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webrtc_m130/pc/channel.cc
Steve Anton 47136ddaea Change RtpSenders to interact with the media channel directly 
Similar to the change for RtpReceivers, this removes the BaseChannel
methods that would just proxy calls to the MediaChannel and instead
gives the MediaChannel directly to the RtpSenders to make the calls
directly.

Bug: webrtc:8587
Change-Id: Ibab98d75ff1641e902281ad9e31ffdad36caff35
Reviewed-on: https://webrtc-review.googlesource.com/38983
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21608}
2018-01-13 01:44:04 +00:00

1883 lines
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/*
* Copyright 2004 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 <algorithm>
#include <iterator>
#include <utility>
#include "pc/channel.h"
#include "api/call/audio_sink.h"
#include "media/base/mediaconstants.h"
#include "media/base/rtputils.h"
#include "rtc_base/bind.h"
#include "rtc_base/byteorder.h"
#include "rtc_base/checks.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/dscp.h"
#include "rtc_base/logging.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/trace_event.h"
// Adding 'nogncheck' to disable the gn include headers check to support modular
// WebRTC build targets.
#include "media/engine/webrtcvoiceengine.h" // nogncheck
#include "p2p/base/packettransportinternal.h"
#include "pc/channelmanager.h"
#include "pc/rtpmediautils.h"
namespace cricket {
using rtc::Bind;
using webrtc::SdpType;
namespace {
struct SendPacketMessageData : public rtc::MessageData {
rtc::CopyOnWriteBuffer packet;
rtc::PacketOptions options;
};
} // namespace
enum {
MSG_EARLYMEDIATIMEOUT = 1,
MSG_SEND_RTP_PACKET,
MSG_SEND_RTCP_PACKET,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
MSG_FIRSTPACKETRECEIVED,
};
static void SafeSetError(const std::string& message, std::string* error_desc) {
if (error_desc) {
*error_desc = message;
}
}
static bool ValidPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) {
// Check the packet size. We could check the header too if needed.
return packet && IsValidRtpRtcpPacketSize(rtcp, packet->size());
}
template <class Codec>
void RtpParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
RtpParameters<Codec>* params) {
// TODO(pthatcher): Remove this once we're sure no one will give us
// a description without codecs. Currently the ORTC implementation is relying
// on this.
if (desc->has_codecs()) {
params->codecs = desc->codecs();
}
// TODO(pthatcher): See if we really need
// rtp_header_extensions_set() and remove it if we don't.
if (desc->rtp_header_extensions_set()) {
params->extensions = extensions;
}
params->rtcp.reduced_size = desc->rtcp_reduced_size();
}
template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
RtpSendParameters<Codec>* send_params) {
RtpParametersFromMediaDescription(desc, extensions, send_params);
send_params->max_bandwidth_bps = desc->bandwidth();
}
BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: worker_thread_(worker_thread),
network_thread_(network_thread),
signaling_thread_(signaling_thread),
content_name_(content_name),
rtcp_mux_required_(rtcp_mux_required),
unencrypted_rtp_transport_(
rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required)),
srtp_required_(srtp_required),
media_channel_(std::move(media_channel)) {
RTC_DCHECK_RUN_ON(worker_thread_);
rtp_transport_ = unencrypted_rtp_transport_.get();
ConnectToRtpTransport();
RTC_LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK_RUN_ON(worker_thread_);
Deinit();
StopConnectionMonitor();
// Eats any outstanding messages or packets.
worker_thread_->Clear(&invoker_);
worker_thread_->Clear(this);
// We must destroy the media channel before the transport channel, otherwise
// the media channel may try to send on the dead transport channel. NULLing
// is not an effective strategy since the sends will come on another thread.
media_channel_.reset();
RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_;
}
void BaseChannel::ConnectToRtpTransport() {
RTC_DCHECK(rtp_transport_);
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
// TODO(zstein): RtpTransport::SignalPacketReceived will probably be replaced
// with a callback interface later so that the demuxer can select which
// channel to signal.
rtp_transport_->SignalPacketReceived.connect(this,
&BaseChannel::OnPacketReceived);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&BaseChannel::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this,
&BaseChannel::SignalSentPacket_n);
}
void BaseChannel::DisconnectFromRtpTransport() {
RTC_DCHECK(rtp_transport_);
rtp_transport_->SignalReadyToSend.disconnect(this);
rtp_transport_->SignalPacketReceived.disconnect(this);
rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
rtp_transport_->SignalWritableState.disconnect(this);
rtp_transport_->SignalSentPacket.disconnect(this);
}
void BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
RTC_DCHECK_RUN_ON(worker_thread_);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport);
if (rtcp_mux_required_) {
rtcp_mux_filter_.SetActive();
}
});
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this);
}
void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
RTC_DCHECK_RUN_ON(worker_thread_);
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
SetRtpTransport(rtp_transport);
if (rtcp_mux_required_) {
rtcp_mux_filter_.SetActive();
}
});
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this);
}
void BaseChannel::Deinit() {
RTC_DCHECK(worker_thread_->IsCurrent());
media_channel_->SetInterface(NULL);
// Packets arrive on the network thread, processing packets calls virtual
// functions, so need to stop this process in Deinit that is called in
// derived classes destructor.
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
FlushRtcpMessages_n();
if (dtls_srtp_transport_) {
dtls_srtp_transport_->SetDtlsTransports(nullptr, nullptr);
} else {
rtp_transport_->SetRtpPacketTransport(nullptr);
rtp_transport_->SetRtcpPacketTransport(nullptr);
}
// Clear pending read packets/messages.
network_thread_->Clear(&invoker_);
network_thread_->Clear(this);
});
}
void BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
if (!network_thread_->IsCurrent()) {
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
SetRtpTransport(rtp_transport);
return;
});
}
RTC_DCHECK(rtp_transport);
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
rtp_transport_ = rtp_transport;
RTC_LOG(LS_INFO) << "Setting the RtpTransport for " << content_name();
ConnectToRtpTransport();
UpdateWritableState_n();
}
void BaseChannel::SetTransports(DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport) {
network_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetTransports_n, this, rtp_dtls_transport,
rtcp_dtls_transport, rtp_dtls_transport, rtcp_dtls_transport));
}
void BaseChannel::SetTransports(
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
network_thread_->Invoke<void>(
RTC_FROM_HERE, Bind(&BaseChannel::SetTransports_n, this, nullptr, nullptr,
rtp_packet_transport, rtcp_packet_transport));
}
void BaseChannel::SetTransports_n(
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
// Validate some assertions about the input.
RTC_DCHECK(rtp_packet_transport);
RTC_DCHECK_EQ(NeedsRtcpTransport(), rtcp_packet_transport != nullptr);
if (rtp_dtls_transport || rtcp_dtls_transport) {
// DTLS/non-DTLS pointers should be to the same object.
RTC_DCHECK(rtp_dtls_transport == rtp_packet_transport);
RTC_DCHECK(rtcp_dtls_transport == rtcp_packet_transport);
// Can't go from non-DTLS to DTLS.
RTC_DCHECK(!rtp_transport_->rtp_packet_transport() || rtp_dtls_transport_);
} else {
// Can't go from DTLS to non-DTLS.
RTC_DCHECK(!rtp_dtls_transport_);
}
// Transport names should be the same.
if (rtp_dtls_transport && rtcp_dtls_transport) {
RTC_DCHECK(rtp_dtls_transport->transport_name() ==
rtcp_dtls_transport->transport_name());
}
if (rtp_packet_transport == rtp_transport_->rtp_packet_transport()) {
// Nothing to do if transport isn't changing.
return;
}
std::string debug_name;
if (rtp_dtls_transport) {
transport_name_ = rtp_dtls_transport->transport_name();
debug_name = transport_name_;
} else {
debug_name = rtp_packet_transport->transport_name();
}
// If this BaseChannel doesn't require RTCP mux and we haven't fully
// negotiated RTCP mux, we need an RTCP transport.
if (rtcp_packet_transport) {
RTC_LOG(LS_INFO) << "Setting RTCP Transport for " << content_name()
<< " on " << debug_name << " transport "
<< rtcp_packet_transport;
SetTransport_n(/*rtcp=*/true, rtcp_dtls_transport, rtcp_packet_transport);
}
RTC_LOG(LS_INFO) << "Setting RTP Transport for " << content_name() << " on "
<< debug_name << " transport " << rtp_packet_transport;
SetTransport_n(/*rtcp=*/false, rtp_dtls_transport, rtp_packet_transport);
// Set DtlsTransport/PacketTransport for RTP-level transport.
if ((rtp_dtls_transport_ || rtcp_dtls_transport_) && dtls_srtp_transport_) {
// When setting the transport with non-null |dtls_srtp_transport_|, we are
// using DTLS-SRTP. This could happen for bundling. If the
// |dtls_srtp_transport| is null, we cannot tell if it doing DTLS-SRTP or
// SDES until the description is set. So don't call |EnableDtlsSrtp_n| here.
dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport,
rtcp_dtls_transport);
} else {
rtp_transport_->SetRtpPacketTransport(rtp_packet_transport);
rtp_transport_->SetRtcpPacketTransport(rtcp_packet_transport);
}
// Update aggregate writable/ready-to-send state between RTP and RTCP upon
// setting new transport channels.
UpdateWritableState_n();
}
void BaseChannel::SetTransport_n(
bool rtcp,
DtlsTransportInternal* new_dtls_transport,
rtc::PacketTransportInternal* new_packet_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
if (new_dtls_transport) {
RTC_DCHECK(new_dtls_transport == new_packet_transport);
}
DtlsTransportInternal*& old_dtls_transport =
rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_;
rtc::PacketTransportInternal* old_packet_transport =
rtcp ? rtp_transport_->rtcp_packet_transport()
: rtp_transport_->rtp_packet_transport();
if (!old_packet_transport && !new_packet_transport) {
// Nothing to do.
return;
}
RTC_DCHECK(old_packet_transport != new_packet_transport);
old_dtls_transport = new_dtls_transport;
// If there's no new transport, we're done.
if (!new_packet_transport) {
return;
}
if (rtcp && new_dtls_transport) {
RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_active()))
<< "Setting RTCP for DTLS/SRTP after the DTLS is active "
<< "should never happen.";
}
auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_;
for (const auto& pair : socket_options) {
new_packet_transport->SetOption(pair.first, pair.second);
}
}
bool BaseChannel::Enable(bool enable) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
this));
return true;
}
bool BaseChannel::AddRecvStream(const StreamParams& sp) {
return InvokeOnWorker<bool>(RTC_FROM_HERE,
Bind(&BaseChannel::AddRecvStream_w, this, sp));
}
bool BaseChannel::RemoveRecvStream(uint32_t ssrc) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
}
bool BaseChannel::AddSendStream(const StreamParams& sp) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&MediaChannel::AddSendStream, media_channel(), sp));
}
bool BaseChannel::RemoveSendStream(uint32_t ssrc) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
}
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetLocalContent_w, this, content, type, error_desc));
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetRemoteContent_w, this, content, type, error_desc));
}
void BaseChannel::StartConnectionMonitor(int cms) {
// We pass in the BaseChannel instead of the rtp_dtls_transport_
// because if the rtp_dtls_transport_ changes, the ConnectionMonitor
// would be pointing to the wrong TransportChannel.
// We pass in the network thread because on that thread connection monitor
// will call BaseChannel::GetConnectionStats which must be called on the
// network thread.
connection_monitor_.reset(
new ConnectionMonitor(this, network_thread(), rtc::Thread::Current()));
connection_monitor_->SignalUpdate.connect(
this, &BaseChannel::OnConnectionMonitorUpdate);
connection_monitor_->Start(cms);
}
void BaseChannel::StopConnectionMonitor() {
if (connection_monitor_) {
connection_monitor_->Stop();
connection_monitor_.reset();
}
}
bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) {
RTC_DCHECK(network_thread_->IsCurrent());
if (!rtp_dtls_transport_) {
return false;
}
return rtp_dtls_transport_->ice_transport()->GetStats(infos);
}
bool BaseChannel::NeedsRtcpTransport() {
// If this BaseChannel doesn't require RTCP mux and we haven't fully
// negotiated RTCP mux, we need an RTCP transport.
return !rtcp_mux_required_ && !rtcp_mux_filter_.IsFullyActive();
}
bool BaseChannel::IsReadyToReceiveMedia_w() const {
// Receive data if we are enabled and have local content,
return enabled() &&
webrtc::RtpTransceiverDirectionHasRecv(local_content_direction_);
}
bool BaseChannel::IsReadyToSendMedia_w() const {
// Need to access some state updated on the network thread.
return network_thread_->Invoke<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::IsReadyToSendMedia_n, this));
}
bool BaseChannel::IsReadyToSendMedia_n() const {
// Send outgoing data if we are enabled, have local and remote content,
// and we have had some form of connectivity.
return enabled() &&
webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) &&
webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) &&
was_ever_writable() && (srtp_active() || !ShouldSetupDtlsSrtp_n());
}
bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return SendPacket(false, packet, options);
}
bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
return SendPacket(true, packet, options);
}
int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
int value) {
return network_thread_->Invoke<int>(
RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value));
}
int BaseChannel::SetOption_n(SocketType type,
rtc::Socket::Option opt,
int value) {
RTC_DCHECK(network_thread_->IsCurrent());
rtc::PacketTransportInternal* transport = nullptr;
switch (type) {
case ST_RTP:
transport = rtp_transport_->rtp_packet_transport();
socket_options_.push_back(
std::pair<rtc::Socket::Option, int>(opt, value));
break;
case ST_RTCP:
transport = rtp_transport_->rtcp_packet_transport();
rtcp_socket_options_.push_back(
std::pair<rtc::Socket::Option, int>(opt, value));
break;
}
return transport ? transport->SetOption(opt, value) : -1;
}
void BaseChannel::OnWritableState(bool writable) {
RTC_DCHECK(network_thread_->IsCurrent());
if (writable) {
// This is used to cover the scenario when the DTLS handshake is completed
// and DtlsTransport becomes writable before the remote description is set.
if (ShouldSetupDtlsSrtp_n()) {
EnableDtlsSrtp_n();
}
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::OnNetworkRouteChanged(
rtc::Optional<rtc::NetworkRoute> network_route) {
RTC_DCHECK(network_thread_->IsCurrent());
rtc::NetworkRoute new_route;
if (network_route) {
new_route = *(network_route);
}
// Note: When the RTCP-muxing is not enabled, RTCP transport and RTP transport
// use the same transport name and MediaChannel::OnNetworkRouteChanged cannot
// work correctly. Intentionally leave it broken to simplify the code and
// encourage the users to stop using non-muxing RTCP.
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] {
media_channel_->OnNetworkRouteChanged(transport_name_, new_route);
});
}
void BaseChannel::OnTransportReadyToSend(bool ready) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[=] { media_channel_->OnReadyToSend(ready); });
}
bool BaseChannel::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
// SendPacket gets called from MediaEngine, on a pacer or an encoder thread.
// If the thread is not our network thread, we will post to our network
// so that the real work happens on our network. This avoids us having to
// synchronize access to all the pieces of the send path, including
// SRTP and the inner workings of the transport channels.
// The only downside is that we can't return a proper failure code if
// needed. Since UDP is unreliable anyway, this should be a non-issue.
if (!network_thread_->IsCurrent()) {
// Avoid a copy by transferring the ownership of the packet data.
int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET;
SendPacketMessageData* data = new SendPacketMessageData;
data->packet = std::move(*packet);
data->options = options;
network_thread_->Post(RTC_FROM_HERE, this, message_id, data);
return true;
}
TRACE_EVENT0("webrtc", "BaseChannel::SendPacket");
// Now that we are on the correct thread, ensure we have a place to send this
// packet before doing anything. (We might get RTCP packets that we don't
// intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
// transport.
if (!rtp_transport_->IsWritable(rtcp)) {
return false;
}
// Protect ourselves against crazy data.
if (!ValidPacket(rtcp, packet)) {
RTC_LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
<< RtpRtcpStringLiteral(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
if (!srtp_active()) {
if (srtp_required_) {
// The audio/video engines may attempt to send RTCP packets as soon as the
// streams are created, so don't treat this as an error for RTCP.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
if (rtcp) {
return false;
}
// However, there shouldn't be any RTP packets sent before SRTP is set up
// (and SetSend(true) is called).
RTC_LOG(LS_ERROR)
<< "Can't send outgoing RTP packet when SRTP is inactive"
<< " and crypto is required";
RTC_NOTREACHED();
return false;
}
std::string packet_type = rtcp ? "RTCP" : "RTP";
RTC_LOG(LS_WARNING) << "Sending an " << packet_type
<< " packet without encryption.";
} else {
// Make sure we didn't accidentally send any packets without encryption.
RTC_DCHECK(rtp_transport_ == sdes_transport_.get() ||
rtp_transport_ == dtls_srtp_transport_.get());
}
// Bon voyage.
return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
: rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
}
bool BaseChannel::HandlesPayloadType(int packet_type) const {
return rtp_transport_->HandlesPayloadType(packet_type);
}
void BaseChannel::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
if (!has_received_packet_ && !rtcp) {
has_received_packet_ = true;
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
if (!srtp_active() && srtp_required_) {
// Our session description indicates that SRTP is required, but we got a
// packet before our SRTP filter is active. This means either that
// a) we got SRTP packets before we received the SDES keys, in which case
// we can't decrypt it anyway, or
// b) we got SRTP packets before DTLS completed on both the RTP and RTCP
// transports, so we haven't yet extracted keys, even if DTLS did
// complete on the transport that the packets are being sent on. It's
// really good practice to wait for both RTP and RTCP to be good to go
// before sending media, to prevent weird failure modes, so it's fine
// for us to just eat packets here. This is all sidestepped if RTCP mux
// is used anyway.
RTC_LOG(LS_WARNING)
<< "Can't process incoming " << RtpRtcpStringLiteral(rtcp)
<< " packet when SRTP is inactive and crypto is required";
return;
}
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time));
}
void BaseChannel::ProcessPacket(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
const rtc::PacketTime& packet_time) {
RTC_DCHECK(worker_thread_->IsCurrent());
// Need to copy variable because OnRtcpReceived/OnPacketReceived
// requires non-const pointer to buffer. This doesn't memcpy the actual data.
rtc::CopyOnWriteBuffer data(packet);
if (rtcp) {
media_channel_->OnRtcpReceived(&data, packet_time);
} else {
media_channel_->OnPacketReceived(&data, packet_time);
}
}
void BaseChannel::EnableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (enabled_)
return;
RTC_LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
UpdateMediaSendRecvState_w();
}
void BaseChannel::DisableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (!enabled_)
return;
RTC_LOG(LS_INFO) << "Channel disabled";
enabled_ = false;
UpdateMediaSendRecvState_w();
}
void BaseChannel::UpdateWritableState_n() {
rtc::PacketTransportInternal* rtp_packet_transport =
rtp_transport_->rtp_packet_transport();
rtc::PacketTransportInternal* rtcp_packet_transport =
rtp_transport_->rtcp_packet_transport();
if (rtp_packet_transport && rtp_packet_transport->writable() &&
(!rtcp_packet_transport || rtcp_packet_transport->writable())) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::ChannelWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (writable_) {
return;
}
RTC_LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
<< (was_ever_writable_ ? "" : " for the first time");
was_ever_writable_ = true;
writable_ = true;
UpdateMediaSendRecvState();
}
bool BaseChannel::ShouldSetupDtlsSrtp_n() const {
// Since DTLS is applied to all transports, checking RTP should be enough.
return rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive();
}
void BaseChannel::ChannelNotWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (!writable_)
return;
RTC_LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
writable_ = false;
UpdateMediaSendRecvState();
}
bool BaseChannel::SetRtpTransportParameters(
const MediaContentDescription* content,
SdpType type,
ContentSource src,
const RtpHeaderExtensions& extensions,
std::string* error_desc) {
std::vector<int> encrypted_extension_ids;
for (const webrtc::RtpExtension& extension : extensions) {
if (extension.encrypt) {
RTC_LOG(LS_INFO) << "Using " << (src == CS_LOCAL ? "local" : "remote")
<< " encrypted extension: " << extension.ToString();
encrypted_extension_ids.push_back(extension.id);
}
}
// Cache srtp_required_ for belt and suspenders check on SendPacket
return network_thread_->Invoke<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetRtpTransportParameters_n, this, content, type, src,
encrypted_extension_ids, error_desc));
}
bool BaseChannel::SetRtpTransportParameters_n(
const MediaContentDescription* content,
SdpType type,
ContentSource src,
const std::vector<int>& encrypted_extension_ids,
std::string* error_desc) {
RTC_DCHECK(network_thread_->IsCurrent());
if (!SetSrtp_n(content->cryptos(), type, src, encrypted_extension_ids,
error_desc)) {
return false;
}
if (!SetRtcpMux_n(content->rtcp_mux(), type, src, error_desc)) {
return false;
}
return true;
}
// |dtls| will be set to true if DTLS is active for transport and crypto is
// empty.
bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
bool* dtls,
std::string* error_desc) {
*dtls = rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive();
if (*dtls && !cryptos.empty()) {
SafeSetError("Cryptos must be empty when DTLS is active.", error_desc);
return false;
}
return true;
}
void BaseChannel::EnableSdes_n() {
if (sdes_transport_) {
return;
}
// DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
// time.
RTC_DCHECK(!dtls_srtp_transport_);
RTC_DCHECK(unencrypted_rtp_transport_);
sdes_transport_ = rtc::MakeUnique<webrtc::SrtpTransport>(
std::move(unencrypted_rtp_transport_));
#if defined(ENABLE_EXTERNAL_AUTH)
sdes_transport_->EnableExternalAuth();
#endif
SetRtpTransport(sdes_transport_.get());
RTC_LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport.";
}
void BaseChannel::EnableDtlsSrtp_n() {
if (dtls_srtp_transport_) {
return;
}
// DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
// time.
RTC_DCHECK(!sdes_transport_);
RTC_DCHECK(unencrypted_rtp_transport_);
auto srtp_transport = rtc::MakeUnique<webrtc::SrtpTransport>(
std::move(unencrypted_rtp_transport_));
#if defined(ENABLE_EXTERNAL_AUTH)
srtp_transport->EnableExternalAuth();
#endif
dtls_srtp_transport_ =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
SetRtpTransport(dtls_srtp_transport_.get());
if (cached_send_extension_ids_) {
dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds(
*cached_send_extension_ids_);
}
if (cached_recv_extension_ids_) {
dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds(
*cached_recv_extension_ids_);
}
// Set the DtlsTransport and the |dtls_srtp_transport_| will handle the DTLS
// relate signal internally.
RTC_DCHECK(rtp_dtls_transport_);
dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport_,
rtcp_dtls_transport_);
RTC_LOG(LS_INFO) << "Wrapping SrtpTransport in DtlsSrtpTransport.";
}
bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
SdpType type,
ContentSource src,
const std::vector<int>& encrypted_extension_ids,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w");
bool ret = false;
bool dtls = false;
ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc);
if (!ret) {
return false;
}
// If SRTP was not required, but we're setting a description that uses SDES,
// we need to upgrade to an SrtpTransport.
if (!sdes_transport_ && !dtls && !cryptos.empty()) {
EnableSdes_n();
}
if ((type == SdpType::kAnswer || type == SdpType::kPrAnswer) && dtls) {
EnableDtlsSrtp_n();
}
UpdateEncryptedHeaderExtensionIds(src, encrypted_extension_ids);
if (!dtls) {
switch (type) {
case SdpType::kOffer:
ret = sdes_negotiator_.SetOffer(cryptos, src);
break;
case SdpType::kPrAnswer:
ret = sdes_negotiator_.SetProvisionalAnswer(cryptos, src);
break;
case SdpType::kAnswer:
ret = sdes_negotiator_.SetAnswer(cryptos, src);
break;
default:
break;
}
// If setting an SDES answer succeeded, apply the negotiated parameters
// to the SRTP transport.
if ((type == SdpType::kPrAnswer || type == SdpType::kAnswer) && ret) {
if (sdes_negotiator_.send_cipher_suite() &&
sdes_negotiator_.recv_cipher_suite()) {
RTC_DCHECK(cached_send_extension_ids_);
RTC_DCHECK(cached_recv_extension_ids_);
ret = sdes_transport_->SetRtpParams(
*(sdes_negotiator_.send_cipher_suite()),
sdes_negotiator_.send_key().data(),
static_cast<int>(sdes_negotiator_.send_key().size()),
*(cached_send_extension_ids_),
*(sdes_negotiator_.recv_cipher_suite()),
sdes_negotiator_.recv_key().data(),
static_cast<int>(sdes_negotiator_.recv_key().size()),
*(cached_recv_extension_ids_));
} else {
RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES.";
if (type == SdpType::kAnswer && sdes_transport_) {
// Explicitly reset the |sdes_transport_| if no crypto param is
// provided in the answer. No need to call |ResetParams()| for
// |sdes_negotiator_| because it resets the params inside |SetAnswer|.
sdes_transport_->ResetParams();
}
}
}
}
if (!ret) {
SafeSetError("Failed to setup SRTP.", error_desc);
return false;
}
return true;
}
bool BaseChannel::SetRtcpMux_n(bool enable,
SdpType type,
ContentSource src,
std::string* error_desc) {
// Provide a more specific error message for the RTCP mux "require" policy
// case.
if (rtcp_mux_required_ && !enable) {
SafeSetError(
"rtcpMuxPolicy is 'require', but media description does not "
"contain 'a=rtcp-mux'.",
error_desc);
return false;
}
bool ret = false;
switch (type) {
case SdpType::kOffer:
ret = rtcp_mux_filter_.SetOffer(enable, src);
break;
case SdpType::kPrAnswer:
// This may activate RTCP muxing, but we don't yet destroy the transport
// because the final answer may deactivate it.
ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
break;
case SdpType::kAnswer:
ret = rtcp_mux_filter_.SetAnswer(enable, src);
if (ret && rtcp_mux_filter_.IsActive()) {
ActivateRtcpMux();
}
break;
default:
break;
}
if (!ret) {
SafeSetError("Failed to setup RTCP mux filter.", error_desc);
return false;
}
rtp_transport_->SetRtcpMuxEnabled(rtcp_mux_filter_.IsActive());
// |rtcp_mux_filter_| can be active if |action| is SdpType::kPrAnswer or
// SdpType::kAnswer, but we only want to tear down the RTCP transport if we
// received a final answer.
if (rtcp_mux_filter_.IsActive()) {
// If the RTP transport is already writable, then so are we.
if (rtp_transport_->rtp_packet_transport()->writable()) {
ChannelWritable_n();
}
}
return true;
}
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->AddRecvStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->RemoveRecvStream(ssrc);
}
bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
SdpType type,
std::string* error_desc) {
// Check for streams that have been removed.
bool ret = true;
for (StreamParamsVec::const_iterator it = local_streams_.begin();
it != local_streams_.end(); ++it) {
if (!GetStreamBySsrc(streams, it->first_ssrc())) {
if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
std::ostringstream desc;
desc << "Failed to remove send stream with ssrc "
<< it->first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
// Check for new streams.
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
if (media_channel()->AddSendStream(*it)) {
RTC_LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
} else {
std::ostringstream desc;
desc << "Failed to add send stream ssrc: " << it->first_ssrc();
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
local_streams_ = streams;
return ret;
}
bool BaseChannel::UpdateRemoteStreams_w(
const std::vector<StreamParams>& streams,
SdpType type,
std::string* error_desc) {
// Check for streams that have been removed.
bool ret = true;
for (StreamParamsVec::const_iterator it = remote_streams_.begin();
it != remote_streams_.end(); ++it) {
if (!GetStreamBySsrc(streams, it->first_ssrc())) {
if (!RemoveRecvStream_w(it->first_ssrc())) {
std::ostringstream desc;
desc << "Failed to remove remote stream with ssrc "
<< it->first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
// Check for new streams.
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) {
if (AddRecvStream_w(*it)) {
RTC_LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
} else {
std::ostringstream desc;
desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
remote_streams_ = streams;
return ret;
}
RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
if (!rtp_dtls_transport_ ||
!rtp_dtls_transport_->crypto_options()
.enable_encrypted_rtp_header_extensions) {
RtpHeaderExtensions filtered;
auto pred = [](const webrtc::RtpExtension& extension) {
return !extension.encrypt;
};
std::copy_if(extensions.begin(), extensions.end(),
std::back_inserter(filtered), pred);
return filtered;
}
return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions);
}
void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
const std::vector<webrtc::RtpExtension>& extensions) {
// Absolute Send Time extension id is used only with external auth,
// so do not bother searching for it and making asyncronious call to set
// something that is not used.
#if defined(ENABLE_EXTERNAL_AUTH)
const webrtc::RtpExtension* send_time_extension =
webrtc::RtpExtension::FindHeaderExtensionByUri(
extensions, webrtc::RtpExtension::kAbsSendTimeUri);
int rtp_abs_sendtime_extn_id =
send_time_extension ? send_time_extension->id : -1;
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, network_thread_,
Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this,
rtp_abs_sendtime_extn_id));
#endif
}
void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
int rtp_abs_sendtime_extn_id) {
if (sdes_transport_) {
sdes_transport_->CacheRtpAbsSendTimeHeaderExtension(
rtp_abs_sendtime_extn_id);
} else if (dtls_srtp_transport_) {
dtls_srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
rtp_abs_sendtime_extn_id);
} else {
RTC_LOG(LS_WARNING)
<< "Trying to cache the Absolute Send Time extension id "
"but the SRTP is not active.";
}
}
void BaseChannel::OnMessage(rtc::Message *pmsg) {
TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
switch (pmsg->message_id) {
case MSG_SEND_RTP_PACKET:
case MSG_SEND_RTCP_PACKET: {
RTC_DCHECK(network_thread_->IsCurrent());
SendPacketMessageData* data =
static_cast<SendPacketMessageData*>(pmsg->pdata);
bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET;
SendPacket(rtcp, &data->packet, data->options);
delete data;
break;
}
case MSG_FIRSTPACKETRECEIVED: {
SignalFirstPacketReceived(this);
break;
}
}
}
void BaseChannel::AddHandledPayloadType(int payload_type) {
rtp_transport_->AddHandledPayloadType(payload_type);
}
void BaseChannel::FlushRtcpMessages_n() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
RTC_DCHECK(network_thread_->IsCurrent());
rtc::MessageList rtcp_messages;
network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages);
for (const auto& message : rtcp_messages) {
network_thread_->Send(RTC_FROM_HERE, this, MSG_SEND_RTCP_PACKET,
message.pdata);
}
}
void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
rtc::Bind(&BaseChannel::SignalSentPacket_w, this, sent_packet));
}
void BaseChannel::SignalSentPacket_w(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(worker_thread_->IsCurrent());
SignalSentPacket(sent_packet);
}
void BaseChannel::UpdateEncryptedHeaderExtensionIds(
cricket::ContentSource source,
const std::vector<int>& extension_ids) {
if (source == ContentSource::CS_LOCAL) {
cached_recv_extension_ids_ = std::move(extension_ids);
if (dtls_srtp_transport_) {
dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds(
extension_ids);
}
} else {
cached_send_extension_ids_ = std::move(extension_ids);
if (dtls_srtp_transport_) {
dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds(
extension_ids);
}
}
}
void BaseChannel::ActivateRtcpMux() {
// We permanently activated RTCP muxing; signal that we no longer need
// the RTCP transport.
std::string debug_name =
transport_name_.empty()
? rtp_transport_->rtp_packet_transport()->transport_name()
: transport_name_;
RTC_LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
<< "; no longer need RTCP transport for " << debug_name;
if (rtp_transport_->rtcp_packet_transport()) {
SetTransport_n(/*rtcp=*/true, nullptr, nullptr);
if (dtls_srtp_transport_) {
RTC_DCHECK(rtp_dtls_transport_);
dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport_,
/*rtcp_dtls_transport_=*/nullptr);
} else {
rtp_transport_->SetRtcpPacketTransport(nullptr);
}
SignalRtcpMuxFullyActive(transport_name_);
}
UpdateWritableState_n();
}
VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
MediaEngineInterface* media_engine,
std::unique_ptr<VoiceMediaChannel> media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
rtcp_mux_required,
srtp_required),
media_engine_(media_engine) {}
VoiceChannel::~VoiceChannel() {
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
StopAudioMonitor();
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
bool VoiceChannel::SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
ssrc, enable, options, source));
}
// TODO(juberti): Handle early media the right way. We should get an explicit
// ringing message telling us to start playing local ringback, which we cancel
// if any early media actually arrives. For now, we do the opposite, which is
// to wait 1 second for early media, and start playing local ringback if none
// arrives.
void VoiceChannel::SetEarlyMedia(bool enable) {
if (enable) {
// Start the early media timeout
worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this,
MSG_EARLYMEDIATIMEOUT);
} else {
// Stop the timeout if currently going.
worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
}
}
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
}
void VoiceChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
media_monitor_->SignalUpdate.connect(
this, &VoiceChannel::OnMediaMonitorUpdate);
media_monitor_->Start(cms);
}
void VoiceChannel::StopMediaMonitor() {
if (media_monitor_) {
media_monitor_->Stop();
media_monitor_->SignalUpdate.disconnect(this);
media_monitor_.reset();
}
}
void VoiceChannel::StartAudioMonitor(int cms) {
audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
audio_monitor_
->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
audio_monitor_->Start(cms);
}
void VoiceChannel::StopAudioMonitor() {
if (audio_monitor_) {
audio_monitor_->Stop();
audio_monitor_.reset();
}
}
bool VoiceChannel::IsAudioMonitorRunning() const {
return (audio_monitor_.get() != NULL);
}
int VoiceChannel::GetInputLevel_w() {
return media_engine_->GetInputLevel();
}
int VoiceChannel::GetOutputLevel_w() {
return media_channel()->GetOutputLevel();
}
void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
media_channel()->GetActiveStreams(actives);
}
void VoiceChannel::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
BaseChannel::OnPacketReceived(rtcp, packet, packet_time);
// Set a flag when we've received an RTP packet. If we're waiting for early
// media, this will disable the timeout.
if (!received_media_ && !rtcp) {
received_media_ = true;
}
}
void BaseChannel::UpdateMediaSendRecvState() {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&BaseChannel::UpdateMediaSendRecvState_w, this));
}
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
bool recv = IsReadyToReceiveMedia_w();
media_channel()->SetPlayout(recv);
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSendMedia_w();
media_channel()->SetSend(send);
RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
}
bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
return false;
}
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(audio, rtp_header_extensions, &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set local audio description recv parameters.",
error_desc);
return false;
}
for (const AudioCodec& codec : audio->codecs()) {
AddHandledPayloadType(codec.id);
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into AudioSendParameters, and
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(audio->streams(), type, error_desc)) {
SafeSetError("Failed to set local audio description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
}
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
&send_params);
bool parameters_applied = media_channel()->SetSendParameters(send_params);
if (!parameters_applied) {
SafeSetError("Failed to set remote audio description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
// TODO(pthatcher): Move remote streams into AudioRecvParameters,
// and only give it to the media channel once we have a local
// description too (without a local description, we won't be able to
// recv them anyway).
if (!UpdateRemoteStreams_w(audio->streams(), type, error_desc)) {
SafeSetError("Failed to set remote audio description streams.", error_desc);
return false;
}
if (audio->rtp_header_extensions_set()) {
MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions);
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
void VoiceChannel::HandleEarlyMediaTimeout() {
// This occurs on the main thread, not the worker thread.
if (!received_media_) {
RTC_LOG(LS_INFO) << "No early media received before timeout";
SignalEarlyMediaTimeout(this);
}
}
void VoiceChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_EARLYMEDIATIMEOUT:
HandleEarlyMediaTimeout();
break;
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void VoiceChannel::OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
SignalConnectionMonitor(this, infos);
}
void VoiceChannel::OnMediaMonitorUpdate(
VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
RTC_DCHECK(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
const AudioInfo& info) {
SignalAudioMonitor(this, info);
}
VideoChannel::VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
rtcp_mux_required,
srtp_required) {}
VideoChannel::~VideoChannel() {
TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void VideoChannel::UpdateMediaSendRecvState_w() {
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSendMedia_w();
if (!media_channel()->SetSend(send)) {
RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel";
// TODO(gangji): Report error back to server.
}
RTC_LOG(LS_INFO) << "Changing video state, send=" << send;
}
void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo,
media_channel(), bwe_info));
}
bool VideoChannel::GetStats(VideoMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats,
media_channel(), stats));
}
void VideoChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
media_monitor_->SignalUpdate.connect(
this, &VideoChannel::OnMediaMonitorUpdate);
media_monitor_->Start(cms);
}
void VideoChannel::StopMediaMonitor() {
if (media_monitor_) {
media_monitor_->Stop();
media_monitor_.reset();
}
}
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local video description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
return false;
}
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(video, rtp_header_extensions, &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set local video description recv parameters.",
error_desc);
return false;
}
for (const VideoCodec& codec : video->codecs()) {
AddHandledPayloadType(codec.id);
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into VideoSendParameters, and
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(video->streams(), type, error_desc)) {
SafeSetError("Failed to set local video description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote video description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
}
VideoSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(video, rtp_header_extensions,
&send_params);
if (video->conference_mode()) {
send_params.conference_mode = true;
}
bool parameters_applied = media_channel()->SetSendParameters(send_params);
if (!parameters_applied) {
SafeSetError("Failed to set remote video description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
// TODO(pthatcher): Move remote streams into VideoRecvParameters,
// and only give it to the media channel once we have a local
// description too (without a local description, we won't be able to
// recv them anyway).
if (!UpdateRemoteStreams_w(video->streams(), type, error_desc)) {
SafeSetError("Failed to set remote video description streams.", error_desc);
return false;
}
if (video->rtp_header_extensions_set()) {
MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions);
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
void VideoChannel::OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) {
SignalConnectionMonitor(this, infos);
}
// TODO(pthatcher): Look into removing duplicate code between
// audio, video, and data, perhaps by using templates.
void VideoChannel::OnMediaMonitorUpdate(
VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
RTC_DCHECK(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<DataMediaChannel> media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
rtcp_mux_required,
srtp_required) {}
RtpDataChannel::~RtpDataChannel() {
TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel");
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void RtpDataChannel::Init_w(
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
this, &RtpDataChannel::OnDataChannelReadyToSend);
}
void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
BaseChannel::Init_w(rtp_transport);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
this, &RtpDataChannel::OnDataChannelReadyToSend);
}
bool RtpDataChannel::SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&DataMediaChannel::SendData, media_channel(), params,
payload, result));
}
bool RtpDataChannel::CheckDataChannelTypeFromContent(
const DataContentDescription* content,
std::string* error_desc) {
bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
(content->protocol() == kMediaProtocolDtlsSctp));
// It's been set before, but doesn't match. That's bad.
if (is_sctp) {
SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.",
error_desc);
return false;
}
return true;
}
bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local data description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in local description.", error_desc);
return false;
}
const DataContentDescription* data = content->as_data();
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
return false;
}
DataRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(data, rtp_header_extensions, &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set remote data description recv parameters.",
error_desc);
return false;
}
for (const DataCodec& codec : data->codecs()) {
AddHandledPayloadType(codec.id);
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into DataSendParameters, and
// only give it to the media channel once we have a remote
// description too (without a remote description, we won't be able
// to send them anyway).
if (!UpdateLocalStreams_w(data->streams(), type, error_desc)) {
SafeSetError("Failed to set local data description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote data description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in remote description.", error_desc);
return false;
}
const DataContentDescription* data = content->as_data();
// If the remote data doesn't have codecs, it must be empty, so ignore it.
if (!data->has_codecs()) {
return true;
}
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
RTC_LOG(LS_INFO) << "Setting remote data description";
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
}
DataSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription<DataCodec>(data, rtp_header_extensions,
&send_params);
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set remote data description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
// TODO(pthatcher): Move remote streams into DataRecvParameters,
// and only give it to the media channel once we have a local
// description too (without a local description, we won't be able to
// recv them anyway).
if (!UpdateRemoteStreams_w(data->streams(), type, error_desc)) {
SafeSetError("Failed to set remote data description streams.",
error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
void RtpDataChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
bool recv = IsReadyToReceiveMedia_w();
if (!media_channel()->SetReceive(recv)) {
RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel";
}
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSendMedia_w();
if (!media_channel()->SetSend(send)) {
RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel";
}
// Trigger SignalReadyToSendData asynchronously.
OnDataChannelReadyToSend(send);
RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
}
void RtpDataChannel::OnMessage(rtc::Message* pmsg) {
switch (pmsg->message_id) {
case MSG_READYTOSENDDATA: {
DataChannelReadyToSendMessageData* data =
static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
ready_to_send_data_ = data->data();
SignalReadyToSendData(ready_to_send_data_);
delete data;
break;
}
case MSG_DATARECEIVED: {
DataReceivedMessageData* data =
static_cast<DataReceivedMessageData*>(pmsg->pdata);
SignalDataReceived(data->params, data->payload);
delete data;
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void RtpDataChannel::OnConnectionMonitorUpdate(
ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) {
SignalConnectionMonitor(this, infos);
}
void RtpDataChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
media_monitor_->SignalUpdate.connect(this,
&RtpDataChannel::OnMediaMonitorUpdate);
media_monitor_->Start(cms);
}
void RtpDataChannel::StopMediaMonitor() {
if (media_monitor_) {
media_monitor_->Stop();
media_monitor_->SignalUpdate.disconnect(this);
media_monitor_.reset();
}
}
void RtpDataChannel::OnMediaMonitorUpdate(DataMediaChannel* media_channel,
const DataMediaInfo& info) {
RTC_DCHECK(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params,
const char* data,
size_t len) {
DataReceivedMessageData* msg = new DataReceivedMessageData(
params, data, len);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg);
}
void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
// This is usded for congestion control to indicate that the stream is ready
// to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
// that the transport channel is ready.
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
new DataChannelReadyToSendMessageData(writable));
}
} // namespace cricket
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