool enable,
242 const cricket::AudioOptions& options, 242 const cricket::AudioOptions& options,
243 cricket::AudioSource* source) override; 243 cricket::AudioSource* source) override;
244 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override; 244 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
245 void SetRawAudioSink(uint32_t ssrc, 245 void SetRawAudioSink(uint32_t ssrc,
246 rtc::scoped_ptr<AudioSinkInterface> sink) override; 246 rtc::scoped_ptr<AudioSinkInterface> sink) override;
247 247
248 RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
249 bool SetAudioRtpParameters(uint32_t ssrc,
250 const RtpParameters& parameters) override;
251
248 // Implements VideoMediaProviderInterface. 252 // Implements VideoMediaProviderInterface.
249 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override; 253 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
250 void SetVideoPlayout( 254 void SetVideoPlayout(
251 uint32_t ssrc, 255 uint32_t ssrc,
252 bool enable, 256 bool enable,
253 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override; 257 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
254 void SetVideoSend(uint32_t ssrc, 258 void SetVideoSend(uint32_t ssrc,
255 bool enable, 259 bool enable,
256 const cricket::VideoOptions* options) override; 260 const cricket::VideoOptions* options) override;
257 261
262 RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
263 bool SetVideoRtpParameters(uint32_t ssrc,
264 const RtpParameters& parameters) override;
265
258 // Implements DtmfProviderInterface. 266 // Implements DtmfProviderInterface.
259 virtual bool CanInsertDtmf(const std::string& track_id); 267 virtual bool CanInsertDtmf(const std::string& track_id);
260 virtual bool InsertDtmf(const std::string& track_id, 268 virtual bool InsertDtmf(const std::string& track_id,
261 int code, int duration); 269 int code, int duration);
262 virtual sigslot::signal0<>* GetOnDestroyedSignal(); 270 virtual sigslot::signal0<>* GetOnDestroyedSignal();
263 271
264 // Implements DataChannelProviderInterface. 272 // Implements DataChannelProviderInterface.
265 bool SendData(const cricket::SendDataParams& params, 273 bool SendData(const cricket::SendDataParams& params,
266 const rtc::Buffer& payload, 274 const rtc::Buffer& payload,
267 cricket::SendDataResult* result) override; 275 cricket::SendDataResult* result) override;
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505 PeerConnectionInterface::BundlePolicy bundle_policy_; 513 PeerConnectionInterface::BundlePolicy bundle_policy_;
506 514
507 // Declares the RTCP mux policy for the WebRTCSession. 515 // Declares the RTCP mux policy for the WebRTCSession.
508 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; 516 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
509 517
510 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); 518 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
511 }; 519 };
512 } // namespace webrtc 520 } // namespace webrtc
513 521
514 #endif // WEBRTC_API_WEBRTCSESSION_H_ 522 #endif // WEBRTC_API_WEBRTCSESSION_H_
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