webrtc_m130/pc/srtptransport.h
zhihuang eb23e17798 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )
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}
2017-09-19 08:12:52 +00:00

109 lines
3.3 KiB
C++

/*
* Copyright 2017 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 PC_SRTPTRANSPORT_H_
#define PC_SRTPTRANSPORT_H_
#include <memory>
#include <string>
#include <utility>
#include "pc/rtptransportinternal.h"
#include "pc/srtpfilter.h"
#include "rtc_base/checks.h"
namespace webrtc {
// This class will eventually be a wrapper around RtpTransportInternal
// that protects and unprotects sent and received RTP packets. This
// functionality is currently implemented by SrtpFilter and BaseChannel, but
// will be moved here in the future.
class SrtpTransport : public RtpTransportInternal {
public:
SrtpTransport(bool rtcp_mux_enabled, const std::string& content_name);
// TODO(zstein): Consider taking an RtpTransport instead of an
// RtpTransportInternal.
SrtpTransport(std::unique_ptr<RtpTransportInternal> transport,
const std::string& content_name);
void SetRtcpMuxEnabled(bool enable) override {
rtp_transport_->SetRtcpMuxEnabled(enable);
}
rtc::PacketTransportInternal* rtp_packet_transport() const override {
return rtp_transport_->rtp_packet_transport();
}
void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) override {
rtp_transport_->SetRtpPacketTransport(rtp);
}
PacketTransportInterface* GetRtpPacketTransport() const override {
return rtp_transport_->GetRtpPacketTransport();
}
rtc::PacketTransportInternal* rtcp_packet_transport() const override {
return rtp_transport_->rtcp_packet_transport();
}
void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override {
rtp_transport_->SetRtcpPacketTransport(rtcp);
}
PacketTransportInterface* GetRtcpPacketTransport() const override {
return rtp_transport_->GetRtcpPacketTransport();
}
bool IsWritable(bool rtcp) const override {
return rtp_transport_->IsWritable(rtcp);
}
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) override;
bool HandlesPayloadType(int payload_type) const override {
return rtp_transport_->HandlesPayloadType(payload_type);
}
void AddHandledPayloadType(int payload_type) override {
rtp_transport_->AddHandledPayloadType(payload_type);
}
RTCError SetParameters(const RtpTransportParameters& parameters) override {
return rtp_transport_->SetParameters(parameters);
}
RtpTransportParameters GetParameters() const override {
return rtp_transport_->GetParameters();
}
// TODO(zstein): Remove this when we remove RtpTransportAdapter.
RtpTransportAdapter* GetInternal() override { return nullptr; }
private:
void ConnectToRtpTransport();
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
const std::string content_name_;
std::unique_ptr<RtpTransportInternal> rtp_transport_;
};
} // namespace webrtc
#endif // PC_SRTPTRANSPORT_H_