| /* |
| * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/channel.h" |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <functional> |
| #include <memory> |
| #include <optional> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/strings/string_view.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/jsep.h" |
| #include "api/media_types.h" |
| #include "api/rtp_headers.h" |
| #include "api/rtp_parameters.h" |
| #include "api/sequence_checker.h" |
| #include "api/task_queue/pending_task_safety_flag.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "media/base/codec.h" |
| #include "media/base/media_channel.h" |
| #include "media/base/rid_description.h" |
| #include "media/base/rtp_utils.h" |
| #include "media/base/stream_params.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "p2p/base/dtls_transport_internal.h" |
| #include "pc/rtp_media_utils.h" |
| #include "pc/rtp_transport_internal.h" |
| #include "pc/session_description.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/containers/flat_set.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/network/sent_packet.h" |
| #include "rtc_base/network_route.h" |
| #include "rtc_base/socket.h" |
| #include "rtc_base/strings/string_format.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/trace_event.h" |
| #include "rtc_base/unique_id_generator.h" |
| |
| namespace cricket { |
| namespace { |
| |
| using ::rtc::StringFormat; |
| using ::rtc::UniqueRandomIdGenerator; |
| using ::webrtc::PendingTaskSafetyFlag; |
| using ::webrtc::SdpType; |
| |
| // Finds a stream based on target's Primary SSRC or RIDs. |
| // This struct is used in BaseChannel::UpdateLocalStreams_w. |
| struct StreamFinder { |
| explicit StreamFinder(const StreamParams* target) : target_(target) { |
| RTC_DCHECK(target); |
| } |
| |
| bool operator()(const StreamParams& sp) const { |
| if (target_->has_ssrcs() && sp.has_ssrcs()) { |
| return sp.has_ssrc(target_->first_ssrc()); |
| } |
| |
| if (!target_->has_rids() && !sp.has_rids()) { |
| return false; |
| } |
| |
| const std::vector<RidDescription>& target_rids = target_->rids(); |
| const std::vector<RidDescription>& source_rids = sp.rids(); |
| if (source_rids.size() != target_rids.size()) { |
| return false; |
| } |
| |
| // Check that all RIDs match. |
| return std::equal(source_rids.begin(), source_rids.end(), |
| target_rids.begin(), |
| [](const RidDescription& lhs, const RidDescription& rhs) { |
| return lhs.rid == rhs.rid; |
| }); |
| } |
| |
| const StreamParams* target_; |
| }; |
| |
| } // namespace |
| |
| void MediaChannelParametersFromMediaDescription( |
| const MediaContentDescription* desc, |
| const RtpHeaderExtensions& extensions, |
| bool is_stream_active, |
| MediaChannelParameters* params) { |
| RTC_DCHECK(desc->type() == MEDIA_TYPE_AUDIO || |
| desc->type() == MEDIA_TYPE_VIDEO); |
| params->is_stream_active = is_stream_active; |
| params->codecs = desc->codecs(); |
| // TODO(bugs.webrtc.org/11513): See if we really need |
| // rtp_header_extensions_set() and remove it if we don't. |
| if (desc->rtp_header_extensions_set()) { |
| params->extensions = extensions; |
| } |
| params->rtcp.reduced_size = desc->rtcp_reduced_size(); |
| params->rtcp.remote_estimate = desc->remote_estimate(); |
| } |
| |
| void RtpSendParametersFromMediaDescription( |
| const MediaContentDescription* desc, |
| webrtc::RtpExtension::Filter extensions_filter, |
| SenderParameters* send_params) { |
| RtpHeaderExtensions extensions = |
| webrtc::RtpExtension::DeduplicateHeaderExtensions( |
| desc->rtp_header_extensions(), extensions_filter); |
| const bool is_stream_active = |
| webrtc::RtpTransceiverDirectionHasRecv(desc->direction()); |
| MediaChannelParametersFromMediaDescription(desc, extensions, is_stream_active, |
| send_params); |
| send_params->max_bandwidth_bps = desc->bandwidth(); |
| send_params->extmap_allow_mixed = desc->extmap_allow_mixed(); |
| } |
| |
| BaseChannel::BaseChannel( |
| webrtc::TaskQueueBase* worker_thread, |
| rtc::Thread* network_thread, |
| webrtc::TaskQueueBase* signaling_thread, |
| std::unique_ptr<MediaSendChannelInterface> send_media_channel_impl, |
| std::unique_ptr<MediaReceiveChannelInterface> receive_media_channel_impl, |
| absl::string_view mid, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : media_send_channel_(std::move(send_media_channel_impl)), |
| media_receive_channel_(std::move(receive_media_channel_impl)), |
| worker_thread_(worker_thread), |
| network_thread_(network_thread), |
| signaling_thread_(signaling_thread), |
| alive_(PendingTaskSafetyFlag::Create()), |
| srtp_required_(srtp_required), |
| extensions_filter_( |
| crypto_options.srtp.enable_encrypted_rtp_header_extensions |
| ? webrtc::RtpExtension::kPreferEncryptedExtension |
| : webrtc::RtpExtension::kDiscardEncryptedExtension), |
| demuxer_criteria_(mid), |
| ssrc_generator_(ssrc_generator) { |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| RTC_DCHECK(media_send_channel_); |
| RTC_DCHECK(media_receive_channel_); |
| RTC_DCHECK(ssrc_generator_); |
| RTC_DLOG(LS_INFO) << "Created channel: " << ToString(); |
| } |
| |
| BaseChannel::~BaseChannel() { |
| TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel"); |
| RTC_DCHECK_RUN_ON(worker_thread_); |
| |
| // Eats any outstanding messages or packets. |
| alive_->SetNotAlive(); |
| // The media channel is destroyed at the end of the destructor, since it |
| // is a std::unique_ptr. The transport channel (rtp_transport) must outlive |
| // the media channel. |
| } |
| |
| std::string BaseChannel::ToString() const { |
| return StringFormat( |
| "{mid: %s, media_type: %s}", mid().c_str(), |
| MediaTypeToString(media_send_channel_->media_type()).c_str()); |
| } |
| |
| bool BaseChannel::ConnectToRtpTransport_n() { |
| RTC_DCHECK(rtp_transport_); |
| RTC_DCHECK(media_send_channel()); |
| |
| // We don't need to call OnDemuxerCriteriaUpdatePending/Complete because |
| // there's no previous criteria to worry about. |
| if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) { |
| return false; |
| } |
| rtp_transport_->SubscribeReadyToSend( |
| this, [this](bool ready) { OnTransportReadyToSend(ready); }); |
| rtp_transport_->SubscribeNetworkRouteChanged( |
| this, [this](std::optional<rtc::NetworkRoute> route) { |
| OnNetworkRouteChanged(route); |
| }); |
| rtp_transport_->SubscribeWritableState( |
| this, [this](bool state) { OnWritableState(state); }); |
| rtp_transport_->SubscribeSentPacket( |
| this, |
| [this](const rtc::SentPacket& packet) { SignalSentPacket_n(packet); }); |
| return true; |
| } |
| |
| void BaseChannel::DisconnectFromRtpTransport_n() { |
| RTC_DCHECK(rtp_transport_); |
| RTC_DCHECK(media_send_channel()); |
| rtp_transport_->UnregisterRtpDemuxerSink(this); |
| rtp_transport_->UnsubscribeReadyToSend(this); |
| rtp_transport_->UnsubscribeNetworkRouteChanged(this); |
| rtp_transport_->UnsubscribeWritableState(this); |
| rtp_transport_->UnsubscribeSentPacket(this); |
| rtp_transport_ = nullptr; |
| media_send_channel()->SetInterface(nullptr); |
| media_receive_channel()->SetInterface(nullptr); |
| } |
| |
| bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) { |
| TRACE_EVENT0("webrtc", "BaseChannel::SetRtpTransport"); |
| RTC_DCHECK_RUN_ON(network_thread()); |
| if (rtp_transport == rtp_transport_) { |
| return true; |
| } |
| |
| if (rtp_transport_) { |
| DisconnectFromRtpTransport_n(); |
| // Clear the cached header extensions on the worker. |
| worker_thread_->PostTask(SafeTask(alive_, [this] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| rtp_header_extensions_.clear(); |
| })); |
| } |
| |
| rtp_transport_ = rtp_transport; |
| if (rtp_transport_) { |
| if (!ConnectToRtpTransport_n()) { |
| return false; |
| } |
| |
| RTC_DCHECK(!media_send_channel()->HasNetworkInterface()); |
| media_send_channel()->SetInterface(this); |
| media_receive_channel()->SetInterface(this); |
| |
| media_send_channel()->OnReadyToSend(rtp_transport_->IsReadyToSend()); |
| UpdateWritableState_n(); |
| |
| // Set the cached socket options. |
| for (const auto& pair : socket_options_) { |
| rtp_transport_->SetRtpOption(pair.first, pair.second); |
| } |
| if (!rtp_transport_->rtcp_mux_enabled()) { |
| for (const auto& pair : rtcp_socket_options_) { |
| rtp_transport_->SetRtcpOption(pair.first, pair.second); |
| } |
| } |
| } |
| |
| return true; |
| } |
| |
| void BaseChannel::Enable(bool enable) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| |
| if (enable == enabled_s_) |
| return; |
| |
| enabled_s_ = enable; |
| |
| worker_thread_->PostTask(SafeTask(alive_, [this, enable] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| // Sanity check to make sure that enabled_ and enabled_s_ |
| // stay in sync. |
| RTC_DCHECK_NE(enabled_, enable); |
| if (enable) { |
| EnableMedia_w(); |
| } else { |
| DisableMedia_w(); |
| } |
| })); |
| } |
| |
| bool BaseChannel::SetLocalContent(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent"); |
| return SetLocalContent_w(content, type, error_desc); |
| } |
| |
| bool BaseChannel::SetRemoteContent(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent"); |
| return SetRemoteContent_w(content, type, error_desc); |
| } |
| |
| bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) { |
| // TODO(bugs.webrtc.org/11993): The demuxer state needs to be managed on the |
| // network thread. At the moment there's a workaround for inconsistent state |
| // between the worker and network thread because of this (see |
| // OnDemuxerCriteriaUpdatePending elsewhere in this file) and |
| // SetPayloadTypeDemuxingEnabled_w has a BlockingCall over to the network |
| // thread to apply state updates. |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled"); |
| return SetPayloadTypeDemuxingEnabled_w(enabled); |
| } |
| |
| bool BaseChannel::IsReadyToSendMedia_w() const { |
| // Send outgoing data if we are enabled, have local and remote content, |
| // and we have had some form of connectivity. |
| return enabled_ && |
| webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) && |
| webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) && |
| was_ever_writable_; |
| } |
| |
| bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| return SendPacket(false, packet, options); |
| } |
| |
| bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| return SendPacket(true, packet, options); |
| } |
| |
| int BaseChannel::SetOption(SocketType type, |
| rtc::Socket::Option opt, |
| int value) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| RTC_DCHECK(rtp_transport_); |
| switch (type) { |
| case ST_RTP: |
| socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| return rtp_transport_->SetRtpOption(opt, value); |
| case ST_RTCP: |
| rtcp_socket_options_.push_back( |
| std::pair<rtc::Socket::Option, int>(opt, value)); |
| return rtp_transport_->SetRtcpOption(opt, value); |
| } |
| return -1; |
| } |
| |
| void BaseChannel::OnWritableState(bool writable) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| if (writable) { |
| ChannelWritable_n(); |
| } else { |
| ChannelNotWritable_n(); |
| } |
| } |
| |
| void BaseChannel::OnNetworkRouteChanged( |
| std::optional<rtc::NetworkRoute> network_route) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| |
| RTC_LOG(LS_INFO) << "Network route changed for " << ToString(); |
| |
| rtc::NetworkRoute new_route; |
| if (network_route) { |
| new_route = *(network_route); |
| } |
| // Note: When the RTCP-muxing is not enabled, RTCP transport and RTP transport |
| // use the same transport name and MediaChannel::OnNetworkRouteChanged cannot |
| // work correctly. Intentionally leave it broken to simplify the code and |
| // encourage the users to stop using non-muxing RTCP. |
| media_send_channel()->OnNetworkRouteChanged(transport_name(), new_route); |
| } |
| |
| void BaseChannel::SetFirstPacketReceivedCallback( |
| std::function<void()> callback) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(!on_first_packet_received_ || !callback); |
| |
| // TODO(bugs.webrtc.org/11992): Rename SetFirstPacketReceivedCallback to |
| // something that indicates network thread initialization/uninitialization and |
| // call Init_n() / Deinit_n() respectively. |
| // if (!callback) |
| // Deinit_n(); |
| |
| on_first_packet_received_ = std::move(callback); |
| } |
| |
| void BaseChannel::SetFirstPacketSentCallback(std::function<void()> callback) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(!on_first_packet_sent_ || !callback); |
| |
| on_first_packet_sent_ = std::move(callback); |
| } |
| |
| void BaseChannel::OnTransportReadyToSend(bool ready) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| media_send_channel()->OnReadyToSend(ready); |
| } |
| |
| bool BaseChannel::SendPacket(bool rtcp, |
| rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| TRACE_EVENT0("webrtc", "BaseChannel::SendPacket"); |
| |
| // Until all the code is migrated to use RtpPacketType instead of bool. |
| RtpPacketType packet_type = rtcp ? RtpPacketType::kRtcp : RtpPacketType::kRtp; |
| |
| // Ensure we have a place to send this packet before doing anything. We might |
| // get RTCP packets that we don't intend to send. If we've negotiated RTCP |
| // mux, send RTCP over the RTP transport. |
| if (!rtp_transport_ || !rtp_transport_->IsWritable(rtcp)) { |
| return false; |
| } |
| |
| // Protect ourselves against crazy data. |
| if (!IsValidRtpPacketSize(packet_type, packet->size())) { |
| RTC_LOG(LS_ERROR) << "Dropping outgoing " << ToString() << " " |
| << RtpPacketTypeToString(packet_type) |
| << " packet: wrong size=" << packet->size(); |
| return false; |
| } |
| |
| if (!srtp_active()) { |
| if (srtp_required_) { |
| // The audio/video engines may attempt to send RTCP packets as soon as the |
| // streams are created, so don't treat this as an error for RTCP. |
| // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6809 |
| // However, there shouldn't be any RTP packets sent before SRTP is set |
| // up (and SetSend(true) is called). |
| RTC_DCHECK(rtcp) << "Can't send outgoing RTP packet for " << ToString() |
| << " when SRTP is inactive and crypto is required"; |
| return false; |
| } |
| |
| RTC_DLOG(LS_WARNING) << "Sending an " << (rtcp ? "RTCP" : "RTP") |
| << " packet without encryption for " << ToString() |
| << "."; |
| } |
| |
| if (on_first_packet_sent_ && options.info_signaled_after_sent.is_media) { |
| on_first_packet_sent_(); |
| on_first_packet_sent_ = nullptr; |
| } |
| |
| return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS) |
| : rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS); |
| } |
| |
| void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| |
| if (on_first_packet_received_) { |
| on_first_packet_received_(); |
| on_first_packet_received_ = nullptr; |
| } |
| |
| if (!srtp_active() && srtp_required_) { |
| // Our session description indicates that SRTP is required, but we got a |
| // packet before our SRTP filter is active. This means either that |
| // a) we got SRTP packets before we received the SDES keys, in which case |
| // we can't decrypt it anyway, or |
| // b) we got SRTP packets before DTLS completed on both the RTP and RTCP |
| // transports, so we haven't yet extracted keys, even if DTLS did |
| // complete on the transport that the packets are being sent on. It's |
| // really good practice to wait for both RTP and RTCP to be good to go |
| // before sending media, to prevent weird failure modes, so it's fine |
| // for us to just eat packets here. This is all sidestepped if RTCP mux |
| // is used anyway. |
| RTC_LOG(LS_WARNING) << "Can't process incoming RTP packet when " |
| "SRTP is inactive and crypto is required " |
| << ToString(); |
| return; |
| } |
| media_receive_channel()->OnPacketReceived(parsed_packet); |
| } |
| |
| bool BaseChannel::MaybeUpdateDemuxerAndRtpExtensions_w( |
| bool update_demuxer, |
| std::optional<RtpHeaderExtensions> extensions, |
| std::string& error_desc) { |
| if (extensions) { |
| if (rtp_header_extensions_ == extensions) { |
| extensions.reset(); // No need to update header extensions. |
| } else { |
| rtp_header_extensions_ = *extensions; |
| } |
| } |
| |
| if (!update_demuxer && !extensions) |
| return true; // No update needed. |
| |
| // TODO(bugs.webrtc.org/13536): See if we can do this asynchronously. |
| |
| if (update_demuxer) |
| media_receive_channel()->OnDemuxerCriteriaUpdatePending(); |
| |
| bool success = network_thread()->BlockingCall([&]() mutable { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| // NOTE: This doesn't take the BUNDLE case in account meaning the RTP header |
| // extension maps are not merged when BUNDLE is enabled. This is fine |
| // because the ID for MID should be consistent among all the RTP transports. |
| if (extensions) |
| rtp_transport_->UpdateRtpHeaderExtensionMap(*extensions); |
| |
| if (!update_demuxer) |
| return true; |
| |
| if (!rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this)) { |
| error_desc = |
| StringFormat("Failed to apply demuxer criteria for '%s': '%s'.", |
| mid().c_str(), demuxer_criteria_.ToString().c_str()); |
| return false; |
| } |
| return true; |
| }); |
| |
| if (update_demuxer) |
| media_receive_channel()->OnDemuxerCriteriaUpdateComplete(); |
| |
| return success; |
| } |
| |
| bool BaseChannel::RegisterRtpDemuxerSink_w() { |
| media_receive_channel()->OnDemuxerCriteriaUpdatePending(); |
| // Copy demuxer criteria, since they're a worker-thread variable |
| // and we want to pass them to the network thread |
| bool ret = network_thread_->BlockingCall( |
| [this, demuxer_criteria = demuxer_criteria_] { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| if (!rtp_transport_) { |
| // Transport was disconnected before attempting to update the |
| // criteria. This can happen while setting the remote description. |
| // See chromium:1295469 for an example. |
| return false; |
| } |
| // Note that RegisterRtpDemuxerSink first unregisters the sink if |
| // already registered. So this will change the state of the class |
| // whether the call succeeds or not. |
| return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria, this); |
| }); |
| |
| media_receive_channel()->OnDemuxerCriteriaUpdateComplete(); |
| |
| return ret; |
| } |
| |
| void BaseChannel::EnableMedia_w() { |
| if (enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel enabled: " << ToString(); |
| enabled_ = true; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::DisableMedia_w() { |
| if (!enabled_) |
| return; |
| |
| RTC_LOG(LS_INFO) << "Channel disabled: " << ToString(); |
| enabled_ = false; |
| UpdateMediaSendRecvState_w(); |
| } |
| |
| void BaseChannel::UpdateWritableState_n() { |
| TRACE_EVENT0("webrtc", "BaseChannel::UpdateWritableState_n"); |
| if (rtp_transport_->IsWritable(/*rtcp=*/true) && |
| rtp_transport_->IsWritable(/*rtcp=*/false)) { |
| ChannelWritable_n(); |
| } else { |
| ChannelNotWritable_n(); |
| } |
| } |
| |
| void BaseChannel::ChannelWritable_n() { |
| TRACE_EVENT0("webrtc", "BaseChannel::ChannelWritable_n"); |
| if (writable_) { |
| return; |
| } |
| writable_ = true; |
| RTC_LOG(LS_INFO) << "Channel writable (" << ToString() << ")" |
| << (was_ever_writable_n_ ? "" : " for the first time"); |
| // We only have to do this PostTask once, when first transitioning to |
| // writable. |
| if (!was_ever_writable_n_) { |
| worker_thread_->PostTask(SafeTask(alive_, [this] { |
| RTC_DCHECK_RUN_ON(worker_thread()); |
| was_ever_writable_ = true; |
| UpdateMediaSendRecvState_w(); |
| })); |
| } |
| was_ever_writable_n_ = true; |
| } |
| |
| void BaseChannel::ChannelNotWritable_n() { |
| TRACE_EVENT0("webrtc", "BaseChannel::ChannelNotWritable_n"); |
| if (!writable_) { |
| return; |
| } |
| writable_ = false; |
| RTC_LOG(LS_INFO) << "Channel not writable (" << ToString() << ")"; |
| } |
| |
| bool BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) { |
| RTC_LOG_THREAD_BLOCK_COUNT(); |
| |
| if (enabled == payload_type_demuxing_enabled_) { |
| return true; |
| } |
| |
| payload_type_demuxing_enabled_ = enabled; |
| |
| bool config_changed = false; |
| |
| if (!enabled) { |
| // TODO(crbug.com/11477): This will remove *all* unsignaled streams (those |
| // without an explicitly signaled SSRC), which may include streams that |
| // were matched to this channel by MID or RID. Ideally we'd remove only the |
| // streams that were matched based on payload type alone, but currently |
| // there is no straightforward way to identify those streams. |
| media_receive_channel()->ResetUnsignaledRecvStream(); |
| if (!demuxer_criteria_.payload_types().empty()) { |
| config_changed = true; |
| demuxer_criteria_.payload_types().clear(); |
| } |
| } else if (!payload_types_.empty()) { |
| for (const auto& type : payload_types_) { |
| if (demuxer_criteria_.payload_types().insert(type).second) { |
| config_changed = true; |
| } |
| } |
| } else { |
| RTC_DCHECK(demuxer_criteria_.payload_types().empty()); |
| } |
| |
| RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0); |
| |
| if (!config_changed) |
| return true; |
| |
| // Note: This synchronously hops to the network thread. |
| return RegisterRtpDemuxerSink_w(); |
| } |
| |
| bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams, |
| SdpType type, |
| std::string& error_desc) { |
| // In the case of RIDs (where SSRCs are not negotiated), this method will |
| // generate an SSRC for each layer in StreamParams. That representation will |
| // be stored internally in `local_streams_`. |
| // In subsequent offers, the same stream can appear in `streams` again |
| // (without the SSRCs), so it should be looked up using RIDs (if available) |
| // and then by primary SSRC. |
| // In both scenarios, it is safe to assume that the media channel will be |
| // created with a StreamParams object with SSRCs. However, it is not safe to |
| // assume that `local_streams_` will always have SSRCs as there are scenarios |
| // in which niether SSRCs or RIDs are negotiated. |
| |
| // Check for streams that have been removed. |
| bool ret = true; |
| for (const StreamParams& old_stream : local_streams_) { |
| if (!old_stream.has_ssrcs() || |
| GetStream(streams, StreamFinder(&old_stream))) { |
| continue; |
| } |
| if (!media_send_channel()->RemoveSendStream(old_stream.first_ssrc())) { |
| error_desc = StringFormat( |
| "Failed to remove send stream with ssrc %u from m-section with " |
| "mid='%s'.", |
| old_stream.first_ssrc(), mid().c_str()); |
| ret = false; |
| } |
| } |
| // Check for new streams. |
| std::vector<StreamParams> all_streams; |
| for (const StreamParams& stream : streams) { |
| StreamParams* existing = GetStream(local_streams_, StreamFinder(&stream)); |
| if (existing) { |
| // Parameters cannot change for an existing stream. |
| all_streams.push_back(*existing); |
| continue; |
| } |
| |
| all_streams.push_back(stream); |
| StreamParams& new_stream = all_streams.back(); |
| |
| if (!new_stream.has_ssrcs() && !new_stream.has_rids()) { |
| continue; |
| } |
| |
| RTC_DCHECK(new_stream.has_ssrcs() || new_stream.has_rids()); |
| if (new_stream.has_ssrcs() && new_stream.has_rids()) { |
| error_desc = StringFormat( |
| "Failed to add send stream: %u into m-section with mid='%s'. Stream " |
| "has both SSRCs and RIDs.", |
| new_stream.first_ssrc(), mid().c_str()); |
| ret = false; |
| continue; |
| } |
| |
| // At this point we use the legacy simulcast group in StreamParams to |
| // indicate that we want multiple layers to the media channel. |
| if (!new_stream.has_ssrcs()) { |
| // TODO(bugs.webrtc.org/10250): Indicate if flex is desired here. |
| new_stream.GenerateSsrcs(new_stream.rids().size(), /* rtx = */ true, |
| /* flex_fec = */ false, ssrc_generator_); |
| } |
| |
| if (media_send_channel()->AddSendStream(new_stream)) { |
| RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0] |
| << " into " << ToString(); |
| } else { |
| error_desc = StringFormat( |
| "Failed to add send stream ssrc: %u into m-section with mid='%s'", |
| new_stream.first_ssrc(), mid().c_str()); |
| ret = false; |
| } |
| } |
| local_streams_ = all_streams; |
| return ret; |
| } |
| |
| bool BaseChannel::UpdateRemoteStreams_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| RTC_LOG_THREAD_BLOCK_COUNT(); |
| bool needs_re_registration = false; |
| if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) { |
| RTC_DLOG(LS_VERBOSE) << "UpdateRemoteStreams_w: remote side will not send " |
| "- disable payload type demuxing for " |
| << ToString(); |
| if (ClearHandledPayloadTypes()) { |
| needs_re_registration = payload_type_demuxing_enabled_; |
| } |
| } |
| |
| const std::vector<StreamParams>& streams = content->streams(); |
| const bool new_has_unsignaled_ssrcs = HasStreamWithNoSsrcs(streams); |
| const bool old_has_unsignaled_ssrcs = HasStreamWithNoSsrcs(remote_streams_); |
| |
| // Check for streams that have been removed. |
| for (const StreamParams& old_stream : remote_streams_) { |
| // If we no longer have an unsignaled stream, we would like to remove |
| // the unsignaled stream params that are cached. |
| if (!old_stream.has_ssrcs() && !new_has_unsignaled_ssrcs) { |
| media_receive_channel()->ResetUnsignaledRecvStream(); |
| RTC_LOG(LS_INFO) << "Reset unsignaled remote stream for " << ToString() |
| << "."; |
| } else if (old_stream.has_ssrcs() && |
| !GetStreamBySsrc(streams, old_stream.first_ssrc())) { |
| if (media_receive_channel()->RemoveRecvStream(old_stream.first_ssrc())) { |
| RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc() |
| << " from " << ToString() << "."; |
| } else { |
| error_desc = StringFormat( |
| "Failed to remove remote stream with ssrc %u from m-section with " |
| "mid='%s'.", |
| old_stream.first_ssrc(), mid().c_str()); |
| return false; |
| } |
| } |
| } |
| |
| // Check for new streams. |
| webrtc::flat_set<uint32_t> ssrcs; |
| for (const StreamParams& new_stream : streams) { |
| // We allow a StreamParams with an empty list of SSRCs, in which case the |
| // MediaChannel will cache the parameters and use them for any unsignaled |
| // stream received later. |
| if ((!new_stream.has_ssrcs() && !old_has_unsignaled_ssrcs) || |
| !GetStreamBySsrc(remote_streams_, new_stream.first_ssrc())) { |
| if (media_receive_channel()->AddRecvStream(new_stream)) { |
| RTC_LOG(LS_INFO) << "Add remote ssrc: " |
| << (new_stream.has_ssrcs() |
| ? std::to_string(new_stream.first_ssrc()) |
| : "unsignaled") |
| << " to " << ToString(); |
| } else { |
| error_desc = |
| StringFormat("Failed to add remote stream ssrc: %s to %s", |
| new_stream.has_ssrcs() |
| ? std::to_string(new_stream.first_ssrc()).c_str() |
| : "unsignaled", |
| ToString().c_str()); |
| return false; |
| } |
| } |
| // Update the receiving SSRCs. |
| ssrcs.insert(new_stream.ssrcs.begin(), new_stream.ssrcs.end()); |
| } |
| |
| if (demuxer_criteria_.ssrcs() != ssrcs) { |
| demuxer_criteria_.ssrcs() = std::move(ssrcs); |
| needs_re_registration = true; |
| } |
| |
| RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0); |
| |
| // Re-register the sink to update after changing the demuxer criteria. |
| if (needs_re_registration && !RegisterRtpDemuxerSink_w()) { |
| error_desc = StringFormat("Failed to set up audio demuxing for mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| |
| remote_streams_ = streams; |
| |
| set_remote_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| |
| RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1); |
| |
| return true; |
| } |
| |
| RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions( |
| const RtpHeaderExtensions& extensions) { |
| return webrtc::RtpExtension::DeduplicateHeaderExtensions(extensions, |
| extensions_filter_); |
| } |
| |
| bool BaseChannel::MaybeAddHandledPayloadType(int payload_type) { |
| bool demuxer_criteria_modified = false; |
| if (payload_type_demuxing_enabled_) { |
| demuxer_criteria_modified = demuxer_criteria_.payload_types() |
| .insert(static_cast<uint8_t>(payload_type)) |
| .second; |
| } |
| // Even if payload type demuxing is currently disabled, we need to remember |
| // the payload types in case it's re-enabled later. |
| payload_types_.insert(static_cast<uint8_t>(payload_type)); |
| return demuxer_criteria_modified; |
| } |
| |
| bool BaseChannel::ClearHandledPayloadTypes() { |
| const bool was_empty = demuxer_criteria_.payload_types().empty(); |
| demuxer_criteria_.payload_types().clear(); |
| payload_types_.clear(); |
| return !was_empty; |
| } |
| |
| void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| RTC_DCHECK(network_initialized()); |
| media_send_channel()->OnPacketSent(sent_packet); |
| } |
| |
| VoiceChannel::VoiceChannel( |
| webrtc::TaskQueueBase* worker_thread, |
| rtc::Thread* network_thread, |
| webrtc::TaskQueueBase* signaling_thread, |
| std::unique_ptr<VoiceMediaSendChannelInterface> media_send_channel, |
| std::unique_ptr<VoiceMediaReceiveChannelInterface> media_receive_channel, |
| absl::string_view mid, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_send_channel), |
| std::move(media_receive_channel), |
| mid, |
| srtp_required, |
| crypto_options, |
| ssrc_generator) {} |
| |
| VoiceChannel::~VoiceChannel() { |
| TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel"); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| } |
| |
| void VoiceChannel::UpdateMediaSendRecvState_w() { |
| // Render incoming data if we're the active call, and we have the local |
| // content. We receive data on the default channel and multiplexed streams. |
| bool receive = enabled() && webrtc::RtpTransceiverDirectionHasRecv( |
| local_content_direction()); |
| media_receive_channel()->SetPlayout(receive); |
| |
| // Send outgoing data if we're the active call, we have the remote content, |
| // and we have had some form of connectivity. |
| bool send = IsReadyToSendMedia_w(); |
| media_send_channel()->SetSend(send); |
| |
| RTC_LOG(LS_INFO) << "Changing voice state, recv=" << receive |
| << " send=" << send << " for " << ToString(); |
| } |
| |
| bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w"); |
| RTC_DLOG(LS_INFO) << "Setting local voice description for " << ToString(); |
| |
| RTC_LOG_THREAD_BLOCK_COUNT(); |
| |
| RtpHeaderExtensions header_extensions = |
| GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions()); |
| bool update_header_extensions = true; |
| media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed()); |
| |
| AudioReceiverParameters recv_params = last_recv_params_; |
| MediaChannelParametersFromMediaDescription( |
| content, header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(content->direction()), |
| &recv_params); |
| recv_params.mid = mid(); |
| |
| if (!media_receive_channel()->SetReceiverParameters(recv_params)) { |
| error_desc = StringFormat( |
| "Failed to set local audio description recv parameters for m-section " |
| "with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| |
| bool criteria_modified = false; |
| if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) { |
| for (const Codec& codec : content->codecs()) { |
| if (MaybeAddHandledPayloadType(codec.id)) { |
| criteria_modified = true; |
| } |
| } |
| } |
| |
| last_recv_params_ = recv_params; |
| |
| if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) { |
| RTC_DCHECK(!error_desc.empty()); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| |
| // Disabled because suggeting PTs takes thread jumps. |
| // TODO: https://issues.webrtc.org/360058654 - reenable after cleanup |
| // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0); |
| |
| bool success = MaybeUpdateDemuxerAndRtpExtensions_w( |
| criteria_modified, |
| update_header_extensions |
| ? std::optional<RtpHeaderExtensions>(std::move(header_extensions)) |
| : std::nullopt, |
| error_desc); |
| |
| // RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1); |
| |
| return success; |
| } |
| |
| bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w"); |
| RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString(); |
| |
| AudioSenderParameter send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription(content, extensions_filter(), |
| &send_params); |
| send_params.mid = mid(); |
| |
| bool parameters_applied = |
| media_send_channel()->SetSenderParameters(send_params); |
| if (!parameters_applied) { |
| error_desc = StringFormat( |
| "Failed to set remote audio description send parameters for m-section " |
| "with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| // The receive channel can send RTCP packets in the reverse direction. It |
| // should use the reduced size mode if a peer has requested it through the |
| // remote content. |
| media_receive_channel()->SetRtcpMode(content->rtcp_reduced_size() |
| ? webrtc::RtcpMode::kReducedSize |
| : webrtc::RtcpMode::kCompound); |
| // Update Receive channel based on Send channel's codec information. |
| // TODO(bugs.webrtc.org/14911): This is silly. Stop doing it. |
| media_receive_channel()->SetReceiveNackEnabled( |
| media_send_channel()->SenderNackEnabled()); |
| media_receive_channel()->SetReceiveNonSenderRttEnabled( |
| media_send_channel()->SenderNonSenderRttEnabled()); |
| last_send_params_ = send_params; |
| |
| return UpdateRemoteStreams_w(content, type, error_desc); |
| } |
| |
| VideoChannel::VideoChannel( |
| webrtc::TaskQueueBase* worker_thread, |
| rtc::Thread* network_thread, |
| webrtc::TaskQueueBase* signaling_thread, |
| std::unique_ptr<VideoMediaSendChannelInterface> media_send_channel, |
| std::unique_ptr<VideoMediaReceiveChannelInterface> media_receive_channel, |
| absl::string_view mid, |
| bool srtp_required, |
| webrtc::CryptoOptions crypto_options, |
| UniqueRandomIdGenerator* ssrc_generator) |
| : BaseChannel(worker_thread, |
| network_thread, |
| signaling_thread, |
| std::move(media_send_channel), |
| std::move(media_receive_channel), |
| mid, |
| srtp_required, |
| crypto_options, |
| ssrc_generator) { |
| // TODO(bugs.webrtc.org/13931): Remove when values are set |
| // in a more sensible fashion |
| send_channel()->SetSendCodecChangedCallback([this]() { |
| // Adjust receive streams based on send codec. |
| receive_channel()->SetReceiverFeedbackParameters( |
| send_channel()->SendCodecHasLntf(), send_channel()->SendCodecHasNack(), |
| send_channel()->SendCodecRtcpMode(), |
| send_channel()->SendCodecRtxTime()); |
| }); |
| } |
| |
| VideoChannel::~VideoChannel() { |
| TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel"); |
| // this can't be done in the base class, since it calls a virtual |
| DisableMedia_w(); |
| } |
| |
| void VideoChannel::UpdateMediaSendRecvState_w() { |
| // Send outgoing data if we're the active call, we have the remote content, |
| // and we have had some form of connectivity. |
| bool receive = enabled() && webrtc::RtpTransceiverDirectionHasRecv( |
| local_content_direction()); |
| media_receive_channel()->SetReceive(receive); |
| |
| bool send = IsReadyToSendMedia_w(); |
| media_send_channel()->SetSend(send); |
| RTC_LOG(LS_INFO) << "Changing video state, recv=" << receive |
| << " send=" << send << " for " << ToString(); |
| } |
| |
| bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w"); |
| RTC_DLOG(LS_INFO) << "Setting local video description for " << ToString(); |
| |
| RTC_LOG_THREAD_BLOCK_COUNT(); |
| |
| RtpHeaderExtensions header_extensions = |
| GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions()); |
| bool update_header_extensions = true; |
| media_send_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed()); |
| |
| VideoReceiverParameters recv_params = last_recv_params_; |
| |
| MediaChannelParametersFromMediaDescription( |
| content, header_extensions, |
| webrtc::RtpTransceiverDirectionHasRecv(content->direction()), |
| &recv_params); |
| |
| VideoSenderParameters send_params = last_send_params_; |
| |
| // Ensure that there is a matching packetization for each send codec. If the |
| // other peer offered to exclusively send non-standard packetization but we |
| // only accept to receive standard packetization we effectively amend their |
| // offer by ignoring the packetiztion and fall back to standard packetization |
| // instead. |
| bool needs_send_params_update = false; |
| if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) { |
| webrtc::flat_set<const Codec*> matched_codecs; |
| for (Codec& send_codec : send_params.codecs) { |
| if (absl::c_any_of(matched_codecs, [&](const Codec* c) { |
| return send_codec.Matches(*c); |
| })) { |
| continue; |
| } |
| |
| std::vector<const Codec*> recv_codecs = |
| FindAllMatchingCodecs(recv_params.codecs, send_codec); |
| if (recv_codecs.empty()) { |
| continue; |
| } |
| |
| bool may_ignore_packetization = false; |
| bool has_matching_packetization = false; |
| for (const Codec* recv_codec : recv_codecs) { |
| if (!recv_codec->packetization.has_value() && |
| send_codec.packetization.has_value()) { |
| may_ignore_packetization = true; |
| } else if (recv_codec->packetization == send_codec.packetization) { |
| has_matching_packetization = true; |
| break; |
| } |
| } |
| |
| if (may_ignore_packetization) { |
| send_codec.packetization = std::nullopt; |
| needs_send_params_update = true; |
| } else if (!has_matching_packetization) { |
| error_desc = StringFormat( |
| "Failed to set local answer due to incompatible codec " |
| "packetization for pt='%d' specified in m-section with mid='%s'.", |
| send_codec.id, mid().c_str()); |
| return false; |
| } |
| |
| if (has_matching_packetization) { |
| matched_codecs.insert(&send_codec); |
| } |
| } |
| } |
| |
| if (!media_receive_channel()->SetReceiverParameters(recv_params)) { |
| error_desc = StringFormat( |
| "Failed to set local video description recv parameters for m-section " |
| "with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| |
| bool criteria_modified = false; |
| if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) { |
| for (const Codec& codec : content->codecs()) { |
| if (MaybeAddHandledPayloadType(codec.id)) |
| criteria_modified = true; |
| } |
| } |
| |
| last_recv_params_ = recv_params; |
| |
| if (needs_send_params_update) { |
| if (!media_send_channel()->SetSenderParameters(send_params)) { |
| error_desc = StringFormat( |
| "Failed to set send parameters for m-section with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| last_send_params_ = send_params; |
| } |
| |
| if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) { |
| RTC_DCHECK(!error_desc.empty()); |
| return false; |
| } |
| |
| set_local_content_direction(content->direction()); |
| UpdateMediaSendRecvState_w(); |
| |
| RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0); |
| |
| bool success = MaybeUpdateDemuxerAndRtpExtensions_w( |
| criteria_modified, |
| update_header_extensions |
| ? std::optional<RtpHeaderExtensions>(std::move(header_extensions)) |
| : std::nullopt, |
| error_desc); |
| |
| RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1); |
| |
| return success; |
| } |
| |
| bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content, |
| SdpType type, |
| std::string& error_desc) { |
| TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w"); |
| RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString(); |
| |
| VideoSenderParameters send_params = last_send_params_; |
| RtpSendParametersFromMediaDescription(content, extensions_filter(), |
| &send_params); |
| send_params.mid = mid(); |
| send_params.conference_mode = content->conference_mode(); |
| |
| VideoReceiverParameters recv_params = last_recv_params_; |
| |
| // Ensure that there is a matching packetization for each receive codec. If we |
| // offered to exclusively receive a non-standard packetization but the other |
| // peer only accepts to send standard packetization we effectively amend our |
| // offer by ignoring the packetiztion and fall back to standard packetization |
| // instead. |
| bool needs_recv_params_update = false; |
| if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) { |
| webrtc::flat_set<const Codec*> matched_codecs; |
| for (Codec& recv_codec : recv_params.codecs) { |
| if (absl::c_any_of(matched_codecs, [&](const Codec* c) { |
| return recv_codec.Matches(*c); |
| })) { |
| continue; |
| } |
| |
| std::vector<const Codec*> send_codecs = |
| FindAllMatchingCodecs(send_params.codecs, recv_codec); |
| if (send_codecs.empty()) { |
| continue; |
| } |
| |
| bool may_ignore_packetization = false; |
| bool has_matching_packetization = false; |
| for (const Codec* send_codec : send_codecs) { |
| if (!send_codec->packetization.has_value() && |
| recv_codec.packetization.has_value()) { |
| may_ignore_packetization = true; |
| } else if (send_codec->packetization == recv_codec.packetization) { |
| has_matching_packetization = true; |
| break; |
| } |
| } |
| |
| if (may_ignore_packetization) { |
| recv_codec.packetization = std::nullopt; |
| needs_recv_params_update = true; |
| } else if (!has_matching_packetization) { |
| error_desc = StringFormat( |
| "Failed to set remote answer due to incompatible codec " |
| "packetization for pt='%d' specified in m-section with mid='%s'.", |
| recv_codec.id, mid().c_str()); |
| return false; |
| } |
| |
| if (has_matching_packetization) { |
| matched_codecs.insert(&recv_codec); |
| } |
| } |
| } |
| |
| if (!media_send_channel()->SetSenderParameters(send_params)) { |
| error_desc = StringFormat( |
| "Failed to set remote video description send parameters for m-section " |
| "with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| // adjust receive streams based on send codec |
| media_receive_channel()->SetReceiverFeedbackParameters( |
| media_send_channel()->SendCodecHasLntf(), |
| media_send_channel()->SendCodecHasNack(), |
| media_send_channel()->SendCodecRtcpMode(), |
| media_send_channel()->SendCodecRtxTime()); |
| last_send_params_ = send_params; |
| |
| if (needs_recv_params_update) { |
| if (!media_receive_channel()->SetReceiverParameters(recv_params)) { |
| error_desc = StringFormat( |
| "Failed to set recv parameters for m-section with mid='%s'.", |
| mid().c_str()); |
| return false; |
| } |
| last_recv_params_ = recv_params; |
| } |
| |
| return UpdateRemoteStreams_w(content, type, error_desc); |
| } |
| |
| } // namespace cricket |
