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/*
* Copyright 2017 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/rtp_transceiver.h"
#include <stdint.h>
#include <algorithm>
#include <cstddef>
#include <functional>
#include <iterator>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/field_trials_view.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video_codecs/scalability_mode.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
#include "pc/channel_interface.h"
#include "pc/connection_context.h"
#include "pc/rtp_media_utils.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_receiver_proxy.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_sender_proxy.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread.h"
namespace webrtc {
namespace {
RTCError VerifyCodecPreferences(
const std::vector<RtpCodecCapability>& unfiltered_codecs,
const std::vector<cricket::Codec>& recv_codecs,
const FieldTrialsView& field_trials) {
// If the intersection between codecs and
// RTCRtpReceiver.getCapabilities(kind).codecs only contains RTX, RED, FEC
// codecs or Comfort Noise codecs or is an empty set, throw
// InvalidModificationError.
// This ensures that we always have something to offer, regardless of
// transceiver.direction.
// TODO(fippo): clean up the filtering killswitch
std::vector<RtpCodecCapability> codecs = unfiltered_codecs;
if (!absl::c_any_of(codecs, [&recv_codecs](const RtpCodecCapability& codec) {
return codec.IsMediaCodec() &&
absl::c_any_of(recv_codecs,
[&codec](const cricket::Codec& recv_codec) {
return recv_codec.MatchesRtpCodec(codec);
});
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Invalid codec preferences: Missing codec from recv "
"codec capabilities.");
}
// Let codecCapabilities RTCRtpReceiver.getCapabilities(kind).codecs.
// For each codec in codecs, If
// codec is not in codecCapabilities, throw InvalidModificationError.
for (const auto& codec_preference : codecs) {
bool is_recv_codec = absl::c_any_of(
recv_codecs, [&codec_preference](const cricket::Codec& codec) {
return codec.MatchesRtpCodec(codec_preference);
});
if (!is_recv_codec) {
if (!field_trials.IsDisabled(
"WebRTC-SetCodecPreferences-ReceiveOnlyFilterInsteadOfThrow")) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
std::string(
"Invalid codec preferences: invalid codec with name \"") +
codec_preference.name + "\".");
} else {
// Killswitch behavior: filter out any codec not in receive codecs.
codecs.erase(std::remove_if(
codecs.begin(), codecs.end(),
[&recv_codecs](const RtpCodecCapability& codec) {
return codec.IsMediaCodec() &&
!absl::c_any_of(
recv_codecs,
[&codec](const cricket::Codec& recv_codec) {
return recv_codec.MatchesRtpCodec(codec);
});
}));
}
}
}
// Check we have a real codec (not just rtx, red, fec or CN)
if (absl::c_all_of(codecs, [](const RtpCodecCapability& codec) {
return !codec.IsMediaCodec();
})) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_MODIFICATION,
"Invalid codec preferences: codec list must have a non "
"RTX, RED or FEC entry.");
}
return RTCError::OK();
}
TaskQueueBase* GetCurrentTaskQueueOrThread() {
TaskQueueBase* current = TaskQueueBase::Current();
if (!current)
current = rtc::ThreadManager::Instance()->CurrentThread();
return current;
}
} // namespace
RtpTransceiver::RtpTransceiver(cricket::MediaType media_type,
ConnectionContext* context)
: thread_(GetCurrentTaskQueueOrThread()),
unified_plan_(false),
media_type_(media_type),
context_(context) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
}
RtpTransceiver::RtpTransceiver(
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver,
ConnectionContext* context,
std::vector<RtpHeaderExtensionCapability> header_extensions_to_negotiate,
std::function<void()> on_negotiation_needed)
: thread_(GetCurrentTaskQueueOrThread()),
unified_plan_(true),
media_type_(sender->media_type()),
context_(context),
header_extensions_to_negotiate_(
std::move(header_extensions_to_negotiate)),
on_negotiation_needed_(std::move(on_negotiation_needed)) {
RTC_DCHECK(media_type_ == cricket::MEDIA_TYPE_AUDIO ||
media_type_ == cricket::MEDIA_TYPE_VIDEO);
RTC_DCHECK_EQ(sender->media_type(), receiver->media_type());
sender->internal()->SetSendCodecs(
sender->media_type() == cricket::MEDIA_TYPE_VIDEO
? media_engine()->video().send_codecs(false)
: media_engine()->voice().send_codecs());
senders_.push_back(sender);
receivers_.push_back(receiver);
// Set default header extensions depending on whether simulcast/SVC is used.
RtpParameters parameters = sender->internal()->GetParametersInternal();
bool uses_simulcast = parameters.encodings.size() > 1;
bool uses_svc = !parameters.encodings.empty() &&
parameters.encodings[0].scalability_mode.has_value() &&
parameters.encodings[0].scalability_mode !=
ScalabilityModeToString(ScalabilityMode::kL1T1);
if (uses_simulcast || uses_svc) {
// Enable DD and VLA extensions, can be deactivated by the API.
// Skip this if the GFD extension was enabled via field trial
// for backward compability reasons.
bool uses_gfd =
absl::c_find_if(
header_extensions_to_negotiate_,
[](const RtpHeaderExtensionCapability& ext) {
return ext.uri == RtpExtension::kGenericFrameDescriptorUri00 &&
ext.direction != webrtc::RtpTransceiverDirection::kStopped;
}) != header_extensions_to_negotiate_.end();
if (!uses_gfd) {
for (RtpHeaderExtensionCapability& ext :
header_extensions_to_negotiate_) {
if (ext.uri == RtpExtension::kVideoLayersAllocationUri ||
ext.uri == RtpExtension::kDependencyDescriptorUri) {
ext.direction = RtpTransceiverDirection::kSendRecv;
}
}
}
}
}
RtpTransceiver::~RtpTransceiver() {
// TODO(tommi): On Android, when running PeerConnectionClientTest (e.g.
// PeerConnectionClientTest#testCameraSwitch), the instance doesn't get
// deleted on `thread_`. See if we can fix that.
if (!stopped_) {
RTC_DCHECK_RUN_ON(thread_);
StopInternal();
}
RTC_CHECK(!channel_) << "Missing call to ClearChannel?";
}
RTCError RtpTransceiver::CreateChannel(
absl::string_view mid,
Call* call_ptr,
const cricket::MediaConfig& media_config,
bool srtp_required,
CryptoOptions crypto_options,
const cricket::AudioOptions& audio_options,
const cricket::VideoOptions& video_options,
VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
std::function<RtpTransportInternal*(absl::string_view)> transport_lookup) {
RTC_DCHECK_RUN_ON(thread_);
if (!media_engine()) {
// TODO(hta): Must be a better way
return RTCError(RTCErrorType::INTERNAL_ERROR,
"No media engine for mid=" + std::string(mid));
}
std::unique_ptr<cricket::ChannelInterface> new_channel;
if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
// simply be on the worker thread and use `call_` (update upstream code).
RTC_DCHECK(call_ptr);
RTC_DCHECK(media_engine());
// TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
// PeerConnection and add the expectation that we're already on the right
// thread.
context()->worker_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(context()->worker_thread());
AudioCodecPairId codec_pair_id = AudioCodecPairId::Create();
std::unique_ptr<cricket::VoiceMediaSendChannelInterface>
media_send_channel = media_engine()->voice().CreateSendChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_send_channel) {
// TODO(bugs.webrtc.org/14912): Consider CHECK or reporting failure
return;
}
std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface>
media_receive_channel = media_engine()->voice().CreateReceiveChannel(
call_ptr, media_config, audio_options, crypto_options,
codec_pair_id);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel =
media_receive_channel.get()](const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VoiceChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required, crypto_options,
context()->ssrc_generator());
});
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type());
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to
// the worker thread. We shouldn't be using the `call_ptr_` hack here but
// simply be on the worker thread and use `call_` (update upstream code).
context()->worker_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(context()->worker_thread());
std::unique_ptr<cricket::VideoMediaSendChannelInterface>
media_send_channel = media_engine()->video().CreateSendChannel(
call_ptr, media_config, video_options, crypto_options,
video_bitrate_allocator_factory);
if (!media_send_channel) {
return;
}
std::unique_ptr<cricket::VideoMediaReceiveChannelInterface>
media_receive_channel = media_engine()->video().CreateReceiveChannel(
call_ptr, media_config, video_options, crypto_options);
if (!media_receive_channel) {
return;
}
// Note that this is safe because both sending and
// receiving channels will be deleted at the same time.
media_send_channel->SetSsrcListChangedCallback(
[receive_channel =
media_receive_channel.get()](const std::set<uint32_t>& choices) {
receive_channel->ChooseReceiverReportSsrc(choices);
});
new_channel = std::make_unique<cricket::VideoChannel>(
context()->worker_thread(), context()->network_thread(),
context()->signaling_thread(), std::move(media_send_channel),
std::move(media_receive_channel), mid, srtp_required, crypto_options,
context()->ssrc_generator());
});
}
if (!new_channel) {
// TODO(hta): Must be a better way
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Failed to create channel for mid=" + std::string(mid));
}
SetChannel(std::move(new_channel), transport_lookup);
return RTCError::OK();
}
void RtpTransceiver::SetChannel(
std::unique_ptr<cricket::ChannelInterface> channel,
std::function<RtpTransportInternal*(const std::string&)> transport_lookup) {
RTC_DCHECK_RUN_ON(thread_);
RTC_DCHECK(channel);
RTC_DCHECK(transport_lookup);
RTC_DCHECK(!channel_);
// Cannot set a channel on a stopped transceiver.
if (stopped_) {
return;
}
RTC_LOG_THREAD_BLOCK_COUNT();
RTC_DCHECK_EQ(media_type(), channel->media_type());
signaling_thread_safety_ = PendingTaskSafetyFlag::Create();
std::unique_ptr<cricket::ChannelInterface> channel_to_delete;
// An alternative to this, could be to require SetChannel to be called
// on the network thread. The channel object operates for the most part
// on the network thread, as part of its initialization being on the network
// thread is required, so setting a channel object as part of the construction
// (without thread hopping) might be the more efficient thing to do than
// how SetChannel works today.
// Similarly, if the channel() accessor is limited to the network thread, that
// helps with keeping the channel implementation requirements being met and
// avoids synchronization for accessing the pointer or network related state.
context()->network_thread()->BlockingCall([&]() {
if (channel_) {
channel_->SetFirstPacketReceivedCallback(nullptr);
channel_->SetRtpTransport(nullptr);
channel_to_delete = std::move(channel_);
}
channel_ = std::move(channel);
channel_->SetRtpTransport(transport_lookup(channel_->mid()));
channel_->SetFirstPacketReceivedCallback(
[thread = thread_, flag = signaling_thread_safety_, this]() mutable {
thread->PostTask(
SafeTask(std::move(flag), [this]() { OnFirstPacketReceived(); }));
});
});
PushNewMediaChannelAndDeleteChannel(nullptr);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void RtpTransceiver::ClearChannel() {
RTC_DCHECK_RUN_ON(thread_);
if (!channel_) {
return;
}
RTC_LOG_THREAD_BLOCK_COUNT();
if (channel_) {
signaling_thread_safety_->SetNotAlive();
signaling_thread_safety_ = nullptr;
}
std::unique_ptr<cricket::ChannelInterface> channel_to_delete;
context()->network_thread()->BlockingCall([&]() {
if (channel_) {
channel_->SetFirstPacketReceivedCallback(nullptr);
channel_->SetRtpTransport(nullptr);
channel_to_delete = std::move(channel_);
}
});
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
PushNewMediaChannelAndDeleteChannel(std::move(channel_to_delete));
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void RtpTransceiver::PushNewMediaChannelAndDeleteChannel(
std::unique_ptr<cricket::ChannelInterface> channel_to_delete) {
// The clumsy combination of pushing down media channel and deleting
// the channel is due to the desire to do both things in one Invoke().
if (!channel_to_delete && senders_.empty() && receivers_.empty()) {
return;
}
context()->worker_thread()->BlockingCall([&]() {
// Push down the new media_channel, if any, otherwise clear it.
auto* media_send_channel =
channel_ ? channel_->media_send_channel() : nullptr;
for (const auto& sender : senders_) {
sender->internal()->SetMediaChannel(media_send_channel);
}
auto* media_receive_channel =
channel_ ? channel_->media_receive_channel() : nullptr;
for (const auto& receiver : receivers_) {
receiver->internal()->SetMediaChannel(media_receive_channel);
}
// Destroy the channel, if we had one, now _after_ updating the receivers
// who might have had references to the previous channel.
if (channel_to_delete) {
channel_to_delete.reset(nullptr);
}
});
}
void RtpTransceiver::AddSender(
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender) {
RTC_DCHECK_RUN_ON(thread_);
RTC_DCHECK(!stopped_);
RTC_DCHECK(!unified_plan_);
RTC_DCHECK(sender);
RTC_DCHECK_EQ(media_type(), sender->media_type());
RTC_DCHECK(!absl::c_linear_search(senders_, sender));
std::vector<cricket::Codec> send_codecs =
media_type() == cricket::MEDIA_TYPE_VIDEO
? media_engine()->video().send_codecs(false)
: media_engine()->voice().send_codecs();
sender->internal()->SetSendCodecs(send_codecs);
senders_.push_back(sender);
}
bool RtpTransceiver::RemoveSender(RtpSenderInterface* sender) {
RTC_DCHECK(!unified_plan_);
if (sender) {
RTC_DCHECK_EQ(media_type(), sender->media_type());
}
auto it = absl::c_find(senders_, sender);
if (it == senders_.end()) {
return false;
}
(*it)->internal()->Stop();
senders_.erase(it);
return true;
}
void RtpTransceiver::AddReceiver(
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver) {
RTC_DCHECK_RUN_ON(thread_);
RTC_DCHECK(!stopped_);
RTC_DCHECK(!unified_plan_);
RTC_DCHECK(receiver);
RTC_DCHECK_EQ(media_type(), receiver->media_type());
RTC_DCHECK(!absl::c_linear_search(receivers_, receiver));
receivers_.push_back(receiver);
}
bool RtpTransceiver::RemoveReceiver(RtpReceiverInterface* receiver) {
RTC_DCHECK_RUN_ON(thread_);
RTC_DCHECK(!unified_plan_);
if (receiver) {
RTC_DCHECK_EQ(media_type(), receiver->media_type());
}
auto it = absl::c_find(receivers_, receiver);
if (it == receivers_.end()) {
return false;
}
(*it)->internal()->Stop();
context()->worker_thread()->BlockingCall([&]() {
// `Stop()` will clear the receiver's pointer to the media channel.
(*it)->internal()->SetMediaChannel(nullptr);
});
receivers_.erase(it);
return true;
}
rtc::scoped_refptr<RtpSenderInternal> RtpTransceiver::sender_internal() const {
RTC_DCHECK(unified_plan_);
RTC_CHECK_EQ(1u, senders_.size());
return rtc::scoped_refptr<RtpSenderInternal>(senders_[0]->internal());
}
rtc::scoped_refptr<RtpReceiverInternal> RtpTransceiver::receiver_internal()
const {
RTC_DCHECK(unified_plan_);
RTC_CHECK_EQ(1u, receivers_.size());
return rtc::scoped_refptr<RtpReceiverInternal>(receivers_[0]->internal());
}
cricket::MediaType RtpTransceiver::media_type() const {
return media_type_;
}
absl::optional<std::string> RtpTransceiver::mid() const {
return mid_;
}
void RtpTransceiver::OnFirstPacketReceived() {
for (const auto& receiver : receivers_) {
receiver->internal()->NotifyFirstPacketReceived();
}
}
rtc::scoped_refptr<RtpSenderInterface> RtpTransceiver::sender() const {
RTC_DCHECK(unified_plan_);
RTC_CHECK_EQ(1u, senders_.size());
return senders_[0];
}
rtc::scoped_refptr<RtpReceiverInterface> RtpTransceiver::receiver() const {
RTC_DCHECK(unified_plan_);
RTC_CHECK_EQ(1u, receivers_.size());
return receivers_[0];
}
void RtpTransceiver::set_current_direction(RtpTransceiverDirection direction) {
RTC_LOG(LS_INFO) << "Changing transceiver (MID=" << mid_.value_or("<not set>")
<< ") current direction from "
<< (current_direction_ ? RtpTransceiverDirectionToString(
*current_direction_)
: "<not set>")
<< " to " << RtpTransceiverDirectionToString(direction)
<< ".";
current_direction_ = direction;
if (RtpTransceiverDirectionHasSend(*current_direction_)) {
has_ever_been_used_to_send_ = true;
}
}
void RtpTransceiver::set_fired_direction(
absl::optional<RtpTransceiverDirection> direction) {
fired_direction_ = direction;
}
bool RtpTransceiver::stopped() const {
RTC_DCHECK_RUN_ON(thread_);
return stopped_;
}
bool RtpTransceiver::stopping() const {
RTC_DCHECK_RUN_ON(thread_);
return stopping_;
}
RtpTransceiverDirection RtpTransceiver::direction() const {
if (unified_plan_ && stopping())
return RtpTransceiverDirection::kStopped;
return direction_;
}
RTCError RtpTransceiver::SetDirectionWithError(
RtpTransceiverDirection new_direction) {
if (unified_plan_ && stopping()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"Cannot set direction on a stopping transceiver.");
}
if (new_direction == direction_)
return RTCError::OK();
if (new_direction == RtpTransceiverDirection::kStopped) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"The set direction 'stopped' is invalid.");
}
direction_ = new_direction;
on_negotiation_needed_();
return RTCError::OK();
}
absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
const {
if (unified_plan_ && stopped())
return RtpTransceiverDirection::kStopped;
return current_direction_;
}
absl::optional<RtpTransceiverDirection> RtpTransceiver::fired_direction()
const {
return fired_direction_;
}
void RtpTransceiver::StopSendingAndReceiving() {
// 1. Let sender be transceiver.[[Sender]].
// 2. Let receiver be transceiver.[[Receiver]].
//
// 3. Stop sending media with sender.
//
RTC_DCHECK_RUN_ON(thread_);
// 4. Send an RTCP BYE for each RTP stream that was being sent by sender, as
// specified in [RFC3550].
for (const auto& sender : senders_)
sender->internal()->Stop();
// Signal to receiver sources that we're stopping.
for (const auto& receiver : receivers_)
receiver->internal()->Stop();
context()->worker_thread()->BlockingCall([&]() {
// 5 Stop receiving media with receiver.
for (const auto& receiver : receivers_)
receiver->internal()->SetMediaChannel(nullptr);
});
stopping_ = true;
direction_ = RtpTransceiverDirection::kInactive;
}
RTCError RtpTransceiver::StopStandard() {
RTC_DCHECK_RUN_ON(thread_);
// If we're on Plan B, do what Stop() used to do there.
if (!unified_plan_) {
StopInternal();
return RTCError::OK();
}
// 1. Let transceiver be the RTCRtpTransceiver object on which the method is
// invoked.
//
// 2. Let connection be the RTCPeerConnection object associated with
// transceiver.
//
// 3. If connection.[[IsClosed]] is true, throw an InvalidStateError.
if (is_pc_closed_) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
// 4. If transceiver.[[Stopping]] is true, abort these steps.
if (stopping_)
return RTCError::OK();
// 5. Stop sending and receiving given transceiver, and update the
// negotiation-needed flag for connection.
StopSendingAndReceiving();
on_negotiation_needed_();
return RTCError::OK();
}
void RtpTransceiver::StopInternal() {
RTC_DCHECK_RUN_ON(thread_);
StopTransceiverProcedure();
}
void RtpTransceiver::StopTransceiverProcedure() {
RTC_DCHECK_RUN_ON(thread_);
// As specified in the "Stop the RTCRtpTransceiver" procedure
// 1. If transceiver.[[Stopping]] is false, stop sending and receiving given
// transceiver.
if (!stopping_)
StopSendingAndReceiving();
// 2. Set transceiver.[[Stopped]] to true.
stopped_ = true;
// Signal the updated change to the senders.
for (const auto& sender : senders_)
sender->internal()->SetTransceiverAsStopped();
// 3. Set transceiver.[[Receptive]] to false.
// 4. Set transceiver.[[CurrentDirection]] to null.
current_direction_ = absl::nullopt;
}
RTCError RtpTransceiver::SetCodecPreferences(
rtc::ArrayView<RtpCodecCapability> codec_capabilities) {
RTC_DCHECK(unified_plan_);
// 3. If codecs is an empty list, set transceiver's [[PreferredCodecs]] slot
// to codecs and abort these steps.
if (codec_capabilities.empty()) {
codec_preferences_.clear();
return RTCError::OK();
}
// 4. Remove any duplicate values in codecs.
std::vector<RtpCodecCapability> codecs;
absl::c_remove_copy_if(codec_capabilities, std::back_inserter(codecs),
[&codecs](const RtpCodecCapability& codec) {
return absl::c_linear_search(codecs, codec);
});
// 6. to 8.
RTCError result;
std::vector<cricket::Codec> recv_codecs;
if (media_type_ == cricket::MEDIA_TYPE_AUDIO) {
recv_codecs = media_engine()->voice().recv_codecs();
} else if (media_type_ == cricket::MEDIA_TYPE_VIDEO) {
recv_codecs = media_engine()->video().recv_codecs(context()->use_rtx());
}
result = VerifyCodecPreferences(codecs, recv_codecs,
context()->env().field_trials());
if (result.ok()) {
codec_preferences_ = codecs;
}
return result;
}
std::vector<RtpHeaderExtensionCapability>
RtpTransceiver::GetHeaderExtensionsToNegotiate() const {
return header_extensions_to_negotiate_;
}
std::vector<RtpHeaderExtensionCapability>
RtpTransceiver::GetNegotiatedHeaderExtensions() const {
RTC_DCHECK_RUN_ON(thread_);
std::vector<RtpHeaderExtensionCapability> result;
result.reserve(header_extensions_to_negotiate_.size());
for (const auto& ext : header_extensions_to_negotiate_) {
auto negotiated = absl::c_find_if(negotiated_header_extensions_,
[&ext](const RtpExtension& negotiated) {
return negotiated.uri == ext.uri;
});
RtpHeaderExtensionCapability capability(ext.uri);
// TODO(bugs.webrtc.org/7477): extend when header extensions support
// direction.
capability.direction = negotiated != negotiated_header_extensions_.end()
? RtpTransceiverDirection::kSendRecv
: RtpTransceiverDirection::kStopped;
result.push_back(capability);
}
return result;
}
// Helper function to determine mandatory-to-negotiate extensions.
// Since BUNDLE is offered by default, MID is mandatory and can not be turned
// off via this API.
bool IsMandatoryHeaderExtension(const std::string& uri) {
return uri == RtpExtension::kMidUri;
}
RTCError RtpTransceiver::SetHeaderExtensionsToNegotiate(
rtc::ArrayView<const RtpHeaderExtensionCapability> header_extensions) {
if (header_extensions.size() != header_extensions_to_negotiate_.size()) {
return RTCError(RTCErrorType::INVALID_MODIFICATION,
"Size of extensions to negotiate does not match.");
}
// For each index i of extensions, run the following steps: ...
for (size_t i = 0; i < header_extensions.size(); i++) {
const auto& extension = header_extensions[i];
if (extension.uri != header_extensions_to_negotiate_[i].uri) {
return RTCError(RTCErrorType::INVALID_MODIFICATION,
"Reordering extensions is not allowed.");
}
if (IsMandatoryHeaderExtension(extension.uri) &&
extension.direction != RtpTransceiverDirection::kSendRecv) {
return RTCError(RTCErrorType::INVALID_MODIFICATION,
"Attempted to stop a mandatory extension.");
}
// TODO(bugs.webrtc.org/7477): Currently there are no recvonly extensions so
// this can not be checked: "When there exists header extension capabilities
// that have directions other than kSendRecv, restrict extension.direction
// as to not exceed that capability."
}
// Apply mutation after error checking.
for (size_t i = 0; i < header_extensions.size(); i++) {
header_extensions_to_negotiate_[i].direction =
header_extensions[i].direction;
}
return RTCError::OK();
}
void RtpTransceiver::OnNegotiationUpdate(
SdpType sdp_type,
const cricket::MediaContentDescription* content) {
RTC_DCHECK_RUN_ON(thread_);
RTC_DCHECK(content);
if (sdp_type == SdpType::kAnswer)
negotiated_header_extensions_ = content->rtp_header_extensions();
}
void RtpTransceiver::SetPeerConnectionClosed() {
is_pc_closed_ = true;
}
} // namespace webrtc