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webrtc / src / 4eb0188cb6a63da5fe5960ad1634c2ba5e8ea2bd / . / pc / channel.cc
blob: 544ef4fddd3ef5fe166f46a5b2289add62637418 [file] [log] [blame]
/*
* 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/rtptransport.h"
#include "pc/srtptransport.h"
namespace cricket {
using rtc::Bind;
namespace {
// See comment below for why we need to use a pointer to a unique_ptr.
bool SetRawAudioSink_w(VoiceMediaChannel* channel,
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
channel->SetRawAudioSink(ssrc, std::move(*sink));
return true;
}
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_CHANNEL_ERROR,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
MSG_FIRSTPACKETRECEIVED,
};
// Value specified in RFC 5764.
static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
static const int kAgcMinus10db = -10;
static void SafeSetError(const std::string& message, std::string* error_desc) {
if (error_desc) {
*error_desc = message;
}
}
struct VoiceChannelErrorMessageData : public rtc::MessageData {
VoiceChannelErrorMessageData(uint32_t in_ssrc,
VoiceMediaChannel::Error in_error)
: ssrc(in_ssrc), error(in_error) {}
uint32_t ssrc;
VoiceMediaChannel::Error error;
};
struct VideoChannelErrorMessageData : public rtc::MessageData {
VideoChannelErrorMessageData(uint32_t in_ssrc,
VideoMediaChannel::Error in_error)
: ssrc(in_ssrc), error(in_error) {}
uint32_t ssrc;
VideoMediaChannel::Error error;
};
struct DataChannelErrorMessageData : public rtc::MessageData {
DataChannelErrorMessageData(uint32_t in_ssrc,
DataMediaChannel::Error in_error)
: ssrc(in_ssrc), error(in_error) {}
uint32_t ssrc;
DataMediaChannel::Error error;
};
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());
}
static bool IsReceiveContentDirection(MediaContentDirection direction) {
return direction == MD_SENDRECV || direction == MD_RECVONLY;
}
static bool IsSendContentDirection(MediaContentDirection direction) {
return direction == MD_SENDRECV || direction == MD_SENDONLY;
}
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),
srtp_required_(srtp_required),
media_channel_(std::move(media_channel)) {
RTC_DCHECK_RUN_ON(worker_thread_);
if (srtp_required) {
auto transport =
rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_required, content_name);
srtp_transport_ = transport.get();
rtp_transport_ = std::move(transport);
#if defined(ENABLE_EXTERNAL_AUTH)
srtp_transport_->EnableExternalAuth();
#endif
} else {
rtp_transport_ = rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required);
srtp_transport_ = nullptr;
}
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);
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::DisconnectTransportChannels_n() {
// Send any outstanding RTCP packets.
FlushRtcpMessages_n();
// Stop signals from transport channels, but keep them alive because
// media_channel may use them from a different thread.
if (rtp_dtls_transport_) {
DisconnectFromDtlsTransport(rtp_dtls_transport_);
} else if (rtp_transport_->rtp_packet_transport()) {
DisconnectFromPacketTransport(rtp_transport_->rtp_packet_transport());
}
if (rtcp_dtls_transport_) {
DisconnectFromDtlsTransport(rtcp_dtls_transport_);
} else if (rtp_transport_->rtcp_packet_transport()) {
DisconnectFromPacketTransport(rtp_transport_->rtcp_packet_transport());
}
rtp_transport_->SetRtpPacketTransport(nullptr);
rtp_transport_->SetRtcpPacketTransport(nullptr);
// Clear pending read packets/messages.
network_thread_->Clear(&invoker_);
network_thread_->Clear(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, [&] {
return InitNetwork_n(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport);
});
// 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::InitNetwork_n(
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport, rtp_packet_transport,
rtcp_packet_transport);
if (rtcp_mux_required_) {
rtcp_mux_filter_.SetActive();
}
}
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, Bind(&BaseChannel::DisconnectTransportChannels_n, this));
}
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());
}
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 (rtp_packet_transport == rtp_transport_->rtp_packet_transport()) {
// Nothing to do if transport isn't changing.
return;
}
// When using DTLS-SRTP, we must reset the SrtpTransport every time the
// DtlsTransport changes and wait until the DTLS handshake is complete to set
// the newly negotiated parameters.
if (ShouldSetupDtlsSrtp_n()) {
// Set |writable_| to false such that UpdateWritableState_w can set up
// DTLS-SRTP when |writable_| becomes true again.
writable_ = false;
dtls_active_ = false;
if (srtp_transport_) {
srtp_transport_->ResetParams();
}
}
// 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(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(false, rtp_dtls_transport, rtp_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);
if (old_dtls_transport) {
DisconnectFromDtlsTransport(old_dtls_transport);
} else if (old_packet_transport) {
DisconnectFromPacketTransport(old_packet_transport);
}
if (rtcp) {
rtp_transport_->SetRtcpPacketTransport(new_packet_transport);
} else {
rtp_transport_->SetRtpPacketTransport(new_packet_transport);
}
old_dtls_transport = new_dtls_transport;
// If there's no new transport, we're done after disconnecting from old one.
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.";
}
if (new_dtls_transport) {
ConnectToDtlsTransport(new_dtls_transport);
} else {
ConnectToPacketTransport(new_packet_transport);
}
auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_;
for (const auto& pair : socket_options) {
new_packet_transport->SetOption(pair.first, pair.second);
}
}
void BaseChannel::ConnectToDtlsTransport(DtlsTransportInternal* transport) {
RTC_DCHECK(network_thread_->IsCurrent());
// TODO(zstein): de-dup with ConnectToPacketTransport
transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
transport->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState);
transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n);
}
void BaseChannel::DisconnectFromDtlsTransport(
DtlsTransportInternal* transport) {
RTC_DCHECK(network_thread_->IsCurrent());
transport->SignalWritableState.disconnect(this);
transport->SignalDtlsState.disconnect(this);
transport->SignalSentPacket.disconnect(this);
}
void BaseChannel::ConnectToPacketTransport(
rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(network_thread_);
transport->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
transport->SignalSentPacket.connect(this, &BaseChannel::SignalSentPacket_n);
}
void BaseChannel::DisconnectFromPacketTransport(
rtc::PacketTransportInternal* transport) {
RTC_DCHECK_RUN_ON(network_thread_);
transport->SignalWritableState.disconnect(this);
transport->SignalSentPacket.disconnect(this);
}
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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetLocalContent_w, this, content, action, error_desc));
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, this, content,
action, 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() && IsReceiveContentDirection(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() && IsReceiveContentDirection(remote_content_direction_) &&
IsSendContentDirection(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(rtc::PacketTransportInternal* transport) {
RTC_DCHECK(transport == rtp_transport_->rtp_packet_transport() ||
transport == rtp_transport_->rtcp_packet_transport());
RTC_DCHECK(network_thread_->IsCurrent());
UpdateWritableState_n();
}
void BaseChannel::OnDtlsState(DtlsTransportInternal* transport,
DtlsTransportState state) {
if (!ShouldSetupDtlsSrtp_n()) {
return;
}
// Reset the SrtpTransport if it's not the CONNECTED state. For the CONNECTED
// state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to
// cover other scenarios like the whole transport is writable (not just this
// TransportChannel) or when TransportChannel is attached after DTLS is
// negotiated.
if (state != DTLS_TRANSPORT_CONNECTED) {
dtls_active_ = false;
if (srtp_transport_) {
srtp_transport_->ResetParams();
}
}
}
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;
}
// Bon voyage.
return rtcp
? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
: rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
}
RTC_DCHECK(srtp_transport_);
RTC_DCHECK(srtp_transport_->IsActive());
// Bon voyage.
return rtcp ? srtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
: srtp_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;
MaybeSetupDtlsSrtp_n();
writable_ = true;
UpdateMediaSendRecvState();
}
void BaseChannel::SignalDtlsSrtpSetupFailure_n(bool rtcp) {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(),
Bind(&BaseChannel::SignalDtlsSrtpSetupFailure_s, this, rtcp));
}
void BaseChannel::SignalDtlsSrtpSetupFailure_s(bool rtcp) {
RTC_DCHECK(signaling_thread() == rtc::Thread::Current());
SignalDtlsSrtpSetupFailure(this, rtcp);
}
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();
}
// This function returns true if either DTLS-SRTP is not in use
// *or* DTLS-SRTP is successfully set up.
bool BaseChannel::SetupDtlsSrtp_n(bool rtcp) {
RTC_DCHECK(network_thread_->IsCurrent());
bool ret = false;
DtlsTransportInternal* transport =
rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_;
RTC_DCHECK(transport);
RTC_DCHECK(transport->IsDtlsActive());
int selected_crypto_suite;
if (!transport->GetSrtpCryptoSuite(&selected_crypto_suite)) {
RTC_LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
return false;
}
RTC_LOG(LS_INFO) << "Installing keys from DTLS-SRTP on " << content_name()
<< " " << RtpRtcpStringLiteral(rtcp);
int key_len;
int salt_len;
if (!rtc::GetSrtpKeyAndSaltLengths(selected_crypto_suite, &key_len,
&salt_len)) {
RTC_LOG(LS_ERROR) << "Unknown DTLS-SRTP crypto suite"
<< selected_crypto_suite;
return false;
}
// OK, we're now doing DTLS (RFC 5764)
std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2);
// RFC 5705 exporter using the RFC 5764 parameters
if (!transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0, false,
&dtls_buffer[0], dtls_buffer.size())) {
RTC_LOG(LS_WARNING) << "DTLS-SRTP key export failed";
RTC_NOTREACHED(); // This should never happen
return false;
}
// Sync up the keys with the DTLS-SRTP interface
std::vector<unsigned char> client_write_key(key_len + salt_len);
std::vector<unsigned char> server_write_key(key_len + salt_len);
size_t offset = 0;
memcpy(&client_write_key[0], &dtls_buffer[offset], key_len);
offset += key_len;
memcpy(&server_write_key[0], &dtls_buffer[offset], key_len);
offset += key_len;
memcpy(&client_write_key[key_len], &dtls_buffer[offset], salt_len);
offset += salt_len;
memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len);
std::vector<unsigned char> *send_key, *recv_key;
rtc::SSLRole role;
if (!transport->GetSslRole(&role)) {
RTC_LOG(LS_WARNING) << "GetSslRole failed";
return false;
}
if (role == rtc::SSL_SERVER) {
send_key = &server_write_key;
recv_key = &client_write_key;
} else {
send_key = &client_write_key;
recv_key = &server_write_key;
}
// Use an empty encrypted header extension ID vector if not set. This could
// happen when the DTLS handshake is completed before processing the
// Offer/Answer which contains the encrypted header extension IDs.
std::vector<int> send_extension_ids;
std::vector<int> recv_extension_ids;
if (catched_send_extension_ids_) {
send_extension_ids = *catched_send_extension_ids_;
}
if (catched_recv_extension_ids_) {
recv_extension_ids = *catched_recv_extension_ids_;
}
if (rtcp) {
if (!dtls_active()) {
RTC_DCHECK(srtp_transport_);
ret = srtp_transport_->SetRtcpParams(
selected_crypto_suite, &(*send_key)[0],
static_cast<int>(send_key->size()), send_extension_ids,
selected_crypto_suite, &(*recv_key)[0],
static_cast<int>(recv_key->size()), recv_extension_ids);
} else {
// RTCP doesn't need to call SetRtpParam because it is only used
// to make the updated encrypted RTP header extension IDs take effect.
ret = true;
}
} else {
RTC_DCHECK(srtp_transport_);
ret = srtp_transport_->SetRtpParams(
selected_crypto_suite, &(*send_key)[0],
static_cast<int>(send_key->size()), send_extension_ids,
selected_crypto_suite, &(*recv_key)[0],
static_cast<int>(recv_key->size()), recv_extension_ids);
dtls_active_ = ret;
}
if (!ret) {
RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
}
return ret;
}
void BaseChannel::MaybeSetupDtlsSrtp_n() {
if (dtls_active()) {
return;
}
if (!ShouldSetupDtlsSrtp_n()) {
return;
}
if (!srtp_transport_) {
EnableSrtpTransport_n();
}
if (!SetupDtlsSrtp_n(false)) {
SignalDtlsSrtpSetupFailure_n(false);
return;
}
if (rtcp_dtls_transport_) {
if (!SetupDtlsSrtp_n(true)) {
SignalDtlsSrtpSetupFailure_n(true);
return;
}
}
}
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,
ContentAction action,
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, action, src, encrypted_extension_ids,
error_desc));
}
bool BaseChannel::SetRtpTransportParameters_n(
const MediaContentDescription* content,
ContentAction action,
ContentSource src,
const std::vector<int>& encrypted_extension_ids,
std::string* error_desc) {
RTC_DCHECK(network_thread_->IsCurrent());
if (!SetSrtp_n(content->cryptos(), action, src, encrypted_extension_ids,
error_desc)) {
return false;
}
if (!SetRtcpMux_n(content->rtcp_mux(), action, 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::EnableSrtpTransport_n() {
if (srtp_transport_ == nullptr) {
rtp_transport_->SignalReadyToSend.disconnect(this);
rtp_transport_->SignalPacketReceived.disconnect(this);
rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
auto transport = rtc::MakeUnique<webrtc::SrtpTransport>(
std::move(rtp_transport_), content_name_);
srtp_transport_ = transport.get();
rtp_transport_ = std::move(transport);
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
rtp_transport_->SignalPacketReceived.connect(
this, &BaseChannel::OnPacketReceived);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
RTC_LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport.";
}
}
bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
ContentAction action,
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 (!srtp_transport_ && !dtls && !cryptos.empty()) {
EnableSrtpTransport_n();
}
bool encrypted_header_extensions_id_changed =
EncryptedHeaderExtensionIdsChanged(src, encrypted_extension_ids);
CacheEncryptedHeaderExtensionIds(src, encrypted_extension_ids);
switch (action) {
case CA_OFFER:
// If DTLS is already active on the channel, we could be renegotiating
// here. We don't update the srtp filter.
if (!dtls) {
ret = sdes_negotiator_.SetOffer(cryptos, src);
}
break;
case CA_PRANSWER:
// If we're doing DTLS-SRTP, we don't want to update the filter
// with an answer, because we already have SRTP parameters.
if (!dtls) {
ret = sdes_negotiator_.SetProvisionalAnswer(cryptos, src);
}
break;
case CA_ANSWER:
// If we're doing DTLS-SRTP, we don't want to update the filter
// with an answer, because we already have SRTP parameters.
if (!dtls) {
ret = sdes_negotiator_.SetAnswer(cryptos, src);
}
break;
default:
break;
}
// If setting an SDES answer succeeded, apply the negotiated parameters
// to the SRTP transport.
if ((action == CA_PRANSWER || action == CA_ANSWER) && !dtls && ret) {
if (sdes_negotiator_.send_cipher_suite() &&
sdes_negotiator_.recv_cipher_suite()) {
RTC_DCHECK(catched_send_extension_ids_);
RTC_DCHECK(catched_recv_extension_ids_);
ret = srtp_transport_->SetRtpParams(
*(sdes_negotiator_.send_cipher_suite()),
sdes_negotiator_.send_key().data(),
static_cast<int>(sdes_negotiator_.send_key().size()),
*(catched_send_extension_ids_),
*(sdes_negotiator_.recv_cipher_suite()),
sdes_negotiator_.recv_key().data(),
static_cast<int>(sdes_negotiator_.recv_key().size()),
*(catched_recv_extension_ids_));
} else {
RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES.";
if (action == CA_ANSWER && srtp_transport_) {
// Explicitly reset the |srtp_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|.
srtp_transport_->ResetParams();
}
}
}
// Only update SRTP transport if using DTLS. SDES is handled internally
// by the SRTP filter.
if (ret && dtls_active() && rtp_dtls_transport_ &&
rtp_dtls_transport_->dtls_state() == DTLS_TRANSPORT_CONNECTED &&
encrypted_header_extensions_id_changed) {
ret = SetupDtlsSrtp_n(/*rtcp=*/false);
}
if (!ret) {
SafeSetError("Failed to setup SRTP.", error_desc);
return false;
}
return true;
}
bool BaseChannel::SetRtcpMux_n(bool enable,
ContentAction action,
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 (action) {
case CA_OFFER:
ret = rtcp_mux_filter_.SetOffer(enable, src);
break;
case CA_PRANSWER:
// 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 CA_ANSWER:
ret = rtcp_mux_filter_.SetAnswer(enable, src);
if (ret && rtcp_mux_filter_.IsActive()) {
// 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(true, nullptr, nullptr);
SignalRtcpMuxFullyActive(transport_name_);
}
UpdateWritableState_n();
}
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 CA_PRANSWER or
// CA_ANSWER, 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,
ContentAction action,
std::string* error_desc) {
if (!(action == CA_OFFER || action == CA_ANSWER || action == CA_PRANSWER))
return false;
// 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,
ContentAction action,
std::string* error_desc) {
if (!(action == CA_OFFER || action == CA_ANSWER || action == CA_PRANSWER))
return false;
// 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 (srtp_transport_) {
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(
rtc::PacketTransportInternal* /* transport */,
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::CacheEncryptedHeaderExtensionIds(
cricket::ContentSource source,
const std::vector<int>& extension_ids) {
source == ContentSource::CS_LOCAL
? catched_recv_extension_ids_.emplace(extension_ids)
: catched_send_extension_ids_.emplace(extension_ids);
}
bool BaseChannel::EncryptedHeaderExtensionIdsChanged(
cricket::ContentSource source,
const std::vector<int>& new_extension_ids) {
if (source == ContentSource::CS_LOCAL) {
return !catched_recv_extension_ids_ ||
(*catched_recv_extension_ids_) != new_extension_ids;
} else {
return !catched_send_extension_ids_ ||
(*catched_send_extension_ids_) != new_extension_ids;
}
}
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::CanInsertDtmf() {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&VoiceMediaChannel::CanInsertDtmf, media_channel()));
}
bool VoiceChannel::InsertDtmf(uint32_t ssrc,
int event_code,
int duration) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VoiceChannel::InsertDtmf_w, this, ssrc, event_code, duration));
}
bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VoiceMediaChannel::SetOutputVolume, media_channel(), ssrc, volume));
}
void VoiceChannel::SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
// We need to work around Bind's lack of support for unique_ptr and ownership
// passing. So we invoke to our own little routine that gets a pointer to
// our local variable. This is OK since we're synchronously invoking.
InvokeOnWorker<bool>(RTC_FROM_HERE,
Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
}
webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const {
return worker_thread()->Invoke<webrtc::RtpParameters>(
RTC_FROM_HERE, Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc));
}
webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w(
uint32_t ssrc) const {
return media_channel()->GetRtpSendParameters(ssrc);
}
bool VoiceChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters));
}
bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters) {
return media_channel()->SetRtpSendParameters(ssrc, parameters);
}
webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
return worker_thread()->Invoke<webrtc::RtpParameters>(
RTC_FROM_HERE,
Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc));
}
webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w(
uint32_t ssrc) const {
return media_channel()->GetRtpReceiveParameters(ssrc);
}
bool VoiceChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
}
bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters) {
return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
}
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
}
std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const {
return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>(
RTC_FROM_HERE, Bind(&VoiceChannel::GetSources_w, this, ssrc));
}
std::vector<webrtc::RtpSource> VoiceChannel::GetSources_w(uint32_t ssrc) const {
RTC_DCHECK(worker_thread()->IsCurrent());
return media_channel()->GetSources(ssrc);
}
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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
RTC_LOG(LS_INFO) << "Setting local voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
RTC_DCHECK(audio != NULL);
if (!audio) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
if (!SetRtpTransportParameters(content, action, 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(), action, 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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
RTC_LOG(LS_INFO) << "Setting remote voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
RTC_DCHECK(audio != NULL);
if (!audio) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
if (!SetRtpTransportParameters(content, action, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
}
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
&send_params);
if (audio->agc_minus_10db()) {
send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
}
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(), action, 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);
}
}
bool VoiceChannel::InsertDtmf_w(uint32_t ssrc,
int event,
int duration) {
if (!enabled()) {
return false;
}
return media_channel()->InsertDtmf(ssrc, event, duration);
}
void VoiceChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_EARLYMEDIATIMEOUT:
HandleEarlyMediaTimeout();
break;
case MSG_CHANNEL_ERROR: {
VoiceChannelErrorMessageData* data =
static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
delete data;
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();
}
bool VideoChannel::SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink));
return true;
}
bool VideoChannel::SetVideoSend(
uint32_t ssrc,
bool mute,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
ssrc, mute, options, source));
}
webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const {
return worker_thread()->Invoke<webrtc::RtpParameters>(
RTC_FROM_HERE, Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc));
}
webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w(
uint32_t ssrc) const {
return media_channel()->GetRtpSendParameters(ssrc);
}
bool VideoChannel::SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters));
}
bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters) {
return media_channel()->SetRtpSendParameters(ssrc, parameters);
}
webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters(
uint32_t ssrc) const {
return worker_thread()->Invoke<webrtc::RtpParameters>(
RTC_FROM_HERE,
Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc));
}
webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w(
uint32_t ssrc) const {
return media_channel()->GetRtpReceiveParameters(ssrc);
}
bool VideoChannel::SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
}
bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters) {
return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
}
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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
RTC_LOG(LS_INFO) << "Setting local video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
RTC_DCHECK(video != NULL);
if (!video) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
if (!SetRtpTransportParameters(content, action, 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(), action, 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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
RTC_LOG(LS_INFO) << "Setting remote video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
RTC_DCHECK(video != NULL);
if (!video) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
if (!SetRtpTransportParameters(content, action, 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(), action, 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::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_CHANNEL_ERROR: {
const VideoChannelErrorMessageData* data =
static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
delete data;
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
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);
}
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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
RTC_LOG(LS_INFO) << "Setting local data description";
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
RTC_DCHECK(data != NULL);
if (!data) {
SafeSetError("Can't find data content in local description.", error_desc);
return false;
}
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
if (!SetRtpTransportParameters(content, action, 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(), action, 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,
ContentAction action,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w");
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
RTC_DCHECK(data != NULL);
if (!data) {
SafeSetError("Can't find data content in remote description.", error_desc);
return false;
}
// 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, action, 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(), action, 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;
}
case MSG_CHANNEL_ERROR: {
const DataChannelErrorMessageData* data =
static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
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::OnDataChannelError(uint32_t ssrc,
DataMediaChannel::Error err) {
DataChannelErrorMessageData* data = new DataChannelErrorMessageData(
ssrc, err);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_CHANNEL_ERROR, data);
}
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