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1b83a9e4004a7da8546024909e4a7f274b6304e1
platform-external-webrtc/pc/channel.cc
Sebastian Jansson 1b83a9e400 Only handle each RTCP once. 
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.

The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.

Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
2019-09-18 16:54:39 +00:00

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/*
* 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 <iterator>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/call/audio_sink.h"
#include "api/transport/media/media_transport_config.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/channel_manager.h"
#include "pc/rtp_media_utils.h"
#include "rtc_base/bind.h"
#include "rtc_base/byte_order.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/dscp.h"
#include "rtc_base/logging.h"
#include "rtc_base/network_route.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/trace_event.h"
namespace cricket {
using rtc::Bind;
using rtc::UniqueRandomIdGenerator;
using webrtc::SdpType;
namespace {
struct SendPacketMessageData : public rtc::MessageData {
rtc::CopyOnWriteBuffer packet;
rtc::PacketOptions options;
};
// 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
enum {
MSG_SEND_RTP_PACKET = 1,
MSG_SEND_RTCP_PACKET,
MSG_READYTOSENDDATA,
MSG_DATARECEIVED,
MSG_FIRSTPACKETRECEIVED,
};
static void SafeSetError(const std::string& message, std::string* error_desc) {
if (error_desc) {
*error_desc = message;
}
}
template <class Codec>
void RtpParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
RtpParameters<Codec>* params) {
// TODO(pthatcher): Remove this once we're sure no one will give us
// a description without codecs. Currently the ORTC implementation is relying
// on this.
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 = extensions;
}
params->rtcp.reduced_size = desc->rtcp_reduced_size();
params->rtcp.remote_estimate = desc->remote_estimate();
}
template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
RtpSendParameters<Codec>* send_params) {
RtpParametersFromMediaDescription(desc, extensions, send_params);
send_params->max_bandwidth_bps = desc->bandwidth();
send_params->extmap_allow_mixed = desc->extmap_allow_mixed();
}
BaseChannel::BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: worker_thread_(worker_thread),
network_thread_(network_thread),
signaling_thread_(signaling_thread),
content_name_(content_name),
srtp_required_(srtp_required),
crypto_options_(crypto_options),
media_channel_(std::move(media_channel)),
ssrc_generator_(ssrc_generator) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(ssrc_generator_);
demuxer_criteria_.mid = content_name;
RTC_LOG(LS_INFO) << "Created channel for " << content_name;
}
BaseChannel::~BaseChannel() {
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK_RUN_ON(worker_thread_);
if (media_transport_config_.media_transport) {
media_transport_config_.media_transport->RemoveNetworkChangeCallback(this);
}
// Eats any outstanding messages or packets.
worker_thread_->Clear(&invoker_);
worker_thread_->Clear(this);
// 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.
media_channel_.reset();
RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_;
}
bool BaseChannel::ConnectToRtpTransport() {
RTC_DCHECK(rtp_transport_);
if (!RegisterRtpDemuxerSink()) {
return false;
}
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
// If media transport is used, it's responsible for providing network
// route changed callbacks.
if (!media_transport_config_.media_transport) {
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
}
// TODO(bugs.webrtc.org/9719): Media transport should also be used to provide
// 'writable' state here.
rtp_transport_->SignalWritableState.connect(this,
&BaseChannel::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this,
&BaseChannel::SignalSentPacket_n);
return true;
}
void BaseChannel::DisconnectFromRtpTransport() {
RTC_DCHECK(rtp_transport_);
rtp_transport_->UnregisterRtpDemuxerSink(this);
rtp_transport_->SignalReadyToSend.disconnect(this);
rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
rtp_transport_->SignalWritableState.disconnect(this);
rtp_transport_->SignalSentPacket.disconnect(this);
}
void BaseChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
RTC_DCHECK_RUN_ON(worker_thread_);
media_transport_config_ = media_transport_config;
network_thread_->Invoke<void>(
RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this, media_transport_config);
RTC_LOG(LS_INFO) << "BaseChannel::Init_w, media_transport_config="
<< media_transport_config.DebugString();
if (media_transport_config_.media_transport) {
media_transport_config_.media_transport->AddNetworkChangeCallback(this);
}
}
void BaseChannel::Deinit() {
RTC_DCHECK(worker_thread_->IsCurrent());
media_channel_->SetInterface(/*iface=*/nullptr,
webrtc::MediaTransportConfig());
// Packets arrive on the network thread, processing packets calls virtual
// functions, so need to stop this process in Deinit that is called in
// derived classes destructor.
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
FlushRtcpMessages_n();
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
// Clear pending read packets/messages.
network_thread_->Clear(&invoker_);
network_thread_->Clear(this);
});
}
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
if (rtp_transport == rtp_transport_) {
return true;
}
if (!network_thread_->IsCurrent()) {
return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, rtp_transport] {
return SetRtpTransport(rtp_transport);
});
}
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
rtp_transport_ = rtp_transport;
if (rtp_transport_) {
transport_name_ = rtp_transport_->transport_name();
if (!ConnectToRtpTransport()) {
RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport.";
return false;
}
OnTransportReadyToSend(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;
}
bool BaseChannel::Enable(bool enable) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
this));
return true;
}
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetLocalContent_w, this, content, type, error_desc));
}
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
return InvokeOnWorker<bool>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetRemoteContent_w, this, content, type, error_desc));
}
bool BaseChannel::IsReadyToReceiveMedia_w() const {
// Receive data if we are enabled and have local content,
return enabled() &&
webrtc::RtpTransceiverDirectionHasRecv(local_content_direction_);
}
bool BaseChannel::IsReadyToSendMedia_w() const {
// Need to access some state updated on the network thread.
return network_thread_->Invoke<bool>(
RTC_FROM_HERE, Bind(&BaseChannel::IsReadyToSendMedia_n, this));
}
bool BaseChannel::IsReadyToSendMedia_n() 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) {
return network_thread_->Invoke<int>(
RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value));
}
int BaseChannel::SetOption_n(SocketType type,
rtc::Socket::Option opt,
int value) {
RTC_DCHECK(network_thread_->IsCurrent());
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(network_thread_->IsCurrent());
if (writable) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::OnNetworkRouteChanged(
absl::optional<rtc::NetworkRoute> network_route) {
RTC_LOG(LS_INFO) << "Network route was changed.";
RTC_DCHECK(network_thread_->IsCurrent());
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.
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_, [=] {
media_channel_->OnNetworkRouteChanged(transport_name_, new_route);
});
}
void BaseChannel::OnTransportReadyToSend(bool ready) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[=] { media_channel_->OnReadyToSend(ready); });
}
bool BaseChannel::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
// Until all the code is migrated to use RtpPacketType instead of bool.
RtpPacketType packet_type = rtcp ? RtpPacketType::kRtcp : RtpPacketType::kRtp;
// SendPacket gets called from MediaEngine, on a pacer or an encoder thread.
// If the thread is not our network thread, we will post to our network
// so that the real work happens on our network. 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 (!network_thread_->IsCurrent()) {
// Avoid a copy by transferring the ownership of the packet data.
int message_id = rtcp ? MSG_SEND_RTCP_PACKET : MSG_SEND_RTP_PACKET;
SendPacketMessageData* data = new SendPacketMessageData;
data->packet = std::move(*packet);
data->options = options;
network_thread_->Post(RTC_FROM_HERE, this, message_id, data);
return true;
}
TRACE_EVENT0("webrtc", "BaseChannel::SendPacket");
// 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.
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 " << content_name_ << " "
<< 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
if (rtcp) {
return false;
}
// However, there shouldn't be any RTP packets sent before SRTP is set up
// (and SetSend(true) is called).
RTC_LOG(LS_ERROR)
<< "Can't send outgoing RTP packet when SRTP is inactive"
<< " and crypto is required";
RTC_NOTREACHED();
return false;
}
std::string packet_type = rtcp ? "RTCP" : "RTP";
RTC_LOG(LS_WARNING) << "Sending an " << packet_type
<< " packet without encryption.";
}
// Bon voyage.
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) {
// Take packet time from the |parsed_packet|.
// RtpPacketReceived.arrival_time_ms = (timestamp_us + 500) / 1000;
int64_t packet_time_us = -1;
if (parsed_packet.arrival_time_ms() > 0) {
packet_time_us = parsed_packet.arrival_time_ms() * 1000;
}
if (!has_received_packet_) {
has_received_packet_ = true;
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
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";
return;
}
auto packet_buffer = parsed_packet.Buffer();
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_, [this, packet_buffer, packet_time_us] {
RTC_DCHECK(worker_thread_->IsCurrent());
media_channel_->OnPacketReceived(packet_buffer, packet_time_us);
});
}
void BaseChannel::UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions) {
RTC_DCHECK(rtp_transport_);
// Update the header extension map on network thread in case there is data
// race.
// TODO(zhihuang): Add an rtc::ThreadChecker make sure to RtpTransport won't
// be accessed from different threads.
//
// 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.
network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &header_extensions] {
rtp_transport_->UpdateRtpHeaderExtensionMap(header_extensions);
});
}
bool BaseChannel::RegisterRtpDemuxerSink() {
RTC_DCHECK(rtp_transport_);
return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this] {
return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
});
}
void BaseChannel::EnableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (enabled_)
return;
RTC_LOG(LS_INFO) << "Channel enabled";
enabled_ = true;
UpdateMediaSendRecvState_w();
}
void BaseChannel::DisableMedia_w() {
RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
if (!enabled_)
return;
RTC_LOG(LS_INFO) << "Channel disabled";
enabled_ = false;
UpdateMediaSendRecvState_w();
}
void BaseChannel::UpdateWritableState_n() {
if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
rtp_transport_->IsWritable(/*rtcp=*/false)) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::ChannelWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (writable_) {
return;
}
RTC_LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
<< (was_ever_writable_ ? "" : " for the first time");
was_ever_writable_ = true;
writable_ = true;
UpdateMediaSendRecvState();
}
void BaseChannel::ChannelNotWritable_n() {
RTC_DCHECK(network_thread_->IsCurrent());
if (!writable_)
return;
RTC_LOG(LS_INFO) << "Channel not writable (" << content_name_ << ")";
writable_ = false;
UpdateMediaSendRecvState();
}
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->AddRecvStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
RTC_DCHECK(worker_thread() == rtc::Thread::Current());
return media_channel()->RemoveRecvStream(ssrc);
}
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_channel()->RemoveSendStream(old_stream.first_ssrc())) {
rtc::StringBuilder desc;
desc << "Failed to remove send stream with ssrc "
<< old_stream.first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
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()) {
rtc::StringBuilder desc;
desc << "Failed to add send stream: " << new_stream.first_ssrc()
<< ". Stream has both SSRCs and RIDs.";
SafeSetError(desc.str(), error_desc);
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_channel()->AddSendStream(new_stream)) {
RTC_LOG(LS_INFO) << "Add send stream ssrc: " << new_stream.ssrcs[0];
} else {
rtc::StringBuilder desc;
desc << "Failed to add send stream ssrc: " << new_stream.first_ssrc();
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
local_streams_ = all_streams;
return ret;
}
bool BaseChannel::UpdateRemoteStreams_w(
const std::vector<StreamParams>& streams,
SdpType type,
std::string* error_desc) {
// Check for streams that have been removed.
bool ret = true;
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() && !HasStreamWithNoSsrcs(streams)) ||
!GetStreamBySsrc(streams, old_stream.first_ssrc())) {
if (RemoveRecvStream_w(old_stream.first_ssrc())) {
RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc();
} else {
rtc::StringBuilder desc;
desc << "Failed to remove remote stream with ssrc "
<< old_stream.first_ssrc() << ".";
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
}
demuxer_criteria_.ssrcs.clear();
// Check for new streams.
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() && !HasStreamWithNoSsrcs(remote_streams_)) ||
!GetStreamBySsrc(remote_streams_, new_stream.first_ssrc())) {
if (AddRecvStream_w(new_stream)) {
RTC_LOG(LS_INFO) << "Add remote ssrc: " << new_stream.first_ssrc();
} else {
rtc::StringBuilder desc;
desc << "Failed to add remote stream ssrc: " << new_stream.first_ssrc();
SafeSetError(desc.str(), error_desc);
ret = false;
}
}
// Update the receiving SSRCs.
demuxer_criteria_.ssrcs.insert(new_stream.ssrcs.begin(),
new_stream.ssrcs.end());
}
// Re-register the sink to update the receiving ssrcs.
RegisterRtpDemuxerSink();
remote_streams_ = streams;
return ret;
}
RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
RTC_DCHECK(rtp_transport_);
if (crypto_options_.srtp.enable_encrypted_rtp_header_extensions) {
RtpHeaderExtensions filtered;
absl::c_copy_if(extensions, std::back_inserter(filtered),
[](const webrtc::RtpExtension& extension) {
return !extension.encrypt;
});
return filtered;
}
return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions);
}
void BaseChannel::OnMessage(rtc::Message* pmsg) {
TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
switch (pmsg->message_id) {
case MSG_SEND_RTP_PACKET:
case MSG_SEND_RTCP_PACKET: {
RTC_DCHECK(network_thread_->IsCurrent());
SendPacketMessageData* data =
static_cast<SendPacketMessageData*>(pmsg->pdata);
bool rtcp = pmsg->message_id == MSG_SEND_RTCP_PACKET;
SendPacket(rtcp, &data->packet, data->options);
delete data;
break;
}
case MSG_FIRSTPACKETRECEIVED: {
SignalFirstPacketReceived_(this);
break;
}
}
}
void BaseChannel::AddHandledPayloadType(int payload_type) {
demuxer_criteria_.payload_types.insert(static_cast<uint8_t>(payload_type));
}
void BaseChannel::ClearHandledPayloadTypes() {
demuxer_criteria_.payload_types.clear();
}
void BaseChannel::FlushRtcpMessages_n() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
RTC_DCHECK(network_thread_->IsCurrent());
rtc::MessageList rtcp_messages;
network_thread_->Clear(this, MSG_SEND_RTCP_PACKET, &rtcp_messages);
for (const auto& message : rtcp_messages) {
network_thread_->Send(RTC_FROM_HERE, this, MSG_SEND_RTCP_PACKET,
message.pdata);
}
}
void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[this, sent_packet] {
RTC_DCHECK(worker_thread_->IsCurrent());
SignalSentPacket(sent_packet);
});
}
VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
srtp_required,
crypto_options,
ssrc_generator) {}
VoiceChannel::~VoiceChannel() {
if (media_transport()) {
media_transport()->SetFirstAudioPacketReceivedObserver(nullptr);
}
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void BaseChannel::UpdateMediaSendRecvState() {
RTC_DCHECK(network_thread_->IsCurrent());
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
[this] { UpdateMediaSendRecvState_w(); });
}
void BaseChannel::OnNetworkRouteChanged(
const rtc::NetworkRoute& network_route) {
OnNetworkRouteChanged(absl::make_optional(network_route));
}
void VoiceChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
BaseChannel::Init_w(rtp_transport, media_transport_config);
if (media_transport_config.media_transport) {
media_transport_config.media_transport->SetFirstAudioPacketReceivedObserver(
this);
}
}
void VoiceChannel::OnFirstAudioPacketReceived(int64_t channel_id) {
has_received_packet_ = true;
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
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 recv = IsReadyToReceiveMedia_w();
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 = IsReadyToSendMedia_w();
media_channel()->SetSend(send);
RTC_LOG(LS_INFO) << "Changing voice state, recv=" << recv << " send=" << send;
}
bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(audio->extmap_allow_mixed());
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(audio, rtp_header_extensions, &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set local audio description recv parameters.",
error_desc);
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(audio->direction())) {
for (const AudioCodec& codec : audio->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing.";
return false;
}
}
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(), type, error_desc)) {
SafeSetError("Failed to set local audio description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote voice description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
&send_params);
send_params.mid = content_name();
bool parameters_applied = media_channel()->SetSendParameters(send_params);
if (!parameters_applied) {
SafeSetError("Failed to set remote audio description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) {
RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - "
"disable payload type demuxing";
ClearHandledPayloadTypes();
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to update audio demuxing.";
return false;
}
}
// 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(), type, error_desc)) {
SafeSetError("Failed to set remote audio description streams.", error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
VideoChannel::VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
srtp_required,
crypto_options,
ssrc_generator) {}
VideoChannel::~VideoChannel() {
TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
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 send = IsReadyToSendMedia_w();
if (!media_channel()->SetSend(send)) {
RTC_LOG(LS_ERROR) << "Failed to SetSend on video channel";
// TODO(gangji): Report error back to server.
}
RTC_LOG(LS_INFO) << "Changing video state, send=" << send;
}
void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo,
media_channel(), bwe_info));
}
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local video description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(video->extmap_allow_mixed());
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(video, rtp_header_extensions, &recv_params);
VideoSendParameters send_params = last_send_params_;
bool needs_send_params_update = false;
if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
for (auto& send_codec : send_params.codecs) {
auto* recv_codec = FindMatchingCodec(recv_params.codecs, send_codec);
if (recv_codec) {
if (!recv_codec->packetization && send_codec.packetization) {
send_codec.packetization.reset();
needs_send_params_update = true;
} else if (recv_codec->packetization != send_codec.packetization) {
SafeSetError(
"Failed to set local answer due to invalid codec packetization.",
error_desc);
return false;
}
}
}
}
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set local video description recv parameters.",
error_desc);
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(video->direction())) {
for (const VideoCodec& codec : video->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up video demuxing.";
return false;
}
}
last_recv_params_ = recv_params;
if (needs_send_params_update) {
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set send parameters.", error_desc);
return false;
}
last_send_params_ = send_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(), type, error_desc)) {
SafeSetError("Failed to set local video description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote video description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
VideoSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(video, rtp_header_extensions,
&send_params);
if (video->conference_mode()) {
send_params.conference_mode = true;
}
send_params.mid = content_name();
VideoRecvParameters recv_params = last_recv_params_;
bool needs_recv_params_update = false;
if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
for (auto& recv_codec : recv_params.codecs) {
auto* send_codec = FindMatchingCodec(send_params.codecs, recv_codec);
if (send_codec) {
if (!send_codec->packetization && recv_codec.packetization) {
recv_codec.packetization.reset();
needs_recv_params_update = true;
} else if (send_codec->packetization != recv_codec.packetization) {
SafeSetError(
"Failed to set remote answer due to invalid codec packetization.",
error_desc);
return false;
}
}
}
}
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set remote video description send parameters.",
error_desc);
return false;
}
last_send_params_ = send_params;
if (needs_recv_params_update) {
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set recv parameters.", error_desc);
return false;
}
last_recv_params_ = recv_params;
}
if (!webrtc::RtpTransceiverDirectionHasSend(content->direction())) {
RTC_DLOG(LS_VERBOSE) << "SetRemoteContent_w: remote side will not send - "
"disable payload type demuxing";
ClearHandledPayloadTypes();
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to update video demuxing.";
return false;
}
}
// 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(), type, error_desc)) {
SafeSetError("Failed to set remote video description streams.", error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<DataMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
UniqueRandomIdGenerator* ssrc_generator)
: BaseChannel(worker_thread,
network_thread,
signaling_thread,
std::move(media_channel),
content_name,
srtp_required,
crypto_options,
ssrc_generator) {}
RtpDataChannel::~RtpDataChannel() {
TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
void RtpDataChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
BaseChannel::Init_w(rtp_transport, media_transport_config);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
this, &RtpDataChannel::OnDataChannelReadyToSend);
}
bool RtpDataChannel::SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
return InvokeOnWorker<bool>(
RTC_FROM_HERE, Bind(&DataMediaChannel::SendData, media_channel(), params,
payload, result));
}
bool RtpDataChannel::CheckDataChannelTypeFromContent(
const RtpDataContentDescription* content,
std::string* error_desc) {
bool is_sctp = ((content->protocol() == kMediaProtocolSctp) ||
(content->protocol() == kMediaProtocolDtlsSctp));
// It's been set before, but doesn't match. That's bad.
if (is_sctp) {
SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.",
error_desc);
return false;
}
return true;
}
bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetLocalContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local data description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in local description.", error_desc);
return false;
}
const RtpDataContentDescription* data = content->as_rtp_data();
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
DataRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(data, rtp_header_extensions, &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError("Failed to set remote data description recv parameters.",
error_desc);
return false;
}
for (const DataCodec& codec : data->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up data demuxing.";
return false;
}
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(), type, error_desc)) {
SafeSetError("Failed to set local data description streams.", error_desc);
return false;
}
set_local_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "RtpDataChannel::SetRemoteContent_w");
RTC_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote data description";
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in remote description.", error_desc);
return false;
}
const RtpDataContentDescription* data = content->as_rtp_data();
if (!data) {
RTC_LOG(LS_INFO) << "Accepting and ignoring non-RTP content description";
return true;
}
// If the remote data doesn't have codecs, it must be empty, so ignore it.
if (!data->has_codecs()) {
return true;
}
if (!CheckDataChannelTypeFromContent(data, error_desc)) {
return false;
}
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
RTC_LOG(LS_INFO) << "Setting remote data description";
DataSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription<DataCodec>(data, rtp_header_extensions,
&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(), type, error_desc)) {
SafeSetError("Failed to set remote data description streams.", error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
void RtpDataChannel::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 recv = IsReadyToReceiveMedia_w();
if (!media_channel()->SetReceive(recv)) {
RTC_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 = IsReadyToSendMedia_w();
if (!media_channel()->SetSend(send)) {
RTC_LOG(LS_ERROR) << "Failed to SetSend on data channel";
}
// Trigger SignalReadyToSendData asynchronously.
OnDataChannelReadyToSend(send);
RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send;
}
void RtpDataChannel::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(data->params, data->payload);
delete data;
break;
}
default:
BaseChannel::OnMessage(pmsg);
break;
}
}
void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params,
const char* data,
size_t len) {
DataReceivedMessageData* msg = new DataReceivedMessageData(params, data, len);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg);
}
void RtpDataChannel::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(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
new DataChannelReadyToSendMessageData(writable));
}
} // namespace cricket
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