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webrtc_m130/pc/channel.cc
Tomas Gunnarsson b496c32901 Enhance thread checks in BaseChannel classes. 
Improve consistency between using DCHECK checkers and compile time.
For virtual methods, we were sometimes using both and in other cases
we could be using compile time checks but were using runtime.

Added annotation for last_send_params_, last_recv_params_ in
audio/video channel classes.

Also removing redundant logging for when registration with the
transport fails. This is already being logged in the demuxer.

Bug: webrtc:12230
Change-Id: I48e156c9996dec26a990151301dabc06673541d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244095
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35630}
2022-01-05 10:35:44 +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 <algorithm>
#include <cstdint>
#include <iterator>
#include <map>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/rtp_parameters.h"
#include "api/sequence_checker.h"
#include "api/task_queue/queued_task.h"
#include "media/base/codec.h"
#include "media/base/rid_description.h"
#include "media/base/rtp_utils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "pc/rtp_media_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/network_route.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/trace_event.h"
namespace cricket {
namespace {
using ::rtc::UniqueRandomIdGenerator;
using ::webrtc::PendingTaskSafetyFlag;
using ::webrtc::SdpType;
using ::webrtc::ToQueuedTask;
// 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
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,
bool is_stream_active,
RtpParameters<Codec>* params) {
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();
}
template <class Codec>
void RtpSendParametersFromMediaDescription(
const MediaContentDescriptionImpl<Codec>* desc,
const RtpHeaderExtensions& extensions,
bool is_stream_active,
RtpSendParameters<Codec>* send_params) {
RtpParametersFromMediaDescription(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(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),
alive_(PendingTaskSafetyFlag::Create()),
srtp_required_(srtp_required),
crypto_options_(crypto_options),
media_channel_(std::move(media_channel)),
demuxer_criteria_(content_name),
ssrc_generator_(ssrc_generator) {
RTC_DCHECK_RUN_ON(worker_thread_);
RTC_DCHECK(ssrc_generator_);
RTC_LOG(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 {
rtc::StringBuilder sb;
sb << "{mid: " << content_name();
if (media_channel_) {
sb << ", media_type: " << MediaTypeToString(media_channel_->media_type());
}
sb << "}";
return sb.Release();
}
bool BaseChannel::ConnectToRtpTransport_n() {
RTC_DCHECK(rtp_transport_);
RTC_DCHECK(media_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_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&BaseChannel::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this,
&BaseChannel::SignalSentPacket_n);
return true;
}
void BaseChannel::DisconnectFromRtpTransport_n() {
RTC_DCHECK(rtp_transport_);
RTC_DCHECK(media_channel());
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) {
RTC_DCHECK_RUN_ON(worker_thread());
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);
});
}
void BaseChannel::Deinit() {
RTC_DCHECK_RUN_ON(worker_thread());
// 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, [&] {
RTC_DCHECK_RUN_ON(network_thread());
media_channel_->SetInterface(/*iface=*/nullptr);
if (rtp_transport_) {
DisconnectFromRtpTransport_n();
}
});
}
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();
}
rtp_transport_ = rtp_transport;
if (rtp_transport_) {
if (!ConnectToRtpTransport_n()) {
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;
}
void BaseChannel::Enable(bool enable) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (enable == enabled_s_)
return;
enabled_s_ = enable;
worker_thread_->PostTask(ToQueuedTask(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 an Invoke 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(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());
if (writable) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void BaseChannel::OnNetworkRouteChanged(
absl::optional<rtc::NetworkRoute> network_route) {
RTC_LOG(LS_INFO) << "Network route changed for " << ToString();
RTC_DCHECK_RUN_ON(network_thread());
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_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);
on_first_packet_received_ = std::move(callback);
}
void BaseChannel::OnTransportReadyToSend(bool ready) {
RTC_DCHECK_RUN_ON(network_thread());
media_channel_->OnReadyToSend(ready);
}
bool BaseChannel::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) {
RTC_DCHECK_RUN_ON(network_thread());
// 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.
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 " << 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
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 for " << ToString()
<< " when SRTP is inactive and crypto is required";
RTC_DCHECK_NOTREACHED();
return false;
}
std::string packet_type = rtcp ? "RTCP" : "RTP";
RTC_DLOG(LS_WARNING) << "Sending an " << packet_type
<< " packet without encryption for " << ToString()
<< ".";
}
// 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) {
RTC_DCHECK_RUN_ON(network_thread());
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;
}
webrtc::Timestamp packet_time = parsed_packet.arrival_time();
media_channel_->OnPacketReceived(
parsed_packet.Buffer(),
packet_time.IsMinusInfinity() ? -1 : packet_time.us());
}
void BaseChannel::UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions) {
// Update the header extension map on network thread in case there is data
// race.
//
// 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] {
RTC_DCHECK_RUN_ON(network_thread());
rtp_transport_->UpdateRtpHeaderExtensionMap(header_extensions);
});
}
bool BaseChannel::RegisterRtpDemuxerSink_w() {
media_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_->Invoke<bool>(
RTC_FROM_HERE, [this, demuxer_criteria = demuxer_criteria_] {
RTC_DCHECK_RUN_ON(network_thread());
// 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_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(ToQueuedTask(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::AddRecvStream_w(const StreamParams& sp) {
return media_channel()->AddRecvStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
return media_channel()->RemoveRecvStream(ssrc);
}
void BaseChannel::ResetUnsignaledRecvStream_w() {
media_channel()->ResetUnsignaledRecvStream();
}
bool BaseChannel::SetPayloadTypeDemuxingEnabled_w(bool enabled) {
if (enabled == payload_type_demuxing_enabled_) {
return true;
}
payload_type_demuxing_enabled_ = enabled;
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_channel()->ResetUnsignaledRecvStream();
demuxer_criteria_.payload_types().clear();
} else if (!payload_types_.empty()) {
// TODO(tommi): Instead of 'insert', should this simply overwrite the value
// of the criteria?
demuxer_criteria_.payload_types().insert(payload_types_.begin(),
payload_types_.end());
}
// 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_channel()->RemoveSendStream(old_stream.first_ssrc())) {
rtc::StringBuilder desc;
desc << "Failed to remove send stream with ssrc "
<< old_stream.first_ssrc() << " from m-section with mid='"
<< content_name() << "'.";
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()
<< " into m-section with mid='" << content_name()
<< "'. 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]
<< " into " << ToString();
} else {
rtc::StringBuilder desc;
desc << "Failed to add send stream ssrc: " << new_stream.first_ssrc()
<< " into m-section with mid='" << content_name() << "'";
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)) {
ResetUnsignaledRecvStream_w();
RTC_LOG(LS_INFO) << "Reset unsignaled remote stream for " << ToString()
<< ".";
} else if (old_stream.has_ssrcs() &&
!GetStreamBySsrc(streams, old_stream.first_ssrc())) {
if (RemoveRecvStream_w(old_stream.first_ssrc())) {
RTC_LOG(LS_INFO) << "Remove remote ssrc: " << old_stream.first_ssrc()
<< " from " << ToString() << ".";
} else {
rtc::StringBuilder desc;
desc << "Failed to remove remote stream with ssrc "
<< old_stream.first_ssrc() << " from m-section with mid='"
<< content_name() << "'.";
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.has_ssrcs()
? std::to_string(new_stream.first_ssrc())
: "unsignaled")
<< " to " << ToString();
} else {
rtc::StringBuilder desc;
desc << "Failed to add remote stream ssrc: "
<< (new_stream.has_ssrcs()
? std::to_string(new_stream.first_ssrc())
: "unsignaled")
<< " to " << ToString();
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.
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString();
ret = false;
}
remote_streams_ = streams;
return ret;
}
RtpHeaderExtensions BaseChannel::GetDeduplicatedRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
return webrtc::RtpExtension::DeduplicateHeaderExtensions(
extensions, crypto_options_.srtp.enable_encrypted_rtp_header_extensions
? webrtc::RtpExtension::kPreferEncryptedExtension
: webrtc::RtpExtension::kDiscardEncryptedExtension);
}
void BaseChannel::MaybeAddHandledPayloadType(int payload_type) {
if (payload_type_demuxing_enabled_) {
demuxer_criteria_.payload_types().insert(
static_cast<uint8_t>(payload_type));
}
// 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));
}
void BaseChannel::ClearHandledPayloadTypes() {
demuxer_criteria_.payload_types().clear();
payload_types_.clear();
}
void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(network_thread());
media_channel()->OnPacketSent(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() {
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
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 ready_to_receive = enabled() && webrtc::RtpTransceiverDirectionHasRecv(
local_content_direction());
media_channel()->SetPlayout(ready_to_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_channel()->SetSend(send);
RTC_LOG(LS_INFO) << "Changing voice state, recv=" << ready_to_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_LOG(LS_INFO) << "Setting local voice description for " << ToString();
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
// TODO(tommi): There's a hop to the network thread here.
// some of the below is also network thread related.
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
AudioRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
content->as_audio(), rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError(
"Failed to set local audio description recv parameters for m-section "
"with mid='" +
content_name() + "'.",
error_desc);
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) {
for (const AudioCodec& codec : content->as_audio()->codecs()) {
MaybeAddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing for " << ToString();
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(content->as_audio()->streams(), type, error_desc)) {
SafeSetError(
"Failed to set local audio description streams for m-section with "
"mid='" +
content_name() + "'.",
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_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
const AudioContentDescription* audio = content->as_audio();
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(audio->rtp_header_extensions());
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(
audio, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(audio->direction()), &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 for m-section "
"with mid='" +
content_name() + "'.",
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 for "
<< ToString();
ClearHandledPayloadTypes();
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to update audio demuxing for " << ToString();
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 for m-section with "
"mid='" +
content_name() + "'.",
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: " + ToString();
// TODO(gangji): Report error back to server.
}
RTC_LOG(LS_INFO) << "Changing video state, send=" << send << " for "
<< ToString();
}
void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
RTC_DCHECK_RUN_ON(worker_thread());
VideoMediaChannel* mc = media_channel();
mc->FillBitrateInfo(bwe_info);
}
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
SdpType type,
std::string* error_desc) {
TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
RTC_LOG(LS_INFO) << "Setting local video description for " << ToString();
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(content->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
media_channel()->SetExtmapAllowMixed(content->extmap_allow_mixed());
VideoRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
content->as_video(), rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
&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 "
"specified in m-section with mid='" +
content_name() + "'.",
error_desc);
return false;
}
}
}
}
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError(
"Failed to set local video description recv parameters for m-section "
"with mid='" +
content_name() + "'.",
error_desc);
return false;
}
if (webrtc::RtpTransceiverDirectionHasRecv(content->direction())) {
for (const VideoCodec& codec : content->as_video()->codecs()) {
MaybeAddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to set up video demuxing for " << ToString();
return false;
}
}
last_recv_params_ = recv_params;
if (needs_send_params_update) {
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set send parameters for m-section with mid='" +
content_name() + "'.",
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(content->as_video()->streams(), type, error_desc)) {
SafeSetError(
"Failed to set local video description streams for m-section with "
"mid='" +
content_name() + "'.",
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_LOG(LS_INFO) << "Setting remote video description for " << ToString();
const VideoContentDescription* video = content->as_video();
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(video->rtp_header_extensions());
VideoSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(
video, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(video->direction()), &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 "
"specifid in m-section with mid='" +
content_name() + "'.",
error_desc);
return false;
}
}
}
}
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError(
"Failed to set remote video description send parameters for m-section "
"with mid='" +
content_name() + "'.",
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 for m-section with mid='" +
content_name() + "'.",
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 for "
<< ToString();
ClearHandledPayloadTypes();
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to update video demuxing for " << ToString();
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 for m-section with "
"mid='" +
content_name() + "'.",
error_desc);
return false;
}
set_remote_content_direction(content->direction());
UpdateMediaSendRecvState_w();
return true;
}
} // namespace cricket
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