This CL removes copy and assign support from Buffer and changes various parameters from Buffer to CopyOnWriteBuffer so they can be passed along and copied without actually copying the underlying data. With this changed some parameters to be "const" and fixed an issue when creating a CopyOnWriteBuffer with empty data. BUG=webrtc:5155 Review URL: https://codereview.webrtc.org/1823503002 Cr-Commit-Position: refs/heads/master@{#12062}
126 lines
3.9 KiB
C++
126 lines
3.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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#define WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/base/timing.h"
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#include "webrtc/media/base/mediachannel.h"
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#include "webrtc/media/base/mediaconstants.h"
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#include "webrtc/media/base/mediaengine.h"
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namespace cricket {
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struct DataCodec;
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class RtpDataEngine : public DataEngineInterface {
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public:
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RtpDataEngine();
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virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
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virtual const std::vector<DataCodec>& data_codecs() {
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return data_codecs_;
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}
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// Mostly for testing with a fake clock. Ownership is passed in.
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void SetTiming(rtc::Timing* timing) {
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timing_.reset(timing);
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}
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private:
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std::vector<DataCodec> data_codecs_;
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std::unique_ptr<rtc::Timing> timing_;
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};
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// Keep track of sequence number and timestamp of an RTP stream. The
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// sequence number starts with a "random" value and increments. The
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// timestamp starts with a "random" value and increases monotonically
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// according to the clockrate.
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class RtpClock {
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public:
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RtpClock(int clockrate, uint16_t first_seq_num, uint32_t timestamp_offset)
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: clockrate_(clockrate),
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last_seq_num_(first_seq_num),
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timestamp_offset_(timestamp_offset) {}
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// Given the current time (in number of seconds which must be
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// monotonically increasing), Return the next sequence number and
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// timestamp.
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void Tick(double now, int* seq_num, uint32_t* timestamp);
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private:
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int clockrate_;
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uint16_t last_seq_num_;
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uint32_t timestamp_offset_;
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};
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class RtpDataMediaChannel : public DataMediaChannel {
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public:
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// Timing* Used for the RtpClock
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explicit RtpDataMediaChannel(rtc::Timing* timing);
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// Sets Timing == NULL, so you'll need to call set_timer() before
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// using it. This is needed by FakeMediaEngine.
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RtpDataMediaChannel();
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virtual ~RtpDataMediaChannel();
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void set_timing(rtc::Timing* timing) {
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timing_ = timing;
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}
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virtual bool SetSendParameters(const DataSendParameters& params);
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virtual bool SetRecvParameters(const DataRecvParameters& params);
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virtual bool AddSendStream(const StreamParams& sp);
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virtual bool RemoveSendStream(uint32_t ssrc);
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virtual bool AddRecvStream(const StreamParams& sp);
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virtual bool RemoveRecvStream(uint32_t ssrc);
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virtual bool SetSend(bool send) {
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sending_ = send;
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return true;
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}
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virtual bool SetReceive(bool receive) {
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receiving_ = receive;
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return true;
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}
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virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time);
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virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) {}
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virtual void OnReadyToSend(bool ready) {}
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virtual bool SendData(
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const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& payload,
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SendDataResult* result);
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private:
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void Construct(rtc::Timing* timing);
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bool SetMaxSendBandwidth(int bps);
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bool SetSendCodecs(const std::vector<DataCodec>& codecs);
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bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
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bool sending_;
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bool receiving_;
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rtc::Timing* timing_;
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std::vector<DataCodec> send_codecs_;
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std::vector<DataCodec> recv_codecs_;
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std::vector<StreamParams> send_streams_;
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std::vector<StreamParams> recv_streams_;
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std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
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std::unique_ptr<rtc::RateLimiter> send_limiter_;
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};
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} // namespace cricket
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#endif // WEBRTC_MEDIA_BASE_RTPDATAENGINE_H_
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