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a743303211b89bbcf4cea438ee797bbbc7b59e80
platform-external-webrtc/pc/channel.cc
Lennart Grahl a743303211 Fix RTP header extension encryption 
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.

In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.

Further changes:

- If RTP header encryption enabled, prefer encrypted extensions over
  non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
  is not supported for that extension

Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.

Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
2021-04-14 08:53:45 +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;
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,
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();
signaling_thread_->Clear(this);
// 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_);
if (!RegisterRtpDemuxerSink_n()) {
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_);
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());
media_channel_->SetInterface(/*iface=*/nullptr);
// 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());
FlushRtcpMessages_n();
if (rtp_transport_) {
DisconnectFromRtpTransport();
}
// Clear pending read packets/messages.
network_thread_->Clear(this);
});
}
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;
}
bool BaseChannel::Enable(bool enable) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [this, enable] {
RTC_DCHECK_RUN_ON(worker_thread());
if (enable) {
EnableMedia_w();
} else {
DisableMedia_w();
}
});
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, [this, content, type, error_desc] {
RTC_DCHECK_RUN_ON(worker_thread());
return SetLocalContent_w(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, [this, content, type, error_desc] {
RTC_DCHECK_RUN_ON(worker_thread());
return SetRemoteContent_w(content, type, error_desc);
});
}
bool BaseChannel::SetPayloadTypeDemuxingEnabled(bool enabled) {
TRACE_EVENT0("webrtc", "BaseChannel::SetPayloadTypeDemuxingEnabled");
return InvokeOnWorker<bool>(RTC_FROM_HERE, [this, enabled] {
RTC_DCHECK_RUN_ON(worker_thread());
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) {
return network_thread_->Invoke<int>(RTC_FROM_HERE, [this, type, opt, value] {
RTC_DCHECK_RUN_ON(network_thread());
return SetOption_n(type, opt, value);
});
}
int BaseChannel::SetOption_n(SocketType type,
rtc::Socket::Option opt,
int value) {
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);
}
sigslot::signal1<ChannelInterface*>& BaseChannel::SignalFirstPacketReceived() {
RTC_DCHECK_RUN_ON(signaling_thread_);
return SignalFirstPacketReceived_;
}
sigslot::signal1<const rtc::SentPacket&>& BaseChannel::SignalSentPacket() {
// TODO(bugs.webrtc.org/11994): Uncomment this check once callers have been
// fixed to access this variable from the correct thread.
// RTC_DCHECK_RUN_ON(worker_thread_);
return SignalSentPacket_;
}
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) {
// 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;
}
RTC_DCHECK_RUN_ON(network_thread());
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) {
// 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 "
<< ToString();
return;
}
media_channel_->OnPacketReceived(parsed_packet.Buffer(), 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() {
// 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_);
return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria, this);
});
}
bool BaseChannel::RegisterRtpDemuxerSink_n() {
RTC_DCHECK(rtp_transport_);
// TODO(bugs.webrtc.org/12230): This accesses demuxer_criteria_ on the
// networking thread.
return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
}
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::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::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_RUN_ON(network_thread());
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: {
RTC_DCHECK_RUN_ON(signaling_thread_);
SignalFirstPacketReceived_(this);
break;
}
}
}
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::FlushRtcpMessages_n() {
// Flush all remaining RTCP messages. This should only be called in
// destructor.
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) {
worker_thread_->PostTask(ToQueuedTask(alive_, [this, sent_packet] {
RTC_DCHECK_RUN_ON(worker_thread());
SignalSentPacket()(sent_packet);
}));
}
void BaseChannel::SetNegotiatedHeaderExtensions_w(
const RtpHeaderExtensions& extensions) {
TRACE_EVENT0("webrtc", __func__);
webrtc::MutexLock lock(&negotiated_header_extensions_lock_);
negotiated_header_extensions_ = extensions;
}
RtpHeaderExtensions BaseChannel::GetNegotiatedRtpHeaderExtensions() const {
RTC_DCHECK_RUN_ON(signaling_thread());
webrtc::MutexLock lock(&negotiated_header_extensions_lock_);
return negotiated_header_extensions_;
}
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();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in local description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(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,
webrtc::RtpTransceiverDirectionHasRecv(audio->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(audio->direction())) {
for (const AudioCodec& codec : 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(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();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find audio content in remote description.", error_desc);
return false;
}
const AudioContentDescription* audio = content->as_audio();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(audio->rtp_header_extensions());
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.
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) {
VideoMediaChannel* mc = media_channel();
InvokeOnWorker<void>(RTC_FROM_HERE,
[mc, bwe_info] { 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();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in local description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions());
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(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,
webrtc::RtpTransceiverDirectionHasRecv(video->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(video->direction())) {
for (const VideoCodec& codec : 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(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();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find video content in remote description.", error_desc);
return false;
}
const VideoContentDescription* video = content->as_video();
if (type == SdpType::kAnswer)
SetNegotiatedHeaderExtensions_w(video->rtp_header_extensions());
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;
}
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) {
BaseChannel::Init_w(rtp_transport);
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) {
DataMediaChannel* mc = media_channel();
return InvokeOnWorker<bool>(RTC_FROM_HERE, [mc, &params, &payload, result] {
return mc->SendData(params, payload, result);
});
}
bool RtpDataChannel::CheckDataChannelTypeFromContent(
const MediaContentDescription* content,
std::string* error_desc) {
if (!content->as_rtp_data()) {
if (content->as_sctp()) {
SafeSetError("Data channel type mismatch. Expected RTP, got SCTP.",
error_desc);
} else {
SafeSetError("Data channel is not RTP or 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 for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in local description.", error_desc);
return false;
}
if (!CheckDataChannelTypeFromContent(content, error_desc)) {
return false;
}
const RtpDataContentDescription* data = content->as_rtp_data();
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(data->rtp_header_extensions());
DataRecvParameters recv_params = last_recv_params_;
RtpParametersFromMediaDescription(
data, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &recv_params);
if (!media_channel()->SetRecvParameters(recv_params)) {
SafeSetError(
"Failed to set remote data description recv parameters for m-section "
"with mid='" +
content_name() + "'.",
error_desc);
return false;
}
for (const DataCodec& codec : data->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 data demuxing for " << ToString();
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 for m-section with "
"mid='" +
content_name() + "'.",
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 for " << ToString();
RTC_DCHECK(content);
if (!content) {
SafeSetError("Can't find data content in remote description.", error_desc);
return false;
}
if (!CheckDataChannelTypeFromContent(content, error_desc)) {
return false;
}
const RtpDataContentDescription* data = content->as_rtp_data();
// If the remote data doesn't have codecs, it must be empty, so ignore it.
if (!data->has_codecs()) {
return true;
}
RtpHeaderExtensions rtp_header_extensions =
GetDeduplicatedRtpHeaderExtensions(data->rtp_header_extensions());
RTC_LOG(LS_INFO) << "Setting remote data description for " << ToString();
DataSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription<DataCodec>(
data, rtp_header_extensions,
webrtc::RtpTransceiverDirectionHasRecv(data->direction()), &send_params);
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError(
"Failed to set remote data description send parameters for m-section "
"with mid='" +
content_name() + "'.",
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 for m-section with "
"mid='" +
content_name() + "'.",
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.
RTC_DCHECK_RUN_ON(worker_thread());
bool recv = IsReadyToReceiveMedia_w();
if (!media_channel()->SetReceive(recv)) {
RTC_LOG(LS_ERROR) << "Failed to SetReceive on data channel: " << ToString();
}
// 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: " << ToString();
}
// Trigger SignalReadyToSendData asynchronously.
OnDataChannelReadyToSend(send);
RTC_LOG(LS_INFO) << "Changing data state, recv=" << recv << " send=" << send
<< " for " << ToString();
}
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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