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webrtc_m130/webrtc/pc/channel.cc
zhihuang 38ede13042 Support building WebRTC without audio and video. 
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).

The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.

The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.

Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)

BUG=webrtc:7613

Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
2017-06-15 19:52:32 +00:00

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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 <utility>
#include "webrtc/pc/channel.h"
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/rtputils.h"
// Adding 'nogncheck' to disable the gn include headers check to support modular
// WebRTC build targets.
#include "webrtc/media/engine/webrtcvoiceengine.h" // nogncheck
#include "webrtc/p2p/base/packettransportinternal.h"
#include "webrtc/pc/channelmanager.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;
};
#if defined(ENABLE_EXTERNAL_AUTH)
// Returns the named header extension if found among all extensions,
// nullptr otherwise.
const webrtc::RtpExtension* FindHeaderExtension(
const std::vector<webrtc::RtpExtension>& extensions,
const std::string& uri) {
for (const auto& extension : extensions) {
if (extension.uri == uri)
return &extension;
}
return nullptr;
}
#endif
} // 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;
}
static const MediaContentDescription* GetContentDescription(
const ContentInfo* cinfo) {
if (cinfo == NULL)
return NULL;
return static_cast<const MediaContentDescription*>(cinfo->description);
}
template <class Codec>
void RtpParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
RtpParameters<Codec>* params) {
// TODO(pthatcher): Remove this once we're sure no one will give us
// a description without codecs (currently a CA_UPDATE with just
// streams can).
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 = desc->rtp_header_extensions();
}
params->rtcp.reduced_size = desc->rtcp_reduced_size();
}
template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
RtpSendParameters<Codec>* send_params) {
RtpParametersFromMediaDescription(desc, send_params);
send_params->max_bandwidth_bps = desc->bandwidth();
}
BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
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),
rtp_transport_(rtcp_mux_required),
srtp_required_(srtp_required),
media_channel_(media_channel),
selected_candidate_pair_(nullptr) {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
#if defined(ENABLE_EXTERNAL_AUTH)
srtp_filter_.EnableExternalAuth();
#endif
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);
LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
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.
delete media_channel_;
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);
}
bool BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
if (!network_thread_->Invoke<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::InitNetwork_n, this,
rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport))) {
return false;
}
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
RTC_DCHECK_RUN_ON(worker_thread_);
media_channel_->SetInterface(this);
return true;
}
bool 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();
}
return true;
}
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->debug_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 SrtpFilter every time the transport
// 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;
srtp_filter_.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) {
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);
}
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());
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_filter_.IsActive()))
<< "Setting RTCP for DTLS/SRTP after SrtpFilter 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);
transport->ice_transport()->SignalSelectedCandidatePairChanged.connect(
this, &BaseChannel::OnSelectedCandidatePairChanged);
}
void BaseChannel::DisconnectFromDtlsTransport(
DtlsTransportInternal* transport) {
RTC_DCHECK(network_thread_->IsCurrent());
OnSelectedCandidatePairChanged(transport->ice_transport(), nullptr, -1,
false);
transport->SignalWritableState.disconnect(this);
transport->SignalDtlsState.disconnect(this);
transport->SignalSentPacket.disconnect(this);
transport->ice_transport()->SignalSelectedCandidatePairChanged.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_filter_.IsActive() || !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 srtp filter 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) {
srtp_filter_.ResetParams();
}
}
void BaseChannel::OnSelectedCandidatePairChanged(
IceTransportInternal* ice_transport,
CandidatePairInterface* selected_candidate_pair,
int last_sent_packet_id,
bool ready_to_send) {
RTC_DCHECK((rtp_dtls_transport_ &&
ice_transport == rtp_dtls_transport_->ice_transport()) ||
(rtcp_dtls_transport_ &&
ice_transport == rtcp_dtls_transport_->ice_transport()));
RTC_DCHECK(network_thread_->IsCurrent());
selected_candidate_pair_ = selected_candidate_pair;
std::string transport_name = ice_transport->transport_name();
rtc::NetworkRoute network_route;
if (selected_candidate_pair) {
network_route = rtc::NetworkRoute(
ready_to_send, selected_candidate_pair->local_candidate().network_id(),
selected_candidate_pair->remote_candidate().network_id(),
last_sent_packet_id);
UpdateTransportOverhead();
}
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&MediaChannel::OnNetworkRouteChanged, media_channel_, transport_name,
network_route));
}
void BaseChannel::OnTransportReadyToSend(bool ready) {
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&MediaChannel::OnReadyToSend, media_channel_, 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)) {
LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
<< RtpRtcpStringLiteral(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
rtc::PacketOptions updated_options;
updated_options = options;
// Protect if needed.
if (srtp_filter_.IsActive()) {
TRACE_EVENT0("webrtc", "SRTP Encode");
bool res;
uint8_t* data = packet->data();
int len = static_cast<int>(packet->size());
if (!rtcp) {
// If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
// inside libsrtp for a RTP packet. A external HMAC module will be writing
// a fake HMAC value. This is ONLY done for a RTP packet.
// Socket layer will update rtp sendtime extension header if present in
// packet with current time before updating the HMAC.
#if !defined(ENABLE_EXTERNAL_AUTH)
res = srtp_filter_.ProtectRtp(
data, len, static_cast<int>(packet->capacity()), &len);
#else
if (!srtp_filter_.IsExternalAuthActive()) {
res = srtp_filter_.ProtectRtp(
data, len, static_cast<int>(packet->capacity()), &len);
} else {
updated_options.packet_time_params.rtp_sendtime_extension_id =
rtp_abs_sendtime_extn_id_;
res = srtp_filter_.ProtectRtp(
data, len, static_cast<int>(packet->capacity()), &len,
&updated_options.packet_time_params.srtp_packet_index);
// If protection succeeds, let's get auth params from srtp.
if (res) {
uint8_t* auth_key = NULL;
int key_len;
res = srtp_filter_.GetRtpAuthParams(
&auth_key, &key_len,
&updated_options.packet_time_params.srtp_auth_tag_len);
if (res) {
updated_options.packet_time_params.srtp_auth_key.resize(key_len);
updated_options.packet_time_params.srtp_auth_key.assign(
auth_key, auth_key + key_len);
}
}
}
#endif
if (!res) {
int seq_num = -1;
uint32_t ssrc = 0;
GetRtpSeqNum(data, len, &seq_num);
GetRtpSsrc(data, len, &ssrc);
LOG(LS_ERROR) << "Failed to protect " << content_name_
<< " RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return false;
}
} else {
res = srtp_filter_.ProtectRtcp(data, len,
static_cast<int>(packet->capacity()),
&len);
if (!res) {
int type = -1;
GetRtcpType(data, len, &type);
LOG(LS_ERROR) << "Failed to protect " << content_name_
<< " RTCP packet: size=" << len << ", type=" << type;
return false;
}
}
// Update the length of the packet now that we've added the auth tag.
packet->SetSize(len);
} else 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).
LOG(LS_ERROR) << "Can't send outgoing RTP packet when SRTP is inactive"
<< " and crypto is required";
RTC_NOTREACHED();
return false;
}
// Bon voyage.
int flags = (secure() && secure_dtls()) ? PF_SRTP_BYPASS : PF_NORMAL;
return rtp_transport_.SendPacket(rtcp, packet, updated_options, flags);
}
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);
}
// Unprotect the packet, if needed.
if (srtp_filter_.IsActive()) {
TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet.data<char>();
int len = static_cast<int>(packet.size());
bool res;
if (!rtcp) {
res = srtp_filter_.UnprotectRtp(data, len, &len);
if (!res) {
int seq_num = -1;
uint32_t ssrc = 0;
GetRtpSeqNum(data, len, &seq_num);
GetRtpSsrc(data, len, &ssrc);
LOG(LS_ERROR) << "Failed to unprotect " << content_name_
<< " RTP packet: size=" << len << ", seqnum=" << seq_num
<< ", SSRC=" << ssrc;
return;
}
} else {
res = srtp_filter_.UnprotectRtcp(data, len, &len);
if (!res) {
int type = -1;
GetRtcpType(data, len, &type);
LOG(LS_ERROR) << "Failed to unprotect " << content_name_
<< " RTCP packet: size=" << len << ", type=" << type;
return;
}
}
packet.SetSize(len);
} else if (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.
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);
}
}
bool BaseChannel::PushdownLocalDescription(
const SessionDescription* local_desc, ContentAction action,
std::string* error_desc) {
const ContentInfo* content_info = GetFirstContent(local_desc);
const MediaContentDescription* content_desc =
GetContentDescription(content_info);
if (content_desc && content_info && !content_info->rejected &&
!SetLocalContent(content_desc, action, error_desc)) {
LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
return false;
}
return true;
}
bool BaseChannel::PushdownRemoteDescription(
const SessionDescription* remote_desc, ContentAction action,
std::string* error_desc) {
const ContentInfo* content_info = GetFirstContent(remote_desc);
const MediaContentDescription* content_desc =
GetContentDescription(content_info);
if (content_desc && content_info && !content_info->rejected &&
!SetRemoteContent(content_desc, action, error_desc)) {
LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
return false;
}
return true;
}
void BaseChannel::EnableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (enabled_)
return;
LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
UpdateMediaSendRecvState_w();
}
void BaseChannel::DisableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (!enabled_)
return;
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;
}
LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
<< (was_ever_writable_ ? "" : " for the first time");
if (selected_candidate_pair_)
LOG(LS_INFO)
<< "Using "
<< selected_candidate_pair_->local_candidate().ToSensitiveString()
<< "->"
<< selected_candidate_pair_->remote_candidate().ToSensitiveString();
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)) {
LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
return false;
}
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)) {
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())) {
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)) {
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;
}
if (rtcp) {
ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0],
static_cast<int>(send_key->size()),
selected_crypto_suite, &(*recv_key)[0],
static_cast<int>(recv_key->size()));
} else {
ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0],
static_cast<int>(send_key->size()),
selected_crypto_suite, &(*recv_key)[0],
static_cast<int>(recv_key->size()));
}
if (!ret) {
LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
} else {
dtls_keyed_ = true;
UpdateTransportOverhead();
}
return ret;
}
void BaseChannel::MaybeSetupDtlsSrtp_n() {
if (srtp_filter_.IsActive()) {
return;
}
if (!ShouldSetupDtlsSrtp_n()) {
return;
}
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;
LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
writable_ = false;
UpdateMediaSendRecvState();
}
bool BaseChannel::SetRtpTransportParameters(
const MediaContentDescription* content,
ContentAction action,
ContentSource src,
std::string* error_desc) {
if (action == CA_UPDATE) {
// These parameters never get changed by a CA_UDPATE.
return true;
}
// 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, error_desc));
}
bool BaseChannel::SetRtpTransportParameters_n(
const MediaContentDescription* content,
ContentAction action,
ContentSource src,
std::string* error_desc) {
RTC_DCHECK(network_thread_->IsCurrent());
if (!SetSrtp_n(content->cryptos(), action, src, 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;
}
bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
ContentAction action,
ContentSource src,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w");
if (action == CA_UPDATE) {
// no crypto params.
return true;
}
bool ret = false;
bool dtls = false;
ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc);
if (!ret) {
return false;
}
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 = srtp_filter_.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 = srtp_filter_.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 = srtp_filter_.SetAnswer(cryptos, src);
}
break;
default:
break;
}
if (!ret) {
SafeSetError("Failed to setup SRTP filter.", 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()->debug_name()
: transport_name_;
;
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;
case CA_UPDATE:
// No RTCP mux info.
ret = true;
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 || action == CA_UPDATE))
return false;
// If this is an update, streams only contain streams that have changed.
if (action == CA_UPDATE) {
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
const StreamParams* existing_stream =
GetStreamByIds(local_streams_, it->groupid, it->id);
if (!existing_stream && it->has_ssrcs()) {
if (media_channel()->AddSendStream(*it)) {
local_streams_.push_back(*it);
LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
} else {
std::ostringstream desc;
desc << "Failed to add send stream ssrc: " << it->first_ssrc();
SafeSetError(desc.str(), error_desc);
return false;
}
} else if (existing_stream && !it->has_ssrcs()) {
if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) {
std::ostringstream desc;
desc << "Failed to remove send stream with ssrc "
<< it->first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
return false;
}
RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc());
} else {
LOG(LS_WARNING) << "Ignore unsupported stream update";
}
}
return true;
}
// Else streams are all the streams we want to send.
// 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)) {
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 || action == CA_UPDATE))
return false;
// If this is an update, streams only contain streams that have changed.
if (action == CA_UPDATE) {
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
const StreamParams* existing_stream =
GetStreamByIds(remote_streams_, it->groupid, it->id);
if (!existing_stream && it->has_ssrcs()) {
if (AddRecvStream_w(*it)) {
remote_streams_.push_back(*it);
LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
} else {
std::ostringstream desc;
desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
SafeSetError(desc.str(), error_desc);
return false;
}
} else if (existing_stream && !it->has_ssrcs()) {
if (!RemoveRecvStream_w(existing_stream->first_ssrc())) {
std::ostringstream desc;
desc << "Failed to remove remote stream with ssrc "
<< it->first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
return false;
}
RemoveStreamBySsrc(&remote_streams_, existing_stream->first_ssrc());
} else {
LOG(LS_WARNING) << "Ignore unsupported stream update."
<< " Stream exists? " << (existing_stream != nullptr)
<< " new stream = " << it->ToString();
}
}
return true;
}
// Else streams are all the streams we want to receive.
// 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)) {
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;
}
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 =
FindHeaderExtension(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) {
rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
}
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);
}
VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
MediaEngineInterface* media_engine,
VoiceMediaChannel* media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
media_channel,
content_name,
rtcp_mux_required,
srtp_required),
media_engine_(media_engine),
received_media_(false) {}
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));
}
int BaseChannel::GetTransportOverheadPerPacket() const {
RTC_DCHECK(network_thread_->IsCurrent());
if (!selected_candidate_pair_)
return 0;
int transport_overhead_per_packet = 0;
constexpr int kIpv4Overhaed = 20;
constexpr int kIpv6Overhaed = 40;
transport_overhead_per_packet +=
selected_candidate_pair_->local_candidate().address().family() == AF_INET
? kIpv4Overhaed
: kIpv6Overhaed;
constexpr int kUdpOverhaed = 8;
constexpr int kTcpOverhaed = 20;
transport_overhead_per_packet +=
selected_candidate_pair_->local_candidate().protocol() ==
TCP_PROTOCOL_NAME
? kTcpOverhaed
: kUdpOverhaed;
if (secure()) {
int srtp_overhead = 0;
if (srtp_filter_.GetSrtpOverhead(&srtp_overhead))
transport_overhead_per_packet += srtp_overhead;
}
return transport_overhead_per_packet;
}
void BaseChannel::UpdateTransportOverhead() {
int transport_overhead_per_packet = GetTransportOverheadPerPacket();
if (transport_overhead_per_packet)
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&MediaChannel::OnTransportOverheadChanged, media_channel_,
transport_overhead_per_packet));
}
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);
LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
}
const ContentInfo* VoiceChannel::GetFirstContent(
const SessionDescription* sdesc) {
return GetFirstAudioContent(sdesc);
}
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());
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;
}
if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
return false;
}
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(audio, &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());
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;
}
if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
return false;
}
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(audio, &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(audio->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_) {
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,
VideoMediaChannel* media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
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)) {
LOG(LS_ERROR) << "Failed to SetSend on video channel";
// TODO(gangji): Report error back to server.
}
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();
}
}
const ContentInfo* VideoChannel::GetFirstContent(
const SessionDescription* sdesc) {
return GetFirstVideoContent(sdesc);
}
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());
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;
}
if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
return false;
}
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(video, &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());
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;
}
if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
return false;
}
VideoSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(video, &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(video->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,
DataMediaChannel* media_channel,
const std::string& content_name,
bool rtcp_mux_required,
bool srtp_required)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
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();
}
bool RtpDataChannel::Init_w(
DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
if (!BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport,
rtp_packet_transport, rtcp_packet_transport)) {
return false;
}
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
this, &RtpDataChannel::OnDataChannelReadyToSend);
return true;
}
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));
}
const ContentInfo* RtpDataChannel::GetFirstContent(
const SessionDescription* sdesc) {
return GetFirstDataContent(sdesc);
}
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());
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;
}
if (!SetRtpTransportParameters(content, action, CS_LOCAL, error_desc)) {
return false;
}
DataRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(data, &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 and isn't an update, it
// must be empty, so ignore it.
if (!data->has_codecs() && action != CA_UPDATE) {
return true;
}
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
LOG(LS_INFO) << "Setting remote data description";
if (!SetRtpTransportParameters(content, action, CS_REMOTE, error_desc)) {
return false;
}
DataSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription<DataCodec>(data, &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)) {
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)) {
LOG(LS_ERROR) << "Failed to SetSend on data channel";
}
// Trigger SignalReadyToSendData asynchronously.
OnDataChannelReadyToSend(send);
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
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