eb23e17798
Reason for revert:
This seems to be causing some video freezes. See https://bugs.chromium.org/p/webrtc/issues/detail?id=8251
Original issue's description:
> Completed the functionalities of SrtpTransport.
>
> The SrtpTransport takes the SRTP responsibilities from the BaseChannel
> and SrtpFilter. SrtpTransport is now responsible for setting the crypto
> keys, protecting and unprotecting the packets. SrtpTransport doesn't know
> if the keys are from SDES or DTLS handshake.
>
> BaseChannel is now only responsible setting the offer/answer for SDES
> or extracting the key from DtlsTransport and configuring the
> SrtpTransport.
>
> SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
>
> BUG=webrtc:7013
>
> Review-Url: https://codereview.webrtc.org/2997983002
> Cr-Commit-Position: refs/heads/master@{#19636}
> Committed: e683c6871f
TBR=deadbeef@webrtc.org,pthatcher@google.com,zhihuang@webrtc.org
Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/3018513002
Cr-Commit-Position: refs/heads/master@{#19895}
108 lines
3.2 KiB
C++
108 lines
3.2 KiB
C++
/*
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* Copyright 2017 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 PC_RTPTRANSPORT_H_
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#define PC_RTPTRANSPORT_H_
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#include "pc/bundlefilter.h"
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#include "pc/rtptransportinternal.h"
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#include "rtc_base/sigslot.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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struct PacketTime;
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class PacketTransportInternal;
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} // namespace rtc
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namespace webrtc {
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class RtpTransport : public RtpTransportInternal {
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public:
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RtpTransport(const RtpTransport&) = delete;
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RtpTransport& operator=(const RtpTransport&) = delete;
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explicit RtpTransport(bool rtcp_mux_enabled)
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: rtcp_mux_enabled_(rtcp_mux_enabled) {}
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bool rtcp_mux_enabled() const { return rtcp_mux_enabled_; }
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void SetRtcpMuxEnabled(bool enable) override;
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rtc::PacketTransportInternal* rtp_packet_transport() const override {
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return rtp_packet_transport_;
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}
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void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) override;
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rtc::PacketTransportInternal* rtcp_packet_transport() const override {
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return rtcp_packet_transport_;
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}
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void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override;
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PacketTransportInterface* GetRtpPacketTransport() const override;
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PacketTransportInterface* GetRtcpPacketTransport() const override;
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// TODO(zstein): Use these RtcpParameters for configuration elsewhere.
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RTCError SetParameters(const RtpTransportParameters& parameters) override;
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RtpTransportParameters GetParameters() const override;
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bool IsWritable(bool rtcp) const override;
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bool SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool HandlesPayloadType(int payload_type) const override;
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void AddHandledPayloadType(int payload_type) override;
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protected:
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// TODO(zstein): Remove this when we remove RtpTransportAdapter.
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RtpTransportAdapter* GetInternal() override;
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private:
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bool HandlesPacket(const uint8_t* data, size_t len);
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void OnReadyToSend(rtc::PacketTransportInternal* transport);
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// Updates "ready to send" for an individual channel and fires
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// SignalReadyToSend.
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void SetReadyToSend(bool rtcp, bool ready);
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void MaybeSignalReadyToSend();
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void OnReadPacket(rtc::PacketTransportInternal* transport,
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const char* data,
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size_t len,
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const rtc::PacketTime& packet_time,
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int flags);
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bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
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bool rtcp_mux_enabled_;
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rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
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rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
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bool ready_to_send_ = false;
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bool rtp_ready_to_send_ = false;
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bool rtcp_ready_to_send_ = false;
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RtpTransportParameters parameters_;
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cricket::BundleFilter bundle_filter_;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORT_H_
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