TeamPGM/Nagram
2
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
ab221dafad
Nagram/TMessagesProj/jni/voip/webrtc/pc/channel.cc
DrKLO e8d88e56e4 Update to 7.8.0 (2359)
2021-06-25 03:43:10 +03:00

1169 lines
42 KiB
C++
Raw Blame History

/*
* 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()),
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: " << 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() {
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.
bool result = rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
if (result) {
previous_demuxer_criteria_ = demuxer_criteria_;
} else {
previous_demuxer_criteria_ = {};
RTC_LOG(LS_ERROR) << "Failed to set up demuxing for " << ToString();
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() {
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();
}
});
}
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
RTC_DCHECK_RUN_ON(network_thread());
if (rtp_transport == rtp_transport_) {
return true;
}
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 for "
<< ToString() << ".";
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) {
RTC_DCHECK_RUN_ON(worker_thread());
TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
return SetPayloadTypeDemuxingEnabled_w(enabled);
}
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 {
// 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_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() {
if (demuxer_criteria_ == previous_demuxer_criteria_) {
return true;
}
media_channel_->OnDemuxerCriteriaUpdatePending();
// Copy demuxer criteria, since they're a worker-thread variable
// and we want to pass them to the network thread
return network_thread_->Invoke<bool>(
RTC_FROM_HERE, [this, demuxer_criteria = demuxer_criteria_] {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(rtp_transport_);
bool result =
rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria, this);
if (result) {
previous_demuxer_criteria_ = demuxer_criteria;
} else {
previous_demuxer_criteria_ = {};
}
media_channel_->OnDemuxerCriteriaUpdateComplete();
return result;
});
}
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() {
if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
rtp_transport_->IsWritable(/*rtcp=*/false)) {
ChannelWritable_n();
} else {
ChannelNotWritable_n();
}
}
void 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() {
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();
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to disable payload type demuxing for "
<< ToString();
return false;
}
} else if (!payload_types_.empty()) {
demuxer_criteria_.payload_types.insert(payload_types_.begin(),
payload_types_.end());
if (!RegisterRtpDemuxerSink_w()) {
RTC_LOG(LS_ERROR) << "Failed to enable payload type demuxing for "
<< ToString();
return false;
}
}
return true;
}
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::GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions) {
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::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.
RTC_DCHECK_RUN_ON(worker_thread());
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
<< " for " << ToString();
}
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 for " << ToString();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(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_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
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,
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.
RTC_DCHECK_RUN_ON(worker_thread());
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_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting local video description for " << ToString();
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(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_DCHECK_RUN_ON(worker_thread());
RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString();
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,
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
Reference in New Issue View Git Blame Copy Permalink