| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MEDIA_BASE_RTPDATAENGINE_H_ |
| #define MEDIA_BASE_RTPDATAENGINE_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "media/base/mediachannel.h" |
| #include "media/base/mediaconstants.h" |
| #include "media/base/mediaengine.h" |
| |
| namespace cricket { |
| |
| struct DataCodec; |
| |
| class RtpDataEngine : public DataEngineInterface { |
| public: |
| RtpDataEngine(); |
| |
| virtual DataMediaChannel* CreateChannel(const MediaConfig& config); |
| |
| virtual const std::vector<DataCodec>& data_codecs() { |
| return data_codecs_; |
| } |
| |
| private: |
| std::vector<DataCodec> data_codecs_; |
| }; |
| |
| // Keep track of sequence number and timestamp of an RTP stream. The |
| // sequence number starts with a "random" value and increments. The |
| // timestamp starts with a "random" value and increases monotonically |
| // according to the clockrate. |
| class RtpClock { |
| public: |
| RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset) |
| : clockrate_(clockrate), |
| last_seq_num_(first_seq_num), |
| timestamp_offset_(timestamp_offset) {} |
| |
| // Given the current time (in number of seconds which must be |
| // monotonically increasing), Return the next sequence number and |
| // timestamp. |
| void Tick(double now, int* seq_num, uint32_t* timestamp); |
| |
| private: |
| int clockrate_; |
| uint16_t last_seq_num_; |
| uint32_t timestamp_offset_; |
| }; |
| |
| class RtpDataMediaChannel : public DataMediaChannel { |
| public: |
| explicit RtpDataMediaChannel(const MediaConfig& config); |
| virtual ~RtpDataMediaChannel(); |
| |
| virtual bool SetSendParameters(const DataSendParameters& params); |
| virtual bool SetRecvParameters(const DataRecvParameters& params); |
| virtual bool AddSendStream(const StreamParams& sp); |
| virtual bool RemoveSendStream(uint32_t ssrc); |
| virtual bool AddRecvStream(const StreamParams& sp); |
| virtual bool RemoveRecvStream(uint32_t ssrc); |
| virtual bool SetSend(bool send) { |
| sending_ = send; |
| return true; |
| } |
| virtual bool SetReceive(bool receive) { |
| receiving_ = receive; |
| return true; |
| } |
| virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time); |
| virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketTime& packet_time) {} |
| virtual void OnReadyToSend(bool ready) {} |
| virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {} |
| virtual bool SendData( |
| const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result); |
| virtual rtc::DiffServCodePoint PreferredDscp() const; |
| |
| private: |
| void Construct(); |
| bool SetMaxSendBandwidth(int bps); |
| bool SetSendCodecs(const std::vector<DataCodec>& codecs); |
| bool SetRecvCodecs(const std::vector<DataCodec>& codecs); |
| |
| bool sending_; |
| bool receiving_; |
| std::vector<DataCodec> send_codecs_; |
| std::vector<DataCodec> recv_codecs_; |
| std::vector<StreamParams> send_streams_; |
| std::vector<StreamParams> recv_streams_; |
| std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_; |
| std::unique_ptr<rtc::RateLimiter> send_limiter_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_BASE_RTPDATAENGINE_H_ |