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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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| 43 // dropped otherwise. | 43 // dropped otherwise. |
| 44 void set_active(bool active); | 44 void set_active(bool active); |
| 45 bool active(); | 45 bool active(); |
| 46 | 46 |
| 47 // Implements EncodedImageCallback. | 47 // Implements EncodedImageCallback. |
| 48 // Returns 0 if the packet was routed / sent, -1 otherwise. | 48 // Returns 0 if the packet was routed / sent, -1 otherwise. |
| 49 int32_t Encoded(const EncodedImage& encoded_image, | 49 int32_t Encoded(const EncodedImage& encoded_image, |
| 50 const CodecSpecificInfo* codec_specific_info, | 50 const CodecSpecificInfo* codec_specific_info, |
| 51 const RTPFragmentationHeader* fragmentation) override; | 51 const RTPFragmentationHeader* fragmentation) override; |
| 52 | 52 |
| 53 // Configures current target bitrate. |
| 54 void SetTargetSendBitrate(uint32_t bitrate_bps); |
| 55 |
| 53 // Returns the maximum allowed data payload length, given the configured MTU | 56 // Returns the maximum allowed data payload length, given the configured MTU |
| 54 // and RTP headers. | 57 // and RTP headers. |
| 55 size_t MaxPayloadLength() const; | 58 size_t MaxPayloadLength() const; |
| 56 | 59 |
| 57 private: | 60 private: |
| 58 void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_); | 61 void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| 59 | 62 |
| 60 rtc::CriticalSection crit_; | 63 rtc::CriticalSection crit_; |
| 61 bool active_ GUARDED_BY(crit_); | 64 bool active_ GUARDED_BY(crit_); |
| 62 std::vector<VideoStream> streams_ GUARDED_BY(crit_); | 65 std::vector<VideoStream> streams_ GUARDED_BY(crit_); |
| 63 size_t num_sending_modules_ GUARDED_BY(crit_); | 66 size_t num_sending_modules_ GUARDED_BY(crit_); |
| 64 | 67 |
| 65 // Rtp modules are assumed to be sorted in simulcast index order. Not owned. | 68 // Rtp modules are assumed to be sorted in simulcast index order. Not owned. |
| 66 const std::vector<RtpRtcp*> rtp_modules_; | 69 const std::vector<RtpRtcp*> rtp_modules_; |
| 67 const int payload_type_; | 70 const int payload_type_; |
| 68 | 71 |
| 69 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter); | 72 RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter); |
| 70 }; | 73 }; |
| 71 | 74 |
| 72 } // namespace webrtc | 75 } // namespace webrtc |
| 73 | 76 |
| 74 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ | 77 #endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_ |
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