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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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417 // Returns true if the module is configured to store packets. | 417 // Returns true if the module is configured to store packets. |
418 virtual bool StorePackets() const = 0; | 418 virtual bool StorePackets() const = 0; |
419 | 419 |
420 // Called on receipt of RTCP report block from remote side. | 420 // Called on receipt of RTCP report block from remote side. |
421 virtual void RegisterRtcpStatisticsCallback( | 421 virtual void RegisterRtcpStatisticsCallback( |
422 RtcpStatisticsCallback* callback) = 0; | 422 RtcpStatisticsCallback* callback) = 0; |
423 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; | 423 virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; |
424 // BWE feedback packets. | 424 // BWE feedback packets. |
425 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; | 425 virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; |
426 | 426 |
| 427 virtual void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) = 0; |
| 428 |
427 // ************************************************************************** | 429 // ************************************************************************** |
428 // Audio | 430 // Audio |
429 // ************************************************************************** | 431 // ************************************************************************** |
430 | 432 |
431 // Sets audio packet size, used to determine when it's time to send a DTMF | 433 // Sets audio packet size, used to determine when it's time to send a DTMF |
432 // packet in silence (CNG). | 434 // packet in silence (CNG). |
433 // Returns -1 on failure else 0. | 435 // Returns -1 on failure else 0. |
434 virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; | 436 virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; |
435 | 437 |
436 // Sends a TelephoneEvent tone using RFC 2833 (4733). | 438 // Sends a TelephoneEvent tone using RFC 2833 (4733). |
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477 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 479 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
478 | 480 |
479 // Sends a request for a keyframe. | 481 // Sends a request for a keyframe. |
480 // Returns -1 on failure else 0. | 482 // Returns -1 on failure else 0. |
481 virtual int32_t RequestKeyFrame() = 0; | 483 virtual int32_t RequestKeyFrame() = 0; |
482 }; | 484 }; |
483 | 485 |
484 } // namespace webrtc | 486 } // namespace webrtc |
485 | 487 |
486 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 488 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
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