c0dad89bed
Reason for revert:
Hitting DCHECK in chromium's WebrtcTransportTest.TerminateDataChannel and WebrtcTransportTest.DataStreamLate. Will investigate and reland.
Original issue's description:
> Separating SCTP code from BaseChannel/MediaChannel.
>
> The BaseChannel code is geared around RTP; the presence of media engines,
> send and receive streams, SRTP, SDP directional attribute negotiation, etc.
> It doesn't make sense to use it for SCTP as well. This separation should make
> future work both on BaseChannel and the SCTP code paths easier.
>
> SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
> directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
> doesn't get confused with webrtc::DataChannel any more.
>
> Beyond just moving code around, some consequences of this CL:
> - We'll now stop using the worker thread for SCTP. Packets will be
> processed right on the network thread instead.
> - The SDP directional attribute is ignored, as it's supposed to be.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2564333002
> Cr-Commit-Position: refs/heads/master@{#15906}
> Committed: 67b3bbe639
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2614813003
Cr-Commit-Position: refs/heads/master@{#15908}
113 lines
3.6 KiB
C++
113 lines
3.6 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/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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const MediaConfig& config);
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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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private:
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std::vector<DataCodec> data_codecs_;
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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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RtpDataMediaChannel(const MediaConfig& config);
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virtual ~RtpDataMediaChannel();
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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 void OnTransportOverheadChanged(int transport_overhead_per_packet) {}
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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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virtual rtc::DiffServCodePoint PreferredDscp() const;
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private:
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void Construct();
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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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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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