| /* |
| * libjingle |
| * Copyright 2004--2007, 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/phone/channel.h" |
| |
| #include "talk/base/buffer.h" |
| #include "talk/base/byteorder.h" |
| #include "talk/base/common.h" |
| #include "talk/base/logging.h" |
| #include "talk/p2p/base/transportchannel.h" |
| #include "talk/session/phone/channelmanager.h" |
| #include "talk/session/phone/mediasessionclient.h" |
| #include "talk/session/phone/rtcpmuxfilter.h" |
| #include "talk/session/phone/rtputils.h" |
| |
| namespace cricket { |
| |
| struct PacketMessageData : public talk_base::MessageData { |
| talk_base::Buffer packet; |
| }; |
| |
| 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; |
| }; |
| |
| 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); |
| } |
| |
| 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), |
| was_ever_writable_(false), |
| has_local_content_(false), |
| has_remote_content_(false), |
| muted_(false) { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| LOG(LS_INFO) << "Created channel"; |
| } |
| |
| BaseChannel::~BaseChannel() { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| 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_->name()); |
| 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; |
| media_channel_->SetInterface(this); |
| transport_channel_->SignalWritableState.connect( |
| this, &BaseChannel::OnWritableState); |
| transport_channel_->SignalReadPacket.connect( |
| this, &BaseChannel::OnChannelRead); |
| |
| session_->SignalState.connect(this, &BaseChannel::OnSessionState); |
| session_->SignalRemoteDescriptionUpdate.connect(this, |
| &BaseChannel::OnRemoteDescriptionUpdate); |
| |
| OnSessionState(session(), session()->state()); |
| set_rtcp_transport_channel(rtcp_transport_channel); |
| return true; |
| } |
| |
| // 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::Mute(bool mute) { |
| Clear(MSG_UNMUTE); // Clear any penging auto-unmutes. |
| Send(mute ? MSG_MUTE : MSG_UNMUTE); |
| return true; |
| } |
| |
| bool BaseChannel::RemoveStream(uint32 ssrc) { |
| StreamMessageData data(ssrc, 0); |
| Send(MSG_REMOVESTREAM, &data); |
| ssrc_filter()->RemoveStream(ssrc); |
| return true; |
| } |
| |
| bool BaseChannel::SetRtcpCName(const std::string& cname) { |
| SetRtcpCNameData data(cname); |
| Send(MSG_SETRTCPCNAME, &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_.get()) { |
| 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_->name()); |
| } |
| rtcp_transport_channel_ = channel; |
| if (rtcp_transport_channel_) { |
| rtcp_transport_channel_->SignalWritableState.connect( |
| this, &BaseChannel::OnWritableState); |
| rtcp_transport_channel_->SignalReadPacket.connect( |
| this, &BaseChannel::OnChannelRead); |
| } |
| } |
| } |
| |
| bool BaseChannel::SendPacket(talk_base::Buffer* packet) { |
| return SendPacket(false, packet); |
| } |
| |
| bool BaseChannel::SendRtcp(talk_base::Buffer* packet) { |
| return SendPacket(true, packet); |
| } |
| |
| int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt, |
| int value) { |
| switch (type) { |
| case ST_RTP: return transport_channel_->SetOption(opt, value); |
| case ST_RTCP: return rtcp_transport_channel_->SetOption(opt, value); |
| default: return -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) { |
| // 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); |
| } |
| |
| bool BaseChannel::PacketIsRtcp(const TransportChannel* channel, |
| const char* data, size_t len) { |
| return (channel == rtcp_transport_channel_ || |
| rtcp_mux_filter_.DemuxRtcp(data, len)); |
| } |
| |
| bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) { |
| // Ensure we have a path capable of sending packets. |
| if (!writable_) { |
| return false; |
| } |
| |
| // 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); |
| 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; |
| } |
| |
| // Protect if needed. |
| if (srtp_filter_.IsActive()) { |
| bool res; |
| char* data = packet->data(); |
| int len = packet->length(); |
| if (!rtcp) { |
| res = srtp_filter_.ProtectRtp(data, len, 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, 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); |
| } |
| |
| // Signal to the media sink after protecting the packet. TODO: |
| // Separate APIs to record unprotected media and protected header. |
| { |
| talk_base::CritScope cs(&signal_send_packet_cs_); |
| SignalSendPacket(packet->data(), packet->length(), rtcp); |
| } |
| |
| // Bon voyage. |
| return (channel->SendPacket(packet->data(), packet->length()) |
| == static_cast<int>(packet->length())); |
| } |
| |
| void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet) { |
| // Protect ourselvs against crazy data. |
| if (!ValidPacket(rtcp, packet)) { |
| LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " " |
| << PacketType(rtcp) << " packet: wrong size=" |
| << packet->length(); |
| return; |
| } |
| |
| // 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; |
| } |
| |
| // Signal to the media sink before unprotecting the packet. TODO: |
| // Separate APIs to record unprotected media and protected header. |
| { |
| talk_base::CritScope cs(&signal_recv_packet_cs_); |
| SignalRecvPacket(packet->data(), packet->length(), rtcp); |
| } |
| |
| // Unprotect the packet, if needed. |
| if (srtp_filter_.IsActive()) { |
| char* data = packet->data(); |
| int len = 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); |
| } |
| |
| // Push it down to the media channel. |
| if (!rtcp) { |
| media_channel_->OnPacketReceived(packet); |
| } else { |
| media_channel_->OnRtcpReceived(packet); |
| } |
| } |
| |
| void BaseChannel::OnSessionState(BaseSession* session, |
| BaseSession::State state) { |
| const MediaContentDescription* content = NULL; |
| switch (state) { |
| case Session::STATE_SENTINITIATE: |
| content = GetFirstContent(session->local_description()); |
| if (content && !SetLocalContent(content, CA_OFFER)) { |
| LOG(LS_ERROR) << "Failure in SetLocalContent with CA_OFFER"; |
| session->SetError(BaseSession::ERROR_CONTENT); |
| } |
| break; |
| case Session::STATE_SENTACCEPT: |
| content = GetFirstContent(session->local_description()); |
| if (content && !SetLocalContent(content, CA_ANSWER)) { |
| LOG(LS_ERROR) << "Failure in SetLocalContent with CA_ANSWER"; |
| session->SetError(BaseSession::ERROR_CONTENT); |
| } |
| break; |
| case Session::STATE_RECEIVEDINITIATE: |
| content = GetFirstContent(session->remote_description()); |
| if (content && !SetRemoteContent(content, CA_OFFER)) { |
| LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_OFFER"; |
| session->SetError(BaseSession::ERROR_CONTENT); |
| } |
| break; |
| case Session::STATE_RECEIVEDACCEPT: |
| content = GetFirstContent(session->remote_description()); |
| if (content && !SetRemoteContent(content, CA_ANSWER)) { |
| LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_ANSWER"; |
| session->SetError(BaseSession::ERROR_CONTENT); |
| } |
| break; |
| default: |
| break; |
| } |
| } |
| |
| void BaseChannel::OnRemoteDescriptionUpdate(BaseSession* session) { |
| const MediaContentDescription* content = |
| GetFirstContent(session->remote_description()); |
| |
| if (content && !SetRemoteContent(content, CA_UPDATE)) { |
| LOG(LS_ERROR) << "Failure in SetRemoteContent with CA_UPDATE"; |
| 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(); |
| } |
| |
| void BaseChannel::MuteMedia_w() { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| if (muted_) |
| return; |
| |
| if (media_channel()->Mute(true)) { |
| LOG(LS_INFO) << "Channel muted"; |
| muted_ = true; |
| } |
| } |
| |
| void BaseChannel::UnmuteMedia_w() { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| if (!muted_) |
| return; |
| |
| if (media_channel()->Mute(false)) { |
| LOG(LS_INFO) << "Channel unmuted"; |
| muted_ = false; |
| } |
| } |
| |
| void BaseChannel::ChannelWritable_w() { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| if (writable_) |
| return; |
| LOG(LS_INFO) << "Channel socket writable (" |
| << transport_channel_->name().c_str() << ")" |
| << (was_ever_writable_ ? "" : " for the first time"); |
| was_ever_writable_ = true; |
| writable_ = true; |
| ChangeState(); |
| } |
| |
| void BaseChannel::ChannelNotWritable_w() { |
| ASSERT(worker_thread_ == talk_base::Thread::Current()); |
| if (!writable_) |
| return; |
| |
| LOG(LS_INFO) << "Channel socket not writable (" |
| << transport_channel_->name().c_str() << ")"; |
| 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); |
| } |
| |
| bool BaseChannel::SetRtcpCName_w(const std::string& cname) { |
| return media_channel()->SetRtcpCName(cname); |
| } |
| |
| bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos, |
| ContentAction action, ContentSource src) { |
| bool ret; |
| if (action == CA_OFFER) { |
| ret = srtp_filter_.SetOffer(cryptos, src); |
| } else if (action == CA_ANSWER) { |
| ret = srtp_filter_.SetAnswer(cryptos, src); |
| } else { |
| // CA_UPDATE, no crypto params. |
| ret = true; |
| } |
| return ret; |
| } |
| |
| bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action, |
| ContentSource src) { |
| bool ret; |
| if (action == CA_OFFER) { |
| ret = rtcp_mux_filter_.SetOffer(enable, src); |
| } else if (action == 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); |
| // If the RTP transport is already writable, then so are we. |
| if (transport_channel_->writable()) { |
| ChannelWritable_w(); |
| } |
| } |
| } else { |
| // CA_UPDATE, no RTCP mux info. |
| ret = true; |
| } |
| return ret; |
| } |
| |
| // TODO: Check all of the ssrcs in all of the streams in |
| // the content, and not just the first one. |
| bool BaseChannel::SetSsrcMux_w(bool enable, |
| const MediaContentDescription* content, |
| ContentAction action, |
| ContentSource src) { |
| bool ret = true; |
| if (action == CA_OFFER) { |
| ret = ssrc_filter_.SetOffer(enable, src); |
| if (ret && src == CS_REMOTE) { // if received offer with ssrc |
| ret = ssrc_filter_.AddStream(content->first_ssrc()); |
| } |
| } else if (action == CA_ANSWER) { |
| ret = ssrc_filter_.SetAnswer(enable, src); |
| if (ret && src == CS_REMOTE && ssrc_filter_.IsActive()) { |
| ret = ssrc_filter_.AddStream(content->first_ssrc()); |
| } |
| } |
| 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_MUTE: |
| MuteMedia_w(); |
| break; |
| case MSG_UNMUTE: |
| UnmuteMedia_w(); |
| break; |
| |
| case MSG_SETRTCPCNAME: { |
| SetRtcpCNameData* data = static_cast<SetRtcpCNameData*>(pmsg->pdata); |
| data->result = SetRtcpCName_w(data->cname); |
| 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_REMOVESTREAM: { |
| StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata); |
| RemoveStream_w(data->ssrc1); |
| 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); |
| delete data; // because it is Posted |
| 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), |
| mute_on_type_(false), |
| mute_on_type_timeout_(kTypingBlackoutPeriod) { |
| } |
| |
| VoiceChannel::~VoiceChannel() { |
| StopAudioMonitor(); |
| StopMediaMonitor(); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| } |
| |
| bool VoiceChannel::Init() { |
| TransportChannel* rtcp_channel = rtcp() ? |
| session()->CreateChannel(content_name(), "rtcp") : NULL; |
| if (!BaseChannel::Init(session()->CreateChannel(content_name(), "rtp"), |
| rtcp_channel)) { |
| return false; |
| } |
| media_channel()->SignalMediaError.connect( |
| this, &VoiceChannel::OnVoiceChannelError); |
| srtp_filter()->SignalSrtpError.connect( |
| this, &VoiceChannel::OnSrtpError); |
| return true; |
| } |
| |
| bool VoiceChannel::AddStream(uint32 ssrc) { |
| StreamMessageData data(ssrc, 0); |
| Send(MSG_ADDSTREAM, &data); |
| ssrc_filter()->AddStream(ssrc); |
| return true; |
| } |
| |
| bool VoiceChannel::SetRingbackTone(const void* buf, int len) { |
| SetRingbackToneMessageData data(buf, len); |
| Send(MSG_SETRINGBACKTONE, &data); |
| return data.result; |
| } |
| |
| // TODO: 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) { |
| DtmfMessageData data(digit, playout); |
| Send(MSG_PRESSDTMF, &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; |
| } |
| |
| 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_.get()) { |
| 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_.get()) { |
| audio_monitor_->Stop(); |
| audio_monitor_.reset(); |
| } |
| } |
| |
| bool VoiceChannel::IsAudioMonitorRunning() const { |
| return (audio_monitor_.get() != NULL); |
| } |
| |
| int VoiceChannel::GetInputLevel_w() { |
| return media_engine()->GetInputLevel(); |
| } |
| |
| int VoiceChannel::GetOutputLevel_w() { |
| return media_channel()->GetOutputLevel(); |
| } |
| |
| void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) { |
| media_channel()->GetActiveStreams(actives); |
| } |
| |
| void VoiceChannel::OnChannelRead(TransportChannel* channel, |
| const char* data, size_t len) { |
| BaseChannel::OnChannelRead(channel, data, len); |
| |
| // 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 = enabled() && has_local_content(); |
| 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 = enabled() && has_remote_content() && was_ever_writable(); |
| 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 MediaContentDescription* VoiceChannel::GetFirstContent( |
| const SessionDescription* sdesc) { |
| const ContentInfo* cinfo = GetFirstAudioContent(sdesc); |
| if (cinfo == NULL) |
| return NULL; |
| |
| return static_cast<const MediaContentDescription*>(cinfo->description); |
| } |
| |
| 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); |
| |
| bool ret; |
| if (audio->has_ssrcs()) { |
| // TODO: Handle multiple streams and ssrcs here. |
| media_channel()->SetSendSsrc(audio->first_ssrc()); |
| LOG(LS_INFO) << "Set send ssrc for audio: " << audio->first_ssrc(); |
| } |
| // Set local SRTP parameters (what we will encrypt with). |
| ret = SetSrtp_w(audio->cryptos(), action, CS_LOCAL); |
| // Set local RTCP mux parameters. |
| if (ret) { |
| ret = SetRtcpMux_w(audio->rtcp_mux(), action, CS_LOCAL); |
| } |
| // Set SSRC mux filter |
| if (ret) { |
| ret = SetSsrcMux_w(audio->has_ssrcs(), content, action, CS_LOCAL); |
| } |
| // Set local audio codecs (what we want to receive). |
| if (ret) { |
| ret = media_channel()->SetRecvCodecs(audio->codecs()); |
| } |
| // Set local RTP header extensions. |
| if (ret && audio->rtp_header_extensions_set()) { |
| ret = media_channel()->SetRecvRtpHeaderExtensions( |
| audio->rtp_header_extensions()); |
| } |
| // If everything worked, see if we can start receiving. |
| if (ret) { |
| set_has_local_content(true); |
| 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); |
| |
| bool ret; |
| // Set remote SRTP parameters (what the other side will encrypt with). |
| ret = SetSrtp_w(audio->cryptos(), action, CS_REMOTE); |
| // Set remote RTCP mux parameters. |
| if (ret) { |
| ret = SetRtcpMux_w(audio->rtcp_mux(), action, CS_REMOTE); |
| } |
| // Set SSRC mux filter |
| if (ret) { |
| ret = SetSsrcMux_w(audio->has_ssrcs(), content, action, CS_REMOTE); |
| } |
| |
| // Set remote video codecs (what the other side wants to receive). |
| if (ret) { |
| ret = media_channel()->SetSendCodecs(audio->codecs()); |
| } |
| // Set remote RTP header extensions. |
| if (ret && audio->rtp_header_extensions_set()) { |
| ret = media_channel()->SetSendRtpHeaderExtensions( |
| audio->rtp_header_extensions()); |
| } |
| |
| // Tweak our audio processing settings, if needed. |
| int audio_options = 0; |
| if (audio->conference_mode()) { |
| audio_options |= OPT_CONFERENCE; |
| } |
| if (audio->agc_minus_10db()) { |
| audio_options |= OPT_AGC_MINUS_10DB; |
| } |
| 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) { |
| set_has_remote_content(true); |
| ChangeState(); |
| } else { |
| LOG(LS_WARNING) << "Failed to set remote voice description"; |
| } |
| return ret; |
| } |
| |
| void VoiceChannel::AddStream_w(uint32 ssrc) { |
| ASSERT(worker_thread() == talk_base::Thread::Current()); |
| media_channel()->AddStream(ssrc); |
| } |
| |
| void VoiceChannel::RemoveStream_w(uint32 ssrc) { |
| media_channel()->RemoveStream(ssrc); |
| } |
| |
| 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::PressDTMF_w(int digit, bool playout) { |
| if (!enabled() || !writable()) { |
| return false; |
| } |
| |
| return media_channel()->PressDTMF(digit, playout); |
| } |
| |
| bool VoiceChannel::SetOutputScaling_w(uint32 ssrc, double left, double right) { |
| return media_channel()->SetOutputScaling(ssrc, left, right); |
| } |
| |
| void VoiceChannel::OnMessage(talk_base::Message *pmsg) { |
| switch (pmsg->message_id) { |
| case MSG_ADDSTREAM: { |
| StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata); |
| AddStream_w(data->ssrc1); |
| break; |
| } |
| 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_PRESSDTMF: { |
| DtmfMessageData* data = static_cast<DtmfMessageData*>(pmsg->pdata); |
| data->result = PressDTMF_w(data->digit, data->playout); |
| break; |
| } |
| case MSG_SCALEVOLUME: { |
| ScaleVolumeMessageData* data = |
| static_cast<ScaleVolumeMessageData*>(pmsg->pdata); |
| data->result = SetOutputScaling_w(data->ssrc, data->left, data->right); |
| 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( |
| 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) { |
| if (err == VoiceMediaChannel::ERROR_REC_TYPING_NOISE_DETECTED && |
| mute_on_type_ && !muted()) { |
| Mute(true); |
| PostDelayed(mute_on_type_timeout_, MSG_UNMUTE, NULL); |
| } |
| 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; |
| } |
| } |
| |
| 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) { |
| } |
| |
| bool VideoChannel::Init() { |
| TransportChannel* rtcp_channel = rtcp() ? |
| session()->CreateChannel(content_name(), "video_rtcp") : NULL; |
| if (!BaseChannel::Init( |
| session()->CreateChannel(content_name(), "video_rtp"), |
| rtcp_channel)) { |
| return false; |
| } |
| media_channel()->SignalScreencastWindowEvent.connect( |
| this, &VideoChannel::OnScreencastWindowEvent); |
| 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() { |
| StopMediaMonitor(); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| } |
| |
| bool VideoChannel::AddStream(uint32 ssrc, uint32 voice_ssrc) { |
| StreamMessageData data(ssrc, voice_ssrc); |
| Send(MSG_ADDSTREAM, &data); |
| ssrc_filter()->AddStream(ssrc); |
| return true; |
| } |
| |
| bool VideoChannel::SetRenderer(uint32 ssrc, VideoRenderer* renderer) { |
| RenderMessageData data(ssrc, renderer); |
| Send(MSG_SETRENDERER, &data); |
| return true; |
| } |
| |
| bool VideoChannel::AddScreencast(uint32 ssrc, talk_base::WindowId id) { |
| ScreencastMessageData data(ssrc, id); |
| Send(MSG_ADDSCREENCAST, &data); |
| return true; |
| } |
| |
| bool VideoChannel::RemoveScreencast(uint32 ssrc) { |
| ScreencastMessageData data(ssrc, 0); |
| Send(MSG_REMOVESCREENCAST, &data); |
| return true; |
| } |
| |
| bool VideoChannel::SendIntraFrame() { |
| Send(MSG_SENDINTRAFRAME); |
| return true; |
| } |
| |
| bool VideoChannel::RequestIntraFrame() { |
| Send(MSG_REQUESTINTRAFRAME); |
| return true; |
| } |
| |
| void VideoChannel::EnableCpuAdaptation(bool enable) { |
| Send(enable ? MSG_ENABLECPUADAPTATION : MSG_DISABLECPUADAPTATION); |
| } |
| |
| 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 = enabled() && has_local_content(); |
| if (!media_channel()->SetRender(recv)) { |
| LOG(LS_ERROR) << "Failed to SetRender on video channel"; |
| // TODO: 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 = enabled() && has_remote_content() && was_ever_writable(); |
| if (!media_channel()->SetSend(send)) { |
| LOG(LS_ERROR) << "Failed to SetSend on video channel"; |
| // TODO: Report error back to server. |
| } |
| |
| LOG(LS_INFO) << "Changing video state, recv=" << recv << " send=" << send; |
| } |
| |
| 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_.get()) { |
| media_monitor_->Stop(); |
| media_monitor_.reset(); |
| } |
| } |
| |
| const MediaContentDescription* VideoChannel::GetFirstContent( |
| const SessionDescription* sdesc) { |
| const ContentInfo* cinfo = GetFirstVideoContent(sdesc); |
| if (cinfo == NULL) |
| return NULL; |
| |
| return static_cast<const MediaContentDescription*>(cinfo->description); |
| } |
| |
| 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); |
| |
| bool ret; |
| if (video->has_ssrcs()) { |
| // TODO: Handle multiple streams and ssrcs here. |
| media_channel()->SetSendSsrc(video->first_ssrc()); |
| LOG(LS_INFO) << "Set send ssrc for video: " << video->first_ssrc(); |
| } |
| // Set local SRTP parameters (what we will encrypt with). |
| ret = SetSrtp_w(video->cryptos(), action, CS_LOCAL); |
| // Set local RTCP mux parameters. |
| if (ret) { |
| ret = SetRtcpMux_w(video->rtcp_mux(), action, CS_LOCAL); |
| } |
| // Set SSRC mux filter |
| if (ret) { |
| ret = SetSsrcMux_w(video->has_ssrcs(), content, action, CS_LOCAL); |
| } |
| |
| // Set local video codecs (what we want to receive). |
| if (ret) { |
| ret = media_channel()->SetRecvCodecs(video->codecs()); |
| } |
| // Set local RTP header extensions. |
| if (ret && video->rtp_header_extensions_set()) { |
| ret = media_channel()->SetRecvRtpHeaderExtensions( |
| video->rtp_header_extensions()); |
| } |
| // If everything worked, see if we can start receiving. |
| if (ret) { |
| set_has_local_content(true); |
| 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); |
| |
| bool ret; |
| // Set remote SRTP parameters (what the other side will encrypt with). |
| ret = SetSrtp_w(video->cryptos(), action, CS_REMOTE); |
| // Set remote RTCP mux parameters. |
| if (ret) { |
| ret = SetRtcpMux_w(video->rtcp_mux(), action, CS_REMOTE); |
| } |
| // Set SSRC mux filter |
| if (ret) { |
| ret = SetSsrcMux_w(video->has_ssrcs(), content, action, CS_REMOTE); |
| } |
| // Set remote video codecs (what the other side wants to receive). |
| if (ret) { |
| ret = media_channel()->SetSendCodecs(video->codecs()); |
| } |
| // Set remote RTP header extensions. |
| if (ret && video->rtp_header_extensions_set()) { |
| ret = media_channel()->SetSendRtpHeaderExtensions( |
| video->rtp_header_extensions()); |
| } |
| // Set bandwidth parameters (what the other side wants to get, default=auto) |
| if (ret) { |
| int bandwidth_bps = video->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) { |
| set_has_remote_content(true); |
| ChangeState(); |
| } else { |
| LOG(LS_WARNING) << "Failed to set remote video description"; |
| } |
| return ret; |
| } |
| |
| void VideoChannel::AddStream_w(uint32 ssrc, uint32 voice_ssrc) { |
| media_channel()->AddStream(ssrc, voice_ssrc); |
| } |
| |
| void VideoChannel::RemoveStream_w(uint32 ssrc) { |
| media_channel()->RemoveStream(ssrc); |
| } |
| |
| void VideoChannel::SetRenderer_w(uint32 ssrc, VideoRenderer* renderer) { |
| media_channel()->SetRenderer(ssrc, renderer); |
| } |
| |
| void VideoChannel::AddScreencast_w(uint32 ssrc, talk_base::WindowId id) { |
| media_channel()->AddScreencast(ssrc, id); |
| } |
| |
| void VideoChannel::RemoveScreencast_w(uint32 ssrc) { |
| media_channel()->RemoveScreencast(ssrc); |
| } |
| |
| void VideoChannel::OnScreencastWindowEvent_s(uint32 ssrc, |
| talk_base::WindowEvent we) { |
| ASSERT(signaling_thread() == talk_base::Thread::Current()); |
| SignalScreencastWindowEvent(ssrc, we); |
| } |
| |
| void VideoChannel::OnMessage(talk_base::Message *pmsg) { |
| switch (pmsg->message_id) { |
| case MSG_ADDSTREAM: { |
| StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata); |
| AddStream_w(data->ssrc1, data->ssrc2); |
| break; |
| } |
| case MSG_SETRENDERER: { |
| RenderMessageData* data = static_cast<RenderMessageData*>(pmsg->pdata); |
| SetRenderer_w(data->ssrc, data->renderer); |
| break; |
| } |
| case MSG_ADDSCREENCAST: { |
| ScreencastMessageData* data = |
| static_cast<ScreencastMessageData*>(pmsg->pdata); |
| AddScreencast_w(data->ssrc, data->window_id); |
| break; |
| } |
| case MSG_REMOVESCREENCAST: { |
| ScreencastMessageData* data = |
| static_cast<ScreencastMessageData*>(pmsg->pdata); |
| RemoveScreencast_w(data->ssrc); |
| break; |
| } |
| case MSG_SCREENCASTWINDOWEVENT: { |
| ScreencastEventData* data = |
| static_cast<ScreencastEventData*>(pmsg->pdata); |
| OnScreencastWindowEvent_s(data->ssrc, data->event); |
| delete data; |
| break; |
| } |
| case MSG_SENDINTRAFRAME: |
| SendIntraFrame_w(); |
| break; |
| case MSG_REQUESTINTRAFRAME: |
| RequestIntraFrame_w(); |
| break; |
| case MSG_ENABLECPUADAPTATION: |
| EnableCpuAdaptation_w(true); |
| break; |
| case MSG_DISABLECPUADAPTATION: |
| EnableCpuAdaptation_w(false); |
| 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( |
| SocketMonitor *monitor, const std::vector<ConnectionInfo> &infos) { |
| SignalConnectionMonitor(this, infos); |
| } |
| |
| 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) { |
| ScreencastEventData* pdata = new ScreencastEventData(ssrc, event); |
| signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata); |
| } |
| |
| 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: 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; |
| } |
| } |
| |
| } // namespace cricket |