| Index: talk/session/media/channel.cc
|
| diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
|
| deleted file mode 100644
|
| index bee37718add1e6ceaddbe308fa2fd4aad1330808..0000000000000000000000000000000000000000
|
| --- a/talk/session/media/channel.cc
|
| +++ /dev/null
|
| @@ -1,2274 +0,0 @@
|
| -/*
|
| - * 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 <utility>
|
| -
|
| -#include "talk/session/media/channel.h"
|
| -
|
| -#include "talk/session/media/channelmanager.h"
|
| -#include "webrtc/audio/audio_sink.h"
|
| -#include "webrtc/base/bind.h"
|
| -#include "webrtc/base/buffer.h"
|
| -#include "webrtc/base/byteorder.h"
|
| -#include "webrtc/base/common.h"
|
| -#include "webrtc/base/dscp.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/trace_event.h"
|
| -#include "webrtc/media/base/constants.h"
|
| -#include "webrtc/media/base/rtputils.h"
|
| -#include "webrtc/p2p/base/transportchannel.h"
|
| -
|
| -namespace cricket {
|
| -using rtc::Bind;
|
| -
|
| -namespace {
|
| -// See comment below for why we need to use a pointer to a scoped_ptr.
|
| -bool SetRawAudioSink_w(VoiceMediaChannel* channel,
|
| - uint32_t ssrc,
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface>* sink) {
|
| - channel->SetRawAudioSink(ssrc, std::move(*sink));
|
| - return true;
|
| -}
|
| -} // namespace
|
| -
|
| -enum {
|
| - MSG_EARLYMEDIATIMEOUT = 1,
|
| - MSG_SCREENCASTWINDOWEVENT,
|
| - MSG_RTPPACKET,
|
| - MSG_RTCPPACKET,
|
| - MSG_CHANNEL_ERROR,
|
| - MSG_READYTOSENDDATA,
|
| - MSG_DATARECEIVED,
|
| - MSG_FIRSTPACKETRECEIVED,
|
| - MSG_STREAMCLOSEDREMOTELY,
|
| -};
|
| -
|
| -// Value specified in RFC 5764.
|
| -static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
|
| -
|
| -static const int kAgcMinus10db = -10;
|
| -
|
| -static void SafeSetError(const std::string& message, std::string* error_desc) {
|
| - if (error_desc) {
|
| - *error_desc = message;
|
| - }
|
| -}
|
| -
|
| -struct PacketMessageData : public rtc::MessageData {
|
| - rtc::Buffer packet;
|
| - rtc::PacketOptions options;
|
| -};
|
| -
|
| -struct ScreencastEventMessageData : public rtc::MessageData {
|
| - ScreencastEventMessageData(uint32_t s, rtc::WindowEvent we)
|
| - : ssrc(s), event(we) {}
|
| - uint32_t ssrc;
|
| - rtc::WindowEvent event;
|
| -};
|
| -
|
| -struct VoiceChannelErrorMessageData : public rtc::MessageData {
|
| - VoiceChannelErrorMessageData(uint32_t in_ssrc,
|
| - VoiceMediaChannel::Error in_error)
|
| - : ssrc(in_ssrc), error(in_error) {}
|
| - uint32_t ssrc;
|
| - VoiceMediaChannel::Error error;
|
| -};
|
| -
|
| -struct VideoChannelErrorMessageData : public rtc::MessageData {
|
| - VideoChannelErrorMessageData(uint32_t in_ssrc,
|
| - VideoMediaChannel::Error in_error)
|
| - : ssrc(in_ssrc), error(in_error) {}
|
| - uint32_t ssrc;
|
| - VideoMediaChannel::Error error;
|
| -};
|
| -
|
| -struct DataChannelErrorMessageData : public rtc::MessageData {
|
| - DataChannelErrorMessageData(uint32_t in_ssrc,
|
| - DataMediaChannel::Error in_error)
|
| - : ssrc(in_ssrc), error(in_error) {}
|
| - uint32_t ssrc;
|
| - DataMediaChannel::Error error;
|
| -};
|
| -
|
| -static const char* PacketType(bool rtcp) {
|
| - return (!rtcp) ? "RTP" : "RTCP";
|
| -}
|
| -
|
| -static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) {
|
| - // Check the packet size. We could check the header too if needed.
|
| - return (packet &&
|
| - packet->size() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) &&
|
| - packet->size() <= kMaxRtpPacketLen);
|
| -}
|
| -
|
| -static bool IsReceiveContentDirection(MediaContentDirection direction) {
|
| - return direction == MD_SENDRECV || direction == MD_RECVONLY;
|
| -}
|
| -
|
| -static bool IsSendContentDirection(MediaContentDirection direction) {
|
| - return direction == MD_SENDRECV || direction == MD_SENDONLY;
|
| -}
|
| -
|
| -static const MediaContentDescription* GetContentDescription(
|
| - const ContentInfo* cinfo) {
|
| - if (cinfo == NULL)
|
| - return NULL;
|
| - return static_cast<const MediaContentDescription*>(cinfo->description);
|
| -}
|
| -
|
| -template <class Codec>
|
| -void RtpParametersFromMediaDescription(
|
| - const MediaContentDescriptionImpl<Codec>* desc,
|
| - RtpParameters<Codec>* params) {
|
| - // TODO(pthatcher): Remove this once we're sure no one will give us
|
| - // a description without codecs (currently a CA_UPDATE with just
|
| - // streams can).
|
| - if (desc->has_codecs()) {
|
| - params->codecs = desc->codecs();
|
| - }
|
| - // TODO(pthatcher): See if we really need
|
| - // rtp_header_extensions_set() and remove it if we don't.
|
| - if (desc->rtp_header_extensions_set()) {
|
| - params->extensions = desc->rtp_header_extensions();
|
| - }
|
| - params->rtcp.reduced_size = desc->rtcp_reduced_size();
|
| -}
|
| -
|
| -template <class Codec, class Options>
|
| -void RtpSendParametersFromMediaDescription(
|
| - const MediaContentDescriptionImpl<Codec>* desc,
|
| - RtpSendParameters<Codec, Options>* send_params) {
|
| - RtpParametersFromMediaDescription(desc, send_params);
|
| - send_params->max_bandwidth_bps = desc->bandwidth();
|
| -}
|
| -
|
| -BaseChannel::BaseChannel(rtc::Thread* thread,
|
| - MediaChannel* media_channel,
|
| - TransportController* transport_controller,
|
| - const std::string& content_name,
|
| - bool rtcp)
|
| - : worker_thread_(thread),
|
| - transport_controller_(transport_controller),
|
| - media_channel_(media_channel),
|
| - content_name_(content_name),
|
| - rtcp_transport_enabled_(rtcp),
|
| - transport_channel_(nullptr),
|
| - rtcp_transport_channel_(nullptr),
|
| - enabled_(false),
|
| - writable_(false),
|
| - rtp_ready_to_send_(false),
|
| - rtcp_ready_to_send_(false),
|
| - was_ever_writable_(false),
|
| - local_content_direction_(MD_INACTIVE),
|
| - remote_content_direction_(MD_INACTIVE),
|
| - has_received_packet_(false),
|
| - dtls_keyed_(false),
|
| - secure_required_(false),
|
| - rtp_abs_sendtime_extn_id_(-1) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Created channel for " << content_name;
|
| -}
|
| -
|
| -BaseChannel::~BaseChannel() {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - Deinit();
|
| - StopConnectionMonitor();
|
| - FlushRtcpMessages(); // Send any outstanding RTCP packets.
|
| - worker_thread_->Clear(this); // eats any outstanding messages or packets
|
| - // We must destroy the media channel before the transport channel, otherwise
|
| - // the media channel may try to send on the dead transport channel. NULLing
|
| - // is not an effective strategy since the sends will come on another thread.
|
| - delete media_channel_;
|
| - // Note that we don't just call set_transport_channel(nullptr) because that
|
| - // would call a pure virtual method which we can't do from a destructor.
|
| - if (transport_channel_) {
|
| - DisconnectFromTransportChannel(transport_channel_);
|
| - transport_controller_->DestroyTransportChannel_w(
|
| - transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
| - }
|
| - if (rtcp_transport_channel_) {
|
| - DisconnectFromTransportChannel(rtcp_transport_channel_);
|
| - transport_controller_->DestroyTransportChannel_w(
|
| - transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
| - }
|
| - LOG(LS_INFO) << "Destroyed channel";
|
| -}
|
| -
|
| -bool BaseChannel::Init() {
|
| - if (!SetTransport(content_name())) {
|
| - return false;
|
| - }
|
| -
|
| - if (!SetDtlsSrtpCryptoSuites(transport_channel(), false)) {
|
| - return false;
|
| - }
|
| - if (rtcp_transport_enabled() &&
|
| - !SetDtlsSrtpCryptoSuites(rtcp_transport_channel(), true)) {
|
| - return false;
|
| - }
|
| -
|
| - // Both RTP and RTCP channels are set, we can call SetInterface on
|
| - // media channel and it can set network options.
|
| - media_channel_->SetInterface(this);
|
| - return true;
|
| -}
|
| -
|
| -void BaseChannel::Deinit() {
|
| - media_channel_->SetInterface(NULL);
|
| -}
|
| -
|
| -bool BaseChannel::SetTransport(const std::string& transport_name) {
|
| - return worker_thread_->Invoke<bool>(
|
| - Bind(&BaseChannel::SetTransport_w, this, transport_name));
|
| -}
|
| -
|
| -bool BaseChannel::SetTransport_w(const std::string& transport_name) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - if (transport_name == transport_name_) {
|
| - // Nothing to do if transport name isn't changing
|
| - return true;
|
| - }
|
| -
|
| - // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport
|
| - // changes and wait until the DTLS handshake is complete to set the newly
|
| - // negotiated parameters.
|
| - if (ShouldSetupDtlsSrtp()) {
|
| - // Set |writable_| to false such that UpdateWritableState_w can set up
|
| - // DTLS-SRTP when the writable_ becomes true again.
|
| - writable_ = false;
|
| - srtp_filter_.ResetParams();
|
| - }
|
| -
|
| - // TODO(guoweis): Remove this grossness when we remove non-muxed RTCP.
|
| - if (rtcp_transport_enabled()) {
|
| - LOG(LS_INFO) << "Create RTCP TransportChannel for " << content_name()
|
| - << " on " << transport_name << " transport ";
|
| - set_rtcp_transport_channel(
|
| - transport_controller_->CreateTransportChannel_w(
|
| - transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP),
|
| - false /* update_writablity */);
|
| - if (!rtcp_transport_channel()) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - // We're not updating the writablity during the transition state.
|
| - set_transport_channel(transport_controller_->CreateTransportChannel_w(
|
| - transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
|
| - if (!transport_channel()) {
|
| - return false;
|
| - }
|
| -
|
| - // TODO(guoweis): Remove this grossness when we remove non-muxed RTCP.
|
| - if (rtcp_transport_enabled()) {
|
| - // We can only update the RTCP ready to send after set_transport_channel has
|
| - // handled channel writability.
|
| - SetReadyToSend(
|
| - true, rtcp_transport_channel() && rtcp_transport_channel()->writable());
|
| - }
|
| - transport_name_ = transport_name;
|
| - return true;
|
| -}
|
| -
|
| -void BaseChannel::set_transport_channel(TransportChannel* new_tc) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - TransportChannel* old_tc = transport_channel_;
|
| - if (!old_tc && !new_tc) {
|
| - // Nothing to do
|
| - return;
|
| - }
|
| - ASSERT(old_tc != new_tc);
|
| -
|
| - if (old_tc) {
|
| - DisconnectFromTransportChannel(old_tc);
|
| - transport_controller_->DestroyTransportChannel_w(
|
| - transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
| - }
|
| -
|
| - transport_channel_ = new_tc;
|
| -
|
| - if (new_tc) {
|
| - ConnectToTransportChannel(new_tc);
|
| - for (const auto& pair : socket_options_) {
|
| - new_tc->SetOption(pair.first, pair.second);
|
| - }
|
| - }
|
| -
|
| - // Update aggregate writable/ready-to-send state between RTP and RTCP upon
|
| - // setting new channel
|
| - UpdateWritableState_w();
|
| - SetReadyToSend(false, new_tc && new_tc->writable());
|
| -}
|
| -
|
| -void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc,
|
| - bool update_writablity) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - TransportChannel* old_tc = rtcp_transport_channel_;
|
| - if (!old_tc && !new_tc) {
|
| - // Nothing to do
|
| - return;
|
| - }
|
| - ASSERT(old_tc != new_tc);
|
| -
|
| - if (old_tc) {
|
| - DisconnectFromTransportChannel(old_tc);
|
| - transport_controller_->DestroyTransportChannel_w(
|
| - transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
| - }
|
| -
|
| - rtcp_transport_channel_ = new_tc;
|
| -
|
| - if (new_tc) {
|
| - RTC_CHECK(!(ShouldSetupDtlsSrtp() && srtp_filter_.IsActive()))
|
| - << "Setting RTCP for DTLS/SRTP after SrtpFilter is active "
|
| - << "should never happen.";
|
| - ConnectToTransportChannel(new_tc);
|
| - for (const auto& pair : rtcp_socket_options_) {
|
| - new_tc->SetOption(pair.first, pair.second);
|
| - }
|
| - }
|
| -
|
| - if (update_writablity) {
|
| - // Update aggregate writable/ready-to-send state between RTP and RTCP upon
|
| - // setting new channel
|
| - UpdateWritableState_w();
|
| - SetReadyToSend(true, new_tc && new_tc->writable());
|
| - }
|
| -}
|
| -
|
| -void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
|
| - tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
|
| - tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);
|
| - tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState);
|
| -}
|
| -
|
| -void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - tc->SignalWritableState.disconnect(this);
|
| - tc->SignalReadPacket.disconnect(this);
|
| - tc->SignalReadyToSend.disconnect(this);
|
| - tc->SignalDtlsState.disconnect(this);
|
| -}
|
| -
|
| -bool BaseChannel::Enable(bool enable) {
|
| - worker_thread_->Invoke<void>(Bind(
|
| - enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
|
| - this));
|
| - return true;
|
| -}
|
| -
|
| -bool BaseChannel::AddRecvStream(const StreamParams& sp) {
|
| - return InvokeOnWorker(Bind(&BaseChannel::AddRecvStream_w, this, sp));
|
| -}
|
| -
|
| -bool BaseChannel::RemoveRecvStream(uint32_t ssrc) {
|
| - return InvokeOnWorker(Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
|
| -}
|
| -
|
| -bool BaseChannel::AddSendStream(const StreamParams& sp) {
|
| - return InvokeOnWorker(
|
| - Bind(&MediaChannel::AddSendStream, media_channel(), sp));
|
| -}
|
| -
|
| -bool BaseChannel::RemoveSendStream(uint32_t ssrc) {
|
| - return InvokeOnWorker(
|
| - Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
|
| -}
|
| -
|
| -bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
|
| - return InvokeOnWorker(Bind(&BaseChannel::SetLocalContent_w,
|
| - this, content, action, error_desc));
|
| -}
|
| -
|
| -bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
|
| - return InvokeOnWorker(Bind(&BaseChannel::SetRemoteContent_w,
|
| - this, content, action, error_desc));
|
| -}
|
| -
|
| -void BaseChannel::StartConnectionMonitor(int cms) {
|
| - // We pass in the BaseChannel instead of the transport_channel_
|
| - // because if the transport_channel_ changes, the ConnectionMonitor
|
| - // would be pointing to the wrong TransportChannel.
|
| - connection_monitor_.reset(new ConnectionMonitor(
|
| - this, worker_thread(), rtc::Thread::Current()));
|
| - connection_monitor_->SignalUpdate.connect(
|
| - this, &BaseChannel::OnConnectionMonitorUpdate);
|
| - connection_monitor_->Start(cms);
|
| -}
|
| -
|
| -void BaseChannel::StopConnectionMonitor() {
|
| - if (connection_monitor_) {
|
| - connection_monitor_->Stop();
|
| - connection_monitor_.reset();
|
| - }
|
| -}
|
| -
|
| -bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - return transport_channel_->GetStats(infos);
|
| -}
|
| -
|
| -bool BaseChannel::IsReadyToReceive() const {
|
| - // Receive data if we are enabled and have local content,
|
| - return enabled() && IsReceiveContentDirection(local_content_direction_);
|
| -}
|
| -
|
| -bool BaseChannel::IsReadyToSend() const {
|
| - // Send outgoing data if we are enabled, have local and remote content,
|
| - // and we have had some form of connectivity.
|
| - return enabled() && IsReceiveContentDirection(remote_content_direction_) &&
|
| - IsSendContentDirection(local_content_direction_) &&
|
| - was_ever_writable() &&
|
| - (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp());
|
| -}
|
| -
|
| -bool BaseChannel::SendPacket(rtc::Buffer* packet,
|
| - const rtc::PacketOptions& options) {
|
| - return SendPacket(false, packet, options);
|
| -}
|
| -
|
| -bool BaseChannel::SendRtcp(rtc::Buffer* packet,
|
| - const rtc::PacketOptions& options) {
|
| - return SendPacket(true, packet, options);
|
| -}
|
| -
|
| -int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
|
| - int value) {
|
| - TransportChannel* channel = NULL;
|
| - switch (type) {
|
| - case ST_RTP:
|
| - channel = transport_channel_;
|
| - socket_options_.push_back(
|
| - std::pair<rtc::Socket::Option, int>(opt, value));
|
| - break;
|
| - case ST_RTCP:
|
| - channel = rtcp_transport_channel_;
|
| - rtcp_socket_options_.push_back(
|
| - std::pair<rtc::Socket::Option, int>(opt, value));
|
| - break;
|
| - }
|
| - return channel ? channel->SetOption(opt, value) : -1;
|
| -}
|
| -
|
| -void BaseChannel::OnWritableState(TransportChannel* channel) {
|
| - ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
|
| - UpdateWritableState_w();
|
| -}
|
| -
|
| -void BaseChannel::OnChannelRead(TransportChannel* channel,
|
| - const char* data, size_t len,
|
| - const rtc::PacketTime& packet_time,
|
| - int flags) {
|
| - TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead");
|
| - // OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| -
|
| - // When using RTCP multiplexing we might get RTCP packets on the RTP
|
| - // transport. We feed RTP traffic into the demuxer to determine if it is RTCP.
|
| - bool rtcp = PacketIsRtcp(channel, data, len);
|
| - rtc::Buffer packet(data, len);
|
| - HandlePacket(rtcp, &packet, packet_time);
|
| -}
|
| -
|
| -void BaseChannel::OnReadyToSend(TransportChannel* channel) {
|
| - ASSERT(channel == transport_channel_ || channel == rtcp_transport_channel_);
|
| - SetReadyToSend(channel == rtcp_transport_channel_, true);
|
| -}
|
| -
|
| -void BaseChannel::OnDtlsState(TransportChannel* channel,
|
| - DtlsTransportState state) {
|
| - if (!ShouldSetupDtlsSrtp()) {
|
| - return;
|
| - }
|
| -
|
| - // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED
|
| - // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to
|
| - // cover other scenarios like the whole channel is writable (not just this
|
| - // TransportChannel) or when TransportChannel is attached after DTLS is
|
| - // negotiated.
|
| - if (state != DTLS_TRANSPORT_CONNECTED) {
|
| - srtp_filter_.ResetParams();
|
| - }
|
| -}
|
| -
|
| -void BaseChannel::SetReadyToSend(bool rtcp, bool ready) {
|
| - if (rtcp) {
|
| - rtcp_ready_to_send_ = ready;
|
| - } else {
|
| - rtp_ready_to_send_ = ready;
|
| - }
|
| -
|
| - 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);
|
| - } else {
|
| - // Notify the MediaChannel when either rtp or rtcp channel can't send.
|
| - media_channel_->OnReadyToSend(false);
|
| - }
|
| -}
|
| -
|
| -bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
|
| - const char* data, size_t len) {
|
| - return (channel == rtcp_transport_channel_ ||
|
| - rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
|
| -}
|
| -
|
| -bool BaseChannel::SendPacket(bool rtcp,
|
| - rtc::Buffer* packet,
|
| - const rtc::PacketOptions& options) {
|
| - // SendPacket gets called from MediaEngine, typically on an encoder thread.
|
| - // If the thread is not our worker thread, we will post to our worker
|
| - // so that the real work happens on our worker. This avoids us having to
|
| - // synchronize access to all the pieces of the send path, including
|
| - // SRTP and the inner workings of the transport channels.
|
| - // The only downside is that we can't return a proper failure code if
|
| - // needed. Since UDP is unreliable anyway, this should be a non-issue.
|
| - if (rtc::Thread::Current() != worker_thread_) {
|
| - // Avoid a copy by transferring the ownership of the packet data.
|
| - int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
|
| - PacketMessageData* data = new PacketMessageData;
|
| - data->packet = std::move(*packet);
|
| - data->options = options;
|
| - worker_thread_->Post(this, message_id, data);
|
| - return true;
|
| - }
|
| -
|
| - // Now that we are on the correct thread, ensure we have a place to send this
|
| - // packet before doing anything. (We might get RTCP packets that we don't
|
| - // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
|
| - // transport.
|
| - TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
|
| - transport_channel_ : rtcp_transport_channel_;
|
| - if (!channel || !channel->writable()) {
|
| - return false;
|
| - }
|
| -
|
| - // Protect ourselves against crazy data.
|
| - if (!ValidPacket(rtcp, packet)) {
|
| - LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
|
| - << PacketType(rtcp)
|
| - << " packet: wrong size=" << packet->size();
|
| - return false;
|
| - }
|
| -
|
| - rtc::PacketOptions updated_options;
|
| - updated_options = options;
|
| - // Protect if needed.
|
| - if (srtp_filter_.IsActive()) {
|
| - bool res;
|
| - uint8_t* data = packet->data();
|
| - int len = static_cast<int>(packet->size());
|
| - if (!rtcp) {
|
| - // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
|
| - // inside libsrtp for a RTP packet. A external HMAC module will be writing
|
| - // a fake HMAC value. This is ONLY done for a RTP packet.
|
| - // Socket layer will update rtp sendtime extension header if present in
|
| - // packet with current time before updating the HMAC.
|
| -#if !defined(ENABLE_EXTERNAL_AUTH)
|
| - res = srtp_filter_.ProtectRtp(
|
| - data, len, static_cast<int>(packet->capacity()), &len);
|
| -#else
|
| - updated_options.packet_time_params.rtp_sendtime_extension_id =
|
| - rtp_abs_sendtime_extn_id_;
|
| - res = srtp_filter_.ProtectRtp(
|
| - data, len, static_cast<int>(packet->capacity()), &len,
|
| - &updated_options.packet_time_params.srtp_packet_index);
|
| - // If protection succeeds, let's get auth params from srtp.
|
| - if (res) {
|
| - uint8_t* auth_key = NULL;
|
| - int key_len;
|
| - res = srtp_filter_.GetRtpAuthParams(
|
| - &auth_key, &key_len,
|
| - &updated_options.packet_time_params.srtp_auth_tag_len);
|
| - if (res) {
|
| - updated_options.packet_time_params.srtp_auth_key.resize(key_len);
|
| - updated_options.packet_time_params.srtp_auth_key.assign(
|
| - auth_key, auth_key + key_len);
|
| - }
|
| - }
|
| -#endif
|
| - if (!res) {
|
| - int seq_num = -1;
|
| - uint32_t ssrc = 0;
|
| - GetRtpSeqNum(data, len, &seq_num);
|
| - GetRtpSsrc(data, len, &ssrc);
|
| - LOG(LS_ERROR) << "Failed to protect " << content_name_
|
| - << " RTP packet: size=" << len
|
| - << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
|
| - return false;
|
| - }
|
| - } else {
|
| - res = srtp_filter_.ProtectRtcp(data, len,
|
| - static_cast<int>(packet->capacity()),
|
| - &len);
|
| - if (!res) {
|
| - int type = -1;
|
| - GetRtcpType(data, len, &type);
|
| - LOG(LS_ERROR) << "Failed to protect " << content_name_
|
| - << " RTCP packet: size=" << len << ", type=" << type;
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - // Update the length of the packet now that we've added the auth tag.
|
| - packet->SetSize(len);
|
| - } else if (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;
|
| - }
|
| -
|
| - // Bon voyage.
|
| - int ret =
|
| - channel->SendPacket(packet->data<char>(), packet->size(), updated_options,
|
| - (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
|
| - if (ret != static_cast<int>(packet->size())) {
|
| - if (channel->GetError() == EWOULDBLOCK) {
|
| - LOG(LS_WARNING) << "Got EWOULDBLOCK from socket.";
|
| - SetReadyToSend(rtcp, false);
|
| - }
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool BaseChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
|
| - // Protect ourselves against crazy data.
|
| - if (!ValidPacket(rtcp, packet)) {
|
| - LOG(LS_ERROR) << "Dropping incoming " << content_name_ << " "
|
| - << PacketType(rtcp)
|
| - << " packet: wrong size=" << packet->size();
|
| - return false;
|
| - }
|
| - if (rtcp) {
|
| - // Permit all (seemingly valid) RTCP packets.
|
| - return true;
|
| - }
|
| - // Check whether we handle this payload.
|
| - return bundle_filter_.DemuxPacket(packet->data<uint8_t>(), packet->size());
|
| -}
|
| -
|
| -void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
|
| - const rtc::PacketTime& packet_time) {
|
| - if (!WantsPacket(rtcp, packet)) {
|
| - return;
|
| - }
|
| -
|
| - // We are only interested in the first rtp packet because that
|
| - // indicates the media has started flowing.
|
| - if (!has_received_packet_ && !rtcp) {
|
| - has_received_packet_ = true;
|
| - signaling_thread()->Post(this, MSG_FIRSTPACKETRECEIVED);
|
| - }
|
| -
|
| - // Unprotect the packet, if needed.
|
| - if (srtp_filter_.IsActive()) {
|
| - char* data = packet->data<char>();
|
| - int len = static_cast<int>(packet->size());
|
| - bool res;
|
| - if (!rtcp) {
|
| - res = srtp_filter_.UnprotectRtp(data, len, &len);
|
| - if (!res) {
|
| - int seq_num = -1;
|
| - uint32_t ssrc = 0;
|
| - GetRtpSeqNum(data, len, &seq_num);
|
| - GetRtpSsrc(data, len, &ssrc);
|
| - LOG(LS_ERROR) << "Failed to unprotect " << content_name_
|
| - << " RTP packet: size=" << len
|
| - << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
|
| - return;
|
| - }
|
| - } else {
|
| - res = srtp_filter_.UnprotectRtcp(data, len, &len);
|
| - if (!res) {
|
| - int type = -1;
|
| - GetRtcpType(data, len, &type);
|
| - LOG(LS_ERROR) << "Failed to unprotect " << content_name_
|
| - << " RTCP packet: size=" << len << ", type=" << type;
|
| - return;
|
| - }
|
| - }
|
| -
|
| - packet->SetSize(len);
|
| - } else if (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;
|
| - }
|
| -
|
| - // Push it down to the media channel.
|
| - if (!rtcp) {
|
| - media_channel_->OnPacketReceived(packet, packet_time);
|
| - } else {
|
| - media_channel_->OnRtcpReceived(packet, packet_time);
|
| - }
|
| -}
|
| -
|
| -bool BaseChannel::PushdownLocalDescription(
|
| - const SessionDescription* local_desc, ContentAction action,
|
| - std::string* error_desc) {
|
| - const ContentInfo* content_info = GetFirstContent(local_desc);
|
| - const MediaContentDescription* content_desc =
|
| - GetContentDescription(content_info);
|
| - if (content_desc && content_info && !content_info->rejected &&
|
| - !SetLocalContent(content_desc, action, error_desc)) {
|
| - LOG(LS_ERROR) << "Failure in SetLocalContent with action " << action;
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool BaseChannel::PushdownRemoteDescription(
|
| - const SessionDescription* remote_desc, ContentAction action,
|
| - std::string* error_desc) {
|
| - const ContentInfo* content_info = GetFirstContent(remote_desc);
|
| - const MediaContentDescription* content_desc =
|
| - GetContentDescription(content_info);
|
| - if (content_desc && content_info && !content_info->rejected &&
|
| - !SetRemoteContent(content_desc, action, error_desc)) {
|
| - LOG(LS_ERROR) << "Failure in SetRemoteContent with action " << action;
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -void BaseChannel::EnableMedia_w() {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - if (enabled_)
|
| - return;
|
| -
|
| - LOG(LS_INFO) << "Channel enabled";
|
| - enabled_ = true;
|
| - ChangeState();
|
| -}
|
| -
|
| -void BaseChannel::DisableMedia_w() {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - if (!enabled_)
|
| - return;
|
| -
|
| - LOG(LS_INFO) << "Channel disabled";
|
| - enabled_ = false;
|
| - ChangeState();
|
| -}
|
| -
|
| -void BaseChannel::UpdateWritableState_w() {
|
| - if (transport_channel_ && transport_channel_->writable() &&
|
| - (!rtcp_transport_channel_ || rtcp_transport_channel_->writable())) {
|
| - ChannelWritable_w();
|
| - } else {
|
| - ChannelNotWritable_w();
|
| - }
|
| -}
|
| -
|
| -void BaseChannel::ChannelWritable_w() {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - if (writable_) {
|
| - return;
|
| - }
|
| -
|
| - LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
|
| - << (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;
|
| - }
|
| - }
|
| -
|
| - was_ever_writable_ = true;
|
| - MaybeSetupDtlsSrtp_w();
|
| - writable_ = true;
|
| - ChangeState();
|
| -}
|
| -
|
| -void BaseChannel::SignalDtlsSetupFailure_w(bool rtcp) {
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - signaling_thread()->Invoke<void>(Bind(
|
| - &BaseChannel::SignalDtlsSetupFailure_s, this, rtcp));
|
| -}
|
| -
|
| -void BaseChannel::SignalDtlsSetupFailure_s(bool rtcp) {
|
| - ASSERT(signaling_thread() == rtc::Thread::Current());
|
| - SignalDtlsSetupFailure(this, rtcp);
|
| -}
|
| -
|
| -bool BaseChannel::SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp) {
|
| - std::vector<int> crypto_suites;
|
| - // We always use the default SRTP crypto suites for RTCP, but we may use
|
| - // different crypto suites for RTP depending on the media type.
|
| - if (!rtcp) {
|
| - GetSrtpCryptoSuites(&crypto_suites);
|
| - } else {
|
| - GetDefaultSrtpCryptoSuites(&crypto_suites);
|
| - }
|
| - return tc->SetSrtpCryptoSuites(crypto_suites);
|
| -}
|
| -
|
| -bool BaseChannel::ShouldSetupDtlsSrtp() const {
|
| - // Since DTLS is applied to all channels, checking RTP should be enough.
|
| - return transport_channel_ && transport_channel_->IsDtlsActive();
|
| -}
|
| -
|
| -// 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_;
|
| -
|
| - RTC_DCHECK(channel->IsDtlsActive());
|
| -
|
| - int selected_crypto_suite;
|
| -
|
| - if (!channel->GetSrtpCryptoSuite(&selected_crypto_suite)) {
|
| - LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
|
| - return false;
|
| - }
|
| -
|
| - LOG(LS_INFO) << "Installing keys from DTLS-SRTP on "
|
| - << content_name() << " "
|
| - << PacketType(rtcp_channel);
|
| -
|
| - // OK, we're now doing DTLS (RFC 5764)
|
| - std::vector<unsigned char> dtls_buffer(SRTP_MASTER_KEY_KEY_LEN * 2 +
|
| - SRTP_MASTER_KEY_SALT_LEN * 2);
|
| -
|
| - // RFC 5705 exporter using the RFC 5764 parameters
|
| - if (!channel->ExportKeyingMaterial(
|
| - kDtlsSrtpExporterLabel,
|
| - NULL, 0, false,
|
| - &dtls_buffer[0], dtls_buffer.size())) {
|
| - LOG(LS_WARNING) << "DTLS-SRTP key export failed";
|
| - ASSERT(false); // This should never happen
|
| - return false;
|
| - }
|
| -
|
| - // Sync up the keys with the DTLS-SRTP interface
|
| - std::vector<unsigned char> client_write_key(SRTP_MASTER_KEY_KEY_LEN +
|
| - SRTP_MASTER_KEY_SALT_LEN);
|
| - std::vector<unsigned char> server_write_key(SRTP_MASTER_KEY_KEY_LEN +
|
| - SRTP_MASTER_KEY_SALT_LEN);
|
| - size_t offset = 0;
|
| - memcpy(&client_write_key[0], &dtls_buffer[offset],
|
| - SRTP_MASTER_KEY_KEY_LEN);
|
| - offset += SRTP_MASTER_KEY_KEY_LEN;
|
| - memcpy(&server_write_key[0], &dtls_buffer[offset],
|
| - SRTP_MASTER_KEY_KEY_LEN);
|
| - offset += SRTP_MASTER_KEY_KEY_LEN;
|
| - memcpy(&client_write_key[SRTP_MASTER_KEY_KEY_LEN],
|
| - &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
|
| - offset += SRTP_MASTER_KEY_SALT_LEN;
|
| - memcpy(&server_write_key[SRTP_MASTER_KEY_KEY_LEN],
|
| - &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
|
| -
|
| - std::vector<unsigned char> *send_key, *recv_key;
|
| - rtc::SSLRole role;
|
| - if (!channel->GetSslRole(&role)) {
|
| - LOG(LS_WARNING) << "GetSslRole failed";
|
| - return false;
|
| - }
|
| -
|
| - if (role == rtc::SSL_SERVER) {
|
| - send_key = &server_write_key;
|
| - recv_key = &client_write_key;
|
| - } else {
|
| - send_key = &client_write_key;
|
| - recv_key = &server_write_key;
|
| - }
|
| -
|
| - if (rtcp_channel) {
|
| - ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0],
|
| - static_cast<int>(send_key->size()),
|
| - selected_crypto_suite, &(*recv_key)[0],
|
| - static_cast<int>(recv_key->size()));
|
| - } else {
|
| - ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0],
|
| - static_cast<int>(send_key->size()),
|
| - selected_crypto_suite, &(*recv_key)[0],
|
| - static_cast<int>(recv_key->size()));
|
| - }
|
| -
|
| - if (!ret)
|
| - LOG(LS_WARNING) << "DTLS-SRTP key installation failed";
|
| - else
|
| - dtls_keyed_ = true;
|
| -
|
| - return ret;
|
| -}
|
| -
|
| -void BaseChannel::MaybeSetupDtlsSrtp_w() {
|
| - if (srtp_filter_.IsActive()) {
|
| - return;
|
| - }
|
| -
|
| - if (!ShouldSetupDtlsSrtp()) {
|
| - return;
|
| - }
|
| -
|
| - if (!SetupDtlsSrtp(false)) {
|
| - SignalDtlsSetupFailure_w(false);
|
| - return;
|
| - }
|
| -
|
| - if (rtcp_transport_channel_) {
|
| - if (!SetupDtlsSrtp(true)) {
|
| - SignalDtlsSetupFailure_w(true);
|
| - return;
|
| - }
|
| - }
|
| -}
|
| -
|
| -void BaseChannel::ChannelNotWritable_w() {
|
| - ASSERT(worker_thread_ == rtc::Thread::Current());
|
| - if (!writable_)
|
| - return;
|
| -
|
| - LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
|
| - writable_ = false;
|
| - ChangeState();
|
| -}
|
| -
|
| -bool BaseChannel::SetRtpTransportParameters_w(
|
| - const MediaContentDescription* content,
|
| - ContentAction action,
|
| - ContentSource src,
|
| - std::string* error_desc) {
|
| - if (action == CA_UPDATE) {
|
| - // These parameters never get changed by a CA_UDPATE.
|
| - return true;
|
| - }
|
| -
|
| - // Cache secure_required_ for belt and suspenders check on SendPacket
|
| - if (src == CS_LOCAL) {
|
| - set_secure_required(content->crypto_required() != CT_NONE);
|
| - }
|
| -
|
| - if (!SetSrtp_w(content->cryptos(), action, src, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - if (!SetRtcpMux_w(content->rtcp_mux(), action, src, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -// |dtls| will be set to true if DTLS is active for transport channel and
|
| -// crypto is empty.
|
| -bool BaseChannel::CheckSrtpConfig(const std::vector<CryptoParams>& cryptos,
|
| - bool* dtls,
|
| - std::string* error_desc) {
|
| - *dtls = transport_channel_->IsDtlsActive();
|
| - if (*dtls && !cryptos.empty()) {
|
| - SafeSetError("Cryptos must be empty when DTLS is active.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool BaseChannel::SetSrtp_w(const std::vector<CryptoParams>& cryptos,
|
| - ContentAction action,
|
| - ContentSource src,
|
| - std::string* error_desc) {
|
| - if (action == CA_UPDATE) {
|
| - // no crypto params.
|
| - return true;
|
| - }
|
| - bool ret = false;
|
| - bool dtls = false;
|
| - ret = CheckSrtpConfig(cryptos, &dtls, error_desc);
|
| - if (!ret) {
|
| - return false;
|
| - }
|
| - switch (action) {
|
| - case CA_OFFER:
|
| - // If DTLS is already active on the channel, we could be renegotiating
|
| - // here. We don't update the srtp filter.
|
| - if (!dtls) {
|
| - ret = srtp_filter_.SetOffer(cryptos, src);
|
| - }
|
| - break;
|
| - case CA_PRANSWER:
|
| - // If we're doing DTLS-SRTP, we don't want to update the filter
|
| - // with an answer, because we already have SRTP parameters.
|
| - if (!dtls) {
|
| - ret = srtp_filter_.SetProvisionalAnswer(cryptos, src);
|
| - }
|
| - break;
|
| - case CA_ANSWER:
|
| - // If we're doing DTLS-SRTP, we don't want to update the filter
|
| - // with an answer, because we already have SRTP parameters.
|
| - if (!dtls) {
|
| - ret = srtp_filter_.SetAnswer(cryptos, src);
|
| - }
|
| - break;
|
| - default:
|
| - break;
|
| - }
|
| - if (!ret) {
|
| - SafeSetError("Failed to setup SRTP filter.", error_desc);
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -void BaseChannel::ActivateRtcpMux() {
|
| - worker_thread_->Invoke<void>(Bind(
|
| - &BaseChannel::ActivateRtcpMux_w, this));
|
| -}
|
| -
|
| -void BaseChannel::ActivateRtcpMux_w() {
|
| - if (!rtcp_mux_filter_.IsActive()) {
|
| - rtcp_mux_filter_.SetActive();
|
| - set_rtcp_transport_channel(nullptr, true);
|
| - rtcp_transport_enabled_ = false;
|
| - }
|
| -}
|
| -
|
| -bool BaseChannel::SetRtcpMux_w(bool enable, ContentAction action,
|
| - ContentSource src,
|
| - std::string* error_desc) {
|
| - bool ret = false;
|
| - switch (action) {
|
| - case CA_OFFER:
|
| - ret = rtcp_mux_filter_.SetOffer(enable, src);
|
| - break;
|
| - case CA_PRANSWER:
|
| - ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
|
| - break;
|
| - case CA_ANSWER:
|
| - ret = rtcp_mux_filter_.SetAnswer(enable, src);
|
| - if (ret && rtcp_mux_filter_.IsActive()) {
|
| - // We activated RTCP mux, close down the RTCP transport.
|
| - LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
|
| - << " by destroying RTCP transport channel for "
|
| - << transport_name();
|
| - set_rtcp_transport_channel(nullptr, true);
|
| - rtcp_transport_enabled_ = false;
|
| - }
|
| - break;
|
| - case CA_UPDATE:
|
| - // No RTCP mux info.
|
| - ret = true;
|
| - break;
|
| - default:
|
| - break;
|
| - }
|
| - if (!ret) {
|
| - SafeSetError("Failed to setup RTCP mux filter.", error_desc);
|
| - return false;
|
| - }
|
| - // |rtcp_mux_filter_| can be active if |action| is CA_PRANSWER or
|
| - // CA_ANSWER, but we only want to tear down the RTCP transport channel if we
|
| - // received a final answer.
|
| - if (rtcp_mux_filter_.IsActive()) {
|
| - // If the RTP transport is already writable, then so are we.
|
| - if (transport_channel_->writable()) {
|
| - ChannelWritable_w();
|
| - }
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - return media_channel()->AddRecvStream(sp);
|
| -}
|
| -
|
| -bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - return media_channel()->RemoveRecvStream(ssrc);
|
| -}
|
| -
|
| -bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
|
| - action == CA_PRANSWER || action == CA_UPDATE))
|
| - return false;
|
| -
|
| - // If this is an update, streams only contain streams that have changed.
|
| - if (action == CA_UPDATE) {
|
| - for (StreamParamsVec::const_iterator it = streams.begin();
|
| - it != streams.end(); ++it) {
|
| - const StreamParams* existing_stream =
|
| - GetStreamByIds(local_streams_, it->groupid, it->id);
|
| - if (!existing_stream && it->has_ssrcs()) {
|
| - if (media_channel()->AddSendStream(*it)) {
|
| - local_streams_.push_back(*it);
|
| - LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc();
|
| - } else {
|
| - std::ostringstream desc;
|
| - desc << "Failed to add send stream ssrc: " << it->first_ssrc();
|
| - SafeSetError(desc.str(), error_desc);
|
| - return false;
|
| - }
|
| - } else if (existing_stream && !it->has_ssrcs()) {
|
| - if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) {
|
| - std::ostringstream desc;
|
| - desc << "Failed to remove send stream with ssrc "
|
| - << it->first_ssrc() << ".";
|
| - SafeSetError(desc.str(), error_desc);
|
| - return false;
|
| - }
|
| - RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc());
|
| - } else {
|
| - LOG(LS_WARNING) << "Ignore unsupported stream update";
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| - // Else streams are all the streams we want to send.
|
| -
|
| - // Check for streams that have been removed.
|
| - bool ret = true;
|
| - for (StreamParamsVec::const_iterator it = local_streams_.begin();
|
| - it != local_streams_.end(); ++it) {
|
| - if (!GetStreamBySsrc(streams, it->first_ssrc())) {
|
| - if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
|
| - std::ostringstream desc;
|
| - desc << "Failed to remove send stream with ssrc "
|
| - << it->first_ssrc() << ".";
|
| - SafeSetError(desc.str(), error_desc);
|
| - ret = false;
|
| - }
|
| - }
|
| - }
|
| - // Check for new streams.
|
| - for (StreamParamsVec::const_iterator it = streams.begin();
|
| - it != streams.end(); ++it) {
|
| - if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
|
| - if (media_channel()->AddSendStream(*it)) {
|
| - LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
|
| - } else {
|
| - std::ostringstream desc;
|
| - desc << "Failed to add send stream ssrc: " << it->first_ssrc();
|
| - SafeSetError(desc.str(), error_desc);
|
| - ret = false;
|
| - }
|
| - }
|
| - }
|
| - local_streams_ = streams;
|
| - return ret;
|
| -}
|
| -
|
| -bool BaseChannel::UpdateRemoteStreams_w(
|
| - const std::vector<StreamParams>& streams,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
|
| - action == CA_PRANSWER || action == CA_UPDATE))
|
| - return false;
|
| -
|
| - // If this is an update, streams only contain streams that have changed.
|
| - if (action == CA_UPDATE) {
|
| - for (StreamParamsVec::const_iterator it = streams.begin();
|
| - it != streams.end(); ++it) {
|
| - const StreamParams* existing_stream =
|
| - GetStreamByIds(remote_streams_, it->groupid, it->id);
|
| - if (!existing_stream && it->has_ssrcs()) {
|
| - if (AddRecvStream_w(*it)) {
|
| - remote_streams_.push_back(*it);
|
| - LOG(LS_INFO) << "Add remote stream ssrc: " << it->first_ssrc();
|
| - } else {
|
| - std::ostringstream desc;
|
| - desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
|
| - SafeSetError(desc.str(), error_desc);
|
| - return false;
|
| - }
|
| - } else if (existing_stream && !it->has_ssrcs()) {
|
| - if (!RemoveRecvStream_w(existing_stream->first_ssrc())) {
|
| - std::ostringstream desc;
|
| - desc << "Failed to remove remote stream with ssrc "
|
| - << it->first_ssrc() << ".";
|
| - SafeSetError(desc.str(), error_desc);
|
| - return false;
|
| - }
|
| - RemoveStreamBySsrc(&remote_streams_, existing_stream->first_ssrc());
|
| - } else {
|
| - LOG(LS_WARNING) << "Ignore unsupported stream update."
|
| - << " Stream exists? " << (existing_stream != nullptr)
|
| - << " new stream = " << it->ToString();
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| - // Else streams are all the streams we want to receive.
|
| -
|
| - // Check for streams that have been removed.
|
| - bool ret = true;
|
| - for (StreamParamsVec::const_iterator it = remote_streams_.begin();
|
| - it != remote_streams_.end(); ++it) {
|
| - if (!GetStreamBySsrc(streams, it->first_ssrc())) {
|
| - if (!RemoveRecvStream_w(it->first_ssrc())) {
|
| - std::ostringstream desc;
|
| - desc << "Failed to remove remote stream with ssrc "
|
| - << it->first_ssrc() << ".";
|
| - SafeSetError(desc.str(), error_desc);
|
| - ret = false;
|
| - }
|
| - }
|
| - }
|
| - // Check for new streams.
|
| - for (StreamParamsVec::const_iterator it = streams.begin();
|
| - it != streams.end(); ++it) {
|
| - if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) {
|
| - if (AddRecvStream_w(*it)) {
|
| - LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
|
| - } else {
|
| - std::ostringstream desc;
|
| - desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
|
| - SafeSetError(desc.str(), error_desc);
|
| - ret = false;
|
| - }
|
| - }
|
| - }
|
| - remote_streams_ = streams;
|
| - return ret;
|
| -}
|
| -
|
| -void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension(
|
| - const std::vector<RtpHeaderExtension>& extensions) {
|
| - const RtpHeaderExtension* send_time_extension =
|
| - FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
|
| - rtp_abs_sendtime_extn_id_ =
|
| - send_time_extension ? send_time_extension->id : -1;
|
| -}
|
| -
|
| -void BaseChannel::OnMessage(rtc::Message *pmsg) {
|
| - TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
|
| - switch (pmsg->message_id) {
|
| - case MSG_RTPPACKET:
|
| - case MSG_RTCPPACKET: {
|
| - PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
|
| - SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
|
| - data->options);
|
| - delete data; // because it is Posted
|
| - break;
|
| - }
|
| - case MSG_FIRSTPACKETRECEIVED: {
|
| - SignalFirstPacketReceived(this);
|
| - break;
|
| - }
|
| - }
|
| -}
|
| -
|
| -void BaseChannel::FlushRtcpMessages() {
|
| - // Flush all remaining RTCP messages. This should only be called in
|
| - // destructor.
|
| - ASSERT(rtc::Thread::Current() == worker_thread_);
|
| - rtc::MessageList rtcp_messages;
|
| - worker_thread_->Clear(this, MSG_RTCPPACKET, &rtcp_messages);
|
| - for (rtc::MessageList::iterator it = rtcp_messages.begin();
|
| - it != rtcp_messages.end(); ++it) {
|
| - worker_thread_->Send(this, MSG_RTCPPACKET, it->pdata);
|
| - }
|
| -}
|
| -
|
| -VoiceChannel::VoiceChannel(rtc::Thread* thread,
|
| - MediaEngineInterface* media_engine,
|
| - VoiceMediaChannel* media_channel,
|
| - TransportController* transport_controller,
|
| - const std::string& content_name,
|
| - bool rtcp)
|
| - : BaseChannel(thread,
|
| - media_channel,
|
| - transport_controller,
|
| - content_name,
|
| - rtcp),
|
| - media_engine_(media_engine),
|
| - received_media_(false) {}
|
| -
|
| -VoiceChannel::~VoiceChannel() {
|
| - StopAudioMonitor();
|
| - StopMediaMonitor();
|
| - // this can't be done in the base class, since it calls a virtual
|
| - DisableMedia_w();
|
| - Deinit();
|
| -}
|
| -
|
| -bool VoiceChannel::Init() {
|
| - if (!BaseChannel::Init()) {
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool VoiceChannel::SetAudioSend(uint32_t ssrc,
|
| - bool enable,
|
| - const AudioOptions* options,
|
| - AudioRenderer* renderer) {
|
| - return InvokeOnWorker(Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
|
| - ssrc, enable, options, renderer));
|
| -}
|
| -
|
| -// TODO(juberti): Handle early media the right way. We should get an explicit
|
| -// ringing message telling us to start playing local ringback, which we cancel
|
| -// if any early media actually arrives. For now, we do the opposite, which is
|
| -// to wait 1 second for early media, and start playing local ringback if none
|
| -// arrives.
|
| -void VoiceChannel::SetEarlyMedia(bool enable) {
|
| - if (enable) {
|
| - // Start the early media timeout
|
| - worker_thread()->PostDelayed(kEarlyMediaTimeout, this,
|
| - MSG_EARLYMEDIATIMEOUT);
|
| - } else {
|
| - // Stop the timeout if currently going.
|
| - worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
|
| - }
|
| -}
|
| -
|
| -bool VoiceChannel::CanInsertDtmf() {
|
| - return InvokeOnWorker(Bind(&VoiceMediaChannel::CanInsertDtmf,
|
| - media_channel()));
|
| -}
|
| -
|
| -bool VoiceChannel::InsertDtmf(uint32_t ssrc,
|
| - int event_code,
|
| - int duration) {
|
| - return InvokeOnWorker(Bind(&VoiceChannel::InsertDtmf_w, this,
|
| - ssrc, event_code, duration));
|
| -}
|
| -
|
| -bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
|
| - return InvokeOnWorker(Bind(&VoiceMediaChannel::SetOutputVolume,
|
| - media_channel(), ssrc, volume));
|
| -}
|
| -
|
| -void VoiceChannel::SetRawAudioSink(
|
| - uint32_t ssrc,
|
| - rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
|
| - // We need to work around Bind's lack of support for scoped_ptr and ownership
|
| - // passing. So we invoke to our own little routine that gets a pointer to
|
| - // our local variable. This is OK since we're synchronously invoking.
|
| - InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
|
| -}
|
| -
|
| -bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
|
| - return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
|
| - media_channel(), stats));
|
| -}
|
| -
|
| -void VoiceChannel::StartMediaMonitor(int cms) {
|
| - media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
|
| - rtc::Thread::Current()));
|
| - media_monitor_->SignalUpdate.connect(
|
| - this, &VoiceChannel::OnMediaMonitorUpdate);
|
| - media_monitor_->Start(cms);
|
| -}
|
| -
|
| -void VoiceChannel::StopMediaMonitor() {
|
| - if (media_monitor_) {
|
| - media_monitor_->Stop();
|
| - media_monitor_->SignalUpdate.disconnect(this);
|
| - media_monitor_.reset();
|
| - }
|
| -}
|
| -
|
| -void VoiceChannel::StartAudioMonitor(int cms) {
|
| - audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
|
| - audio_monitor_
|
| - ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
|
| - audio_monitor_->Start(cms);
|
| -}
|
| -
|
| -void VoiceChannel::StopAudioMonitor() {
|
| - if (audio_monitor_) {
|
| - audio_monitor_->Stop();
|
| - audio_monitor_.reset();
|
| - }
|
| -}
|
| -
|
| -bool VoiceChannel::IsAudioMonitorRunning() const {
|
| - return (audio_monitor_.get() != NULL);
|
| -}
|
| -
|
| -int VoiceChannel::GetInputLevel_w() {
|
| - return media_engine_->GetInputLevel();
|
| -}
|
| -
|
| -int VoiceChannel::GetOutputLevel_w() {
|
| - return media_channel()->GetOutputLevel();
|
| -}
|
| -
|
| -void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
|
| - media_channel()->GetActiveStreams(actives);
|
| -}
|
| -
|
| -void VoiceChannel::OnChannelRead(TransportChannel* channel,
|
| - const char* data, size_t len,
|
| - const rtc::PacketTime& packet_time,
|
| - int flags) {
|
| - BaseChannel::OnChannelRead(channel, data, len, packet_time, flags);
|
| -
|
| - // Set a flag when we've received an RTP packet. If we're waiting for early
|
| - // media, this will disable the timeout.
|
| - if (!received_media_ && !PacketIsRtcp(channel, data, len)) {
|
| - received_media_ = true;
|
| - }
|
| -}
|
| -
|
| -void VoiceChannel::ChangeState() {
|
| - // Render incoming data if we're the active call, and we have the local
|
| - // content. We receive data on the default channel and multiplexed streams.
|
| - bool recv = IsReadyToReceive();
|
| - media_channel()->SetPlayout(recv);
|
| -
|
| - // Send outgoing data if we're the active call, we have the remote content,
|
| - // and we have had some form of connectivity.
|
| - bool send = 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";
|
| - }
|
| -
|
| - LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
|
| -}
|
| -
|
| -const ContentInfo* VoiceChannel::GetFirstContent(
|
| - const SessionDescription* sdesc) {
|
| - return GetFirstAudioContent(sdesc);
|
| -}
|
| -
|
| -bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Setting local voice description";
|
| -
|
| - const AudioContentDescription* audio =
|
| - static_cast<const AudioContentDescription*>(content);
|
| - ASSERT(audio != NULL);
|
| - if (!audio) {
|
| - SafeSetError("Can't find audio content in local description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - AudioRecvParameters recv_params = last_recv_params_;
|
| - RtpParametersFromMediaDescription(audio, &recv_params);
|
| - if (!media_channel()->SetRecvParameters(recv_params)) {
|
| - SafeSetError("Failed to set local audio description recv parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - for (const AudioCodec& codec : audio->codecs()) {
|
| - bundle_filter()->AddPayloadType(codec.id);
|
| - }
|
| - last_recv_params_ = recv_params;
|
| -
|
| - // TODO(pthatcher): Move local streams into AudioSendParameters, and
|
| - // only give it to the media channel once we have a remote
|
| - // description too (without a remote description, we won't be able
|
| - // to send them anyway).
|
| - if (!UpdateLocalStreams_w(audio->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set local audio description streams.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - set_local_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Setting remote voice description";
|
| -
|
| - const AudioContentDescription* audio =
|
| - static_cast<const AudioContentDescription*>(content);
|
| - ASSERT(audio != NULL);
|
| - if (!audio) {
|
| - SafeSetError("Can't find audio content in remote description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - AudioSendParameters send_params = last_send_params_;
|
| - RtpSendParametersFromMediaDescription(audio, &send_params);
|
| - if (audio->agc_minus_10db()) {
|
| - send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
|
| - }
|
| - if (!media_channel()->SetSendParameters(send_params)) {
|
| - SafeSetError("Failed to set remote audio description send parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - last_send_params_ = send_params;
|
| -
|
| - // TODO(pthatcher): Move remote streams into AudioRecvParameters,
|
| - // and only give it to the media channel once we have a local
|
| - // description too (without a local description, we won't be able to
|
| - // recv them anyway).
|
| - if (!UpdateRemoteStreams_w(audio->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set remote audio description streams.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (audio->rtp_header_extensions_set()) {
|
| - MaybeCacheRtpAbsSendTimeHeaderExtension(audio->rtp_header_extensions());
|
| - }
|
| -
|
| - set_remote_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -void VoiceChannel::HandleEarlyMediaTimeout() {
|
| - // This occurs on the main thread, not the worker thread.
|
| - if (!received_media_) {
|
| - LOG(LS_INFO) << "No early media received before timeout";
|
| - SignalEarlyMediaTimeout(this);
|
| - }
|
| -}
|
| -
|
| -bool VoiceChannel::InsertDtmf_w(uint32_t ssrc,
|
| - int event,
|
| - int duration) {
|
| - if (!enabled()) {
|
| - return false;
|
| - }
|
| - return media_channel()->InsertDtmf(ssrc, event, duration);
|
| -}
|
| -
|
| -void VoiceChannel::OnMessage(rtc::Message *pmsg) {
|
| - switch (pmsg->message_id) {
|
| - case MSG_EARLYMEDIATIMEOUT:
|
| - HandleEarlyMediaTimeout();
|
| - break;
|
| - case MSG_CHANNEL_ERROR: {
|
| - VoiceChannelErrorMessageData* data =
|
| - static_cast<VoiceChannelErrorMessageData*>(pmsg->pdata);
|
| - delete data;
|
| - break;
|
| - }
|
| - default:
|
| - BaseChannel::OnMessage(pmsg);
|
| - break;
|
| - }
|
| -}
|
| -
|
| -void VoiceChannel::OnConnectionMonitorUpdate(
|
| - ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
|
| - SignalConnectionMonitor(this, infos);
|
| -}
|
| -
|
| -void VoiceChannel::OnMediaMonitorUpdate(
|
| - VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
|
| - ASSERT(media_channel == this->media_channel());
|
| - SignalMediaMonitor(this, info);
|
| -}
|
| -
|
| -void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
|
| - const AudioInfo& info) {
|
| - SignalAudioMonitor(this, info);
|
| -}
|
| -
|
| -void VoiceChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
|
| - GetSupportedAudioCryptoSuites(crypto_suites);
|
| -}
|
| -
|
| -VideoChannel::VideoChannel(rtc::Thread* thread,
|
| - VideoMediaChannel* media_channel,
|
| - TransportController* transport_controller,
|
| - const std::string& content_name,
|
| - bool rtcp)
|
| - : BaseChannel(thread,
|
| - media_channel,
|
| - transport_controller,
|
| - content_name,
|
| - rtcp),
|
| - previous_we_(rtc::WE_CLOSE) {}
|
| -
|
| -bool VideoChannel::Init() {
|
| - if (!BaseChannel::Init()) {
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -VideoChannel::~VideoChannel() {
|
| - std::vector<uint32_t> 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::SetSink(uint32_t ssrc,
|
| - rtc::VideoSinkInterface<VideoFrame>* sink) {
|
| - worker_thread()->Invoke<void>(
|
| - Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink));
|
| - return true;
|
| -}
|
| -
|
| -bool VideoChannel::AddScreencast(uint32_t ssrc, VideoCapturer* capturer) {
|
| - return worker_thread()->Invoke<bool>(Bind(
|
| - &VideoChannel::AddScreencast_w, this, ssrc, capturer));
|
| -}
|
| -
|
| -bool VideoChannel::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
|
| - return InvokeOnWorker(Bind(&VideoMediaChannel::SetCapturer,
|
| - media_channel(), ssrc, capturer));
|
| -}
|
| -
|
| -bool VideoChannel::RemoveScreencast(uint32_t ssrc) {
|
| - return InvokeOnWorker(Bind(&VideoChannel::RemoveScreencast_w, this, ssrc));
|
| -}
|
| -
|
| -bool VideoChannel::IsScreencasting() {
|
| - return InvokeOnWorker(Bind(&VideoChannel::IsScreencasting_w, this));
|
| -}
|
| -
|
| -bool VideoChannel::SetVideoSend(uint32_t ssrc,
|
| - bool mute,
|
| - const VideoOptions* options) {
|
| - return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
|
| - ssrc, mute, options));
|
| -}
|
| -
|
| -void VideoChannel::ChangeState() {
|
| - // 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, send=" << send;
|
| -}
|
| -
|
| -bool VideoChannel::GetStats(VideoMediaInfo* stats) {
|
| - return InvokeOnWorker(
|
| - Bind(&VideoMediaChannel::GetStats, media_channel(), stats));
|
| -}
|
| -
|
| -void VideoChannel::StartMediaMonitor(int cms) {
|
| - media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
|
| - rtc::Thread::Current()));
|
| - media_monitor_->SignalUpdate.connect(
|
| - this, &VideoChannel::OnMediaMonitorUpdate);
|
| - media_monitor_->Start(cms);
|
| -}
|
| -
|
| -void VideoChannel::StopMediaMonitor() {
|
| - if (media_monitor_) {
|
| - media_monitor_->Stop();
|
| - media_monitor_.reset();
|
| - }
|
| -}
|
| -
|
| -const ContentInfo* VideoChannel::GetFirstContent(
|
| - const SessionDescription* sdesc) {
|
| - return GetFirstVideoContent(sdesc);
|
| -}
|
| -
|
| -bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Setting local video description";
|
| -
|
| - const VideoContentDescription* video =
|
| - static_cast<const VideoContentDescription*>(content);
|
| - ASSERT(video != NULL);
|
| - if (!video) {
|
| - SafeSetError("Can't find video content in local description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - VideoRecvParameters recv_params = last_recv_params_;
|
| - RtpParametersFromMediaDescription(video, &recv_params);
|
| - if (!media_channel()->SetRecvParameters(recv_params)) {
|
| - SafeSetError("Failed to set local video description recv parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - for (const VideoCodec& codec : video->codecs()) {
|
| - bundle_filter()->AddPayloadType(codec.id);
|
| - }
|
| - last_recv_params_ = recv_params;
|
| -
|
| - // TODO(pthatcher): Move local streams into VideoSendParameters, and
|
| - // only give it to the media channel once we have a remote
|
| - // description too (without a remote description, we won't be able
|
| - // to send them anyway).
|
| - if (!UpdateLocalStreams_w(video->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set local video description streams.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - set_local_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Setting remote video description";
|
| -
|
| - const VideoContentDescription* video =
|
| - static_cast<const VideoContentDescription*>(content);
|
| - ASSERT(video != NULL);
|
| - if (!video) {
|
| - SafeSetError("Can't find video content in remote description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| -
|
| - if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - VideoSendParameters send_params = last_send_params_;
|
| - RtpSendParametersFromMediaDescription(video, &send_params);
|
| - if (video->conference_mode()) {
|
| - send_params.options.conference_mode = rtc::Optional<bool>(true);
|
| - }
|
| - if (!media_channel()->SetSendParameters(send_params)) {
|
| - SafeSetError("Failed to set remote video description send parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - last_send_params_ = send_params;
|
| -
|
| - // TODO(pthatcher): Move remote streams into VideoRecvParameters,
|
| - // and only give it to the media channel once we have a local
|
| - // description too (without a local description, we won't be able to
|
| - // recv them anyway).
|
| - if (!UpdateRemoteStreams_w(video->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set remote video description streams.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (video->rtp_header_extensions_set()) {
|
| - MaybeCacheRtpAbsSendTimeHeaderExtension(video->rtp_header_extensions());
|
| - }
|
| -
|
| - set_remote_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -bool VideoChannel::AddScreencast_w(uint32_t ssrc, VideoCapturer* capturer) {
|
| - if (screencast_capturers_.find(ssrc) != screencast_capturers_.end()) {
|
| - return false;
|
| - }
|
| - capturer->SignalStateChange.connect(this, &VideoChannel::OnStateChange);
|
| - screencast_capturers_[ssrc] = capturer;
|
| - return true;
|
| -}
|
| -
|
| -bool VideoChannel::RemoveScreencast_w(uint32_t 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::OnScreencastWindowEvent_s(uint32_t ssrc,
|
| - rtc::WindowEvent we) {
|
| - ASSERT(signaling_thread() == rtc::Thread::Current());
|
| - SignalScreencastWindowEvent(ssrc, we);
|
| -}
|
| -
|
| -void VideoChannel::OnMessage(rtc::Message *pmsg) {
|
| - switch (pmsg->message_id) {
|
| - case MSG_SCREENCASTWINDOWEVENT: {
|
| - const ScreencastEventMessageData* data =
|
| - static_cast<ScreencastEventMessageData*>(pmsg->pdata);
|
| - OnScreencastWindowEvent_s(data->ssrc, data->event);
|
| - delete data;
|
| - break;
|
| - }
|
| - case MSG_CHANNEL_ERROR: {
|
| - const VideoChannelErrorMessageData* data =
|
| - static_cast<VideoChannelErrorMessageData*>(pmsg->pdata);
|
| - delete data;
|
| - break;
|
| - }
|
| - default:
|
| - BaseChannel::OnMessage(pmsg);
|
| - break;
|
| - }
|
| -}
|
| -
|
| -void VideoChannel::OnConnectionMonitorUpdate(
|
| - ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) {
|
| - SignalConnectionMonitor(this, infos);
|
| -}
|
| -
|
| -// TODO(pthatcher): Look into removing duplicate code between
|
| -// audio, video, and data, perhaps by using templates.
|
| -void VideoChannel::OnMediaMonitorUpdate(
|
| - VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
|
| - ASSERT(media_channel == this->media_channel());
|
| - SignalMediaMonitor(this, info);
|
| -}
|
| -
|
| -void VideoChannel::OnScreencastWindowEvent(uint32_t ssrc,
|
| - rtc::WindowEvent event) {
|
| - ScreencastEventMessageData* pdata =
|
| - new ScreencastEventMessageData(ssrc, event);
|
| - signaling_thread()->Post(this, MSG_SCREENCASTWINDOWEVENT, pdata);
|
| -}
|
| -
|
| -void VideoChannel::OnStateChange(VideoCapturer* capturer, CaptureState ev) {
|
| - // Map capturer events to window events. In the future we may want to simply
|
| - // pass these events up directly.
|
| - rtc::WindowEvent we;
|
| - if (ev == CS_STOPPED) {
|
| - we = rtc::WE_CLOSE;
|
| - } else if (ev == CS_PAUSED) {
|
| - we = rtc::WE_MINIMIZE;
|
| - } else if (ev == CS_RUNNING && previous_we_ == rtc::WE_MINIMIZE) {
|
| - we = rtc::WE_RESTORE;
|
| - } else {
|
| - return;
|
| - }
|
| - previous_we_ = we;
|
| -
|
| - uint32_t ssrc = 0;
|
| - if (!GetLocalSsrc(capturer, &ssrc)) {
|
| - return;
|
| - }
|
| -
|
| - OnScreencastWindowEvent(ssrc, we);
|
| -}
|
| -
|
| -bool VideoChannel::GetLocalSsrc(const VideoCapturer* capturer, uint32_t* 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::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
|
| - GetSupportedVideoCryptoSuites(crypto_suites);
|
| -}
|
| -
|
| -DataChannel::DataChannel(rtc::Thread* thread,
|
| - DataMediaChannel* media_channel,
|
| - TransportController* transport_controller,
|
| - const std::string& content_name,
|
| - bool rtcp)
|
| - : BaseChannel(thread,
|
| - media_channel,
|
| - transport_controller,
|
| - content_name,
|
| - rtcp),
|
| - data_channel_type_(cricket::DCT_NONE),
|
| - ready_to_send_data_(false) {}
|
| -
|
| -DataChannel::~DataChannel() {
|
| - StopMediaMonitor();
|
| - // this can't be done in the base class, since it calls a virtual
|
| - DisableMedia_w();
|
| -
|
| - Deinit();
|
| -}
|
| -
|
| -bool DataChannel::Init() {
|
| - if (!BaseChannel::Init()) {
|
| - return false;
|
| - }
|
| - media_channel()->SignalDataReceived.connect(
|
| - this, &DataChannel::OnDataReceived);
|
| - media_channel()->SignalReadyToSend.connect(
|
| - this, &DataChannel::OnDataChannelReadyToSend);
|
| - media_channel()->SignalStreamClosedRemotely.connect(
|
| - this, &DataChannel::OnStreamClosedRemotely);
|
| - return true;
|
| -}
|
| -
|
| -bool DataChannel::SendData(const SendDataParams& params,
|
| - const rtc::Buffer& payload,
|
| - SendDataResult* result) {
|
| - return InvokeOnWorker(Bind(&DataMediaChannel::SendData,
|
| - media_channel(), params, payload, result));
|
| -}
|
| -
|
| -const ContentInfo* DataChannel::GetFirstContent(
|
| - const SessionDescription* sdesc) {
|
| - return GetFirstDataContent(sdesc);
|
| -}
|
| -
|
| -bool DataChannel::WantsPacket(bool rtcp, rtc::Buffer* packet) {
|
| - if (data_channel_type_ == DCT_SCTP) {
|
| - // TODO(pthatcher): Do this in a more robust way by checking for
|
| - // SCTP or DTLS.
|
| - return !IsRtpPacket(packet->data(), packet->size());
|
| - } else if (data_channel_type_ == DCT_RTP) {
|
| - return BaseChannel::WantsPacket(rtcp, packet);
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -bool DataChannel::SetDataChannelType(DataChannelType new_data_channel_type,
|
| - std::string* error_desc) {
|
| - // It hasn't been set before, so set it now.
|
| - if (data_channel_type_ == DCT_NONE) {
|
| - data_channel_type_ = new_data_channel_type;
|
| - return true;
|
| - }
|
| -
|
| - // It's been set before, but doesn't match. That's bad.
|
| - if (data_channel_type_ != new_data_channel_type) {
|
| - std::ostringstream desc;
|
| - desc << "Data channel type mismatch."
|
| - << " Expected " << data_channel_type_
|
| - << " Got " << new_data_channel_type;
|
| - SafeSetError(desc.str(), error_desc);
|
| - return false;
|
| - }
|
| -
|
| - // It's hasn't changed. Nothing to do.
|
| - return true;
|
| -}
|
| -
|
| -bool DataChannel::SetDataChannelTypeFromContent(
|
| - const DataContentDescription* content,
|
| - std::string* error_desc) {
|
| - bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
|
| - (content->protocol() == kMediaProtocolDtlsSctp));
|
| - DataChannelType data_channel_type = is_sctp ? DCT_SCTP : DCT_RTP;
|
| - return SetDataChannelType(data_channel_type, error_desc);
|
| -}
|
| -
|
| -bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "DataChannel::SetLocalContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| - LOG(LS_INFO) << "Setting local data description";
|
| -
|
| - const DataContentDescription* data =
|
| - static_cast<const DataContentDescription*>(content);
|
| - ASSERT(data != NULL);
|
| - if (!data) {
|
| - SafeSetError("Can't find data content in local description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - if (!SetDataChannelTypeFromContent(data, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - if (data_channel_type_ == DCT_RTP) {
|
| - if (!SetRtpTransportParameters_w(content, action, CS_LOCAL, error_desc)) {
|
| - return false;
|
| - }
|
| - }
|
| -
|
| - // FYI: We send the SCTP port number (not to be confused with the
|
| - // underlying UDP port number) as a codec parameter. So even SCTP
|
| - // data channels need codecs.
|
| - DataRecvParameters recv_params = last_recv_params_;
|
| - RtpParametersFromMediaDescription(data, &recv_params);
|
| - if (!media_channel()->SetRecvParameters(recv_params)) {
|
| - SafeSetError("Failed to set remote data description recv parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - if (data_channel_type_ == DCT_RTP) {
|
| - for (const DataCodec& codec : data->codecs()) {
|
| - bundle_filter()->AddPayloadType(codec.id);
|
| - }
|
| - }
|
| - last_recv_params_ = recv_params;
|
| -
|
| - // TODO(pthatcher): Move local streams into DataSendParameters, and
|
| - // only give it to the media channel once we have a remote
|
| - // description too (without a remote description, we won't be able
|
| - // to send them anyway).
|
| - if (!UpdateLocalStreams_w(data->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set local data description streams.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - set_local_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
|
| - ContentAction action,
|
| - std::string* error_desc) {
|
| - TRACE_EVENT0("webrtc", "DataChannel::SetRemoteContent_w");
|
| - ASSERT(worker_thread() == rtc::Thread::Current());
|
| -
|
| - const DataContentDescription* data =
|
| - static_cast<const DataContentDescription*>(content);
|
| - ASSERT(data != NULL);
|
| - if (!data) {
|
| - SafeSetError("Can't find data content in remote description.", error_desc);
|
| - return false;
|
| - }
|
| -
|
| - // If the remote data doesn't have codecs and isn't an update, it
|
| - // must be empty, so ignore it.
|
| - if (!data->has_codecs() && action != CA_UPDATE) {
|
| - return true;
|
| - }
|
| -
|
| - if (!SetDataChannelTypeFromContent(data, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| - LOG(LS_INFO) << "Setting remote data description";
|
| - if (data_channel_type_ == DCT_RTP &&
|
| - !SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
|
| - return false;
|
| - }
|
| -
|
| -
|
| - DataSendParameters send_params = last_send_params_;
|
| - RtpSendParametersFromMediaDescription<DataCodec>(data, &send_params);
|
| - if (!media_channel()->SetSendParameters(send_params)) {
|
| - SafeSetError("Failed to set remote data description send parameters.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| - last_send_params_ = send_params;
|
| -
|
| - // TODO(pthatcher): Move remote streams into DataRecvParameters,
|
| - // and only give it to the media channel once we have a local
|
| - // description too (without a local description, we won't be able to
|
| - // recv them anyway).
|
| - if (!UpdateRemoteStreams_w(data->streams(), action, error_desc)) {
|
| - SafeSetError("Failed to set remote data description streams.",
|
| - error_desc);
|
| - return false;
|
| - }
|
| -
|
| - set_remote_content_direction(content->direction());
|
| - ChangeState();
|
| - return true;
|
| -}
|
| -
|
| -void DataChannel::ChangeState() {
|
| - // Render incoming data if we're the active call, and we have the local
|
| - // content. We receive data on the default channel and multiplexed streams.
|
| - bool recv = IsReadyToReceive();
|
| - if (!media_channel()->SetReceive(recv)) {
|
| - LOG(LS_ERROR) << "Failed to SetReceive on data channel";
|
| - }
|
| -
|
| - // Send outgoing data if we're the active call, we have the remote content,
|
| - // and we have had some form of connectivity.
|
| - bool send = IsReadyToSend();
|
| - if (!media_channel()->SetSend(send)) {
|
| - LOG(LS_ERROR) << "Failed to SetSend on data channel";
|
| - }
|
| -
|
| - // Trigger SignalReadyToSendData asynchronously.
|
| - OnDataChannelReadyToSend(send);
|
| -
|
| - LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
|
| -}
|
| -
|
| -void DataChannel::OnMessage(rtc::Message *pmsg) {
|
| - switch (pmsg->message_id) {
|
| - case MSG_READYTOSENDDATA: {
|
| - DataChannelReadyToSendMessageData* data =
|
| - static_cast<DataChannelReadyToSendMessageData*>(pmsg->pdata);
|
| - ready_to_send_data_ = data->data();
|
| - SignalReadyToSendData(ready_to_send_data_);
|
| - delete data;
|
| - break;
|
| - }
|
| - case MSG_DATARECEIVED: {
|
| - DataReceivedMessageData* data =
|
| - static_cast<DataReceivedMessageData*>(pmsg->pdata);
|
| - SignalDataReceived(this, data->params, data->payload);
|
| - delete data;
|
| - break;
|
| - }
|
| - case MSG_CHANNEL_ERROR: {
|
| - const DataChannelErrorMessageData* data =
|
| - static_cast<DataChannelErrorMessageData*>(pmsg->pdata);
|
| - delete data;
|
| - break;
|
| - }
|
| - case MSG_STREAMCLOSEDREMOTELY: {
|
| - rtc::TypedMessageData<uint32_t>* data =
|
| - static_cast<rtc::TypedMessageData<uint32_t>*>(pmsg->pdata);
|
| - SignalStreamClosedRemotely(data->data());
|
| - delete data;
|
| - break;
|
| - }
|
| - default:
|
| - BaseChannel::OnMessage(pmsg);
|
| - break;
|
| - }
|
| -}
|
| -
|
| -void DataChannel::OnConnectionMonitorUpdate(
|
| - ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
|
| - SignalConnectionMonitor(this, infos);
|
| -}
|
| -
|
| -void DataChannel::StartMediaMonitor(int cms) {
|
| - media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
|
| - rtc::Thread::Current()));
|
| - media_monitor_->SignalUpdate.connect(
|
| - this, &DataChannel::OnMediaMonitorUpdate);
|
| - media_monitor_->Start(cms);
|
| -}
|
| -
|
| -void DataChannel::StopMediaMonitor() {
|
| - if (media_monitor_) {
|
| - media_monitor_->Stop();
|
| - media_monitor_->SignalUpdate.disconnect(this);
|
| - media_monitor_.reset();
|
| - }
|
| -}
|
| -
|
| -void DataChannel::OnMediaMonitorUpdate(
|
| - DataMediaChannel* media_channel, const DataMediaInfo& info) {
|
| - ASSERT(media_channel == this->media_channel());
|
| - SignalMediaMonitor(this, info);
|
| -}
|
| -
|
| -void DataChannel::OnDataReceived(
|
| - const ReceiveDataParams& params, const char* data, size_t len) {
|
| - DataReceivedMessageData* msg = new DataReceivedMessageData(
|
| - params, data, len);
|
| - signaling_thread()->Post(this, MSG_DATARECEIVED, msg);
|
| -}
|
| -
|
| -void DataChannel::OnDataChannelError(uint32_t 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::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
|
| - GetSupportedDataCryptoSuites(crypto_suites);
|
| -}
|
| -
|
| -bool DataChannel::ShouldSetupDtlsSrtp() const {
|
| - return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
|
| -}
|
| -
|
| -void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
|
| - rtc::TypedMessageData<uint32_t>* message =
|
| - new rtc::TypedMessageData<uint32_t>(sid);
|
| - signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
|
| -}
|
| -
|
| -} // namespace cricket
|
|
|
Chromium Code Reviews
Issue 1691463002:
Move talk/session/media -> webrtc/pc (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master