webrtc/talk/session/media/channel.cc
henrika@webrtc.org aebb1ade9d pRevert 5371 "Revert 5367 "Update talk to 59410372.""
> Revert 5367 "Update talk to 59410372."
> 
> > Update talk to 59410372.
> > 
> > R=jiayl@webrtc.org, wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/6929004
> 
> TBR=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6999004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-14 10:00:58 +00:00

2827 lines
90 KiB
C++

/*
* 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/base/buffer.h"
#include "talk/base/byteorder.h"
#include "talk/base/common.h"
#include "talk/base/dscp.h"
#include "talk/base/logging.h"
#include "talk/media/base/rtputils.h"
#include "talk/p2p/base/transportchannel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/rtcpmuxfilter.h"
#include "talk/session/media/typingmonitor.h"
namespace cricket {
enum {
MSG_ENABLE = 1,
MSG_DISABLE,
MSG_MUTESTREAM,
MSG_ISSTREAMMUTED,
MSG_SETREMOTECONTENT,
MSG_SETLOCALCONTENT,
MSG_EARLYMEDIATIMEOUT,
MSG_CANINSERTDTMF,
MSG_INSERTDTMF,
MSG_GETSTATS,
MSG_SETRENDERER,
MSG_ADDRECVSTREAM,
MSG_REMOVERECVSTREAM,
MSG_ADDSENDSTREAM,
MSG_REMOVESENDSTREAM,
MSG_SETRINGBACKTONE,
MSG_PLAYRINGBACKTONE,
MSG_SETMAXSENDBANDWIDTH,
MSG_ADDSCREENCAST,
MSG_REMOVESCREENCAST,
MSG_SENDINTRAFRAME,
MSG_REQUESTINTRAFRAME,
MSG_SCREENCASTWINDOWEVENT,
MSG_RTPPACKET,
MSG_RTCPPACKET,
MSG_CHANNEL_ERROR,
MSG_SETCHANNELOPTIONS,
MSG_SCALEVOLUME,
MSG_HANDLEVIEWREQUEST,
MSG_READYTOSENDDATA,
MSG_SENDDATA,
MSG_DATARECEIVED,
MSG_SETCAPTURER,
MSG_ISSCREENCASTING,
MSG_GETSCREENCASTDETAILS,
MSG_SETSCREENCASTFACTORY,
MSG_FIRSTPACKETRECEIVED,
MSG_SESSION_ERROR,
};
// Value specified in RFC 5764.
static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
static const int kAgcMinus10db = -10;
// TODO(hellner): use the device manager for creation of screen capturers when
// the cl enabling it has landed.
class NullScreenCapturerFactory : public VideoChannel::ScreenCapturerFactory {
public:
VideoCapturer* CreateScreenCapturer(const ScreencastId& window) {
return NULL;
}
};
VideoChannel::ScreenCapturerFactory* CreateScreenCapturerFactory() {
return new NullScreenCapturerFactory();
}
struct SetContentData : public talk_base::MessageData {
SetContentData(const MediaContentDescription* content, ContentAction action)
: content(content),
action(action),
result(false) {
}
const MediaContentDescription* content;
ContentAction action;
bool result;
};
struct SetBandwidthData : public talk_base::MessageData {
explicit SetBandwidthData(int value) : value(value), result(false) {}
int value;
bool result;
};
struct SetRingbackToneMessageData : public talk_base::MessageData {
SetRingbackToneMessageData(const void* b, int l)
: buf(b),
len(l),
result(false) {
}
const void* buf;
int len;
bool result;
};
struct PlayRingbackToneMessageData : public talk_base::MessageData {
PlayRingbackToneMessageData(uint32 s, bool p, bool l)
: ssrc(s),
play(p),
loop(l),
result(false) {
}
uint32 ssrc;
bool play;
bool loop;
bool result;
};
typedef talk_base::TypedMessageData<bool> BoolMessageData;
struct DtmfMessageData : public talk_base::MessageData {
DtmfMessageData(uint32 ssrc, int event, int duration, int flags)
: ssrc(ssrc),
event(event),
duration(duration),
flags(flags),
result(false) {
}
uint32 ssrc;
int event;
int duration;
int flags;
bool result;
};
struct ScaleVolumeMessageData : public talk_base::MessageData {
ScaleVolumeMessageData(uint32 s, double l, double r)
: ssrc(s),
left(l),
right(r),
result(false) {
}
uint32 ssrc;
double left;
double right;
bool result;
};
struct VoiceStatsMessageData : public talk_base::MessageData {
explicit VoiceStatsMessageData(VoiceMediaInfo* stats)
: result(false),
stats(stats) {
}
bool result;
VoiceMediaInfo* stats;
};
struct VideoStatsMessageData : public talk_base::MessageData {
explicit VideoStatsMessageData(VideoMediaInfo* stats)
: result(false),
stats(stats) {
}
bool result;
VideoMediaInfo* stats;
};
struct PacketMessageData : public talk_base::MessageData {
talk_base::Buffer packet;
talk_base::DiffServCodePoint dscp;
};
struct AudioRenderMessageData: public talk_base::MessageData {
AudioRenderMessageData(uint32 s, AudioRenderer* r, bool l)
: ssrc(s), renderer(r), is_local(l), result(false) {}
uint32 ssrc;
AudioRenderer* renderer;
bool is_local;
bool result;
};
struct VideoRenderMessageData : public talk_base::MessageData {
VideoRenderMessageData(uint32 s, VideoRenderer* r) : ssrc(s), renderer(r) {}
uint32 ssrc;
VideoRenderer* renderer;
};
struct AddScreencastMessageData : public talk_base::MessageData {
AddScreencastMessageData(uint32 s, const ScreencastId& id)
: ssrc(s),
window_id(id),
result(NULL) {
}
uint32 ssrc;
ScreencastId window_id;
VideoCapturer* result;
};
struct RemoveScreencastMessageData : public talk_base::MessageData {
explicit RemoveScreencastMessageData(uint32 s) : ssrc(s), result(false) {}
uint32 ssrc;
bool result;
};
struct ScreencastEventMessageData : public talk_base::MessageData {
ScreencastEventMessageData(uint32 s, talk_base::WindowEvent we)
: ssrc(s),
event(we) {
}
uint32 ssrc;
talk_base::WindowEvent event;
};
struct ViewRequestMessageData : public talk_base::MessageData {
explicit ViewRequestMessageData(const ViewRequest& r)
: request(r),
result(false) {
}
ViewRequest request;
bool result;
};
struct VoiceChannelErrorMessageData : public talk_base::MessageData {
VoiceChannelErrorMessageData(uint32 in_ssrc,
VoiceMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {
}
uint32 ssrc;
VoiceMediaChannel::Error error;
};
struct VideoChannelErrorMessageData : public talk_base::MessageData {
VideoChannelErrorMessageData(uint32 in_ssrc,
VideoMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {
}
uint32 ssrc;
VideoMediaChannel::Error error;
};
struct DataChannelErrorMessageData : public talk_base::MessageData {
DataChannelErrorMessageData(uint32 in_ssrc,
DataMediaChannel::Error in_error)
: ssrc(in_ssrc),
error(in_error) {}
uint32 ssrc;
DataMediaChannel::Error error;
};
struct SessionErrorMessageData : public talk_base::MessageData {
explicit SessionErrorMessageData(cricket::BaseSession::Error error)
: error_(error) {}
BaseSession::Error error_;
};
struct SsrcMessageData : public talk_base::MessageData {
explicit SsrcMessageData(uint32 ssrc) : ssrc(ssrc), result(false) {}
uint32 ssrc;
bool result;
};
struct StreamMessageData : public talk_base::MessageData {
explicit StreamMessageData(const StreamParams& in_sp)
: sp(in_sp),
result(false) {
}
StreamParams sp;
bool result;
};
struct MuteStreamData : public talk_base::MessageData {
MuteStreamData(uint32 ssrc, bool mute)
: ssrc(ssrc), mute(mute), result(false) {}
uint32 ssrc;
bool mute;
bool result;
};
struct AudioOptionsMessageData : public talk_base::MessageData {
explicit AudioOptionsMessageData(const AudioOptions& options)
: options(options),
result(false) {
}
AudioOptions options;
bool result;
};
struct VideoOptionsMessageData : public talk_base::MessageData {
explicit VideoOptionsMessageData(const VideoOptions& options)
: options(options),
result(false) {
}
VideoOptions options;
bool result;
};
struct SetCapturerMessageData : public talk_base::MessageData {
SetCapturerMessageData(uint32 s, VideoCapturer* c)
: ssrc(s),
capturer(c),
result(false) {
}
uint32 ssrc;
VideoCapturer* capturer;
bool result;
};
struct IsScreencastingMessageData : public talk_base::MessageData {
IsScreencastingMessageData()
: result(false) {
}
bool result;
};
struct VideoChannel::ScreencastDetailsMessageData :
public talk_base::MessageData {
explicit ScreencastDetailsMessageData(uint32 s)
: ssrc(s), fps(0), screencast_max_pixels(0) {
}
uint32 ssrc;
int fps;
int screencast_max_pixels;
};
struct SetScreenCaptureFactoryMessageData : public talk_base::MessageData {
explicit SetScreenCaptureFactoryMessageData(
VideoChannel::ScreenCapturerFactory* f)
: screencapture_factory(f) {
}
VideoChannel::ScreenCapturerFactory* screencapture_factory;
};
static const char* PacketType(bool rtcp) {
return (!rtcp) ? "RTP" : "RTCP";
}
static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) {
// Check the packet size. We could check the header too if needed.
return (packet &&
packet->length() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
packet->length() <= 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(talk_base::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) {
ASSERT(worker_thread_ == talk_base::Thread::Current());
LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
ASSERT(worker_thread_ == talk_base::Thread::Current());
Deinit();
StopConnectionMonitor();
FlushRtcpMessages(); // Send any outstanding RTCP packets.
Clear(); // 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_rtcp_transport_channel(NULL);
if (transport_channel_ != NULL)
session_->DestroyChannel(content_name_, transport_channel_->component());
LOG(LS_INFO) << "Destroyed channel";
}
bool BaseChannel::Init(TransportChannel* transport_channel,
TransportChannel* rtcp_transport_channel) {
if (transport_channel == NULL) {
return false;
}
if (rtcp() && rtcp_transport_channel == NULL) {
return false;
}
transport_channel_ = transport_channel;
if (!SetDtlsSrtpCiphers(transport_channel_, false)) {
return false;
}
transport_channel_->SignalWritableState.connect(
this, &BaseChannel::OnWritableState);
transport_channel_->SignalReadPacket.connect(
this, &BaseChannel::OnChannelRead);
transport_channel_->SignalReadyToSend.connect(
this, &BaseChannel::OnReadyToSend);
session_->SignalNewLocalDescription.connect(
this, &BaseChannel::OnNewLocalDescription);
session_->SignalNewRemoteDescription.connect(
this, &BaseChannel::OnNewRemoteDescription);
set_rtcp_transport_channel(rtcp_transport_channel);
// 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);
}
// Can be called from thread other than worker thread
bool BaseChannel::Enable(bool enable) {
Send(enable ? MSG_ENABLE : MSG_DISABLE);
return true;
}
// Can be called from thread other than worker thread
bool BaseChannel::MuteStream(uint32 ssrc, bool mute) {
MuteStreamData data(ssrc, mute);
Send(MSG_MUTESTREAM, &data);
return data.result;
}
bool BaseChannel::IsStreamMuted(uint32 ssrc) {
SsrcMessageData data(ssrc);
Send(MSG_ISSTREAMMUTED, &data);
return data.result;
}
bool BaseChannel::AddRecvStream(const StreamParams& sp) {
StreamMessageData data(sp);
Send(MSG_ADDRECVSTREAM, &data);
return data.result;
}
bool BaseChannel::RemoveRecvStream(uint32 ssrc) {
SsrcMessageData data(ssrc);
Send(MSG_REMOVERECVSTREAM, &data);
return data.result;
}
bool BaseChannel::AddSendStream(const StreamParams& sp) {
StreamMessageData data(sp);
Send(MSG_ADDSENDSTREAM, &data);
return data.result;
}
bool BaseChannel::RemoveSendStream(uint32 ssrc) {
SsrcMessageData data(ssrc);
Send(MSG_REMOVESENDSTREAM, &data);
return data.result;
}
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
ContentAction action) {
SetContentData data(content, action);
Send(MSG_SETLOCALCONTENT, &data);
return data.result;
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
ContentAction action) {
SetContentData data(content, action);
Send(MSG_SETREMOTECONTENT, &data);
return data.result;
}
bool BaseChannel::SetMaxSendBandwidth(int max_bandwidth) {
SetBandwidthData data(max_bandwidth);
Send(MSG_SETMAXSENDBANDWIDTH, &data);
return data.result;
}
void BaseChannel::StartConnectionMonitor(int cms) {
socket_monitor_.reset(new SocketMonitor(transport_channel_,
worker_thread(),
talk_base::Thread::Current()));
socket_monitor_->SignalUpdate.connect(
this, &BaseChannel::OnConnectionMonitorUpdate);
socket_monitor_->Start(cms);
}
void BaseChannel::StopConnectionMonitor() {
if (socket_monitor_) {
socket_monitor_->Stop();
socket_monitor_.reset();
}
}
void BaseChannel::set_rtcp_transport_channel(TransportChannel* channel) {
if (rtcp_transport_channel_ != channel) {
if (rtcp_transport_channel_) {
session_->DestroyChannel(
content_name_, rtcp_transport_channel_->component());
}
rtcp_transport_channel_ = channel;
if (rtcp_transport_channel_) {
// TODO(juberti): Propagate this error code
VERIFY(SetDtlsSrtpCiphers(rtcp_transport_channel_, true));
rtcp_transport_channel_->SignalWritableState.connect(
this, &BaseChannel::OnWritableState);
rtcp_transport_channel_->SignalReadPacket.connect(
this, &BaseChannel::OnChannelRead);
rtcp_transport_channel_->SignalReadyToSend.connect(
this, &BaseChannel::OnReadyToSend);
}
}
}
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(talk_base::Buffer* packet,
talk_base::DiffServCodePoint dscp) {
return SendPacket(false, packet, dscp);
}
bool BaseChannel::SendRtcp(talk_base::Buffer* packet,
talk_base::DiffServCodePoint dscp) {
return SendPacket(true, packet, dscp);
}
int BaseChannel::SetOption(SocketType type, talk_base::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 talk_base::PacketTime& packet_time,
int flags) {
// OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
ASSERT(worker_thread_ == talk_base::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);
talk_base::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, talk_base::Buffer* packet,
talk_base::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 (talk_base::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->length();
return false;
}
// Signal to the media sink before protecting the packet.
{
talk_base::CritScope cs(&signal_send_packet_cs_);
SignalSendPacketPreCrypto(packet->data(), packet->length(), rtcp);
}
// Protect if needed.
if (srtp_filter_.IsActive()) {
bool res;
char* data = packet->data();
int len = static_cast<int>(packet->length());
if (!rtcp) {
res = srtp_filter_.ProtectRtp(data, len,
static_cast<int>(packet->capacity()), &len);
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->SetLength(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.
{
talk_base::CritScope cs(&signal_send_packet_cs_);
SignalSendPacketPostCrypto(packet->data(), packet->length(), rtcp);
}
// Bon voyage.
int ret = channel->SendPacket(packet->data(), packet->length(), dscp,
(secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
if (ret != static_cast<int>(packet->length())) {
if (channel->GetError() == EWOULDBLOCK) {
LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
SetReadyToSend(channel, false);
}
return false;
}
return true;
}
bool BaseChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) {
// Protect ourselves against crazy data.
if (!ValidPacket(rtcp, packet)) {
LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
<< PacketType(rtcp) << " packet: wrong size="
<< packet->length();
return false;
}
// If this channel is suppose to handle RTP data, that is determined by
// checking against ssrc filter. This is necessary to do it here to avoid
// double decryption.
if (ssrc_filter_.IsActive() &&
!ssrc_filter_.DemuxPacket(packet->data(), packet->length(), rtcp)) {
return false;
}
return true;
}
void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet,
const talk_base::PacketTime& packet_time) {
if (!WantsPacket(rtcp, packet)) {
return;
}
if (!has_received_packet_) {
has_received_packet_ = true;
signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
}
// Signal to the media sink before unprotecting the packet.
{
talk_base::CritScope cs(&signal_recv_packet_cs_);
SignalRecvPacketPostCrypto(packet->data(), packet->length(), rtcp);
}
// Unprotect the packet, if needed.
if (srtp_filter_.IsActive()) {
char* data = packet->data();
int len = static_cast<int>(packet->length());
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->SetLength(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.
{
talk_base::CritScope cs(&signal_recv_packet_cs_);
SignalRecvPacketPreCrypto(packet->data(), packet->length(), 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) {
const ContentInfo* content_info =
GetFirstContent(session->local_description());
const MediaContentDescription* content_desc =
GetContentDescription(content_info);
if (content_desc && content_info && !content_info->rejected &&
!SetLocalContent(content_desc, action)) {
LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
session->SetError(BaseSession::ERROR_CONTENT);
}
}
void BaseChannel::OnNewRemoteDescription(
BaseSession* session, ContentAction action) {
const ContentInfo* content_info =
GetFirstContent(session->remote_description());
const MediaContentDescription* content_desc =
GetContentDescription(content_info);
if (content_desc && content_info && !content_info->rejected &&
!SetRemoteContent(content_desc, action)) {
LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
session->SetError(BaseSession::ERROR_CONTENT);
}
}
void BaseChannel::EnableMedia_w() {
ASSERT(worker_thread_ == talk_base::Thread::Current());
if (enabled_)
return;
LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
ChangeState();
}
void BaseChannel::DisableMedia_w() {
ASSERT(worker_thread_ == talk_base::Thread::Current());
if (!enabled_)
return;
LOG(LS_INFO) << "Channel disabled";
enabled_ = false;
ChangeState();
}
bool BaseChannel::MuteStream_w(uint32 ssrc, bool mute) {
ASSERT(worker_thread_ == talk_base::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_ == talk_base::Thread::Current());
return muted_streams_.find(ssrc) != muted_streams_.end();
}
void BaseChannel::ChannelWritable_w() {
ASSERT(worker_thread_ == talk_base::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)) {
LOG(LS_ERROR) << "Couldn't finish DTLS-SRTP on RTP channel";
SessionErrorMessageData data(BaseSession::ERROR_TRANSPORT);
// Sent synchronously.
signaling_thread()->Send(this, MSG_SESSION_ERROR, &data);
return;
}
if (rtcp_transport_channel_) {
if (!SetupDtlsSrtp(true)) {
LOG(LS_ERROR) << "Couldn't finish DTLS-SRTP on RTCP channel";
SessionErrorMessageData data(BaseSession::ERROR_TRANSPORT);
// Sent synchronously.
signaling_thread()->Send(this, MSG_SESSION_ERROR, &data);
return;
}
}
}
was_ever_writable_ = true;
writable_ = true;
ChangeState();
}
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;
talk_base::SSLRole role;
if (!channel->GetSslRole(&role)) {
LOG(LS_WARNING) << "GetSslRole failed";
return false;
}
if (role == talk_base::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_ == talk_base::Thread::Current());
if (!writable_)
return;
LOG(LS_INFO) << "Channel socket not writable ("
<< transport_channel_->content_name() << ", "
<< transport_channel_->component() << ")";
writable_ = false;
ChangeState();
}
// Sets the maximum video bandwidth for automatic bandwidth adjustment.
bool BaseChannel::SetMaxSendBandwidth_w(int max_bandwidth) {
return media_channel()->SetSendBandwidth(true, max_bandwidth);
}
// |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) {
*dtls = transport_channel_->IsDtlsActive();
if (*dtls && !cryptos.empty()) {
LOG(LS_WARNING) << "Cryptos must be empty when DTLS is active.";
return false;
}
return true;
}
bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
ContentAction action, ContentSource src) {
bool ret = false;
bool dtls = false;
ret = CheckSrtpConfig(cryptos, &dtls);
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 (ret && !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 (ret && !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 (ret && !dtls) {
ret = srtp_filter_.SetAnswer(cryptos, src);
}
break;
case CA_UPDATE:
// no crypto params.
ret = true;
break;
default:
break;
}
return ret;
}
bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
ContentSource src) {
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;
}
// |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 (ret && rtcp_mux_filter_.IsActive()) {
// If the RTP transport is already writable, then so are we.
if (transport_channel_->writable()) {
ChannelWritable_w();
}
}
return ret;
}
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
ASSERT(worker_thread() == talk_base::Thread::Current());
if (!media_channel()->AddRecvStream(sp))
return false;
return ssrc_filter_.AddStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32 ssrc) {
ASSERT(worker_thread() == talk_base::Thread::Current());
ssrc_filter_.RemoveStream(ssrc);
return media_channel()->RemoveRecvStream(ssrc);
}
bool BaseChannel::AddSendStream_w(const StreamParams& sp) {
ASSERT(worker_thread() == talk_base::Thread::Current());
return media_channel()->AddSendStream(sp);
}
bool BaseChannel::RemoveSendStream_w(uint32 ssrc) {
ASSERT(worker_thread() == talk_base::Thread::Current());
return media_channel()->RemoveSendStream(ssrc);
}
bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
ContentAction action) {
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) {
StreamParams existing_stream;
bool stream_exist = GetStreamByIds(local_streams_, it->groupid,
it->id, &existing_stream);
if (!stream_exist && it->has_ssrcs()) {
if (media_channel()->AddSendStream(*it)) {
local_streams_.push_back(*it);
LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
} else {
LOG(LS_INFO) << "Failed to add send stream ssrc: "
<< it->first_ssrc();
return false;
}
} else if (stream_exist && !it->has_ssrcs()) {
if (!media_channel()->RemoveSendStream(existing_stream.first_ssrc())) {
LOG(LS_ERROR) << "Failed to remove send stream with ssrc "
<< it->first_ssrc() << ".";
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(), NULL)) {
if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
LOG(LS_ERROR) << "Failed to remove send stream with ssrc "
<< it->first_ssrc() << ".";
ret = false;
}
}
}
// Check for new streams.
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
if (!GetStreamBySsrc(local_streams_, it->first_ssrc(), NULL)) {
if (media_channel()->AddSendStream(*it)) {
LOG(LS_INFO) << "Add send ssrc: " << it->ssrcs[0];
} else {
LOG(LS_INFO) << "Failed to add send stream ssrc: " << it->first_ssrc();
ret = false;
}
}
}
local_streams_ = streams;
return ret;
}
bool BaseChannel::UpdateRemoteStreams_w(
const std::vector<StreamParams>& streams,
ContentAction action) {
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) {
StreamParams existing_stream;
bool stream_exists = GetStreamByIds(remote_streams_, it->groupid,
it->id, &existing_stream);
if (!stream_exists && it->has_ssrcs()) {
if (AddRecvStream_w(*it)) {
remote_streams_.push_back(*it);
LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
} else {
LOG(LS_INFO) << "Failed to add remote stream ssrc: "
<< it->first_ssrc();
return false;
}
} else if (stream_exists && !it->has_ssrcs()) {
if (!RemoveRecvStream_w(existing_stream.first_ssrc())) {
LOG(LS_ERROR) << "Failed to remove remote stream with ssrc "
<< it->first_ssrc() << ".";
return false;
}
RemoveStreamBySsrc(&remote_streams_, existing_stream.first_ssrc());
} else {
LOG(LS_WARNING) << "Ignore unsupported stream update."
<< " Stream exists? " << stream_exists
<< " existing stream = " << existing_stream.ToString()
<< " 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(), NULL)) {
if (!RemoveRecvStream_w(it->first_ssrc())) {
LOG(LS_ERROR) << "Failed to remove remote stream with ssrc "
<< it->first_ssrc() << ".";
ret = false;
}
}
}
// Check for new streams.
for (StreamParamsVec::const_iterator it = streams.begin();
it != streams.end(); ++it) {
if (!GetStreamBySsrc(remote_streams_, it->first_ssrc(), NULL)) {
if (AddRecvStream_w(*it)) {
LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
} else {
LOG(LS_INFO) << "Failed to add remote stream ssrc: "
<< it->first_ssrc();
ret = false;
}
}
}
remote_streams_ = streams;
return ret;
}
bool BaseChannel::SetBaseLocalContent_w(const MediaContentDescription* content,
ContentAction action) {
// Cache secure_required_ for belt and suspenders check on SendPacket
secure_required_ = content->crypto_required();
bool ret = UpdateLocalStreams_w(content->streams(), action);
// Set local SRTP parameters (what we will encrypt with).
ret &= SetSrtp_w(content->cryptos(), action, CS_LOCAL);
// Set local RTCP mux parameters.
ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_LOCAL);
// Set local RTP header extensions.
if (content->rtp_header_extensions_set()) {
ret &= media_channel()->SetRecvRtpHeaderExtensions(
content->rtp_header_extensions());
}
set_local_content_direction(content->direction());
return ret;
}
bool BaseChannel::SetBaseRemoteContent_w(const MediaContentDescription* content,
ContentAction action) {
bool ret = UpdateRemoteStreams_w(content->streams(), action);
// Set remote SRTP parameters (what the other side will encrypt with).
ret &= SetSrtp_w(content->cryptos(), action, CS_REMOTE);
// Set remote RTCP mux parameters.
ret &= SetRtcpMux_w(content->rtcp_mux(), action, CS_REMOTE);
// Set remote RTP header extensions.
if (content->rtp_header_extensions_set()) {
ret &= media_channel()->SetSendRtpHeaderExtensions(
content->rtp_header_extensions());
}
if (content->bandwidth() != kAutoBandwidth) {
ret &= media_channel()->SetSendBandwidth(false, content->bandwidth());
}
set_remote_content_direction(content->direction());
return ret;
}
void BaseChannel::OnMessage(talk_base::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_ENABLE:
EnableMedia_w();
break;
case MSG_DISABLE:
DisableMedia_w();
break;
case MSG_MUTESTREAM: {
MuteStreamData* data = static_cast<MuteStreamData*>(pmsg->pdata);
data->result = MuteStream_w(data->ssrc, data->mute);
break;
}
case MSG_ISSTREAMMUTED: {
SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
data->result = IsStreamMuted_w(data->ssrc);
break;
}
case MSG_SETLOCALCONTENT: {
SetContentData* data = static_cast<SetContentData*>(pmsg->pdata);
data->result = SetLocalContent_w(data->content, data->action);
break;
}
case MSG_SETREMOTECONTENT: {
SetContentData* data = static_cast<SetContentData*>(pmsg->pdata);
data->result = SetRemoteContent_w(data->content, data->action);
break;
}
case MSG_ADDRECVSTREAM: {
StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata);
data->result = AddRecvStream_w(data->sp);
break;
}
case MSG_REMOVERECVSTREAM: {
SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
data->result = RemoveRecvStream_w(data->ssrc);
break;
}
case MSG_ADDSENDSTREAM: {
StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata);
data->result = AddSendStream_w(data->sp);
break;
}
case MSG_REMOVESENDSTREAM: {
SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
data->result = RemoveSendStream_w(data->ssrc);
break;
}
case MSG_SETMAXSENDBANDWIDTH: {
SetBandwidthData* data = static_cast<SetBandwidthData*>(pmsg->pdata);
data->result = SetMaxSendBandwidth_w(data->value);
break;
}
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;
}
case MSG_SESSION_ERROR: {
SessionErrorMessageData* data = static_cast<SessionErrorMessageData*>
(pmsg->pdata);
session_->SetError(data->error_);
break;
}
}
}
void BaseChannel::Send(uint32 id, talk_base::MessageData *pdata) {
worker_thread_->Send(this, id, pdata);
}
void BaseChannel::Post(uint32 id, talk_base::MessageData *pdata) {
worker_thread_->Post(this, id, pdata);
}
void BaseChannel::PostDelayed(int cmsDelay, uint32 id,
talk_base::MessageData *pdata) {
worker_thread_->PostDelayed(cmsDelay, this, id, pdata);
}
void BaseChannel::Clear(uint32 id, talk_base::MessageList* removed) {
worker_thread_->Clear(this, id, removed);
}
void BaseChannel::FlushRtcpMessages() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
ASSERT(talk_base::Thread::Current() == worker_thread_);
talk_base::MessageList rtcp_messages;
Clear(MSG_RTCPPACKET, &rtcp_messages);
for (talk_base::MessageList::iterator it = rtcp_messages.begin();
it != rtcp_messages.end(); ++it) {
Send(MSG_RTCPPACKET, it->pdata);
}
}
VoiceChannel::VoiceChannel(talk_base::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() {
TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
content_name(), "rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
if (!BaseChannel::Init(session()->CreateChannel(
content_name(), "rtp", ICE_CANDIDATE_COMPONENT_RTP),
rtcp_channel)) {
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) {
AudioRenderMessageData data(ssrc, renderer, false);
Send(MSG_SETRENDERER, &data);
return data.result;
}
bool VoiceChannel::SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
AudioRenderMessageData data(ssrc, renderer, true);
Send(MSG_SETRENDERER, &data);
return data.result;
}
bool VoiceChannel::SetRingbackTone(const void* buf, int len) {
SetRingbackToneMessageData data(buf, len);
Send(MSG_SETRINGBACKTONE, &data);
return data.result;
}
// 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
PostDelayed(kEarlyMediaTimeout, MSG_EARLYMEDIATIMEOUT);
} else {
// Stop the timeout if currently going.
Clear(MSG_EARLYMEDIATIMEOUT);
}
}
bool VoiceChannel::PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
PlayRingbackToneMessageData data(ssrc, play, loop);
Send(MSG_PLAYRINGBACKTONE, &data);
return data.result;
}
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() {
BoolMessageData data(false);
Send(MSG_CANINSERTDTMF, &data);
return data.data();
}
bool VoiceChannel::InsertDtmf(uint32 ssrc, int event_code, int duration,
int flags) {
DtmfMessageData data(ssrc, event_code, duration, flags);
Send(MSG_INSERTDTMF, &data);
return data.result;
}
bool VoiceChannel::SetOutputScaling(uint32 ssrc, double left, double right) {
ScaleVolumeMessageData data(ssrc, left, right);
Send(MSG_SCALEVOLUME, &data);
return data.result;
}
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
VoiceStatsMessageData data(stats);
Send(MSG_GETSTATS, &data);
return data.result;
}
void VoiceChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
talk_base::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, talk_base::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 talk_base::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) {
ASSERT(worker_thread() == talk_base::Thread::Current());
LOG(LS_INFO) << "Setting local voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
ASSERT(audio != NULL);
if (!audio) return false;
bool ret = SetBaseLocalContent_w(content, action);
// 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()) {
ret &= media_channel()->SetRecvCodecs(audio->codecs());
}
// If everything worked, see if we can start receiving.
if (ret) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local voice description";
}
return ret;
}
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action) {
ASSERT(worker_thread() == talk_base::Thread::Current());
LOG(LS_INFO) << "Setting remote voice description";
const AudioContentDescription* audio =
static_cast<const AudioContentDescription*>(content);
ASSERT(audio != NULL);
if (!audio) return false;
bool ret = true;
// Set remote video codecs (what the other side wants to receive).
if (action != CA_UPDATE || audio->has_codecs()) {
ret &= media_channel()->SetSendCodecs(audio->codecs());
}
ret &= SetBaseRemoteContent_w(content, action);
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() == talk_base::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() == talk_base::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::CanInsertDtmf_w() {
return media_channel()->CanInsertDtmf();
}
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::SetOutputScaling_w(uint32 ssrc, double left, double right) {
return media_channel()->SetOutputScaling(ssrc, left, right);
}
bool VoiceChannel::GetStats_w(VoiceMediaInfo* stats) {
return media_channel()->GetStats(stats);
}
bool VoiceChannel::SetChannelOptions(const AudioOptions& options) {
AudioOptionsMessageData data(options);
Send(MSG_SETCHANNELOPTIONS, &data);
return data.result;
}
bool VoiceChannel::SetChannelOptions_w(const AudioOptions& options) {
return media_channel()->SetOptions(options);
}
bool VoiceChannel::SetRenderer_w(uint32 ssrc, AudioRenderer* renderer,
bool is_local) {
if (is_local)
return media_channel()->SetLocalRenderer(ssrc, renderer);
return media_channel()->SetRemoteRenderer(ssrc, renderer);
}
void VoiceChannel::OnMessage(talk_base::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_SETRINGBACKTONE: {
SetRingbackToneMessageData* data =
static_cast<SetRingbackToneMessageData*>(pmsg->pdata);
data->result = SetRingbackTone_w(data->buf, data->len);
break;
}
case MSG_PLAYRINGBACKTONE: {
PlayRingbackToneMessageData* data =
static_cast<PlayRingbackToneMessageData*>(pmsg->pdata);
data->result = PlayRingbackTone_w(data->ssrc, data->play, data->loop);
break;
}
case MSG_EARLYMEDIATIMEOUT:
HandleEarlyMediaTimeout();
break;
case MSG_CANINSERTDTMF: {
BoolMessageData* data =
static_cast<BoolMessageData*>(pmsg->pdata);
data->data() = CanInsertDtmf_w();
break;
}
case MSG_INSERTDTMF: {
DtmfMessageData* data =
static_cast<DtmfMessageData*>(pmsg->pdata);
data->result = InsertDtmf_w(data->ssrc, data->event, data->duration,
data->flags);
break;
}
case MSG_SCALEVOLUME: {
ScaleVolumeMessageData* data =
static_cast<ScaleVolumeMessageData*>(pmsg->pdata);
data->result = SetOutputScaling_w(data->ssrc, data->left, data->right);
break;
}
case MSG_GETSTATS: {
VoiceStatsMessageData* data =
static_cast<VoiceStatsMessageData*>(pmsg->pdata);
data->result = GetStats_w(data->stats);
break;
}
case MSG_CHANNEL_ERROR: {
VoiceChannelErrorMessageData* data =
static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
case MSG_SETCHANNELOPTIONS: {
AudioOptionsMessageData* data =
static_cast<AudioOptionsMessageData*>(pmsg->pdata);
data->result = SetChannelOptions_w(data->options);
break;
}
case MSG_SETRENDERER: {
AudioRenderMessageData* data =
static_cast<AudioRenderMessageData*>(pmsg->pdata);
data->result = SetRenderer_w(data->ssrc, data->renderer, data->is_local);
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void VoiceChannel::OnConnectionMonitorUpdate(
SocketMonitor* 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(talk_base::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),
screencapture_factory_(CreateScreenCapturerFactory()),
previous_we_(talk_base::WE_CLOSE) {
}
bool VideoChannel::Init() {
TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
content_name(), "video_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
if (!BaseChannel::Init(session()->CreateChannel(
content_name(), "video_rtp", ICE_CANDIDATE_COMPONENT_RTP),
rtcp_channel)) {
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) {
VideoRenderMessageData data(ssrc, renderer);
Send(MSG_SETRENDERER, &data);
return true;
}
bool VideoChannel::ApplyViewRequest(const ViewRequest& request) {
ViewRequestMessageData data(request);
Send(MSG_HANDLEVIEWREQUEST, &data);
return data.result;
}
VideoCapturer* VideoChannel::AddScreencast(
uint32 ssrc, const ScreencastId& id) {
AddScreencastMessageData data(ssrc, id);
Send(MSG_ADDSCREENCAST, &data);
return data.result;
}
bool VideoChannel::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
SetCapturerMessageData data(ssrc, capturer);
Send(MSG_SETCAPTURER, &data);
return data.result;
}
bool VideoChannel::RemoveScreencast(uint32 ssrc) {
RemoveScreencastMessageData data(ssrc);
Send(MSG_REMOVESCREENCAST, &data);
return data.result;
}
bool VideoChannel::IsScreencasting() {
IsScreencastingMessageData data;
Send(MSG_ISSCREENCASTING, &data);
return data.result;
}
int VideoChannel::GetScreencastFps(uint32 ssrc) {
ScreencastDetailsMessageData data(ssrc);
Send(MSG_GETSCREENCASTDETAILS, &data);
return data.fps;
}
int VideoChannel::GetScreencastMaxPixels(uint32 ssrc) {
ScreencastDetailsMessageData data(ssrc);
Send(MSG_GETSCREENCASTDETAILS, &data);
return data.screencast_max_pixels;
}
bool VideoChannel::SendIntraFrame() {
Send(MSG_SENDINTRAFRAME);
return true;
}
bool VideoChannel::RequestIntraFrame() {
Send(MSG_REQUESTINTRAFRAME);
return true;
}
void VideoChannel::SetScreenCaptureFactory(
ScreenCapturerFactory* screencapture_factory) {
SetScreenCaptureFactoryMessageData data(screencapture_factory);
Send(MSG_SETSCREENCASTFACTORY, &data);
}
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) {
VideoStatsMessageData data(stats);
Send(MSG_GETSTATS, &data);
return data.result;
}
void VideoChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
talk_base::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) {
ASSERT(worker_thread() == talk_base::Thread::Current());
LOG(LS_INFO) << "Setting local video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
ASSERT(video != NULL);
if (!video) return false;
bool ret = SetBaseLocalContent_w(content, action);
// Set local video codecs (what we want to receive).
if (action != CA_UPDATE || video->has_codecs()) {
ret &= media_channel()->SetRecvCodecs(video->codecs());
}
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) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local video description";
}
return ret;
}
bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action) {
ASSERT(worker_thread() == talk_base::Thread::Current());
LOG(LS_INFO) << "Setting remote video description";
const VideoContentDescription* video =
static_cast<const VideoContentDescription*>(content);
ASSERT(video != NULL);
if (!video) return false;
bool ret = true;
// Set remote video codecs (what the other side wants to receive).
if (action != CA_UPDATE || video->has_codecs()) {
ret &= media_channel()->SetSendCodecs(video->codecs());
}
ret &= SetBaseRemoteContent_w(content, action);
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, NULL)) {
LOG(LS_WARNING) << "View request for ("
<< it->selector.ssrc << ", '"
<< it->selector.groupid << "', '"
<< it->selector.streamid << "'"
<< ") is not in the local streams.";
}
}
return ret;
}
void VideoChannel::SetRenderer_w(uint32 ssrc, VideoRenderer* renderer) {
media_channel()->SetRenderer(ssrc, renderer);
}
VideoCapturer* VideoChannel::AddScreencast_w(
uint32 ssrc, const ScreencastId& id) {
if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
return NULL;
}
VideoCapturer* screen_capturer =
screencapture_factory_->CreateScreenCapturer(id);
if (!screen_capturer) {
return NULL;
}
screen_capturer->SignalStateChange.connect(this,
&VideoChannel::OnStateChange);
screencast_capturers_[ssrc] = screen_capturer;
return screen_capturer;
}
bool VideoChannel::SetCapturer_w(uint32 ssrc, VideoCapturer* capturer) {
return media_channel()->SetCapturer(ssrc, capturer);
}
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::ScreencastDetails_w(
ScreencastDetailsMessageData* 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::SetScreenCaptureFactory_w(
ScreenCapturerFactory* screencapture_factory) {
if (screencapture_factory == NULL) {
screencapture_factory_.reset(CreateScreenCapturerFactory());
} else {
screencapture_factory_.reset(screencapture_factory);
}
}
bool VideoChannel::GetStats_w(VideoMediaInfo* stats) {
return media_channel()->GetStats(stats);
}
void VideoChannel::OnScreencastWindowEvent_s(uint32 ssrc,
talk_base::WindowEvent we) {
ASSERT(signaling_thread() == talk_base::Thread::Current());
SignalScreencastWindowEvent(ssrc, we);
}
bool VideoChannel::SetChannelOptions(const VideoOptions &options) {
VideoOptionsMessageData data(options);
Send(MSG_SETCHANNELOPTIONS, &data);
return data.result;
}
bool VideoChannel::SetChannelOptions_w(const VideoOptions &options) {
return media_channel()->SetOptions(options);
}
void VideoChannel::OnMessage(talk_base::Message *pmsg) {
switch (pmsg->message_id) {
case MSG_SETRENDERER: {
const VideoRenderMessageData* data =
static_cast<VideoRenderMessageData*>(pmsg->pdata);
SetRenderer_w(data->ssrc, data->renderer);
break;
}
case MSG_ADDSCREENCAST: {
AddScreencastMessageData* data =
static_cast<AddScreencastMessageData*>(pmsg->pdata);
data->result = AddScreencast_w(data->ssrc, data->window_id);
break;
}
case MSG_SETCAPTURER: {
SetCapturerMessageData* data =
static_cast<SetCapturerMessageData*>(pmsg->pdata);
data->result = SetCapturer_w(data->ssrc, data->capturer);
break;
}
case MSG_REMOVESCREENCAST: {
RemoveScreencastMessageData* data =
static_cast<RemoveScreencastMessageData*>(pmsg->pdata);
data->result = RemoveScreencast_w(data->ssrc);
break;
}
case MSG_SCREENCASTWINDOWEVENT: {
const ScreencastEventMessageData* data =
static_cast<ScreencastEventMessageData*>(pmsg->pdata);
OnScreencastWindowEvent_s(data->ssrc, data->event);
delete data;
break;
}
case MSG_ISSCREENCASTING: {
IsScreencastingMessageData* data =
static_cast<IsScreencastingMessageData*>(pmsg->pdata);
data->result = IsScreencasting_w();
break;
}
case MSG_GETSCREENCASTDETAILS: {
ScreencastDetailsMessageData* data =
static_cast<ScreencastDetailsMessageData*>(pmsg->pdata);
ScreencastDetails_w(data);
break;
}
case MSG_SENDINTRAFRAME: {
SendIntraFrame_w();
break;
}
case MSG_REQUESTINTRAFRAME: {
RequestIntraFrame_w();
break;
}
case MSG_SETCHANNELOPTIONS: {
VideoOptionsMessageData* data =
static_cast<VideoOptionsMessageData*>(pmsg->pdata);
data->result = SetChannelOptions_w(data->options);
break;
}
case MSG_CHANNEL_ERROR: {
const VideoChannelErrorMessageData* data =
static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
SignalMediaError(this, data->ssrc, data->error);
delete data;
break;
}
case MSG_HANDLEVIEWREQUEST: {
ViewRequestMessageData* data =
static_cast<ViewRequestMessageData*>(pmsg->pdata);
data->result = ApplyViewRequest_w(data->request);
break;
}
case MSG_SETSCREENCASTFACTORY: {
SetScreenCaptureFactoryMessageData* data =
static_cast<SetScreenCaptureFactoryMessageData*>(pmsg->pdata);
SetScreenCaptureFactory_w(data->screencapture_factory);
break;
}
case MSG_GETSTATS: {
VideoStatsMessageData* data =
static_cast<VideoStatsMessageData*>(pmsg->pdata);
data->result = GetStats_w(data->stats);
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void VideoChannel::OnConnectionMonitorUpdate(
SocketMonitor *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,
talk_base::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.
talk_base::WindowEvent we;
if (ev == CS_STOPPED) {
we = talk_base::WE_CLOSE;
} else if (ev == CS_PAUSED) {
we = talk_base::WE_MINIMIZE;
} else if (ev == CS_RUNNING && previous_we_ == talk_base::WE_MINIMIZE) {
we = talk_base::WE_RESTORE;
} else {
return;
}
previous_we_ = we;
uint32 ssrc = 0;
if (!GetLocalSsrc(capturer, &ssrc)) {
return;
}
ScreencastEventMessageData* pdata =
new ScreencastEventMessageData(ssrc, we);
signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
}
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(talk_base::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() {
TransportChannel* rtcp_channel = rtcp() ? session()->CreateChannel(
content_name(), "data_rtcp", ICE_CANDIDATE_COMPONENT_RTCP) : NULL;
if (!BaseChannel::Init(session()->CreateChannel(
content_name(), "data_rtp", ICE_CANDIDATE_COMPONENT_RTP),
rtcp_channel)) {
return false;
}
media_channel()->SignalDataReceived.connect(
this, &DataChannel::OnDataReceived);
media_channel()->SignalMediaError.connect(
this, &DataChannel::OnDataChannelError);
media_channel()->SignalReadyToSend.connect(
this, &DataChannel::OnDataChannelReadyToSend);
srtp_filter()->SignalSrtpError.connect(
this, &DataChannel::OnSrtpError);
return true;
}
bool DataChannel::SendData(const SendDataParams& params,
const talk_base::Buffer& payload,
SendDataResult* result) {
SendDataMessageData message_data(params, &payload, result);
Send(MSG_SENDDATA, &message_data);
return message_data.succeeded;
}
const ContentInfo* DataChannel::GetFirstContent(
const SessionDescription* sdesc) {
return GetFirstDataContent(sdesc);
}
static bool IsRtpPacket(const talk_base::Buffer* packet) {
int version;
if (!GetRtpVersion(packet->data(), packet->length(), &version)) {
return false;
}
return version == 2;
}
bool DataChannel::WantsPacket(bool rtcp, talk_base::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);
} else if (data_channel_type_ == DCT_RTP) {
return BaseChannel::WantsPacket(rtcp, packet);
}
return false;
}
// Sets the maximum bandwidth. Anything over this will be dropped.
bool DataChannel::SetMaxSendBandwidth_w(int max_bps) {
LOG(LS_INFO) << "DataChannel: Setting max bandwidth to " << max_bps;
return media_channel()->SetSendBandwidth(false, max_bps);
}
bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type) {
// 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) {
LOG(LS_WARNING) << "Data channel type mismatch."
<< " Expected " << data_channel_type_
<< " Got " << new_data_channel_type;
return false;
}
// It's hasn't changed. Nothing to do.
return true;
}
bool DataChannel::SetDataChannelTypeFromContent(
const DataContentDescription* content) {
bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
(content->protocol() == kMediaProtocolDtlsSctp));
DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
return SetDataChannelType(data_channel_type);
}
bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
ContentAction action) {
ASSERT(worker_thread() == talk_base::Thread::Current());
LOG(LS_INFO) << "Setting local data description";
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
ASSERT(data != NULL);
if (!data) return false;
bool ret = false;
if (!SetDataChannelTypeFromContent(data)) {
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);
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.
ret = media_channel()->SetRecvCodecs(data->codecs());
}
} else {
ret = SetBaseLocalContent_w(content, action);
if (action != CA_UPDATE || data->has_codecs()) {
ret &= media_channel()->SetRecvCodecs(data->codecs());
}
}
// If everything worked, see if we can start receiving.
if (ret) {
ChangeState();
} else {
LOG(LS_WARNING) << "Failed to set local data description";
}
return ret;
}
bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action) {
ASSERT(worker_thread() == talk_base::Thread::Current());
const DataContentDescription* data =
static_cast<const DataContentDescription*>(content);
ASSERT(data != NULL);
if (!data) return false;
bool ret = true;
if (!SetDataChannelTypeFromContent(data)) {
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);
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.
ret = media_channel()->SetSendCodecs(data->codecs());
}
} 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()) {
ret &= media_channel()->SetSendCodecs(data->codecs());
}
if (ret) {
ret &= SetBaseRemoteContent_w(content, action);
}
if (action != CA_UPDATE) {
int bandwidth_bps = data->bandwidth();
bool auto_bandwidth = (bandwidth_bps == kAutoBandwidth);
ret &= media_channel()->SetSendBandwidth(auto_bandwidth, bandwidth_bps);
}
}
// 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";
}
// Post to trigger SignalReadyToSendData.
signaling_thread()->Post(this, MSG_READYTOSENDDATA,
new DataChannelReadyToSendMessageData(send));
LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
}
void DataChannel::OnMessage(talk_base::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_SENDDATA: {
SendDataMessageData* msg =
static_cast<SendDataMessageData*>(pmsg->pdata);
msg->succeeded = media_channel()->SendData(
msg->params, *(msg->payload), msg->result);
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;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void DataChannel::OnConnectionMonitorUpdate(
SocketMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
SignalConnectionMonitor(this, infos);
}
void DataChannel::StartMediaMonitor(int cms) {
media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
talk_base::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);
}
} // namespace cricket