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Side by Side Diff: webrtc/video/payload_router.h

Issue 2131913003: Revert of Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 months ago
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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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