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webrtc / src / eebcab5ce99d3e8641dd92a569916b0d24e29fca / . / talk / session / media / channel.cc
blob: 0750537eab58da6686a55bedc55aa053de1604ca [file] [log] [blame]
/*
* libjingle
* Copyright 2004 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/channel.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/rtputils.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/typingmonitor.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/logging.h"
namespace cricket {
using rtc::Bind;
enum {
MSG_EARLYMEDIATIMEOUT = 1,
MSG_SCREENCASTWINDOWEVENT,
MSG_RTPPACKET,
MSG_RTCPPACKET,
MSG_CHANNEL_ERROR,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
MSG_FIRSTPACKETRECEIVED,
MSG_STREAMCLOSEDREMOTELY,
};
// Value specified in RFC 5764.
static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
static const int kAgcMinus10db = -10;
static void SetSessionError(BaseSession* session, BaseSession::Error error,
const std::string& error_desc) {
session->SetError(error, error_desc);
}
static void SafeSetError(const std::string& message, std::string* error_desc) {
if (error_desc) {
*error_desc = message;
}
}
struct PacketMessageData : public rtc::MessageData {
rtc::Buffer packet;
rtc::DiffServCodePoint dscp;
};
struct ScreencastEventMessageData : public rtc::MessageData {
ScreencastEventMessageData(uint32 s, rtc::WindowEvent we)
: ssrc(s),
event(we) {
}
uint32 ssrc;
rtc::WindowEvent event;
};
struct VoiceChannelErrorMessageData : public rtc::MessageData {
VoiceChannelErrorMessageData(uint32 in_ssrc,
VoiceMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {
}
uint32 ssrc;
VoiceMediaChannel::Error error;
};
struct VideoChannelErrorMessageData : public rtc::MessageData {
VideoChannelErrorMessageData(uint32 in_ssrc,
VideoMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {
}
uint32 ssrc;
VideoMediaChannel::Error error;
};
struct DataChannelErrorMessageData : public rtc::MessageData {
DataChannelErrorMessageData(uint32 in_ssrc,
DataMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {}
uint32 ssrc;
DataMediaChannel::Error error;
};
struct VideoChannel::ScreencastDetailsData {
explicit ScreencastDetailsData(uint32 s)
: ssrc(s), fps(0), screencast_max_pixels(0) {
}
uint32 ssrc;
int fps;
int screencast_max_pixels;
};
static const char* PacketType(bool rtcp) {
return (!rtcp) ? "RTP" : "RTCP";
}
static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) {
// Check the packet size. We could check the header too if needed.
return (packet &&
packet->size() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
packet->size() <= kMaxRtpPacketLen);
}
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);
}
BaseChannel::BaseChannel(rtc::Thread* thread,
MediaEngineInterface* media_engine,
MediaChannel* media_channel, BaseSession* session,
const std::string& content_name, bool rtcp)
: worker_thread_(thread),
media_engine_(media_engine),
session_(session),
media_channel_(media_channel),
content_name_(content_name),
rtcp_(rtcp),
transport_channel_(NULL),
rtcp_transport_channel_(NULL),
enabled_(false),
writable_(false),
rtp_ready_to_send_(false),
rtcp_ready_to_send_(false),
was_ever_writable_(false),
local_content_direction_(MD_INACTIVE),
remote_content_direction_(MD_INACTIVE),
has_received_packet_(false),
dtls_keyed_(false),
secure_required_(false),
rtp_abs_sendtime_extn_id_(-1) {
ASSERT(worker_thread_ == rtc::Thread::Current());
LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
ASSERT(worker_thread_ == rtc::Thread::Current());
Deinit();
StopConnectionMonitor();
FlushRtcpMessages(); // Send any outstanding RTCP packets.
worker_thread_->Clear(this); // eats any outstanding messages or packets
// 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_;
set_transport_channel(nullptr);
set_rtcp_transport_channel(nullptr);
LOG(LS_INFO) << "Destroyed channel";
}
bool BaseChannel::Init() {
if (!SetTransportChannels(session(), rtcp())) {
return false;
}
if (!SetDtlsSrtpCiphers(transport_channel(), false)) {
return false;
}
if (rtcp() && !SetDtlsSrtpCiphers(rtcp_transport_channel(), true)) {
return false;
}
session_->SignalNewLocalDescription.connect(
this, &BaseChannel::OnNewLocalDescription);
session_->SignalNewRemoteDescription.connect(
this, &BaseChannel::OnNewRemoteDescription);
// Both RTP and RTCP channels are set, we can call SetInterface on
// media channel and it can set network options.
media_channel_->SetInterface(this);
return true;
}
void BaseChannel::Deinit() {
media_channel_->SetInterface(NULL);
}
bool BaseChannel::SetTransportChannels(BaseSession* session, bool rtcp) {
return worker_thread_->Invoke<bool>(Bind(
&BaseChannel::SetTransportChannels_w, this, session, rtcp));
}
bool BaseChannel::SetTransportChannels_w(BaseSession* session, bool rtcp) {
ASSERT(worker_thread_ == rtc::Thread::Current());
set_transport_channel(session->CreateChannel(
content_name(), cricket::ICE_CANDIDATE_COMPONENT_RTP));
if (!transport_channel()) {
return false;
}
if (rtcp) {
set_rtcp_transport_channel(session->CreateChannel(
content_name(), cricket::ICE_CANDIDATE_COMPONENT_RTCP));
if (!rtcp_transport_channel()) {
return false;
}
} else {
set_rtcp_transport_channel(nullptr);
}
return true;
}
void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
ASSERT(worker_thread_ == rtc::Thread::Current());
TransportChannel* old_tc = transport_channel_;
if (old_tc == new_tc) {
return;
}
if (old_tc) {
DisconnectFromTransportChannel(old_tc);
session()->DestroyChannel(
content_name(), cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
transport_channel_ = new_tc;
if (new_tc) {
ConnectToTransportChannel(new_tc);
}
}
void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc) {
ASSERT(worker_thread_ == rtc::Thread::Current());
TransportChannel* old_tc = rtcp_transport_channel_;
if (old_tc == new_tc) {
return;
}
if (old_tc) {
DisconnectFromTransportChannel(old_tc);
session()->DestroyChannel(
content_name(), cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
rtcp_transport_channel_ = new_tc;
if (new_tc) {
ConnectToTransportChannel(new_tc);
}
}
void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
ASSERT(worker_thread_ == rtc::Thread::Current());
tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);
}
void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) {
ASSERT(worker_thread_ == rtc::Thread::Current());
tc->SignalWritableState.disconnect(this);
tc->SignalReadPacket.disconnect(this);
tc->SignalReadyToSend.disconnect(this);
}
bool BaseChannel::Enable(bool enable) {
worker_thread_->Invoke<void>(Bind(
enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
this));
return true;
}
bool BaseChannel::MuteStream(uint32 ssrc, bool mute) {
return InvokeOnWorker(Bind(&BaseChannel::MuteStream_w, this, ssrc, mute));
}
bool BaseChannel::IsStreamMuted(uint32 ssrc) {
return InvokeOnWorker(Bind(&BaseChannel::IsStreamMuted_w, this, ssrc));
}
bool BaseChannel::AddRecvStream(const StreamParams& sp) {
return InvokeOnWorker(Bind(&BaseChannel::AddRecvStream_w, this, sp));
}
bool BaseChannel::RemoveRecvStream(uint32 ssrc) {
return InvokeOnWorker(Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
}
bool BaseChannel::AddSendStream(const StreamParams& sp) {
return InvokeOnWorker(
Bind(&MediaChannel::AddSendStream, media_channel(), sp));
}
bool BaseChannel::RemoveSendStream(uint32 ssrc) {
return InvokeOnWorker(
Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
}
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
return InvokeOnWorker(Bind(&BaseChannel::SetLocalContent_w,
this, content, action, error_desc));
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
return InvokeOnWorker(Bind(&BaseChannel::SetRemoteContent_w,
this, content, action, error_desc));
}
void BaseChannel::StartConnectionMonitor(int cms) {
// We pass in the BaseChannel instead of the transport_channel_
// because if the transport_channel_ changes, the ConnectionMonitor
// would be pointing to the wrong TransportChannel.
connection_monitor_.reset(new ConnectionMonitor(
this, worker_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) {
ASSERT(worker_thread_ == rtc::Thread::Current());
return transport_channel_->GetStats(infos);
}
bool BaseChannel::IsReadyToReceive() const {
// Receive data if we are enabled and have local content,
return enabled() && IsReceiveContentDirection(local_content_direction_);
}
bool BaseChannel::IsReadyToSend() 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();
}
bool BaseChannel::SendPacket(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
return SendPacket(false, packet, dscp);
}
bool BaseChannel::SendRtcp(rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
return SendPacket(true, packet, dscp);
}
int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
int value) {
TransportChannel* channel = NULL;
switch (type) {
case ST_RTP:
channel = transport_channel_;
break;
case ST_RTCP:
channel = rtcp_transport_channel_;
break;
}
return channel ? channel->SetOption(opt, value) : -1;
}
void BaseChannel::OnWritableState(TransportChannel* channel) {
ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
if (transport_channel_->writable()
&& (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
ChannelWritable_w();
} else {
ChannelNotWritable_w();
}
}
void BaseChannel::OnChannelRead(TransportChannel* channel,
const char* data, size_t len,
const rtc::PacketTime& packet_time,
int flags) {
// OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
ASSERT(worker_thread_ == rtc::Thread::Current());
// When using RTCP multiplexing we might get RTCP packets on the RTP
// transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
bool rtcp = PacketIsRtcp(channel, data, len);
rtc::Buffer packet(data, len);
HandlePacket(rtcp, &packet, packet_time);
}
void BaseChannel::OnReadyToSend(TransportChannel* channel) {
SetReadyToSend(channel, true);
}
void BaseChannel::SetReadyToSend(TransportChannel* channel, bool ready) {
ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
if (channel == transport_channel_) {
rtp_ready_to_send_ = ready;
}
if (channel == rtcp_transport_channel_) {
rtcp_ready_to_send_ = ready;
}
if (!ready) {
// Notify the MediaChannel when either rtp or rtcp channel can't send.
media_channel_->OnReadyToSend(false);
} else if (rtp_ready_to_send_ &&
// In the case of rtcp mux |rtcp_transport_channel_| will be null.
(rtcp_ready_to_send_ || !rtcp_transport_channel_)) {
// Notify the MediaChannel when both rtp and rtcp channel can send.
media_channel_->OnReadyToSend(true);
}
}
bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
const char* data, size_t len) {
return (channel == rtcp_transport_channel_ ||
rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
}
bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
rtc::DiffServCodePoint dscp) {
// SendPacket gets called from MediaEngine, typically on an encoder thread.
// If the thread is not our worker thread, we will post to our worker
// so that the real work happens on our worker. 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 (rtc::Thread::Current() != worker_thread_) {
// Avoid a copy by transferring the ownership of the packet data.
int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
PacketMessageData* data = new PacketMessageData;
packet->TransferTo(&data->packet);
data->dscp = dscp;
worker_thread_->Post(this, message_id, data);
return true;
}
// 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.
TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
transport_channel_ : rtcp_transport_channel_;
if (!channel || !channel->writable()) {
return false;
}
// Protect ourselves against crazy data.
if (!ValidPacket(rtcp, packet)) {
LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
<< PacketType(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
// Signal to the media sink before protecting the packet.
{
rtc::CritScope cs(&signal_send_packet_cs_);
SignalSendPacketPreCrypto(packet->data(), packet->size(), rtcp);
}
rtc::PacketOptions options(dscp);
// Protect if needed.
if (srtp_filter_.IsActive()) {
bool res;
char* 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
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,
&options.packet_time_params.srtp_packet_index);
// If protection succeeds, let's get auth params from srtp.
if (res) {
uint8* auth_key = NULL;
int key_len;
res = srtp_filter_.GetRtpAuthParams(
&auth_key, &key_len, &options.packet_time_params.srtp_auth_tag_len);
if (res) {
options.packet_time_params.srtp_auth_key.resize(key_len);
options.packet_time_params.srtp_auth_key.assign(auth_key,
auth_key + key_len);
}
}
#endif
if (!res) {
int seq_num = -1;
uint32 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 (secure_required_) {
// This is a double check for something that supposedly can't happen.
LOG(LS_ERROR) << "Can't send outgoing " << PacketType(rtcp)
<< " packet when SRTP is inactive and crypto is required";
ASSERT(false);
return false;
}
// Signal to the media sink after protecting the packet.
{
rtc::CritScope cs(&signal_send_packet_cs_);
SignalSendPacketPostCrypto(packet->data(), packet->size(), rtcp);
}
// Bon voyage.
int ret =
channel->SendPacket(packet->data(), packet->size(), options,
(secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
if (ret != static_cast<int>(packet->size())) {
if (channel->GetError() == EWOULDBLOCK) {
LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
SetReadyToSend(channel, false);
}
return false;
}
return true;
}
bool BaseChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
// Protect ourselves against crazy data.
if (!ValidPacket(rtcp, packet)) {
LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
<< PacketType(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
// Bundle filter handles both rtp and rtcp packets.
return bundle_filter_.DemuxPacket(packet->data(), packet->size(), rtcp);
}
void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
const rtc::PacketTime& packet_time) {
if (!WantsPacket(rtcp, packet)) {
return;
}
// We are only interested in the first rtp packet because that
// indicates the media has started flowing.
if (!has_received_packet_ && !rtcp) {
has_received_packet_ = true;
signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
}
// Signal to the media sink before unprotecting the packet.
{
rtc::CritScope cs(&signal_recv_packet_cs_);
SignalRecvPacketPostCrypto(packet->data(), packet->size(), rtcp);
}
// Unprotect the packet, if needed.
if (srtp_filter_.IsActive()) {
char* data = packet->data();
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 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 (secure_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
// channels, so we haven't yet extracted keys, even if DTLS did complete
// on the channel 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 " << PacketType(rtcp)
<< " packet when SRTP is inactive and crypto is required";
return;
}
// Signal to the media sink after unprotecting the packet.
{
rtc::CritScope cs(&signal_recv_packet_cs_);
SignalRecvPacketPreCrypto(packet->data(), packet->size(), rtcp);
}
// Push it down to the media channel.
if (!rtcp) {
media_channel_->OnPacketReceived(packet, packet_time);
} else {
media_channel_->OnRtcpReceived(packet, packet_time);
}
}
void BaseChannel::OnNewLocalDescription(
BaseSession* session, ContentAction action) {
std::string error_desc;
if (!PushdownLocalDescription(
session->local_description(), action, &error_desc)) {
SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
}
}
void BaseChannel::OnNewRemoteDescription(
BaseSession* session, ContentAction action) {
std::string error_desc;
if (!PushdownRemoteDescription(
session->remote_description(), action, &error_desc)) {
SetSessionError(session_, BaseSession::ERROR_CONTENT, error_desc);
}
}
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() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (enabled_)
return;
LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
ChangeState();
}
void BaseChannel::DisableMedia_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (!enabled_)
return;
LOG(LS_INFO) << "Channel disabled";
enabled_ = false;
ChangeState();
}
bool BaseChannel::MuteStream_w(uint32 ssrc, bool mute) {
ASSERT(worker_thread_ == rtc::Thread::Current());
bool ret = media_channel()->MuteStream(ssrc, mute);
if (ret) {
if (mute)
muted_streams_.insert(ssrc);
else
muted_streams_.erase(ssrc);
}
return ret;
}
bool BaseChannel::IsStreamMuted_w(uint32 ssrc) {
ASSERT(worker_thread_ == rtc::Thread::Current());
return muted_streams_.find(ssrc) != muted_streams_.end();
}
void BaseChannel::ChannelWritable_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (writable_)
return;
LOG(LS_INFO) << "Channel socket writable ("
<< transport_channel_->content_name() << ", "
<< transport_channel_->component() << ")"
<< (was_ever_writable_ ? "" : " for the first time");
std::vector<ConnectionInfo> infos;
transport_channel_->GetStats(&infos);
for (std::vector<ConnectionInfo>::const_iterator it = infos.begin();
it != infos.end(); ++it) {
if (it->best_connection) {
LOG(LS_INFO) << "Using " << it->local_candidate.ToSensitiveString()
<< "->" << it->remote_candidate.ToSensitiveString();
break;
}
}
// If we're doing DTLS-SRTP, now is the time.
if (!was_ever_writable_ && ShouldSetupDtlsSrtp()) {
if (!SetupDtlsSrtp(false)) {
SignalDtlsSetupFailure(this, false);
return;
}
if (rtcp_transport_channel_) {
if (!SetupDtlsSrtp(true)) {
SignalDtlsSetupFailure(this, true);
return;
}
}
}
was_ever_writable_ = true;
writable_ = true;
ChangeState();
}
void BaseChannel::SignalDtlsSetupFailure_w(bool rtcp) {
ASSERT(worker_thread() == rtc::Thread::Current());
signaling_thread()->Invoke<void>(Bind(
&BaseChannel::SignalDtlsSetupFailure_s, this, rtcp));
}
void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) {
ASSERT(signaling_thread() == rtc::Thread::Current());
SignalDtlsSetupFailure(this, rtcp);
}
bool BaseChannel::SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) {
std::vector<std::string> ciphers;
// We always use the default SRTP ciphers for RTCP, but we may use different
// ciphers for RTP depending on the media type.
if (!rtcp) {
GetSrtpCiphers(&ciphers);
} else {
GetSupportedDefaultCryptoSuites(&ciphers);
}
return tc->SetSrtpCiphers(ciphers);
}
bool BaseChannel::ShouldSetupDtlsSrtp() const {
return true;
}
// This function returns true if either DTLS-SRTP is not in use
// *or* DTLS-SRTP is successfully set up.
bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
bool ret = false;
TransportChannel *channel = rtcp_channel ?
rtcp_transport_channel_ : transport_channel_;
// No DTLS
if (!channel->IsDtlsActive())
return true;
std::string selected_cipher;
if (!channel->GetSrtpCipher(&selected_cipher)) {
LOG(LS_ERROR) << "No DTLS-SRTP selected cipher";
return false;
}
LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
<< content_name() << " "
<< PacketType(rtcp_channel);
// OK, we're now doing DTLS (RFC 5764)
std::vector<unsigned char> dtls_buffer(SRTP_MASTER_KEY_KEY_LEN * 2 +
SRTP_MASTER_KEY_SALT_LEN * 2);
// RFC 5705 exporter using the RFC 5764 parameters
if (!channel->ExportKeyingMaterial(
kDtlsSrtpExporterLabel,
NULL, 0, false,
&dtls_buffer[0], dtls_buffer.size())) {
LOG(LS_WARNING) << "DTLS-SRTP key export failed";
ASSERT(false); // This should never happen
return false;
}
// Sync up the keys with the DTLS-SRTP interface
std::vector<unsigned char> client_write_key(SRTP_MASTER_KEY_KEY_LEN +
SRTP_MASTER_KEY_SALT_LEN);
std::vector<unsigned char> server_write_key(SRTP_MASTER_KEY_KEY_LEN +
SRTP_MASTER_KEY_SALT_LEN);
size_t offset = 0;
memcpy(&client_write_key[0], &dtls_buffer[offset],
SRTP_MASTER_KEY_KEY_LEN);
offset += SRTP_MASTER_KEY_KEY_LEN;
memcpy(&server_write_key[0], &dtls_buffer[offset],
SRTP_MASTER_KEY_KEY_LEN);
offset += SRTP_MASTER_KEY_KEY_LEN;
memcpy(&client_write_key[SRTP_MASTER_KEY_KEY_LEN],
&dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
offset += SRTP_MASTER_KEY_SALT_LEN;
memcpy(&server_write_key[SRTP_MASTER_KEY_KEY_LEN],
&dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
std::vector<unsigned char> *send_key, *recv_key;
rtc::SSLRole role;
if (!channel->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_channel) {
ret = srtp_filter_.SetRtcpParams(
selected_cipher,
&(*send_key)[0],
static_cast<int>(send_key->size()),
selected_cipher,
&(*recv_key)[0],
static_cast<int>(recv_key->size()));
} else {
ret = srtp_filter_.SetRtpParams(
selected_cipher,
&(*send_key)[0],
static_cast<int>(send_key->size()),
selected_cipher,
&(*recv_key)[0],
static_cast<int>(recv_key->size()));
}
if (!ret)
LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
else
dtls_keyed_ = true;
return ret;
}
void BaseChannel::ChannelNotWritable_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (!writable_)
return;
LOG(LS_INFO) << "Channel socket not writable ("
<< transport_channel_->content_name() << ", "
<< transport_channel_->component() << ")";
writable_ = false;
ChangeState();
}
// |dtls| will be set to true if DTLS is active for transport channel and
// crypto is empty.
bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
bool* dtls,
std::string* error_desc) {
*dtls = transport_channel_->IsDtlsActive();
if (*dtls && !cryptos.empty()) {
SafeSetError("Cryptos must be empty when DTLS is active.",
error_desc);
return false;
}
return true;
}
bool BaseChannel::SetRecvRtpHeaderExtensions_w(
const MediaContentDescription* content,
MediaChannel* media_channel,
std::string* error_desc) {
if (content->rtp_header_extensions_set()) {
if (!media_channel->SetRecvRtpHeaderExtensions(
content->rtp_header_extensions())) {
std::ostringstream desc;
desc << "Failed to set receive rtp header extensions for "
<< MediaTypeToString(content->type()) << " content.";
SafeSetError(desc.str(), error_desc);
return false;
}
}
return true;
}
bool BaseChannel::SetSendRtpHeaderExtensions_w(
const MediaContentDescription* content,
MediaChannel* media_channel,
std::string* error_desc) {
if (content->rtp_header_extensions_set()) {
if (!media_channel->SetSendRtpHeaderExtensions(
content->rtp_header_extensions())) {
std::ostringstream desc;
desc << "Failed to set send rtp header extensions for "
<< MediaTypeToString(content->type()) << " content.";
SafeSetError(desc.str(), error_desc);
return false;
} else {
MaybeCacheRtpAbsSendTimeHeaderExtension(content->rtp_header_extensions());
}
}
return true;
}
bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
ContentAction action,
ContentSource src,
std::string* error_desc) {
if (action == CA_UPDATE) {
// no crypto params.
return true;
}
bool ret = false;
bool dtls = false;
ret = CheckSrtpConfig(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_w(bool enable, ContentAction action,
ContentSource src,
std::string* error_desc) {
bool ret = false;
switch (action) {
case CA_OFFER:
ret = rtcp_mux_filter_.SetOffer(enable, src);
break;
case CA_PRANSWER:
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 activated RTCP mux, close down the RTCP transport.
set_rtcp_transport_channel(NULL);
}
break;
case CA_UPDATE:
// No RTCP mux info.
ret = true;
default:
break;
}
if (!ret) {
SafeSetError("Failed to setup RTCP mux filter.", error_desc);
return false;
}
// |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or
// CA_ANSWER, but we only want to tear down the RTCP transport channel if we
// received a final answer.
if (rtcp_mux_filter_.IsActive()) {
// If the RTP transport is already writable, then so are we.
if (transport_channel_->writable()) {
ChannelWritable_w();
}
}
return true;
}
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
ASSERT(worker_thread() == rtc::Thread::Current());
if (!media_channel()->AddRecvStream(sp))
return false;
return bundle_filter_.AddStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32 ssrc) {
ASSERT(worker_thread() == rtc::Thread::Current());
bundle_filter_.RemoveStream(ssrc);
return media_channel()->RemoveRecvStream(ssrc);
}
bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
ContentAction action,
std::string* error_desc) {
if (!VERIFY(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 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 (!VERIFY(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;
}
bool BaseChannel::SetBaseLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
// Cache secure_required_ for belt and suspenders check on SendPacket
secure_required_ = content->crypto_required() != CT_NONE;
// Set local RTP header extensions.
bool ret = SetRecvRtpHeaderExtensions_w(content, media_channel(), error_desc);
// Set local SRTP parameters (what we will encrypt with).
ret &= SetSrtp_w(content->cryptos(), action, CS_LOCAL, error_desc);
// Set local RTCP mux parameters.
ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_LOCAL, error_desc);
// Call UpdateLocalStreams_w last to make sure as many settings as possible
// are already set when creating streams.
ret &= UpdateLocalStreams_w(content->streams(), action, error_desc);
set_local_content_direction(content->direction());
return ret;
}
bool BaseChannel::SetBaseRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
// Set remote RTP header extensions.
bool ret = SetSendRtpHeaderExtensions_w(content, media_channel(), error_desc);
// Set remote SRTP parameters (what the other side will encrypt with).
ret &= SetSrtp_w(content->cryptos(), action, CS_REMOTE, error_desc);
// Set remote RTCP mux parameters.
ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_REMOTE, error_desc);
if (!media_channel()->SetMaxSendBandwidth(content->bandwidth())) {
std::ostringstream desc;
desc << "Failed to set max send bandwidth for "
<< MediaTypeToString(content->type()) << " content.";
SafeSetError(desc.str(), error_desc);
ret = false;
}
// Call UpdateRemoteStreams_w last to make sure as many settings as possible
// are already set when creating streams.
ret &= UpdateRemoteStreams_w(content->streams(), action, error_desc);
set_remote_content_direction(content->direction());
return ret;
}
void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
const std::vector<RtpHeaderExtension>& extensions) {
const RtpHeaderExtension* send_time_extension =
FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
rtp_abs_sendtime_extn_id_ =
send_time_extension ? send_time_extension->id : -1;
}
void BaseChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_RTPPACKET:
case MSG_RTCPPACKET: {
PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, data->dscp);
delete data; // because it is Posted
break;
}
case MSG_FIRSTPACKETRECEIVED: {
SignalFirstPacketReceived(this);
break;
}
}
}
void BaseChannel::FlushRtcpMessages() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
ASSERT(rtc::Thread::Current() == worker_thread_);
rtc::MessageList rtcp_messages;
worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages);
for (rtc::MessageList::iterator it = rtcp_messages.begin();
it != rtcp_messages.end(); ++it) {
worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata);
}
}
VoiceChannel::VoiceChannel(rtc::Thread* thread,
MediaEngineInterface* media_engine,
VoiceMediaChannel* media_channel,
BaseSession* session,
const std::string& content_name,
bool rtcp)
: BaseChannel(thread, media_engine, media_channel, session, content_name,
rtcp),
received_media_(false) {
}
VoiceChannel::~VoiceChannel() {
StopAudioMonitor();
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
bool VoiceChannel::Init() {
if (!BaseChannel::Init()) {
return false;
}
media_channel()->SignalMediaError.connect(
this, &VoiceChannel::OnVoiceChannelError);
srtp_filter()->SignalSrtpError.connect(
this, &VoiceChannel::OnSrtpError);
return true;
}
bool VoiceChannel::SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::SetRemoteRenderer,
media_channel(), ssrc, renderer));
}
bool VoiceChannel::SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::SetLocalRenderer,
media_channel(), ssrc, renderer));
}
bool VoiceChannel::SetRingbackTone(const void* buf, int len) {
return InvokeOnWorker(Bind(&VoiceChannel::SetRingbackTone_w, this, buf, len));
}
// 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(kEarlyMediaTimeout, this,
MSG_EARLYMEDIATIMEOUT);
} else {
// Stop the timeout if currently going.
worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
}
}
bool VoiceChannel::PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
return InvokeOnWorker(Bind(&VoiceChannel::PlayRingbackTone_w,
this, ssrc, play, loop));
}
bool VoiceChannel::PressDTMF(int digit, bool playout) {
int flags = DF_SEND;
if (playout) {
flags |= DF_PLAY;
}
int duration_ms = 160;
return InsertDtmf(0, digit, duration_ms, flags);
}
bool VoiceChannel::CanInsertDtmf() {
return InvokeOnWorker(Bind(&VoiceMediaChannel::CanInsertDtmf,
media_channel()));
}
bool VoiceChannel::InsertDtmf(uint32 ssrc, int event_code, int duration,
int flags) {
return InvokeOnWorker(Bind(&VoiceChannel::InsertDtmf_w, this,
ssrc, event_code, duration, flags));
}
bool VoiceChannel::SetOutputScaling(uint32 ssrc, double left, double right) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOutputScaling,
media_channel(), ssrc, left, right));
}
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
}
void VoiceChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
media_monitor_->SignalUpdate.connect(
this, &VoiceChannel::OnMediaMonitorUpdate);
media_monitor_->Start(cms);
}
void VoiceChannel::StopMediaMonitor() {
if (media_monitor_) {
media_monitor_->Stop();
media_monitor_->SignalUpdate.disconnect(this);
media_monitor_.reset();
}
}
void VoiceChannel::StartAudioMonitor(int cms) {
audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
audio_monitor_
->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
audio_monitor_->Start(cms);
}
void VoiceChannel::StopAudioMonitor() {
if (audio_monitor_) {
audio_monitor_->Stop();
audio_monitor_.reset();
}
}
bool VoiceChannel::IsAudioMonitorRunning() const {
return (audio_monitor_.get() != NULL);
}
void VoiceChannel::StartTypingMonitor(const TypingMonitorOptions& settings) {
typing_monitor_.reset(new TypingMonitor(this, worker_thread(), settings));
SignalAutoMuted.repeat(typing_monitor_->SignalMuted);
}
void VoiceChannel::StopTypingMonitor() {
typing_monitor_.reset();
}
bool VoiceChannel::IsTypingMonitorRunning() const {
return typing_monitor_;
}
bool VoiceChannel::MuteStream_w(uint32 ssrc, bool mute) {
bool ret = BaseChannel::MuteStream_w(ssrc, mute);
if (typing_monitor_ && mute)
typing_monitor_->OnChannelMuted();
return ret;
}
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::OnChannelRead(TransportChannel* channel,
const char* data, size_t len,
const rtc::PacketTime& packet_time,
int flags) {
BaseChannel::OnChannelRead(channel, data, len, packet_time, flags);
// 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_ && !PacketIsRtcp(channel, data, len)) {
received_media_ = true;
}
}
void VoiceChannel::ChangeState() {
// 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 = IsReadyToReceive();
if (!media_channel()->SetPlayout(recv)) {
SendLastMediaError();
}
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSend();
SendFlags send_flag = send ? SEND_MICROPHONE : SEND_NOTHING;
if (!media_channel()->SetSend(send_flag)) {
LOG(LS_ERROR) << "Failed to SetSend " << send_flag << " on voice channel";
SendLastMediaError();
}
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) {
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
ASSERT(audio != NULL);
if (!audio) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
bool ret = SetBaseLocalContent_w(content, action, error_desc);
// Set local audio codecs (what we want to receive).
// TODO(whyuan): Change action != CA_UPDATE to !audio->partial() when partial
// is set properly.
if (action != CA_UPDATE || audio->has_codecs()) {
if (!media_channel()->SetRecvCodecs(audio->codecs())) {
SafeSetError("Failed to set audio receive codecs.", error_desc);
ret = false;
}
}
// If everything worked, see if we can start receiving.
if (ret) {
std::vector<AudioCodec>::const_iterator it = audio->codecs().begin();
for (; it != audio->codecs().end(); ++it) {
bundle_filter()->AddPayloadType(it->id);
}
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local voice description";
}
return ret;
}
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting remote voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
ASSERT(audio != NULL);
if (!audio) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
bool ret = true;
// Set remote video codecs (what the other side wants to receive).
if (action != CA_UPDATE || audio->has_codecs()) {
if (!media_channel()->SetSendCodecs(audio->codecs())) {
SafeSetError("Failed to set audio send codecs.", error_desc);
ret = false;
}
}
ret &= SetBaseRemoteContent_w(content, action, error_desc);
if (action != CA_UPDATE) {
// Tweak our audio processing settings, if needed.
AudioOptions audio_options;
if (!media_channel()->GetOptions(&audio_options)) {
LOG(LS_WARNING) << "Can not set audio options from on remote content.";
} else {
if (audio->conference_mode()) {
audio_options.conference_mode.Set(true);
}
if (audio->agc_minus_10db()) {
audio_options.adjust_agc_delta.Set(kAgcMinus10db);
}
if (!media_channel()->SetOptions(audio_options)) {
// Log an error on failure, but don't abort the call.
LOG(LS_ERROR) << "Failed to set voice channel options";
}
}
}
// If everything worked, see if we can start sending.
if (ret) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set remote voice description";
}
return ret;
}
bool VoiceChannel::SetRingbackTone_w(const void* buf, int len) {
ASSERT(worker_thread() == rtc::Thread::Current());
return media_channel()->SetRingbackTone(static_cast<const char*>(buf), len);
}
bool VoiceChannel::PlayRingbackTone_w(uint32 ssrc, bool play, bool loop) {
ASSERT(worker_thread() == rtc::Thread::Current());
if (play) {
LOG(LS_INFO) << "Playing ringback tone, loop=" << loop;
} else {
LOG(LS_INFO) << "Stopping ringback tone";
}
return media_channel()->PlayRingbackTone(ssrc, play, loop);
}
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 ssrc, int event, int duration,
int flags) {
if (!enabled()) {
return false;
}
return media_channel()->InsertDtmf(ssrc, event, duration, flags);
}
bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOptions,
media_channel(), options));
}
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);
SignalMediaError(this, data->ssrc, data->error);
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) {
ASSERT(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
const AudioInfo& info) {
SignalAudioMonitor(this, info);
}
void VoiceChannel::OnVoiceChannelError(
uint32 ssrc, VoiceMediaChannel::Error err) {
VoiceChannelErrorMessageData* data = new VoiceChannelErrorMessageData(
ssrc, err);
signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
}
void VoiceChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
SrtpFilter::Error error) {
switch (error) {
case SrtpFilter::ERROR_FAIL:
OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
VoiceMediaChannel::ERROR_REC_SRTP_ERROR :
VoiceMediaChannel::ERROR_PLAY_SRTP_ERROR);
break;
case SrtpFilter::ERROR_AUTH:
OnVoiceChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
VoiceMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
VoiceMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
break;
case SrtpFilter::ERROR_REPLAY:
// Only receving channel should have this error.
ASSERT(mode == SrtpFilter::UNPROTECT);
OnVoiceChannelError(ssrc, VoiceMediaChannel::ERROR_PLAY_SRTP_REPLAY);
break;
default:
break;
}
}
void VoiceChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
GetSupportedAudioCryptoSuites(ciphers);
}
VideoChannel::VideoChannel(rtc::Thread* thread,
MediaEngineInterface* media_engine,
VideoMediaChannel* media_channel,
BaseSession* session,
const std::string& content_name,
bool rtcp,
VoiceChannel* voice_channel)
: BaseChannel(thread, media_engine, media_channel, session, content_name,
rtcp),
voice_channel_(voice_channel),
renderer_(NULL),
previous_we_(rtc::WE_CLOSE) {
}
bool VideoChannel::Init() {
if (!BaseChannel::Init()) {
return false;
}
media_channel()->SignalMediaError.connect(
this, &VideoChannel::OnVideoChannelError);
srtp_filter()->SignalSrtpError.connect(
this, &VideoChannel::OnSrtpError);
return true;
}
void VoiceChannel::SendLastMediaError() {
uint32 ssrc;
VoiceMediaChannel::Error error;
media_channel()->GetLastMediaError(&ssrc, &error);
SignalMediaError(this, ssrc, error);
}
VideoChannel::~VideoChannel() {
std::vector<uint32> screencast_ssrcs;
ScreencastMap::iterator iter;
while (!screencast_capturers_.empty()) {
if (!RemoveScreencast(screencast_capturers_.begin()->first)) {
LOG(LS_ERROR) << "Unable to delete screencast with ssrc "
<< screencast_capturers_.begin()->first;
ASSERT(false);
break;
}
}
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
bool VideoChannel::SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
worker_thread()->Invoke<void>(Bind(
&VideoMediaChannel::SetRenderer, media_channel(), ssrc, renderer));
return true;
}
bool VideoChannel::ApplyViewRequest(const ViewRequest& request) {
return InvokeOnWorker(Bind(&VideoChannel::ApplyViewRequest_w, this, request));
}
bool VideoChannel::AddScreencast(uint32 ssrc, VideoCapturer* capturer) {
return worker_thread()->Invoke<bool>(Bind(
&VideoChannel::AddScreencast_w, this, ssrc, capturer));
}
bool VideoChannel::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
return InvokeOnWorker(Bind(&VideoMediaChannel::SetCapturer,
media_channel(), ssrc, capturer));
}
bool VideoChannel::RemoveScreencast(uint32 ssrc) {
return InvokeOnWorker(Bind(&VideoChannel::RemoveScreencast_w, this, ssrc));
}
bool VideoChannel::IsScreencasting() {
return InvokeOnWorker(Bind(&VideoChannel::IsScreencasting_w, this));
}
int VideoChannel::GetScreencastFps(uint32 ssrc) {
ScreencastDetailsData data(ssrc);
worker_thread()->Invoke<void>(Bind(
&VideoChannel::GetScreencastDetails_w, this, &data));
return data.fps;
}
int VideoChannel::GetScreencastMaxPixels(uint32 ssrc) {
ScreencastDetailsData data(ssrc);
worker_thread()->Invoke<void>(Bind(
&VideoChannel::GetScreencastDetails_w, this, &data));
return data.screencast_max_pixels;
}
bool VideoChannel::SendIntraFrame() {
worker_thread()->Invoke<void>(Bind(
&VideoMediaChannel::SendIntraFrame, media_channel()));
return true;
}
bool VideoChannel::RequestIntraFrame() {
worker_thread()->Invoke<void>(Bind(
&VideoMediaChannel::RequestIntraFrame, media_channel()));
return true;
}
void VideoChannel::ChangeState() {
// 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 = IsReadyToReceive();
if (!media_channel()->SetRender(recv)) {
LOG(LS_ERROR) << "Failed to SetRender on video channel";
// TODO(gangji): Report error back to server.
}
// Send outgoing data if we're the active call, we have the remote content,
// and we have had some form of connectivity.
bool send = IsReadyToSend();
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, recv=" << recv << " send=" << send;
}
bool VideoChannel::GetStats(VideoMediaInfo* stats) {
return InvokeOnWorker(
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) {
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
ASSERT(video != NULL);
if (!video) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
bool ret = SetBaseLocalContent_w(content, action, error_desc);
// Set local video codecs (what we want to receive).
if (action != CA_UPDATE || video->has_codecs()) {
if (!media_channel()->SetRecvCodecs(video->codecs())) {
SafeSetError("Failed to set video receive codecs.", error_desc);
ret = false;
}
}
if (action != CA_UPDATE) {
VideoOptions video_options;
media_channel()->GetOptions(&video_options);
video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
if (!media_channel()->SetOptions(video_options)) {
// Log an error on failure, but don't abort the call.
LOG(LS_ERROR) << "Failed to set video channel options";
}
}
// If everything worked, see if we can start receiving.
if (ret) {
std::vector<VideoCodec>::const_iterator it = video->codecs().begin();
for (; it != video->codecs().end(); ++it) {
bundle_filter()->AddPayloadType(it->id);
}
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local video description";
}
return ret;
}
bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting remote video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
ASSERT(video != NULL);
if (!video) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
bool ret = true;
// Set remote video codecs (what the other side wants to receive).
if (action != CA_UPDATE || video->has_codecs()) {
if (!media_channel()->SetSendCodecs(video->codecs())) {
SafeSetError("Failed to set video send codecs.", error_desc);
ret = false;
}
}
ret &= SetBaseRemoteContent_w(content, action, error_desc);
if (action != CA_UPDATE) {
// Tweak our video processing settings, if needed.
VideoOptions video_options;
media_channel()->GetOptions(&video_options);
if (video->conference_mode()) {
video_options.conference_mode.Set(true);
}
video_options.buffered_mode_latency.Set(video->buffered_mode_latency());
if (!media_channel()->SetOptions(video_options)) {
// Log an error on failure, but don't abort the call.
LOG(LS_ERROR) << "Failed to set video channel options";
}
}
// If everything worked, see if we can start sending.
if (ret) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set remote video description";
}
return ret;
}
bool VideoChannel::ApplyViewRequest_w(const ViewRequest& request) {
bool ret = true;
// Set the send format for each of the local streams. If the view request
// does not contain a local stream, set its send format to 0x0, which will
// drop all frames.
for (std::vector<StreamParams>::const_iterator it = local_streams().begin();
it != local_streams().end(); ++it) {
VideoFormat format(0, 0, 0, cricket::FOURCC_I420);
StaticVideoViews::const_iterator view;
for (view = request.static_video_views.begin();
view != request.static_video_views.end(); ++view) {
if (view->selector.Matches(*it)) {
format.width = view->width;
format.height = view->height;
format.interval = cricket::VideoFormat::FpsToInterval(view->framerate);
break;
}
}
ret &= media_channel()->SetSendStreamFormat(it->first_ssrc(), format);
}
// Check if the view request has invalid streams.
for (StaticVideoViews::const_iterator it = request.static_video_views.begin();
it != request.static_video_views.end(); ++it) {
if (!GetStream(local_streams(), it->selector)) {
LOG(LS_WARNING) << "View request for ("
<< it->selector.ssrc << ", '"
<< it->selector.groupid << "', '"
<< it->selector.streamid << "'"
<< ") is not in the local streams.";
}
}
return ret;
}
bool VideoChannel::AddScreencast_w(uint32 ssrc, VideoCapturer* capturer) {
if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
return false;
}
capturer->SignalStateChange.connect(this, &VideoChannel::OnStateChange);
screencast_capturers_[ssrc] = capturer;
return true;
}
bool VideoChannel::RemoveScreencast_w(uint32 ssrc) {
ScreencastMap::iterator iter = screencast_capturers_.find(ssrc);
if (iter == screencast_capturers_.end()) {
return false;
}
// Clean up VideoCapturer.
delete iter->second;
screencast_capturers_.erase(iter);
return true;
}
bool VideoChannel::IsScreencasting_w() const {
return !screencast_capturers_.empty();
}
void VideoChannel::GetScreencastDetails_w(
ScreencastDetailsData* data) const {
ScreencastMap::const_iterator iter = screencast_capturers_.find(data->ssrc);
if (iter == screencast_capturers_.end()) {
return;
}
VideoCapturer* capturer = iter->second;
const VideoFormat* video_format = capturer->GetCaptureFormat();
data->fps = VideoFormat::IntervalToFps(video_format->interval);
data->screencast_max_pixels = capturer->screencast_max_pixels();
}
void VideoChannel::OnScreencastWindowEvent_s(uint32 ssrc,
rtc::WindowEvent we) {
ASSERT(signaling_thread() == rtc::Thread::Current());
SignalScreencastWindowEvent(ssrc, we);
}
bool VideoChannel::SetChannelOptions(const VideoOptions &options) {
return InvokeOnWorker(Bind(&VideoMediaChannel::SetOptions,
media_channel(), options));
}
void VideoChannel::OnMessage(rtc::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_SCREENCASTWINDOWEVENT: {
const ScreencastEventMessageData* data =
static_cast<ScreencastEventMessageData*>(pmsg->pdata);
OnScreencastWindowEvent_s(data->ssrc, data->event);
delete data;
break;
}
case MSG_CHANNEL_ERROR: {
const VideoChannelErrorMessageData* data =
static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
SignalMediaError(this, data->ssrc, data->error);
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) {
ASSERT(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
void VideoChannel::OnScreencastWindowEvent(uint32 ssrc,
rtc::WindowEvent event) {
ScreencastEventMessageData* pdata =
new ScreencastEventMessageData(ssrc, event);
signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
}
void VideoChannel::OnStateChange(VideoCapturer* capturer, CaptureState ev) {
// Map capturer events to window events. In the future we may want to simply
// pass these events up directly.
rtc::WindowEvent we;
if (ev == CS_STOPPED) {
we = rtc::WE_CLOSE;
} else if (ev == CS_PAUSED) {
we = rtc::WE_MINIMIZE;
} else if (ev == CS_RUNNING && previous_we_ == rtc::WE_MINIMIZE) {
we = rtc::WE_RESTORE;
} else {
return;
}
previous_we_ = we;
uint32 ssrc = 0;
if (!GetLocalSsrc(capturer, &ssrc)) {
return;
}
OnScreencastWindowEvent(ssrc, we);
}
bool VideoChannel::GetLocalSsrc(const VideoCapturer* capturer, uint32* ssrc) {
*ssrc = 0;
for (ScreencastMap::iterator iter = screencast_capturers_.begin();
iter != screencast_capturers_.end(); ++iter) {
if (iter->second == capturer) {
*ssrc = iter->first;
return true;
}
}
return false;
}
void VideoChannel::OnVideoChannelError(uint32 ssrc,
VideoMediaChannel::Error error) {
VideoChannelErrorMessageData* data = new VideoChannelErrorMessageData(
ssrc, error);
signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
}
void VideoChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
SrtpFilter::Error error) {
switch (error) {
case SrtpFilter::ERROR_FAIL:
OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
VideoMediaChannel::ERROR_REC_SRTP_ERROR :
VideoMediaChannel::ERROR_PLAY_SRTP_ERROR);
break;
case SrtpFilter::ERROR_AUTH:
OnVideoChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
VideoMediaChannel::ERROR_REC_SRTP_AUTH_FAILED :
VideoMediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED);
break;
case SrtpFilter::ERROR_REPLAY:
// Only receving channel should have this error.
ASSERT(mode == SrtpFilter::UNPROTECT);
// TODO(gangji): Turn on the signaling of replay error once we have
// switched to the new mechanism for doing video retransmissions.
// OnVideoChannelError(ssrc, VideoMediaChannel::ERROR_PLAY_SRTP_REPLAY);
break;
default:
break;
}
}
void VideoChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
GetSupportedVideoCryptoSuites(ciphers);
}
DataChannel::DataChannel(rtc::Thread* thread,
DataMediaChannel* media_channel,
BaseSession* session,
const std::string& content_name,
bool rtcp)
// MediaEngine is NULL
: BaseChannel(thread, NULL, media_channel, session, content_name, rtcp),
data_channel_type_(cricket::DCT_NONE),
ready_to_send_data_(false) {
}
DataChannel::~DataChannel() {
StopMediaMonitor();
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
bool DataChannel::Init() {
if (!BaseChannel::Init()) {
return false;
}
media_channel()->SignalDataReceived.connect(
this, &DataChannel::OnDataReceived);
media_channel()->SignalMediaError.connect(
this, &DataChannel::OnDataChannelError);
media_channel()->SignalReadyToSend.connect(
this, &DataChannel::OnDataChannelReadyToSend);
media_channel()->SignalStreamClosedRemotely.connect(
this, &DataChannel::OnStreamClosedRemotely);
srtp_filter()->SignalSrtpError.connect(
this, &DataChannel::OnSrtpError);
return true;
}
bool DataChannel::SendData(const SendDataParams& params,
const rtc::Buffer& payload,
SendDataResult* result) {
return InvokeOnWorker(Bind(&DataMediaChannel::SendData,
media_channel(), params, payload, result));
}
const ContentInfo* DataChannel::GetFirstContent(
const SessionDescription* sdesc) {
return GetFirstDataContent(sdesc);
}
bool DataChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
if (data_channel_type_ == DCT_SCTP) {
// TODO(pthatcher): Do this in a more robust way by checking for
// SCTP or DTLS.
return !IsRtpPacket(packet->data(), packet->size());
} else if (data_channel_type_ == DCT_RTP) {
return BaseChannel::WantsPacket(rtcp, packet);
}
return false;
}
bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type,
std::string* error_desc) {
// It hasn't been set before, so set it now.
if (data_channel_type_ == DCT_NONE) {
data_channel_type_ = new_data_channel_type;
return true;
}
// It's been set before, but doesn't match. That's bad.
if (data_channel_type_ != new_data_channel_type) {
std::ostringstream desc;
desc << "Data channel type mismatch."
<< " Expected " << data_channel_type_
<< " Got " << new_data_channel_type;
SafeSetError(desc.str(), error_desc);
return false;
}
// It's hasn't changed. Nothing to do.
return true;
}
bool DataChannel::SetDataChannelTypeFromContent(
const DataContentDescription* content,
std::string* error_desc) {
bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
(content->protocol() == kMediaProtocolDtlsSctp));
DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
return SetDataChannelType(data_channel_type, error_desc);
}
bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local data description";
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
ASSERT(data != NULL);
if (!data) {
SafeSetError("Can't find data content in local description.", error_desc);
return false;
}
bool ret = false;
if (!SetDataChannelTypeFromContent(data, error_desc)) {
return false;
}
if (data_channel_type_ == DCT_SCTP) {
// SCTP data channels don't need the rest of the stuff.
ret = UpdateLocalStreams_w(data->streams(), action, error_desc);
if (ret) {
set_local_content_direction(content->direction());
// As in SetRemoteContent_w, make sure we set the local SCTP port
// number as specified in our DataContentDescription.
if (!media_channel()->SetRecvCodecs(data->codecs())) {
SafeSetError("Failed to set data receive codecs.", error_desc);
ret = false;
}
}
} else {
ret = SetBaseLocalContent_w(content, action, error_desc);
if (action != CA_UPDATE || data->has_codecs()) {
if (!media_channel()->SetRecvCodecs(data->codecs())) {
SafeSetError("Failed to set data receive codecs.", error_desc);
ret = false;
}
}
}
// If everything worked, see if we can start receiving.
if (ret) {
std::vector<DataCodec>::const_iterator it = data->codecs().begin();
for (; it != data->codecs().end(); ++it) {
bundle_filter()->AddPayloadType(it->id);
}
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local data description";
}
return ret;
}
bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
ASSERT(worker_thread() == rtc::Thread::Current());
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
ASSERT(data != NULL);
if (!data) {
SafeSetError("Can't find data content in remote description.", error_desc);
return false;
}
bool ret = true;
if (!SetDataChannelTypeFromContent(data, error_desc)) {
return false;
}
if (data_channel_type_ == DCT_SCTP) {
LOG(LS_INFO) << "Setting SCTP remote data description";
// SCTP data channels don't need the rest of the stuff.
ret = UpdateRemoteStreams_w(content->streams(), action, error_desc);
if (ret) {
set_remote_content_direction(content->direction());
// We send the SCTP port number (not to be confused with the underlying
// UDP port number) as a codec parameter. Make sure it gets there.
if (!media_channel()->SetSendCodecs(data->codecs())) {
SafeSetError("Failed to set data send codecs.", error_desc);
ret = false;
}
}
} else {
// If the remote data doesn't have codecs and isn't an update, it
// must be empty, so ignore it.
if (action != CA_UPDATE && !data->has_codecs()) {
return true;
}
LOG(LS_INFO) << "Setting remote data description";
// Set remote video codecs (what the other side wants to receive).
if (action != CA_UPDATE || data->has_codecs()) {
if (!media_channel()->SetSendCodecs(data->codecs())) {
SafeSetError("Failed to set data send codecs.", error_desc);
ret = false;
}
}
if (ret) {
ret &= SetBaseRemoteContent_w(content, action, error_desc);
}
if (action != CA_UPDATE) {
int bandwidth_bps = data->bandwidth();
if (!media_channel()->SetMaxSendBandwidth(bandwidth_bps)) {
std::ostringstream desc;
desc << "Failed to set max send bandwidth for data content.";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
// If everything worked, see if we can start sending.
if (ret) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set remote data description";
}
return ret;
}
void DataChannel::ChangeState() {
// 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 = IsReadyToReceive();
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 = IsReadyToSend();
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 DataChannel::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(this, data->params, data->payload);
delete data;
break;
}
case MSG_CHANNEL_ERROR: {
const DataChannelErrorMessageData* data =
static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
case MSG_STREAMCLOSEDREMOTELY: {
rtc::TypedMessageData<uint32>* data =
static_cast<rtc::TypedMessageData<uint32>*>(pmsg->pdata);
SignalStreamClosedRemotely(data->data());
delete data;
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void DataChannel::OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
SignalConnectionMonitor(this, infos);
}
void DataChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
rtc::Thread::Current()));
media_monitor_->SignalUpdate.connect(
this, &DataChannel::OnMediaMonitorUpdate);
media_monitor_->Start(cms);
}
void DataChannel::StopMediaMonitor() {
if (media_monitor_) {
media_monitor_->Stop();
media_monitor_->SignalUpdate.disconnect(this);
media_monitor_.reset();
}
}
void DataChannel::OnMediaMonitorUpdate(
DataMediaChannel* media_channel, const DataMediaInfo& info) {
ASSERT(media_channel == this->media_channel());
SignalMediaMonitor(this, info);
}
void DataChannel::OnDataReceived(
const ReceiveDataParams& params, const char* data, size_t len) {
DataReceivedMessageData* msg = new DataReceivedMessageData(
params, data, len);
signaling_thread()->Post(this, MSG_DATARECEIVED, msg);
}
void DataChannel::OnDataChannelError(
uint32 ssrc, DataMediaChannel::Error err) {
DataChannelErrorMessageData* data = new DataChannelErrorMessageData(
ssrc, err);
signaling_thread()->Post(this, MSG_CHANNEL_ERROR, data);
}
void DataChannel::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(this, MSG_READYTOSENDDATA,
new DataChannelReadyToSendMessageData(writable));
}
void DataChannel::OnSrtpError(uint32 ssrc, SrtpFilter::Mode mode,
SrtpFilter::Error error) {
switch (error) {
case SrtpFilter::ERROR_FAIL:
OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
DataMediaChannel::ERROR_SEND_SRTP_ERROR :
DataMediaChannel::ERROR_RECV_SRTP_ERROR);
break;
case SrtpFilter::ERROR_AUTH:
OnDataChannelError(ssrc, (mode == SrtpFilter::PROTECT) ?
DataMediaChannel::ERROR_SEND_SRTP_AUTH_FAILED :
DataMediaChannel::ERROR_RECV_SRTP_AUTH_FAILED);
break;
case SrtpFilter::ERROR_REPLAY:
// Only receving channel should have this error.
ASSERT(mode == SrtpFilter::UNPROTECT);
OnDataChannelError(ssrc, DataMediaChannel::ERROR_RECV_SRTP_REPLAY);
break;
default:
break;
}
}
void DataChannel::GetSrtpCiphers(std::vector<std::string>* ciphers) const {
GetSupportedDataCryptoSuites(ciphers);
}
bool DataChannel::ShouldSetupDtlsSrtp() const {
return (data_channel_type_ == DCT_RTP);
}
void DataChannel::OnStreamClosedRemotely(uint32 sid) {
rtc::TypedMessageData<uint32>* message =
new rtc::TypedMessageData<uint32>(sid);
signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
}
} // namespace cricket