webrtc_m130/pc/rtptransport.cc
Zhi Huang 95e7dbb7c7 Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""
This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950.

Reason for revert: Broken internal project.

Original change's description:
> Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."
> 
> This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport."
> > 
> > This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e.
> > 
> > Reason for revert: Broke chromium tests.
> > Original change's description:
> > > Replace BundleFilter with RtpDemuxer in RtpTransport.
> > > 
> > > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
> > > type-based demuxing. RtpTransport will support MID-based demuxing later.
> > > 
> > > Each BaseChannel has its own RTP demuxing criteria and when connecting
> > > to the RtpTransport, BaseChannel will register itself as a demuxer sink.
> > > 
> > > The inheritance model is changed. New inheritance chain:
> > > DtlsSrtpTransport->SrtpTransport->RtpTranpsort
> > > 
> > > NOTE:
> > > When RTCP packets are received, Call::DeliverRtcp will be called for
> > > multiple times (webrtc:9035) which is an existing issue. With this CL,
> > > it will become more of a problem and should be fixed.
> > > 
> > > Bug: webrtc:8587
> > > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
> > > Reviewed-on: https://webrtc-review.googlesource.com/61360
> > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22613}
> > 
> > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
> > 
> > Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8587
> > Reviewed-on: https://webrtc-review.googlesource.com/64860
> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22614}
> 
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
> 
> Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8587
> Reviewed-on: https://webrtc-review.googlesource.com/64862
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22615}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8587
Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4
Reviewed-on: https://webrtc-review.googlesource.com/65381
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22665}
2018-03-29 02:45:17 +00:00

284 lines
10 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.
*/
#include "pc/rtptransport.h"
#include "media/base/rtputils.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/packettransportinterface.h"
#include "rtc_base/checks.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
void RtpTransport::SetRtcpMuxEnabled(bool enable) {
rtcp_mux_enabled_ = enable;
MaybeSignalReadyToSend();
}
void RtpTransport::SetRtpPacketTransport(
rtc::PacketTransportInternal* new_packet_transport) {
if (new_packet_transport == rtp_packet_transport_) {
return;
}
if (rtp_packet_transport_) {
rtp_packet_transport_->SignalReadyToSend.disconnect(this);
rtp_packet_transport_->SignalReadPacket.disconnect(this);
rtp_packet_transport_->SignalNetworkRouteChanged.disconnect(this);
rtp_packet_transport_->SignalWritableState.disconnect(this);
rtp_packet_transport_->SignalSentPacket.disconnect(this);
// Reset the network route of the old transport.
SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
this, &RtpTransport::OnReadyToSend);
new_packet_transport->SignalReadPacket.connect(this,
&RtpTransport::OnReadPacket);
new_packet_transport->SignalNetworkRouteChanged.connect(
this, &RtpTransport::OnNetworkRouteChange);
new_packet_transport->SignalWritableState.connect(
this, &RtpTransport::OnWritableState);
new_packet_transport->SignalSentPacket.connect(this,
&RtpTransport::OnSentPacket);
// Set the network route for the new transport.
SignalNetworkRouteChanged(new_packet_transport->network_route());
}
rtp_packet_transport_ = new_packet_transport;
// Assumes the transport is ready to send if it is writable. If we are wrong,
// ready to send will be updated the next time we try to send.
SetReadyToSend(false,
rtp_packet_transport_ && rtp_packet_transport_->writable());
}
void RtpTransport::SetRtcpPacketTransport(
rtc::PacketTransportInternal* new_packet_transport) {
if (new_packet_transport == rtcp_packet_transport_) {
return;
}
if (rtcp_packet_transport_) {
rtcp_packet_transport_->SignalReadyToSend.disconnect(this);
rtcp_packet_transport_->SignalReadPacket.disconnect(this);
rtcp_packet_transport_->SignalNetworkRouteChanged.disconnect(this);
rtcp_packet_transport_->SignalWritableState.disconnect(this);
rtcp_packet_transport_->SignalSentPacket.disconnect(this);
// Reset the network route of the old transport.
SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
}
if (new_packet_transport) {
new_packet_transport->SignalReadyToSend.connect(
this, &RtpTransport::OnReadyToSend);
new_packet_transport->SignalReadPacket.connect(this,
&RtpTransport::OnReadPacket);
new_packet_transport->SignalNetworkRouteChanged.connect(
this, &RtpTransport::OnNetworkRouteChange);
new_packet_transport->SignalWritableState.connect(
this, &RtpTransport::OnWritableState);
new_packet_transport->SignalSentPacket.connect(this,
&RtpTransport::OnSentPacket);
// Set the network route for the new transport.
SignalNetworkRouteChanged(new_packet_transport->network_route());
}
rtcp_packet_transport_ = new_packet_transport;
// Assumes the transport is ready to send if it is writable. If we are wrong,
// ready to send will be updated the next time we try to send.
SetReadyToSend(true,
rtcp_packet_transport_ && rtcp_packet_transport_->writable());
}
bool RtpTransport::IsWritable(bool rtcp) const {
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
return transport && transport->writable();
}
bool RtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(false, packet, options, flags);
}
bool RtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(true, packet, options, flags);
}
bool RtpTransport::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
rtc::PacketTransportInternal* transport = rtcp && !rtcp_mux_enabled_
? rtcp_packet_transport_
: rtp_packet_transport_;
int ret = transport->SendPacket(packet->data<char>(), packet->size(), options,
flags);
if (ret != static_cast<int>(packet->size())) {
if (transport->GetError() == ENOTCONN) {
RTC_LOG(LS_WARNING) << "Got ENOTCONN from transport.";
SetReadyToSend(rtcp, false);
}
return false;
}
return true;
}
bool RtpTransport::HandlesPacket(const uint8_t* data, size_t len) {
return bundle_filter_.DemuxPacket(data, len);
}
bool RtpTransport::HandlesPayloadType(int payload_type) const {
return bundle_filter_.FindPayloadType(payload_type);
}
void RtpTransport::AddHandledPayloadType(int payload_type) {
bundle_filter_.AddPayloadType(payload_type);
}
PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
return rtp_packet_transport_;
}
PacketTransportInterface* RtpTransport::GetRtcpPacketTransport() const {
return rtcp_packet_transport_;
}
RTCError RtpTransport::SetParameters(const RtpTransportParameters& parameters) {
if (parameters_.rtcp.mux && !parameters.rtcp.mux) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Disabling RTCP muxing is not allowed.");
}
if (parameters.keepalive != parameters_.keepalive) {
// TODO(sprang): Wire up support for keep-alive (only ORTC support for now).
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"RTP keep-alive parameters not supported by this channel.");
}
RtpTransportParameters new_parameters = parameters;
if (new_parameters.rtcp.cname.empty()) {
new_parameters.rtcp.cname = parameters_.rtcp.cname;
}
parameters_ = new_parameters;
return RTCError::OK();
}
RtpTransportParameters RtpTransport::GetParameters() const {
return parameters_;
}
RtpTransportAdapter* RtpTransport::GetInternal() {
return nullptr;
}
bool RtpTransport::IsRtpTransportWritable() {
auto rtcp_packet_transport =
rtcp_mux_enabled_ ? nullptr : rtcp_packet_transport_;
return rtp_packet_transport_ && rtp_packet_transport_->writable() &&
(!rtcp_packet_transport || rtcp_packet_transport->writable());
}
void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
SetReadyToSend(transport == rtcp_packet_transport_, true);
}
void RtpTransport::OnNetworkRouteChange(
rtc::Optional<rtc::NetworkRoute> network_route) {
SignalNetworkRouteChanged(network_route);
}
void RtpTransport::OnWritableState(
rtc::PacketTransportInternal* packet_transport) {
RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
packet_transport == rtcp_packet_transport_);
SignalWritableState(IsRtpTransportWritable());
}
void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
const rtc::SentPacket& sent_packet) {
RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
packet_transport == rtcp_packet_transport_);
SignalSentPacket(sent_packet);
}
void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
if (rtcp) {
rtcp_ready_to_send_ = ready;
} else {
rtp_ready_to_send_ = ready;
}
MaybeSignalReadyToSend();
}
void RtpTransport::MaybeSignalReadyToSend() {
bool ready_to_send =
rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
if (ready_to_send != ready_to_send_) {
ready_to_send_ = ready_to_send;
SignalReadyToSend(ready_to_send);
}
}
// Check the RTP payload type. If 63 < payload type < 96, it's RTCP.
// For additional details, see http://tools.ietf.org/html/rfc5761.
bool IsRtcp(const char* data, int len) {
if (len < 2) {
return false;
}
char pt = data[1] & 0x7F;
return (63 < pt) && (pt < 96);
}
void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags) {
TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket");
// When using RTCP multiplexing we might get RTCP packets on the RTP
// transport. We check the RTP payload type to determine if it is RTCP.
bool rtcp = transport == rtcp_packet_transport() ||
IsRtcp(data, static_cast<int>(len));
rtc::CopyOnWriteBuffer packet(data, len);
if (!WantsPacket(rtcp, &packet)) {
return;
}
// This mutates |packet| if it is protected.
SignalPacketReceived(rtcp, &packet, packet_time);
}
bool RtpTransport::WantsPacket(bool rtcp,
const rtc::CopyOnWriteBuffer* packet) {
// Protect ourselves against crazy data.
if (!packet || !cricket::IsValidRtpRtcpPacketSize(rtcp, packet->size())) {
RTC_LOG(LS_ERROR) << "Dropping incoming "
<< cricket::RtpRtcpStringLiteral(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
if (rtcp) {
// Permit all (seemingly valid) RTCP packets.
return true;
}
// Check whether we handle this payload.
return HandlesPacket(packet->data(), packet->size());
}
} // namespace webrtc