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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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86 int GetCsrcs(uint32_t* csrcs) const; | 86 int GetCsrcs(uint32_t* csrcs) const; |
87 | 87 |
88 RtpReceiver* GetRtpReceiver() const; | 88 RtpReceiver* GetRtpReceiver() const; |
89 RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); } | 89 RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); } |
90 | 90 |
91 void StartReceive(); | 91 void StartReceive(); |
92 void StopReceive(); | 92 void StopReceive(); |
93 | 93 |
94 bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length); | 94 bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length); |
95 | 95 |
96 void FrameContinuous(uint16_t seq_num); | 96 void FrameContinuous(int64_t seq_num); |
97 | 97 |
98 void FrameDecoded(uint16_t seq_num); | 98 void FrameDecoded(int64_t seq_num); |
99 | 99 |
100 void SignalNetworkState(NetworkState state); | 100 void SignalNetworkState(NetworkState state); |
101 | 101 |
102 // Implements RtpPacketSinkInterface. | 102 // Implements RtpPacketSinkInterface. |
103 void OnRtpPacket(const RtpPacketReceived& packet) override; | 103 void OnRtpPacket(const RtpPacketReceived& packet) override; |
104 | 104 |
105 // Implements RtpData. | 105 // Implements RtpData. |
106 int32_t OnReceivedPayloadData(const uint8_t* payload_data, | 106 int32_t OnReceivedPayloadData(const uint8_t* payload_data, |
107 size_t payload_size, | 107 size_t payload_size, |
108 const WebRtcRTPHeader* rtp_header) override; | 108 const WebRtcRTPHeader* rtp_header) override; |
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192 const std::unique_ptr<RtpRtcp> rtp_rtcp_; | 192 const std::unique_ptr<RtpRtcp> rtp_rtcp_; |
193 | 193 |
194 // Members for the new jitter buffer experiment. | 194 // Members for the new jitter buffer experiment. |
195 video_coding::OnCompleteFrameCallback* complete_frame_callback_; | 195 video_coding::OnCompleteFrameCallback* complete_frame_callback_; |
196 KeyFrameRequestSender* keyframe_request_sender_; | 196 KeyFrameRequestSender* keyframe_request_sender_; |
197 VCMTiming* timing_; | 197 VCMTiming* timing_; |
198 std::unique_ptr<NackModule> nack_module_; | 198 std::unique_ptr<NackModule> nack_module_; |
199 rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_; | 199 rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_; |
200 std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_; | 200 std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_; |
201 rtc::CriticalSection last_seq_num_cs_; | 201 rtc::CriticalSection last_seq_num_cs_; |
202 std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>> | 202 std::map<int64_t, uint16_t> last_seq_num_for_pic_id_ |
203 last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_); | 203 GUARDED_BY(last_seq_num_cs_); |
204 video_coding::H264SpsPpsTracker tracker_; | 204 video_coding::H264SpsPpsTracker tracker_; |
205 // TODO(johan): Remove pt_codec_params_ once | 205 // TODO(johan): Remove pt_codec_params_ once |
206 // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved. | 206 // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved. |
207 // Maps a payload type to a map of out-of-band supplied codec parameters. | 207 // Maps a payload type to a map of out-of-band supplied codec parameters. |
208 std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_; | 208 std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_; |
209 int16_t last_payload_type_ = -1; | 209 int16_t last_payload_type_ = -1; |
210 | 210 |
211 bool has_received_frame_; | 211 bool has_received_frame_; |
212 | 212 |
213 std::vector<RtpPacketSinkInterface*> secondary_sinks_ | 213 std::vector<RtpPacketSinkInterface*> secondary_sinks_ |
214 GUARDED_BY(worker_task_checker_); | 214 GUARDED_BY(worker_task_checker_); |
215 }; | 215 }; |
216 | 216 |
217 } // namespace webrtc | 217 } // namespace webrtc |
218 | 218 |
219 #endif // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_ | 219 #endif // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_ |
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