Delete media transport integration.
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 5047eef..d118e52 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -619,19 +619,12 @@
// correctly. This flag will be deprecated soon. Do not rely on it.
bool active_reset_srtp_params = false;
- // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
- // informs PeerConnection that it should use the MediaTransportInterface for
- // media (audio/video). It's invalid to set it to |true| if the
- // MediaTransportFactory wasn't provided.
+ // DEPRECATED. Do not use. This option is ignored by peer connection.
+ // TODO(webrtc:9719): Delete this option.
bool use_media_transport = false;
- // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
- // informs PeerConnection that it should use the MediaTransportInterface for
- // data channels. It's invalid to set it to |true| if the
- // MediaTransportFactory wasn't provided. Data channels over media
- // transport are not compatible with RTP or SCTP data channels. Setting
- // both |use_media_transport_for_data_channels| and
- // |enable_rtp_data_channel| is invalid.
+ // DEPRECATED. Do not use. This option is ignored by peer connection.
+ // TODO(webrtc:9719): Delete this option.
bool use_media_transport_for_data_channels = false;
// If MediaTransportFactory is provided in PeerConnectionFactory, this flag
diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index 593135d..ce2d88c 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -22,131 +22,6 @@
namespace webrtc {
-// TODO(sukhanov): For now fake media transport does nothing and is used only
-// in jsepcontroller unittests. In the future we should implement fake media
-// transport, which forwards frames to another fake media transport, so we
-// could unit test audio / video integration.
-class FakeMediaTransport : public MediaTransportInterface {
- public:
- explicit FakeMediaTransport(
- const MediaTransportSettings& settings,
- const absl::optional<std::string>& transport_offer = "",
- const absl::optional<std::string>& remote_transport_parameters = "")
- : settings_(settings),
- transport_offer_(transport_offer),
- remote_transport_parameters_(remote_transport_parameters) {}
- ~FakeMediaTransport() = default;
-
- RTCError SendAudioFrame(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) override {
- return RTCError::OK();
- }
-
- RTCError SendVideoFrame(
- uint64_t channel_id,
- const MediaTransportEncodedVideoFrame& frame) override {
- return RTCError::OK();
- }
-
- RTCError RequestKeyFrame(uint64_t channel_id) override {
- return RTCError::OK();
- }
-
- void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {}
- void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {}
-
- // Returns true if fake media transport was created as a caller.
- bool is_caller() const { return settings_.is_caller; }
- absl::optional<std::string> pre_shared_key() const {
- return settings_.pre_shared_key;
- }
-
- RTCError OpenChannel(int channel_id) override { return RTCError::OK(); }
-
- RTCError SendData(int channel_id,
- const SendDataParams& params,
- const rtc::CopyOnWriteBuffer& buffer) override {
- return RTCError::OK();
- }
-
- RTCError CloseChannel(int channel_id) override { return RTCError::OK(); }
-
- void SetDataSink(DataChannelSink* sink) override {}
-
- bool IsReadyToSend() const override { return false; }
-
- void SetMediaTransportStateCallback(
- MediaTransportStateCallback* callback) override {
- state_callback_ = callback;
- }
-
- void SetState(webrtc::MediaTransportState state) {
- if (state_callback_) {
- state_callback_->OnStateChanged(state);
- }
- }
-
- void AddTargetTransferRateObserver(
- webrtc::TargetTransferRateObserver* observer) override {
- RTC_CHECK(!absl::c_linear_search(target_rate_observers_, observer));
- target_rate_observers_.push_back(observer);
- }
-
- void RemoveTargetTransferRateObserver(
- webrtc::TargetTransferRateObserver* observer) override {
- auto it = absl::c_find(target_rate_observers_, observer);
- if (it != target_rate_observers_.end()) {
- target_rate_observers_.erase(it);
- }
- }
-
- void SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits) override {}
-
- void SetTargetBitrateLimits(const MediaTransportTargetRateConstraints&
- target_rate_constraints) override {
- target_rate_constraints_in_order_.push_back(target_rate_constraints);
- }
-
- const std::vector<MediaTransportTargetRateConstraints>&
- target_rate_constraints_in_order() {
- return target_rate_constraints_in_order_;
- }
-
- int target_rate_observers_size() { return target_rate_observers_.size(); }
-
- // Settings that were passed down to fake media transport.
- const MediaTransportSettings& settings() { return settings_; }
-
- absl::optional<std::string> GetTransportParametersOffer() const override {
- // At least right now, we intend to use GetTransportParametersOffer before
- // the transport is connected. This may change in the future.
- RTC_CHECK(!is_connected_);
- return transport_offer_;
- }
-
- const absl::optional<std::string>& remote_transport_parameters() {
- return remote_transport_parameters_;
- }
-
- void Connect(rtc::PacketTransportInternal* packet_transport) {
- RTC_CHECK(!is_connected_) << "::Connect was called twice";
- is_connected_ = true;
- }
-
- bool is_connected() { return is_connected_; }
-
- private:
- const MediaTransportSettings settings_;
- MediaTransportStateCallback* state_callback_ = nullptr;
- std::vector<webrtc::TargetTransferRateObserver*> target_rate_observers_;
- const absl::optional<std::string> transport_offer_;
- const absl::optional<std::string> remote_transport_parameters_;
- bool is_connected_ = false;
- std::vector<MediaTransportTargetRateConstraints>
- target_rate_constraints_in_order_;
-};
-
// Fake media transport factory creates fake media transport.
// Also creates fake datagram transport, since both media and datagram
// transports are created by |MediaTransportFactory|.
@@ -163,19 +38,13 @@
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings) override {
- std::unique_ptr<MediaTransportInterface> media_transport =
- std::make_unique<FakeMediaTransport>(settings, transport_offer_);
- media_transport->Connect(packet_transport);
- return std::move(media_transport);
+ return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
}
RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
rtc::Thread* network_thread,
const MediaTransportSettings& settings) override {
- std::unique_ptr<MediaTransportInterface> media_transport =
- std::make_unique<FakeMediaTransport>(
- settings, transport_offer_, settings.remote_transport_parameters);
- return std::move(media_transport);
+ return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
}
RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
diff --git a/api/test/loopback_media_transport.cc b/api/test/loopback_media_transport.cc
index f1bce1c..14b28ac 100644
--- a/api/test/loopback_media_transport.cc
+++ b/api/test/loopback_media_transport.cc
@@ -21,87 +21,6 @@
constexpr size_t kLoopbackMaxDatagramSize = 1200;
-// Wrapper used to hand out unique_ptrs to loopback media transports without
-// ownership changes.
-class WrapperMediaTransport : public MediaTransportInterface {
- public:
- explicit WrapperMediaTransport(MediaTransportInterface* wrapped)
- : wrapped_(wrapped) {}
-
- RTCError SendAudioFrame(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) override {
- return wrapped_->SendAudioFrame(channel_id, std::move(frame));
- }
-
- RTCError SendVideoFrame(
- uint64_t channel_id,
- const MediaTransportEncodedVideoFrame& frame) override {
- return wrapped_->SendVideoFrame(channel_id, frame);
- }
-
- void SetKeyFrameRequestCallback(
- MediaTransportKeyFrameRequestCallback* callback) override {
- wrapped_->SetKeyFrameRequestCallback(callback);
- }
-
- RTCError RequestKeyFrame(uint64_t channel_id) override {
- return wrapped_->RequestKeyFrame(channel_id);
- }
-
- void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {
- wrapped_->SetReceiveAudioSink(sink);
- }
-
- void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {
- wrapped_->SetReceiveVideoSink(sink);
- }
-
- void AddTargetTransferRateObserver(
- TargetTransferRateObserver* observer) override {
- wrapped_->AddTargetTransferRateObserver(observer);
- }
-
- void RemoveTargetTransferRateObserver(
- TargetTransferRateObserver* observer) override {
- wrapped_->RemoveTargetTransferRateObserver(observer);
- }
-
- void SetMediaTransportStateCallback(
- MediaTransportStateCallback* callback) override {
- wrapped_->SetMediaTransportStateCallback(callback);
- }
-
- RTCError OpenChannel(int channel_id) override {
- return wrapped_->OpenChannel(channel_id);
- }
-
- RTCError SendData(int channel_id,
- const SendDataParams& params,
- const rtc::CopyOnWriteBuffer& buffer) override {
- return wrapped_->SendData(channel_id, params, buffer);
- }
-
- RTCError CloseChannel(int channel_id) override {
- return wrapped_->CloseChannel(channel_id);
- }
-
- void SetDataSink(DataChannelSink* sink) override {
- wrapped_->SetDataSink(sink);
- }
-
- bool IsReadyToSend() const override { return wrapped_->IsReadyToSend(); }
-
- void SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits) override {}
-
- absl::optional<std::string> GetTransportParametersOffer() const override {
- return wrapped_->GetTransportParametersOffer();
- }
-
- private:
- MediaTransportInterface* wrapped_;
-};
-
class WrapperDatagramTransport : public DatagramTransportInterface {
public:
explicit WrapperDatagramTransport(DatagramTransportInterface* wrapped)
@@ -166,10 +85,8 @@
} // namespace
WrapperMediaTransportFactory::WrapperMediaTransportFactory(
- MediaTransportInterface* wrapped_media_transport,
DatagramTransportInterface* wrapped_datagram_transport)
- : wrapped_media_transport_(wrapped_media_transport),
- wrapped_datagram_transport_(wrapped_datagram_transport) {}
+ : wrapped_datagram_transport_(wrapped_datagram_transport) {}
WrapperMediaTransportFactory::WrapperMediaTransportFactory(
MediaTransportFactory* wrapped)
@@ -180,12 +97,7 @@
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
- created_transport_count_++;
- if (wrapped_factory_) {
- return wrapped_factory_->CreateMediaTransport(packet_transport,
- network_thread, settings);
- }
- return {std::make_unique<WrapperMediaTransport>(wrapped_media_transport_)};
+ return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
}
RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
@@ -215,22 +127,14 @@
WrapperMediaTransportFactory::CreateMediaTransport(
rtc::Thread* network_thread,
const MediaTransportSettings& settings) {
- created_transport_count_++;
- if (wrapped_factory_) {
- return wrapped_factory_->CreateMediaTransport(network_thread, settings);
- }
- return {std::make_unique<WrapperMediaTransport>(wrapped_media_transport_)};
+ return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
}
MediaTransportPair::MediaTransportPair(rtc::Thread* thread)
- : first_(thread),
- second_(thread),
- first_datagram_transport_(thread),
+ : first_datagram_transport_(thread),
second_datagram_transport_(thread),
- first_factory_(&first_, &first_datagram_transport_),
- second_factory_(&second_, &second_datagram_transport_) {
- first_.Connect(&second_);
- second_.Connect(&first_);
+ first_factory_(&first_datagram_transport_),
+ second_factory_(&second_datagram_transport_) {
first_datagram_transport_.Connect(&second_datagram_transport_);
second_datagram_transport_.Connect(&first_datagram_transport_);
}
@@ -251,205 +155,12 @@
other_ = other;
}
-MediaTransportPair::LoopbackMediaTransport::LoopbackMediaTransport(
- rtc::Thread* thread)
- : dc_transport_(thread), thread_(thread), other_(nullptr) {
- RTC_LOG(LS_INFO) << "LoopbackMediaTransport";
-}
-
-MediaTransportPair::LoopbackMediaTransport::~LoopbackMediaTransport() {
- RTC_LOG(LS_INFO) << "~LoopbackMediaTransport";
- rtc::CritScope lock(&sink_lock_);
- RTC_CHECK(audio_sink_ == nullptr);
- RTC_CHECK(video_sink_ == nullptr);
- RTC_CHECK(target_transfer_rate_observers_.empty());
- RTC_CHECK(rtt_observers_.empty());
-}
-
-void MediaTransportPair::LoopbackMediaTransport::Connect(
- LoopbackMediaTransport* other) {
- other_ = other;
- dc_transport_.Connect(&other->dc_transport_);
-}
-
-void MediaTransportPair::LoopbackMediaTransport::Connect(
- rtc::PacketTransportInternal* packet_transport) {
- if (state_after_connect_) {
- SetState(*state_after_connect_);
- }
-}
-
-absl::optional<std::string>
-MediaTransportPair::LoopbackMediaTransport::GetTransportParametersOffer()
- const {
- return "loopback-media-transport-parameters";
-}
-
-RTCError MediaTransportPair::LoopbackMediaTransport::SendAudioFrame(
- uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) {
- {
- rtc::CritScope lock(&stats_lock_);
- ++stats_.sent_audio_frames;
- }
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, channel_id, frame] {
- other_->OnData(channel_id, frame);
- });
- return RTCError::OK();
-}
-
-RTCError MediaTransportPair::LoopbackMediaTransport::SendVideoFrame(
- uint64_t channel_id,
- const MediaTransportEncodedVideoFrame& frame) {
- {
- rtc::CritScope lock(&stats_lock_);
- ++stats_.sent_video_frames;
- }
- // Ensure that we own the referenced data.
- MediaTransportEncodedVideoFrame frame_copy = frame;
- frame_copy.Retain();
- invoker_.AsyncInvoke<void>(
- RTC_FROM_HERE, thread_, [this, channel_id, frame_copy]() mutable {
- other_->OnData(channel_id, std::move(frame_copy));
- });
- return RTCError::OK();
-}
-
-void MediaTransportPair::LoopbackMediaTransport::SetKeyFrameRequestCallback(
- MediaTransportKeyFrameRequestCallback* callback) {
- rtc::CritScope lock(&sink_lock_);
- if (callback) {
- RTC_CHECK(key_frame_callback_ == nullptr);
- }
- key_frame_callback_ = callback;
-}
-
-RTCError MediaTransportPair::LoopbackMediaTransport::RequestKeyFrame(
- uint64_t channel_id) {
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, channel_id] {
- other_->OnKeyFrameRequested(channel_id);
- });
- return RTCError::OK();
-}
-
-void MediaTransportPair::LoopbackMediaTransport::SetReceiveAudioSink(
- MediaTransportAudioSinkInterface* sink) {
- rtc::CritScope lock(&sink_lock_);
- if (sink) {
- RTC_CHECK(audio_sink_ == nullptr);
- }
- audio_sink_ = sink;
-}
-
-void MediaTransportPair::LoopbackMediaTransport::SetReceiveVideoSink(
- MediaTransportVideoSinkInterface* sink) {
- rtc::CritScope lock(&sink_lock_);
- if (sink) {
- RTC_CHECK(video_sink_ == nullptr);
- }
- video_sink_ = sink;
-}
-
-void MediaTransportPair::LoopbackMediaTransport::AddTargetTransferRateObserver(
- TargetTransferRateObserver* observer) {
- RTC_CHECK(observer);
- {
- rtc::CritScope cs(&sink_lock_);
- RTC_CHECK(
- !absl::c_linear_search(target_transfer_rate_observers_, observer));
- target_transfer_rate_observers_.push_back(observer);
- }
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this] {
- RTC_DCHECK_RUN_ON(thread_);
- const DataRate kBitrate = DataRate::kbps(300);
- const Timestamp now = Timestamp::us(rtc::TimeMicros());
-
- TargetTransferRate transfer_rate;
- transfer_rate.at_time = now;
- transfer_rate.target_rate = kBitrate;
- transfer_rate.network_estimate.at_time = now;
- transfer_rate.network_estimate.round_trip_time = TimeDelta::ms(20);
- transfer_rate.network_estimate.bwe_period = TimeDelta::seconds(3);
- transfer_rate.network_estimate.bandwidth = kBitrate;
-
- rtc::CritScope cs(&sink_lock_);
-
- for (auto* o : target_transfer_rate_observers_) {
- o->OnTargetTransferRate(transfer_rate);
- }
- });
-}
-
-void MediaTransportPair::LoopbackMediaTransport::
- RemoveTargetTransferRateObserver(TargetTransferRateObserver* observer) {
- rtc::CritScope cs(&sink_lock_);
- auto it = absl::c_find(target_transfer_rate_observers_, observer);
- if (it == target_transfer_rate_observers_.end()) {
- RTC_LOG(LS_WARNING)
- << "Attempt to remove an unknown TargetTransferRate observer";
- return;
- }
- target_transfer_rate_observers_.erase(it);
-}
-
-void MediaTransportPair::LoopbackMediaTransport::AddRttObserver(
- MediaTransportRttObserver* observer) {
- RTC_CHECK(observer);
- {
- rtc::CritScope cs(&sink_lock_);
- RTC_CHECK(!absl::c_linear_search(rtt_observers_, observer));
- rtt_observers_.push_back(observer);
- }
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this] {
- RTC_DCHECK_RUN_ON(thread_);
-
- rtc::CritScope cs(&sink_lock_);
- for (auto* o : rtt_observers_) {
- o->OnRttUpdated(20);
- }
- });
-}
-
-void MediaTransportPair::LoopbackMediaTransport::RemoveRttObserver(
- MediaTransportRttObserver* observer) {
- rtc::CritScope cs(&sink_lock_);
- auto it = absl::c_find(rtt_observers_, observer);
- if (it == rtt_observers_.end()) {
- RTC_LOG(LS_WARNING) << "Attempt to remove an unknown RTT observer";
- return;
- }
- rtt_observers_.erase(it);
-}
-
-void MediaTransportPair::LoopbackMediaTransport::SetMediaTransportStateCallback(
- MediaTransportStateCallback* callback) {
- rtc::CritScope lock(&sink_lock_);
- state_callback_ = callback;
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this] {
- RTC_DCHECK_RUN_ON(thread_);
- OnStateChanged();
- });
-}
-
-RTCError MediaTransportPair::LoopbackMediaTransport::OpenChannel(
- int channel_id) {
- // No-op. No need to open channels for the loopback.
- return dc_transport_.OpenChannel(channel_id);
-}
-
RTCError MediaTransportPair::LoopbackDataChannelTransport::OpenChannel(
int channel_id) {
// No-op. No need to open channels for the loopback.
return RTCError::OK();
}
-RTCError MediaTransportPair::LoopbackMediaTransport::SendData(
- int channel_id,
- const SendDataParams& params,
- const rtc::CopyOnWriteBuffer& buffer) {
- return dc_transport_.SendData(channel_id, params, buffer);
-}
-
RTCError MediaTransportPair::LoopbackDataChannelTransport::SendData(
int channel_id,
const SendDataParams& params,
@@ -461,11 +172,6 @@
return RTCError::OK();
}
-RTCError MediaTransportPair::LoopbackMediaTransport::CloseChannel(
- int channel_id) {
- return dc_transport_.CloseChannel(channel_id);
-}
-
RTCError MediaTransportPair::LoopbackDataChannelTransport::CloseChannel(
int channel_id) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, channel_id] {
@@ -478,15 +184,6 @@
return RTCError::OK();
}
-void MediaTransportPair::LoopbackMediaTransport::SetDataSink(
- DataChannelSink* sink) {
- dc_transport_.SetDataSink(sink);
-}
-
-bool MediaTransportPair::LoopbackMediaTransport::IsReadyToSend() const {
- return dc_transport_.IsReadyToSend();
-}
-
void MediaTransportPair::LoopbackDataChannelTransport::SetDataSink(
DataChannelSink* sink) {
rtc::CritScope lock(&sink_lock_);
@@ -501,65 +198,10 @@
return ready_to_send_;
}
-void MediaTransportPair::LoopbackMediaTransport::SetState(
- MediaTransportState state) {
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, state] {
- RTC_DCHECK_RUN_ON(thread_);
- state_ = state;
- OnStateChanged();
- });
-}
-
-void MediaTransportPair::LoopbackMediaTransport::SetStateAfterConnect(
- MediaTransportState state) {
- state_after_connect_ = state;
-}
-
-void MediaTransportPair::LoopbackMediaTransport::FlushAsyncInvokes() {
- invoker_.Flush(thread_);
- dc_transport_.FlushAsyncInvokes();
-}
-
void MediaTransportPair::LoopbackDataChannelTransport::FlushAsyncInvokes() {
invoker_.Flush(thread_);
}
-MediaTransportPair::Stats
-MediaTransportPair::LoopbackMediaTransport::GetStats() {
- rtc::CritScope lock(&stats_lock_);
- return stats_;
-}
-
-void MediaTransportPair::LoopbackMediaTransport::OnData(
- uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) {
- {
- rtc::CritScope lock(&sink_lock_);
- if (audio_sink_) {
- audio_sink_->OnData(channel_id, frame);
- }
- }
- {
- rtc::CritScope lock(&stats_lock_);
- ++stats_.received_audio_frames;
- }
-}
-
-void MediaTransportPair::LoopbackMediaTransport::OnData(
- uint64_t channel_id,
- MediaTransportEncodedVideoFrame frame) {
- {
- rtc::CritScope lock(&sink_lock_);
- if (video_sink_) {
- video_sink_->OnData(channel_id, frame);
- }
- }
- {
- rtc::CritScope lock(&stats_lock_);
- ++stats_.received_video_frames;
- }
-}
-
void MediaTransportPair::LoopbackDataChannelTransport::OnData(
int channel_id,
DataMessageType type,
@@ -570,14 +212,6 @@
}
}
-void MediaTransportPair::LoopbackMediaTransport::OnKeyFrameRequested(
- int channel_id) {
- rtc::CritScope lock(&sink_lock_);
- if (key_frame_callback_) {
- key_frame_callback_->OnKeyFrameRequested(channel_id);
- }
-}
-
void MediaTransportPair::LoopbackDataChannelTransport::OnRemoteCloseChannel(
int channel_id) {
rtc::CritScope lock(&sink_lock_);
@@ -587,15 +221,6 @@
}
}
-void MediaTransportPair::LoopbackMediaTransport::OnStateChanged() {
- rtc::CritScope lock(&sink_lock_);
- if (state_callback_) {
- state_callback_->OnStateChanged(state_);
- }
-
- dc_transport_.OnReadyToSend(state_ == MediaTransportState::kWritable);
-}
-
void MediaTransportPair::LoopbackDataChannelTransport::OnReadyToSend(
bool ready_to_send) {
invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, ready_to_send] {
@@ -608,9 +233,6 @@
});
}
-void MediaTransportPair::LoopbackMediaTransport::SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits) {}
-
MediaTransportPair::LoopbackDatagramTransport::LoopbackDatagramTransport(
rtc::Thread* thread)
: thread_(thread), dc_transport_(thread) {}
diff --git a/api/test/loopback_media_transport.h b/api/test/loopback_media_transport.h
index 475c586..f2aed3e 100644
--- a/api/test/loopback_media_transport.h
+++ b/api/test/loopback_media_transport.h
@@ -42,8 +42,7 @@
// CreateMediaTransport();
class WrapperMediaTransportFactory : public MediaTransportFactory {
public:
- WrapperMediaTransportFactory(
- MediaTransportInterface* wrapped_media_transport,
+ explicit WrapperMediaTransportFactory(
DatagramTransportInterface* wrapped_datagram_transport);
explicit WrapperMediaTransportFactory(MediaTransportFactory* wrapped);
@@ -65,7 +64,6 @@
int created_transport_count() const;
private:
- MediaTransportInterface* wrapped_media_transport_ = nullptr;
DatagramTransportInterface* wrapped_datagram_transport_ = nullptr;
MediaTransportFactory* wrapped_factory_ = nullptr;
int created_transport_count_ = 0;
@@ -85,10 +83,6 @@
explicit MediaTransportPair(rtc::Thread* thread);
~MediaTransportPair();
- // Ownership stays with MediaTransportPair
- MediaTransportInterface* first() { return &first_; }
- MediaTransportInterface* second() { return &second_; }
-
DatagramTransportInterface* first_datagram_transport() {
return &first_datagram_transport_;
}
@@ -105,19 +99,15 @@
}
void SetState(MediaTransportState state) {
- first_.SetState(state);
- second_.SetState(state);
first_datagram_transport_.SetState(state);
second_datagram_transport_.SetState(state);
}
void SetFirstState(MediaTransportState state) {
- first_.SetState(state);
first_datagram_transport_.SetState(state);
}
void SetSecondStateAfterConnect(MediaTransportState state) {
- second_.SetState(state);
second_datagram_transport_.SetState(state);
}
@@ -126,13 +116,10 @@
}
void FlushAsyncInvokes() {
- first_.FlushAsyncInvokes();
- second_.FlushAsyncInvokes();
+ first_datagram_transport_.FlushAsyncInvokes();
+ second_datagram_transport_.FlushAsyncInvokes();
}
- Stats FirstStats() { return first_.GetStats(); }
- Stats SecondStats() { return second_.GetStats(); }
-
int first_factory_transport_count() const {
return first_factory_.created_transport_count();
}
@@ -183,116 +170,6 @@
rtc::AsyncInvoker invoker_;
};
- class LoopbackMediaTransport : public MediaTransportInterface {
- public:
- explicit LoopbackMediaTransport(rtc::Thread* thread);
-
- ~LoopbackMediaTransport() override;
-
- // Connects this loopback transport to another loopback transport.
- void Connect(LoopbackMediaTransport* other);
-
- void Connect(rtc::PacketTransportInternal* transport) override;
-
- RTCError SendAudioFrame(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) override;
-
- RTCError SendVideoFrame(
- uint64_t channel_id,
- const MediaTransportEncodedVideoFrame& frame) override;
-
- void SetKeyFrameRequestCallback(
- MediaTransportKeyFrameRequestCallback* callback) override;
-
- RTCError RequestKeyFrame(uint64_t channel_id) override;
-
- void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override;
-
- void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override;
-
- void AddTargetTransferRateObserver(
- TargetTransferRateObserver* observer) override;
-
- void RemoveTargetTransferRateObserver(
- TargetTransferRateObserver* observer) override;
-
- void AddRttObserver(MediaTransportRttObserver* observer) override;
- void RemoveRttObserver(MediaTransportRttObserver* observer) override;
-
- void SetMediaTransportStateCallback(
- MediaTransportStateCallback* callback) override;
-
- void SetState(MediaTransportState state);
-
- // When Connect() is called, the media transport will enter this state.
- // This is useful for mimicking zero-RTT connectivity, for example.
- void SetStateAfterConnect(MediaTransportState state);
-
- RTCError OpenChannel(int channel_id) override;
-
- RTCError SendData(int channel_id,
- const SendDataParams& params,
- const rtc::CopyOnWriteBuffer& buffer) override;
-
- RTCError CloseChannel(int channel_id) override;
-
- void SetDataSink(DataChannelSink* sink) override;
-
- bool IsReadyToSend() const override;
-
- void FlushAsyncInvokes();
-
- Stats GetStats();
-
- void SetAllocatedBitrateLimits(
- const MediaTransportAllocatedBitrateLimits& limits) override;
-
- absl::optional<std::string> GetTransportParametersOffer() const override;
-
- private:
- void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame);
-
- void OnData(uint64_t channel_id, MediaTransportEncodedVideoFrame frame);
-
- void OnKeyFrameRequested(int channel_id);
-
- void OnStateChanged() RTC_RUN_ON(thread_);
-
- // Implementation of the data channel transport.
- LoopbackDataChannelTransport dc_transport_;
-
- rtc::Thread* const thread_;
- rtc::CriticalSection sink_lock_;
- rtc::CriticalSection stats_lock_;
-
- MediaTransportAudioSinkInterface* audio_sink_ RTC_GUARDED_BY(sink_lock_) =
- nullptr;
- MediaTransportVideoSinkInterface* video_sink_ RTC_GUARDED_BY(sink_lock_) =
- nullptr;
-
- MediaTransportKeyFrameRequestCallback* key_frame_callback_
- RTC_GUARDED_BY(sink_lock_) = nullptr;
-
- MediaTransportStateCallback* state_callback_ RTC_GUARDED_BY(sink_lock_) =
- nullptr;
-
- std::vector<TargetTransferRateObserver*> target_transfer_rate_observers_
- RTC_GUARDED_BY(sink_lock_);
- std::vector<MediaTransportRttObserver*> rtt_observers_
- RTC_GUARDED_BY(sink_lock_);
-
- MediaTransportState state_ RTC_GUARDED_BY(thread_) =
- MediaTransportState::kPending;
-
- absl::optional<MediaTransportState> state_after_connect_;
-
- LoopbackMediaTransport* other_;
-
- Stats stats_ RTC_GUARDED_BY(stats_lock_);
-
- rtc::AsyncInvoker invoker_;
- };
-
class LoopbackDatagramTransport : public DatagramTransportInterface {
public:
explicit LoopbackDatagramTransport(rtc::Thread* thread);
@@ -351,8 +228,6 @@
rtc::AsyncInvoker invoker_;
};
- LoopbackMediaTransport first_;
- LoopbackMediaTransport second_;
LoopbackDatagramTransport first_datagram_transport_;
LoopbackDatagramTransport second_datagram_transport_;
WrapperMediaTransportFactory first_factory_;
diff --git a/api/test/loopback_media_transport_unittest.cc b/api/test/loopback_media_transport_unittest.cc
index 346ac5f..f036de3 100644
--- a/api/test/loopback_media_transport_unittest.cc
+++ b/api/test/loopback_media_transport_unittest.cc
@@ -52,109 +52,15 @@
MOCK_METHOD1(OnStateChanged, void(MediaTransportState));
};
-// Test only uses the sequence number.
-MediaTransportEncodedAudioFrame CreateAudioFrame(int sequence_number) {
- static constexpr int kSamplingRateHz = 48000;
- static constexpr int kStartingSampleIndex = 0;
- static constexpr int kSamplesPerChannel = 480;
- static constexpr int kPayloadType = 17;
-
- return MediaTransportEncodedAudioFrame(
- kSamplingRateHz, kStartingSampleIndex, kSamplesPerChannel,
- sequence_number, MediaTransportEncodedAudioFrame::FrameType::kSpeech,
- kPayloadType, std::vector<uint8_t>(kSamplesPerChannel));
-}
-
-MediaTransportEncodedVideoFrame CreateVideoFrame(
- int frame_id,
- const webrtc::EncodedImage& encoded_image) {
- static constexpr int kPayloadType = 18;
- return MediaTransportEncodedVideoFrame(frame_id, /*referenced_frame_ids=*/{},
- kPayloadType, encoded_image);
-}
-
} // namespace
-TEST(LoopbackMediaTransport, AudioWithNoSinkSilentlyIgnored) {
- std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
- thread->Start();
- MediaTransportPair transport_pair(thread.get());
- transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(0));
- transport_pair.second()->SendAudioFrame(2, CreateAudioFrame(0));
- transport_pair.FlushAsyncInvokes();
-}
-
-TEST(LoopbackMediaTransport, AudioDeliveredToSink) {
- std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
- thread->Start();
- MediaTransportPair transport_pair(thread.get());
- ::testing::StrictMock<MockMediaTransportAudioSinkInterface> sink;
- EXPECT_CALL(sink,
- OnData(1, ::testing::Property(
- &MediaTransportEncodedAudioFrame::sequence_number,
- ::testing::Eq(10))));
- transport_pair.second()->SetReceiveAudioSink(&sink);
- transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(10));
-
- transport_pair.FlushAsyncInvokes();
- transport_pair.second()->SetReceiveAudioSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, VideoDeliveredToSink) {
- std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
- thread->Start();
- MediaTransportPair transport_pair(thread.get());
- ::testing::StrictMock<MockMediaTransportVideoSinkInterface> sink;
- constexpr uint8_t encoded_data[] = {1, 2, 3};
- EncodedImage encoded_image;
- encoded_image.SetEncodedData(
- EncodedImageBuffer::Create(encoded_data, sizeof(encoded_data)));
-
- EXPECT_CALL(sink, OnData(1, ::testing::Property(
- &MediaTransportEncodedVideoFrame::frame_id,
- ::testing::Eq(10))))
- .WillOnce(::testing::Invoke(
- [&encoded_image](int frame_id,
- const MediaTransportEncodedVideoFrame& frame) {
- EXPECT_EQ(frame.encoded_image().data(), encoded_image.data());
- EXPECT_EQ(frame.encoded_image().size(), encoded_image.size());
- }));
-
- transport_pair.second()->SetReceiveVideoSink(&sink);
- transport_pair.first()->SendVideoFrame(1,
- CreateVideoFrame(10, encoded_image));
-
- transport_pair.FlushAsyncInvokes();
- transport_pair.second()->SetReceiveVideoSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, VideoKeyFrameRequestDeliveredToCallback) {
- std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
- thread->Start();
- MediaTransportPair transport_pair(thread.get());
- ::testing::StrictMock<MockMediaTransportKeyFrameRequestCallback> callback1;
- ::testing::StrictMock<MockMediaTransportKeyFrameRequestCallback> callback2;
- const uint64_t kFirstChannelId = 1111;
- const uint64_t kSecondChannelId = 2222;
-
- EXPECT_CALL(callback1, OnKeyFrameRequested(kSecondChannelId));
- EXPECT_CALL(callback2, OnKeyFrameRequested(kFirstChannelId));
- transport_pair.first()->SetKeyFrameRequestCallback(&callback1);
- transport_pair.second()->SetKeyFrameRequestCallback(&callback2);
-
- transport_pair.first()->RequestKeyFrame(kFirstChannelId);
- transport_pair.second()->RequestKeyFrame(kSecondChannelId);
-
- transport_pair.FlushAsyncInvokes();
-}
-
TEST(LoopbackMediaTransport, DataDeliveredToSink) {
std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
thread->Start();
MediaTransportPair transport_pair(thread.get());
MockDataChannelSink sink;
- transport_pair.first()->SetDataSink(&sink);
+ transport_pair.first_datagram_transport()->SetDataSink(&sink);
const int channel_id = 1;
EXPECT_CALL(
@@ -166,10 +72,11 @@
SendDataParams params;
params.type = DataMessageType::kText;
rtc::CopyOnWriteBuffer buffer("foo");
- transport_pair.second()->SendData(channel_id, params, buffer);
+ transport_pair.second_datagram_transport()->SendData(channel_id, params,
+ buffer);
transport_pair.FlushAsyncInvokes();
- transport_pair.first()->SetDataSink(nullptr);
+ transport_pair.first_datagram_transport()->SetDataSink(nullptr);
}
TEST(LoopbackMediaTransport, CloseDeliveredToSink) {
@@ -178,10 +85,10 @@
MediaTransportPair transport_pair(thread.get());
MockDataChannelSink first_sink;
- transport_pair.first()->SetDataSink(&first_sink);
+ transport_pair.first_datagram_transport()->SetDataSink(&first_sink);
MockDataChannelSink second_sink;
- transport_pair.second()->SetDataSink(&second_sink);
+ transport_pair.second_datagram_transport()->SetDataSink(&second_sink);
const int channel_id = 1;
{
@@ -191,11 +98,11 @@
EXPECT_CALL(first_sink, OnChannelClosed(channel_id));
}
- transport_pair.first()->CloseChannel(channel_id);
+ transport_pair.first_datagram_transport()->CloseChannel(channel_id);
transport_pair.FlushAsyncInvokes();
- transport_pair.first()->SetDataSink(nullptr);
- transport_pair.second()->SetDataSink(nullptr);
+ transport_pair.first_datagram_transport()->SetDataSink(nullptr);
+ transport_pair.second_datagram_transport()->SetDataSink(nullptr);
}
TEST(LoopbackMediaTransport, InitialStateDeliveredWhenCallbackSet) {
@@ -206,7 +113,10 @@
MockStateCallback state_callback;
EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
- transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
+ thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
+ transport_pair.first_datagram_transport()->SetTransportStateCallback(
+ &state_callback);
+ });
transport_pair.FlushAsyncInvokes();
}
@@ -221,7 +131,10 @@
MockStateCallback state_callback;
EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
- transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
+ thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
+ transport_pair.first_datagram_transport()->SetTransportStateCallback(
+ &state_callback);
+ });
transport_pair.FlushAsyncInvokes();
}
@@ -234,7 +147,10 @@
EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
- transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
+ thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
+ transport_pair.first_datagram_transport()->SetTransportStateCallback(
+ &state_callback);
+ });
transport_pair.SetState(MediaTransportState::kWritable);
transport_pair.FlushAsyncInvokes();
}
@@ -247,9 +163,9 @@
MockDataChannelSink data_channel_sink;
EXPECT_CALL(data_channel_sink, OnReadyToSend()).Times(0);
- transport_pair.first()->SetDataSink(&data_channel_sink);
+ transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
transport_pair.FlushAsyncInvokes();
- transport_pair.first()->SetDataSink(nullptr);
+ transport_pair.first_datagram_transport()->SetDataSink(nullptr);
}
TEST(LoopbackMediaTransport, ReadyToSendWhenDataSinkSet) {
@@ -263,9 +179,9 @@
MockDataChannelSink data_channel_sink;
EXPECT_CALL(data_channel_sink, OnReadyToSend());
- transport_pair.first()->SetDataSink(&data_channel_sink);
+ transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
transport_pair.FlushAsyncInvokes();
- transport_pair.first()->SetDataSink(nullptr);
+ transport_pair.first_datagram_transport()->SetDataSink(nullptr);
}
TEST(LoopbackMediaTransport, StateChangeDeliveredToDataSink) {
@@ -276,10 +192,10 @@
MockDataChannelSink data_channel_sink;
EXPECT_CALL(data_channel_sink, OnReadyToSend());
- transport_pair.first()->SetDataSink(&data_channel_sink);
+ transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
transport_pair.SetState(MediaTransportState::kWritable);
transport_pair.FlushAsyncInvokes();
- transport_pair.first()->SetDataSink(nullptr);
+ transport_pair.first_datagram_transport()->SetDataSink(nullptr);
}
} // namespace webrtc
diff --git a/api/transport/media/media_transport_config.cc b/api/transport/media/media_transport_config.cc
index cea3f16..b9b19cb 100644
--- a/api/transport/media/media_transport_config.cc
+++ b/api/transport/media/media_transport_config.cc
@@ -15,23 +15,14 @@
namespace webrtc {
-MediaTransportConfig::MediaTransportConfig(
- MediaTransportInterface* media_transport)
- : media_transport(media_transport) {
- RTC_DCHECK(media_transport != nullptr);
-}
-
MediaTransportConfig::MediaTransportConfig(size_t rtp_max_packet_size)
: rtp_max_packet_size(rtp_max_packet_size) {
RTC_DCHECK_GT(rtp_max_packet_size, 0);
}
-std::string MediaTransportConfig::DebugString()
- const { // TODO(sukhanov): Add rtp_max_packet_size (requires fixing
- // audio_send/receive_stream_unittest.cc).
+std::string MediaTransportConfig::DebugString() const {
rtc::StringBuilder result;
- result << "{media_transport: "
- << (media_transport != nullptr ? "(Transport)" : "null") << "}";
+ result << "{rtp_max_packet_size: " << rtp_max_packet_size.value_or(0) << "}";
return result.Release();
}
diff --git a/api/transport/media/media_transport_config.h b/api/transport/media/media_transport_config.h
index 6a12630..7ef6545 100644
--- a/api/transport/media/media_transport_config.h
+++ b/api/transport/media/media_transport_config.h
@@ -14,7 +14,6 @@
#include <utility>
#include "absl/types/optional.h"
-#include "api/transport/media/media_transport_interface.h"
namespace webrtc {
@@ -25,18 +24,11 @@
// Default constructor for no-media transport scenarios.
MediaTransportConfig() = default;
- // Constructor for media transport scenarios.
- // Note that |media_transport| may not be nullptr.
- explicit MediaTransportConfig(MediaTransportInterface* media_transport);
-
// Constructor for datagram transport scenarios.
explicit MediaTransportConfig(size_t rtp_max_packet_size);
std::string DebugString() const;
- // If provided, all media is sent through media_transport.
- MediaTransportInterface* media_transport = nullptr;
-
// If provided, limits RTP packet size (excludes ICE, IP or network overhead).
absl::optional<size_t> rtp_max_packet_size;
};
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index e64b76f..a6d7ed4 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -53,7 +53,6 @@
"../api/neteq:neteq_api",
"../api/rtc_event_log",
"../api/task_queue",
- "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../call:bitrate_allocator",
"../call:call_interfaces",
@@ -121,13 +120,11 @@
"mock_voe_channel_proxy.h",
"remix_resample_unittest.cc",
"test/audio_stats_test.cc",
- "test/media_transport_test.cc",
]
deps = [
":audio",
":audio_end_to_end_test",
"../api:libjingle_peerconnection_api",
- "../api:loopback_media_transport",
"../api:mock_audio_mixer",
"../api:mock_frame_decryptor",
"../api:mock_frame_encryptor",
@@ -137,7 +134,6 @@
"../api/audio_codecs/opus:audio_encoder_opus",
"../api/rtc_event_log",
"../api/task_queue:default_task_queue_factory",
- "../api/transport/media:media_transport_interface",
"../api/units:time_delta",
"../call:mock_bitrate_allocator",
"../call:mock_call_interfaces",
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index e1041be..2e02388 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -56,7 +56,6 @@
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_send_transport: "
<< (rtcp_send_transport ? "(Transport)" : "null");
- ss << ", media_transport_config: " << media_transport_config.DebugString();
if (!sync_group.empty()) {
ss << ", sync_group: " << sync_group;
}
@@ -78,9 +77,8 @@
static_cast<internal::AudioState*>(audio_state);
return voe::CreateChannelReceive(
clock, module_process_thread, neteq_factory,
- internal_audio_state->audio_device_module(),
- config.media_transport_config, config.rtcp_send_transport, event_log,
- config.rtp.local_ssrc, config.rtp.remote_ssrc,
+ internal_audio_state->audio_device_module(), config.rtcp_send_transport,
+ event_log, config.rtp.local_ssrc, config.rtp.remote_ssrc,
config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
config.jitter_buffer_min_delay_ms,
config.jitter_buffer_enable_rtx_handling, config.decoder_factory,
@@ -129,16 +127,14 @@
module_process_thread_checker_.Detach();
- if (!config.media_transport_config.media_transport) {
- RTC_DCHECK(receiver_controller);
- RTC_DCHECK(packet_router);
- // Configure bandwidth estimation.
- channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
+ RTC_DCHECK(receiver_controller);
+ RTC_DCHECK(packet_router);
+ // Configure bandwidth estimation.
+ channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
- // Register with transport.
- rtp_stream_receiver_ = receiver_controller->CreateReceiver(
- config.rtp.remote_ssrc, channel_receive_.get());
- }
+ // Register with transport.
+ rtp_stream_receiver_ = receiver_controller->CreateReceiver(
+ config.rtp.remote_ssrc, channel_receive_.get());
ConfigureStream(this, config, true);
}
@@ -147,9 +143,7 @@
RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
Stop();
channel_receive_->SetAssociatedSendChannel(nullptr);
- if (!config_.media_transport_config.media_transport) {
- channel_receive_->ResetReceiverCongestionControlObjects();
- }
+ channel_receive_->ResetReceiverCongestionControlObjects();
}
void AudioReceiveStream::Reconfigure(
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 473b387..b8eff0a 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -223,8 +223,7 @@
"{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
"{rtp_history_ms: 0}, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
- "rtcp_send_transport: null, media_transport_config: {media_transport: "
- "null}}",
+ "rtcp_send_transport: null}",
config.ToString());
}
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 323b7a9..90d72c4 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -22,7 +22,6 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/function_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
-#include "api/transport/media/media_transport_config.h"
#include "audio/audio_state.h"
#include "audio/channel_send.h"
#include "audio/conversion.h"
@@ -119,7 +118,6 @@
voe::CreateChannelSend(clock,
task_queue_factory,
module_process_thread,
- config.media_transport_config,
/*overhead_observer=*/this,
config.send_transport,
rtcp_rtt_stats,
@@ -150,7 +148,7 @@
!field_trial::IsDisabled("WebRTC-Audio-AlrProbing")),
send_side_bwe_with_overhead_(
field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
- config_(Config(/*send_transport=*/nullptr, MediaTransportConfig())),
+ config_(Config(/*send_transport=*/nullptr)),
audio_state_(audio_state),
channel_send_(std::move(channel_send)),
event_log_(event_log),
@@ -165,23 +163,8 @@
RTC_DCHECK(audio_state_);
RTC_DCHECK(channel_send_);
RTC_DCHECK(bitrate_allocator_);
- // Currently we require the rtp transport even when media transport is used.
RTC_DCHECK(rtp_transport);
- // TODO(nisse): Eventually, we should have only media_transport. But for the
- // time being, we can have either. When media transport is injected, there
- // should be no rtp_transport, and below check should be strengthened to XOR
- // (either rtp_transport or media_transport but not both).
- RTC_DCHECK(rtp_transport || config.media_transport_config.media_transport);
- if (config.media_transport_config.media_transport) {
- // TODO(sukhanov): Currently media transport audio overhead is considered
- // constant, we will not get overhead_observer calls when using
- // media_transport. In the future when we introduce RTP media transport we
- // should make audio overhead interface consistent and work for both RTP and
- // non-RTP implementations.
- audio_overhead_per_packet_bytes_ =
- config.media_transport_config.media_transport->GetAudioPacketOverhead();
- }
rtp_rtcp_module_ = channel_send_->GetRtpRtcp();
RTC_DCHECK(rtp_rtcp_module_);
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 6d6ec6a..95d7f73 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -144,7 +144,7 @@
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
: clock_(1000000),
task_queue_factory_(CreateDefaultTaskQueueFactory()),
- stream_config_(/*send_transport=*/nullptr, MediaTransportConfig()),
+ stream_config_(/*send_transport=*/nullptr),
audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
bitrate_allocator_(&limit_observer_),
worker_queue_(task_queue_factory_->CreateTaskQueue(
@@ -347,8 +347,7 @@
} // namespace
TEST(AudioSendStreamTest, ConfigToString) {
- AudioSendStream::Config config(/*send_transport=*/nullptr,
- MediaTransportConfig());
+ AudioSendStream::Config config(/*send_transport=*/nullptr);
config.rtp.ssrc = kSsrc;
config.rtp.c_name = kCName;
config.min_bitrate_bps = 12000;
@@ -367,7 +366,7 @@
"{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
"urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
"c_name: foo_name}, rtcp_report_interval_ms: 2500, "
- "send_transport: null, media_transport_config: {media_transport: null}, "
+ "send_transport: null, "
"min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
"send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
"cng_payload_type: 42, payload_type: 103, "
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 502818f..2ecc3cf 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -58,21 +58,6 @@
constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
-RTPHeader CreateRTPHeaderForMediaTransportFrame(
- const MediaTransportEncodedAudioFrame& frame,
- uint64_t channel_id) {
- webrtc::RTPHeader rtp_header;
- rtp_header.payloadType = frame.payload_type();
- rtp_header.payload_type_frequency = frame.sampling_rate_hz();
- rtp_header.timestamp = frame.starting_sample_index();
- rtp_header.sequenceNumber = frame.sequence_number();
-
- rtp_header.ssrc = static_cast<uint32_t>(channel_id);
-
- // The rest are initialized by the RTPHeader constructor.
- return rtp_header;
-}
-
AudioCodingModule::Config AcmConfig(
NetEqFactory* neteq_factory,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
@@ -90,15 +75,13 @@
return acm_config;
}
-class ChannelReceive : public ChannelReceiveInterface,
- public MediaTransportAudioSinkInterface {
+class ChannelReceive : public ChannelReceiveInterface {
public:
// Used for receive streams.
ChannelReceive(Clock* clock,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
- const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
@@ -177,12 +160,6 @@
// Used for obtaining RTT for a receive-only channel.
void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
- // TODO(sukhanov): Return const pointer. It requires making media transport
- // getters like GetLatestTargetTransferRate to be also const.
- MediaTransportInterface* media_transport() const {
- return media_transport_config_.media_transport;
- }
-
private:
void ReceivePacket(const uint8_t* packet,
size_t packet_length,
@@ -193,10 +170,6 @@
int GetRtpTimestampRateHz() const;
int64_t GetRTT() const;
- // MediaTransportAudioSinkInterface override;
- void OnData(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) override;
-
void OnReceivedPayloadData(rtc::ArrayView<const uint8_t> payload,
const RTPHeader& rtpHeader);
@@ -283,8 +256,6 @@
rtc::ThreadChecker construction_thread_;
- MediaTransportConfig media_transport_config_;
-
// E2EE Audio Frame Decryption
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
webrtc::CryptoOptions crypto_options_;
@@ -293,9 +264,6 @@
void ChannelReceive::OnReceivedPayloadData(
rtc::ArrayView<const uint8_t> payload,
const RTPHeader& rtpHeader) {
- // We should not be receiving any RTP packets if media_transport is set.
- RTC_CHECK(!media_transport());
-
if (!Playing()) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
@@ -320,26 +288,6 @@
}
}
-// MediaTransportAudioSinkInterface override.
-void ChannelReceive::OnData(uint64_t channel_id,
- MediaTransportEncodedAudioFrame frame) {
- RTC_CHECK(media_transport());
-
- if (!Playing()) {
- // Avoid inserting into NetEQ when we are not playing. Count the
- // packet as discarded.
- return;
- }
-
- // Send encoded audio frame to Decoder / NetEq.
- if (acm_receiver_.InsertPacket(
- CreateRTPHeaderForMediaTransportFrame(frame, channel_id),
- frame.encoded_data()) != 0) {
- RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
- "push data to the ACM";
- }
-}
-
AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
int sample_rate_hz,
AudioFrame* audio_frame) {
@@ -460,7 +408,6 @@
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
- const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
@@ -492,7 +439,6 @@
_audioDeviceModulePtr(audio_device_module),
_outputGain(1.0f),
associated_send_channel_(nullptr),
- media_transport_config_(media_transport_config),
frame_decryptor_(frame_decryptor),
crypto_options_(crypto_options) {
// TODO(nisse): Use _moduleProcessThreadPtr instead?
@@ -526,19 +472,11 @@
// Ensure that RTCP is enabled for the created channel.
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
-
- if (media_transport()) {
- media_transport()->SetReceiveAudioSink(this);
- }
}
ChannelReceive::~ChannelReceive() {
RTC_DCHECK(construction_thread_.IsCurrent());
- if (media_transport()) {
- media_transport()->SetReceiveAudioSink(nullptr);
- }
-
StopPlayout();
if (_moduleProcessThreadPtr)
@@ -931,14 +869,6 @@
}
int64_t ChannelReceive::GetRTT() const {
- if (media_transport()) {
- auto target_rate = media_transport()->GetLatestTargetTransferRate();
- if (target_rate.has_value()) {
- return target_rate->network_estimate.round_trip_time.ms();
- }
-
- return 0;
- }
std::vector<RTCPReportBlock> report_blocks;
_rtpRtcpModule->RemoteRTCPStat(&report_blocks);
@@ -973,7 +903,6 @@
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
- const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
@@ -988,8 +917,8 @@
const webrtc::CryptoOptions& crypto_options) {
return std::make_unique<ChannelReceive>(
clock, module_process_thread, neteq_factory, audio_device_module,
- media_transport_config, rtcp_send_transport, rtc_event_log, local_ssrc,
- remote_ssrc, jitter_buffer_max_packets, jitter_buffer_fast_playout,
+ rtcp_send_transport, rtc_event_log, local_ssrc, remote_ssrc,
+ jitter_buffer_max_packets, jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms, jitter_buffer_enable_rtx_handling,
decoder_factory, codec_pair_id, frame_decryptor, crypto_options);
}
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index 3cab489..034ac7b 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -23,8 +23,6 @@
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/neteq/neteq_factory.h"
-#include "api/transport/media/media_transport_config.h"
-#include "api/transport/media/media_transport_interface.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
@@ -146,7 +144,6 @@
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
AudioDeviceModule* audio_device_module,
- const MediaTransportConfig& media_transport_config,
Transport* rtcp_send_transport,
RtcEventLog* rtc_event_log,
uint32_t local_ssrc,
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 184ea41..5bb2cbe 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -52,34 +52,14 @@
constexpr int64_t kMaxRetransmissionWindowMs = 1000;
constexpr int64_t kMinRetransmissionWindowMs = 30;
-MediaTransportEncodedAudioFrame::FrameType
-MediaTransportFrameTypeForWebrtcFrameType(webrtc::AudioFrameType frame_type) {
- switch (frame_type) {
- case AudioFrameType::kAudioFrameSpeech:
- return MediaTransportEncodedAudioFrame::FrameType::kSpeech;
- break;
-
- case AudioFrameType::kAudioFrameCN:
- return MediaTransportEncodedAudioFrame::FrameType::
- kDiscontinuousTransmission;
- break;
-
- default:
- RTC_CHECK(false) << "Unexpected frame type="
- << static_cast<int>(frame_type);
- break;
- }
-}
-
class RtpPacketSenderProxy;
class TransportFeedbackProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
class ChannelSend : public ChannelSendInterface,
- public AudioPacketizationCallback, // receive encoded
- // packets from the ACM
- public TargetTransferRateObserver {
+ public AudioPacketizationCallback { // receive encoded
+ // packets from the ACM
public:
// TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
// declaration.
@@ -88,7 +68,6 @@
ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -188,21 +167,8 @@
rtc::ArrayView<const uint8_t> payload)
RTC_RUN_ON(encoder_queue_);
- int32_t SendMediaTransportAudio(AudioFrameType frameType,
- uint8_t payloadType,
- uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload)
- RTC_RUN_ON(encoder_queue_);
-
- // Return media transport or nullptr if using RTP.
- MediaTransportInterface* media_transport() {
- return media_transport_config_.media_transport;
- }
-
void OnReceivedRtt(int64_t rtt_ms);
- void OnTargetTransferRate(TargetTransferRate) override;
-
// Thread checkers document and lock usage of some methods on voe::Channel to
// specific threads we know about. The goal is to eventually split up
// voe::Channel into parts with single-threaded semantics, and thereby reduce
@@ -251,20 +217,6 @@
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
- MediaTransportConfig media_transport_config_;
- int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
-
- rtc::CriticalSection media_transport_lock_;
- // Currently set to local SSRC at construction.
- uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) =
- 0;
- // Cache payload type and sampling frequency from most recent call to
- // SetEncoder. Needed to set MediaTransportEncodedAudioFrame metadata, and
- // invalidate on encoder change.
- int media_transport_payload_type_ RTC_GUARDED_BY(&media_transport_lock_);
- int media_transport_sampling_frequency_
- RTC_GUARDED_BY(&media_transport_lock_);
-
// E2EE Audio Frame Encryption
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
RTC_GUARDED_BY(encoder_queue_);
@@ -421,18 +373,7 @@
size_t payloadSize) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
-
- if (media_transport() != nullptr) {
- if (frameType == AudioFrameType::kEmptyFrame) {
- // TODO(bugs.webrtc.org/9719): Media transport Send doesn't support
- // sending empty frames.
- return 0;
- }
-
- return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload);
- } else {
- return SendRtpAudio(frameType, payloadType, timeStamp, payload);
- }
+ return SendRtpAudio(frameType, payloadType, timeStamp, payload);
}
int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
@@ -512,64 +453,9 @@
return 0;
}
-int32_t ChannelSend::SendMediaTransportAudio(
- AudioFrameType frameType,
- uint8_t payloadType,
- uint32_t timeStamp,
- rtc::ArrayView<const uint8_t> payload) {
- // TODO(nisse): Use null _transportPtr for MediaTransport.
- // RTC_DCHECK(_transportPtr == nullptr);
- uint64_t channel_id;
- int sampling_rate_hz;
- {
- rtc::CritScope cs(&media_transport_lock_);
- if (media_transport_payload_type_ != payloadType) {
- // Payload type is being changed, media_transport_sampling_frequency_,
- // no longer current.
- return -1;
- }
- sampling_rate_hz = media_transport_sampling_frequency_;
- channel_id = media_transport_channel_id_;
- }
- MediaTransportEncodedAudioFrame frame(
- /*sampling_rate_hz=*/sampling_rate_hz,
-
- // TODO(nisse): Timestamp and sample index are the same for all supported
- // audio codecs except G722. Refactor audio coding module to only use
- // sample index, and leave translation to RTP time, when needed, for
- // RTP-specific code.
- /*starting_sample_index=*/timeStamp,
-
- // Sample count isn't conveniently available from the AudioCodingModule,
- // and needs some refactoring to wire up in a good way. For now, left as
- // zero.
- /*samples_per_channel=*/0,
-
- /*sequence_number=*/media_transport_sequence_number_,
- MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType,
- std::vector<uint8_t>(payload.begin(), payload.end()));
-
- // TODO(nisse): Introduce a MediaTransportSender object bound to a specific
- // channel id.
- RTCError rtc_error =
- media_transport()->SendAudioFrame(channel_id, std::move(frame));
-
- if (!rtc_error.ok()) {
- RTC_LOG(LS_ERROR) << "Failed to send frame, rtc_error="
- << ToString(rtc_error.type()) << ", "
- << rtc_error.message();
- return -1;
- }
-
- ++media_transport_sequence_number_;
-
- return 0;
-}
-
ChannelSend::ChannelSend(Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -591,7 +477,6 @@
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
- media_transport_config_(media_transport_config),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
encoder_queue_(task_queue_factory->CreateTaskQueue(
@@ -603,17 +488,9 @@
audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
RtpRtcp::Configuration configuration;
-
- // We gradually remove codepaths that depend on RTP when using media
- // transport. All of this logic should be moved to the future
- // RTPMediaTransport. In this case it means that overhead and bandwidth
- // observers should not be called when using media transport.
- if (!media_transport_config.media_transport) {
- configuration.overhead_observer = overhead_observer;
- configuration.bandwidth_callback = rtcp_observer_.get();
- configuration.transport_feedback_callback = feedback_observer_proxy_.get();
- }
-
+ configuration.overhead_observer = overhead_observer;
+ configuration.bandwidth_callback = rtcp_observer_.get();
+ configuration.transport_feedback_callback = feedback_observer_proxy_.get();
configuration.clock = clock;
configuration.audio = true;
configuration.clock = Clock::GetRealTimeClock();
@@ -629,10 +506,6 @@
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
configuration.local_media_ssrc = ssrc;
- if (media_transport_config_.media_transport) {
- rtc::CritScope cs(&media_transport_lock_);
- media_transport_channel_id_ = ssrc;
- }
_rtpRtcpModule = RtpRtcp::Create(configuration);
_rtpRtcpModule->SetSendingMediaStatus(false);
@@ -640,17 +513,6 @@
rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(
configuration.clock, _rtpRtcpModule->RtpSender());
- // We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
- // callbacks after the audio_coding_ is fully initialized.
- if (media_transport_config.media_transport) {
- RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
- media_transport_config.media_transport->AddTargetTransferRateObserver(this);
- media_transport_config.media_transport->SetAudioOverheadObserver(
- overhead_observer);
- } else {
- RTC_DLOG(LS_INFO) << "Not setting media_transport_ rate observers.";
- }
-
_moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
// Ensure that RTCP is enabled by default for the created channel.
@@ -663,12 +525,6 @@
ChannelSend::~ChannelSend() {
RTC_DCHECK(construction_thread_.IsCurrent());
- if (media_transport_config_.media_transport) {
- media_transport_config_.media_transport->RemoveTargetTransferRateObserver(
- this);
- media_transport_config_.media_transport->SetAudioOverheadObserver(nullptr);
- }
-
StopSend();
int error = audio_coding_->RegisterTransportCallback(NULL);
RTC_DCHECK_EQ(0, error);
@@ -729,13 +585,6 @@
encoder->RtpTimestampRateHz(),
encoder->NumChannels(), 0);
- if (media_transport_config_.media_transport) {
- rtc::CritScope cs(&media_transport_lock_);
- media_transport_payload_type_ = payload_type;
- // TODO(nisse): Currently broken for G722, since timestamps passed through
- // encoder use RTP clock rather than sample count, and they differ for G722.
- media_transport_sampling_frequency_ = encoder->RtpTimestampRateHz();
- }
audio_coding_->SetEncoder(std::move(encoder));
}
@@ -785,13 +634,6 @@
}
void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
- // May be called on either worker thread or network thread.
- if (media_transport_config_.media_transport) {
- // Ignore RTCP packets while media transport is used.
- // Those packets should not arrive, but we are seeing occasional packets.
- return;
- }
-
// Deliver RTCP packet to RTP/RTCP module for parsing
_rtpRtcpModule->IncomingRtcpPacket(data, length);
@@ -1064,19 +906,6 @@
}
int64_t ChannelSend::GetRTT() const {
- if (media_transport_config_.media_transport) {
- // GetRTT is generally used in the RTCP codepath, where media transport is
- // not present and so it shouldn't be needed. But it's also invoked in
- // 'GetStats' method, and for now returning media transport RTT here gives
- // us "free" rtt stats for media transport.
- auto target_rate =
- media_transport_config_.media_transport->GetLatestTargetTransferRate();
- if (target_rate.has_value()) {
- return target_rate.value().network_estimate.round_trip_time.ms();
- }
-
- return 0;
- }
std::vector<RTCPReportBlock> report_blocks;
_rtpRtcpModule->RemoteRTCPStat(&report_blocks);
@@ -1106,14 +935,6 @@
});
}
-// TODO(sukhanov): Consider moving TargetTransferRate observer to
-// AudioSendStream. Since AudioSendStream owns encoder and configures ANA, it
-// makes sense to consolidate all rate (and overhead) calculation there.
-void ChannelSend::OnTargetTransferRate(TargetTransferRate rate) {
- RTC_DCHECK(media_transport_config_.media_transport);
- OnReceivedRtt(rate.network_estimate.round_trip_time.ms());
-}
-
void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
// Invoke audio encoders OnReceivedRtt().
CallEncoder(
@@ -1126,7 +947,6 @@
Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
@@ -1137,10 +957,9 @@
int rtcp_report_interval_ms,
uint32_t ssrc) {
return std::make_unique<ChannelSend>(
- clock, task_queue_factory, module_process_thread, media_transport_config,
- overhead_observer, rtp_transport, rtcp_rtt_stats, rtc_event_log,
- frame_encryptor, crypto_options, extmap_allow_mixed,
- rtcp_report_interval_ms, ssrc);
+ clock, task_queue_factory, module_process_thread, overhead_observer,
+ rtp_transport, rtcp_rtt_stats, rtc_event_log, frame_encryptor,
+ crypto_options, extmap_allow_mixed, rtcp_report_interval_ms, ssrc);
}
} // namespace voe
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 053b69a..6f73c2b 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -20,8 +20,6 @@
#include "api/crypto/crypto_options.h"
#include "api/function_view.h"
#include "api/task_queue/task_queue_factory.h"
-#include "api/transport/media/media_transport_config.h"
-#include "api/transport/media/media_transport_interface.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
@@ -129,7 +127,6 @@
Clock* clock,
TaskQueueFactory* task_queue_factory,
ProcessThread* module_process_thread,
- const MediaTransportConfig& media_transport_config,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
diff --git a/audio/test/media_transport_test.cc b/audio/test/media_transport_test.cc
deleted file mode 100644
index bee0539..0000000
--- a/audio/test/media_transport_test.cc
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2018 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 <memory>
-
-#include "api/audio_codecs/audio_decoder_factory_template.h"
-#include "api/audio_codecs/audio_encoder_factory_template.h"
-#include "api/audio_codecs/opus/audio_decoder_opus.h"
-#include "api/audio_codecs/opus/audio_encoder_opus.h"
-#include "api/rtc_event_log/rtc_event_log.h"
-#include "api/task_queue/default_task_queue_factory.h"
-#include "api/test/loopback_media_transport.h"
-#include "api/test/mock_audio_mixer.h"
-#include "api/transport/media/media_transport_config.h"
-#include "audio/audio_receive_stream.h"
-#include "audio/audio_send_stream.h"
-#include "call/rtp_transport_controller_send.h"
-#include "call/test/mock_bitrate_allocator.h"
-#include "modules/audio_device/include/test_audio_device.h"
-#include "modules/audio_mixer/audio_mixer_impl.h"
-#include "modules/audio_processing/include/mock_audio_processing.h"
-#include "modules/utility/include/process_thread.h"
-#include "rtc_base/time_utils.h"
-#include "test/gtest.h"
-#include "test/mock_transport.h"
-
-namespace webrtc {
-namespace test {
-
-namespace {
-using ::testing::NiceMock;
-
-constexpr int kPayloadTypeOpus = 17;
-constexpr int kSamplingFrequency = 48000;
-constexpr int kNumChannels = 2;
-constexpr int kWantedSamples = 3000;
-constexpr int kTestTimeoutMs = 2 * rtc::kNumMillisecsPerSec;
-
-class TestRenderer : public TestAudioDeviceModule::Renderer {
- public:
- TestRenderer(int sampling_frequency, int num_channels, size_t wanted_samples)
- : sampling_frequency_(sampling_frequency),
- num_channels_(num_channels),
- wanted_samples_(wanted_samples) {}
- ~TestRenderer() override = default;
-
- int SamplingFrequency() const override { return sampling_frequency_; }
- int NumChannels() const override { return num_channels_; }
-
- bool Render(rtc::ArrayView<const int16_t> data) override {
- if (data.size() >= wanted_samples_) {
- return false;
- }
- wanted_samples_ -= data.size();
- return true;
- }
-
- private:
- const int sampling_frequency_;
- const int num_channels_;
- size_t wanted_samples_;
-};
-
-} // namespace
-
-TEST(AudioWithMediaTransport, DeliversAudio) {
- std::unique_ptr<rtc::Thread> transport_thread = rtc::Thread::Create();
- transport_thread->Start();
- std::unique_ptr<TaskQueueFactory> task_queue_factory =
- CreateDefaultTaskQueueFactory();
- MediaTransportPair transport_pair(transport_thread.get());
- NiceMock<MockTransport> rtcp_send_transport;
- NiceMock<MockTransport> send_transport;
- RtcEventLogNull null_event_log;
- NiceMock<MockBitrateAllocator> bitrate_allocator;
-
- rtc::scoped_refptr<TestAudioDeviceModule> audio_device =
- TestAudioDeviceModule::Create(
- task_queue_factory.get(),
- TestAudioDeviceModule::CreatePulsedNoiseCapturer(
- /* max_amplitude= */ 10000, kSamplingFrequency, kNumChannels),
- std::make_unique<TestRenderer>(kSamplingFrequency, kNumChannels,
- kWantedSamples));
-
- AudioState::Config audio_config;
- audio_config.audio_mixer = AudioMixerImpl::Create();
- // TODO(nisse): Is a mock AudioProcessing enough?
- audio_config.audio_processing =
- new rtc::RefCountedObject<MockAudioProcessing>();
- audio_config.audio_device_module = audio_device;
- rtc::scoped_refptr<AudioState> audio_state = AudioState::Create(audio_config);
-
- // TODO(nisse): Use some lossless codec?
- const SdpAudioFormat audio_format("opus", kSamplingFrequency, kNumChannels);
-
- // Setup receive stream;
- webrtc::AudioReceiveStream::Config receive_config;
- // TODO(nisse): Update AudioReceiveStream to not require rtcp_send_transport
- // when a MediaTransport is provided.
- receive_config.rtcp_send_transport = &rtcp_send_transport;
- receive_config.media_transport_config.media_transport =
- transport_pair.first();
- receive_config.decoder_map.emplace(kPayloadTypeOpus, audio_format);
- receive_config.decoder_factory =
- CreateAudioDecoderFactory<AudioDecoderOpus>();
-
- std::unique_ptr<ProcessThread> receive_process_thread =
- ProcessThread::Create("audio recv thread");
-
- webrtc::internal::AudioReceiveStream receive_stream(
- Clock::GetRealTimeClock(),
- /*receiver_controller=*/nullptr,
- /*packet_router=*/nullptr, receive_process_thread.get(),
- /*neteq_factory=*/nullptr, receive_config, audio_state, &null_event_log);
-
- // TODO(nisse): Update AudioSendStream to not require send_transport when a
- // MediaTransport is provided.
- AudioSendStream::Config send_config(
- &send_transport, webrtc::MediaTransportConfig(transport_pair.second()));
- send_config.send_codec_spec =
- AudioSendStream::Config::SendCodecSpec(kPayloadTypeOpus, audio_format);
- send_config.encoder_factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
- std::unique_ptr<ProcessThread> send_process_thread =
- ProcessThread::Create("audio send thread");
- FieldTrialBasedConfig field_trials;
- RtpTransportControllerSend rtp_transport(
- Clock::GetRealTimeClock(), &null_event_log, nullptr, nullptr,
- BitrateConstraints(), ProcessThread::Create("Pacer"),
- task_queue_factory.get(), &field_trials);
- webrtc::internal::AudioSendStream send_stream(
- Clock::GetRealTimeClock(), send_config, audio_state,
- task_queue_factory.get(), send_process_thread.get(), &rtp_transport,
- &bitrate_allocator, &null_event_log,
- /*rtcp_rtt_stats=*/nullptr, absl::optional<RtpState>());
-
- audio_device->Init(); // Starts thread.
- audio_device->RegisterAudioCallback(audio_state->audio_transport());
-
- receive_stream.Start();
- send_stream.Start();
- audio_device->StartPlayout();
- audio_device->StartRecording();
-
- EXPECT_TRUE(audio_device->WaitForPlayoutEnd(kTestTimeoutMs));
-
- audio_device->StopRecording();
- audio_device->StopPlayout();
- receive_stream.Stop();
- send_stream.Stop();
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 26a0b37..81afe55 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -47,7 +47,6 @@
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/transport:webrtc_key_value_config",
- "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../modules/audio_device",
"../modules/audio_processing",
@@ -292,7 +291,6 @@
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
- "../api/transport/media:media_transport_interface",
"../api/transport/rtp:rtp_source",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
@@ -383,8 +381,6 @@
":rtp_sender",
":simulated_network",
"../api:array_view",
- "../api:fake_media_transport",
- "../api:fake_media_transport",
"../api:mock_audio_mixer",
"../api:rtp_headers",
"../api:rtp_parameters",
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 55c1af7..090fb82 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -23,7 +23,6 @@
#include "api/crypto/frame_decryptor_interface.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
-#include "api/transport/media/media_transport_config.h"
#include "api/transport/rtp/rtp_source.h"
#include "call/rtp_config.h"
@@ -125,8 +124,6 @@
Transport* rtcp_send_transport = nullptr;
- MediaTransportConfig media_transport_config;
-
// NetEq settings.
size_t jitter_buffer_max_packets = 200;
bool jitter_buffer_fast_accelerate = false;
diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc
index 6fdb39c..ddcba03 100644
--- a/call/audio_send_stream.cc
+++ b/call/audio_send_stream.cc
@@ -21,14 +21,8 @@
AudioSendStream::Stats::Stats() = default;
AudioSendStream::Stats::~Stats() = default;
-AudioSendStream::Config::Config(
- Transport* send_transport,
- const MediaTransportConfig& media_transport_config)
- : send_transport(send_transport),
- media_transport_config(media_transport_config) {}
-
AudioSendStream::Config::Config(Transport* send_transport)
- : Config(send_transport, MediaTransportConfig()) {}
+ : send_transport(send_transport) {}
AudioSendStream::Config::~Config() = default;
@@ -38,7 +32,6 @@
ss << "{rtp: " << rtp.ToString();
ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
- ss << ", media_transport_config: " << media_transport_config.DebugString();
ss << ", min_bitrate_bps: " << min_bitrate_bps;
ss << ", max_bitrate_bps: " << max_bitrate_bps;
ss << ", send_codec_spec: "
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index f373055..734be30 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -25,8 +25,6 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
-#include "api/transport/media/media_transport_config.h"
-#include "api/transport/media/media_transport_interface.h"
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
@@ -76,8 +74,6 @@
struct Config {
Config() = delete;
- Config(Transport* send_transport,
- const MediaTransportConfig& media_transport_config);
explicit Config(Transport* send_transport);
~Config();
std::string ToString() const;
@@ -116,8 +112,6 @@
// the entire life of the AudioSendStream and is owned by the API client.
Transport* send_transport = nullptr;
- MediaTransportConfig media_transport_config;
-
// Bitrate limits used for variable audio bitrate streams. Set both to -1 to
// disable audio bitrate adaptation.
// Note: This is still an experimental feature and not ready for real usage.
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index de91b66..7e59020 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -244,8 +244,7 @@
CreateSendConfig(1, 0, 0, video_send_transport.get());
CreateMatchingReceiveConfigs(receive_transport.get());
- AudioSendStream::Config audio_send_config(audio_send_transport.get(),
- MediaTransportConfig());
+ AudioSendStream::Config audio_send_config(audio_send_transport.get());
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"ISAC", 16000, 1});
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index cf2037e..754be81 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -19,7 +19,6 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/default_task_queue_factory.h"
-#include "api/test/fake_media_transport.h"
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/audio_send_stream.h"
@@ -68,7 +67,7 @@
TEST(CallTest, CreateDestroy_AudioSendStream) {
CallHelper call;
MockTransport send_transport;
- AudioSendStream::Config config(&send_transport, MediaTransportConfig());
+ AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = 42;
AudioSendStream* stream = call->CreateAudioSendStream(config);
EXPECT_NE(stream, nullptr);
@@ -91,7 +90,7 @@
TEST(CallTest, CreateDestroy_AudioSendStreams) {
CallHelper call;
MockTransport send_transport;
- AudioSendStream::Config config(&send_transport, MediaTransportConfig());
+ AudioSendStream::Config config(&send_transport);
std::list<AudioSendStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
@@ -150,7 +149,7 @@
EXPECT_NE(recv_stream, nullptr);
MockTransport send_transport;
- AudioSendStream::Config send_config(&send_transport, MediaTransportConfig());
+ AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
@@ -169,7 +168,7 @@
TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
CallHelper call;
MockTransport send_transport;
- AudioSendStream::Config send_config(&send_transport, MediaTransportConfig());
+ AudioSendStream::Config send_config(&send_transport);
send_config.rtp.ssrc = 777;
AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
EXPECT_NE(send_stream, nullptr);
@@ -273,7 +272,7 @@
auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
MockTransport send_transport;
- AudioSendStream::Config config(&send_transport, MediaTransportConfig());
+ AudioSendStream::Config config(&send_transport);
config.rtp.ssrc = ssrc;
AudioSendStream* stream = call->CreateAudioSendStream(config);
const RtpState rtp_state =
diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc
index acda498..c4895e4 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -69,12 +69,8 @@
VideoReceiveStream::Config::Config(const Config&) = default;
VideoReceiveStream::Config::Config(Config&&) = default;
-VideoReceiveStream::Config::Config(Transport* rtcp_send_transport,
- MediaTransportConfig media_transport_config)
- : rtcp_send_transport(rtcp_send_transport),
- media_transport_config(media_transport_config) {}
VideoReceiveStream::Config::Config(Transport* rtcp_send_transport)
- : Config(rtcp_send_transport, MediaTransportConfig()) {}
+ : rtcp_send_transport(rtcp_send_transport) {}
VideoReceiveStream::Config& VideoReceiveStream::Config::operator=(Config&&) =
default;
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 2959f67..96c60b5 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -22,8 +22,6 @@
#include "api/crypto/frame_decryptor_interface.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
-#include "api/transport/media/media_transport_config.h"
-#include "api/transport/media/media_transport_interface.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
@@ -139,8 +137,6 @@
public:
Config() = delete;
Config(Config&&);
- Config(Transport* rtcp_send_transport,
- MediaTransportConfig media_transport_config);
explicit Config(Transport* rtcp_send_transport);
Config& operator=(Config&&);
Config& operator=(const Config&) = delete;
@@ -151,10 +147,6 @@
std::string ToString() const;
- MediaTransportInterface* media_transport() const {
- return media_transport_config.media_transport;
- }
-
// Decoders for every payload that we can receive.
std::vector<Decoder> decoders;
@@ -217,8 +209,6 @@
// Transport for outgoing packets (RTCP).
Transport* rtcp_send_transport = nullptr;
- MediaTransportConfig media_transport_config;
-
// Must always be set.
rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
diff --git a/call/video_send_stream.cc b/call/video_send_stream.cc
index dac4029..f495d08 100644
--- a/call/video_send_stream.cc
+++ b/call/video_send_stream.cc
@@ -75,14 +75,10 @@
VideoSendStream::Config::Config(const Config&) = default;
VideoSendStream::Config::Config(Config&&) = default;
-VideoSendStream::Config::Config(Transport* send_transport,
- MediaTransportInterface* media_transport)
+VideoSendStream::Config::Config(Transport* send_transport)
: rtp(),
encoder_settings(VideoEncoder::Capabilities(rtp.lntf.enabled)),
- send_transport(send_transport),
- media_transport(media_transport) {}
-VideoSendStream::Config::Config(Transport* send_transport)
- : Config(send_transport, nullptr) {}
+ send_transport(send_transport) {}
VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default;
VideoSendStream::Config::Config::~Config() = default;
@@ -95,7 +91,6 @@
ss << ", rtp: " << rtp.ToString();
ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
- ss << ", media_transport: " << (media_transport ? "(Transport)" : "nullptr");
ss << ", render_delay_ms: " << render_delay_ms;
ss << ", target_delay_ms: " << target_delay_ms;
ss << ", suspend_below_min_bitrate: "
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 478d73c..39abdfc 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -21,7 +21,6 @@
#include "api/call/transport.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
-#include "api/transport/media/media_transport_interface.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
@@ -116,7 +115,6 @@
public:
Config() = delete;
Config(Config&&);
- Config(Transport* send_transport, MediaTransportInterface* media_transport);
explicit Config(Transport* send_transport);
Config& operator=(Config&&);
@@ -139,8 +137,6 @@
// Transport for outgoing packets.
Transport* send_transport = nullptr;
- MediaTransportInterface* media_transport = nullptr;
-
// Expected delay needed by the renderer, i.e. the frame will be delivered
// this many milliseconds, if possible, earlier than expected render time.
// Only valid if |local_renderer| is set.
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 009741f..9912d29 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -523,7 +523,6 @@
":rtc_vp9_profile",
"../:webrtc_common",
"../api:create_simulcast_test_fixture_api",
- "../api:fake_media_transport",
"../api:libjingle_peerconnection_api",
"../api:mock_video_bitrate_allocator",
"../api:mock_video_bitrate_allocator_factory",
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 026d371..185c883 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -271,10 +271,6 @@
return media_transport_config_;
}
- webrtc::MediaTransportInterface* media_transport() {
- return media_transport_config_.media_transport;
- }
-
// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
// Set to true if it's allowed to mix one- and two-byte RTP header extensions
// in the same stream. The setter and getter must only be called from
diff --git a/media/base/rtp_data_engine_unittest.cc b/media/base/rtp_data_engine_unittest.cc
index 79fb2b2..e46c83e 100644
--- a/media/base/rtp_data_engine_unittest.cc
+++ b/media/base/rtp_data_engine_unittest.cc
@@ -15,7 +15,6 @@
#include <memory>
#include <string>
-#include "api/transport/media/media_transport_config.h"
#include "media/base/fake_network_interface.h"
#include "media/base/media_constants.h"
#include "media/base/rtp_utils.h"
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 818be4a..f36314f 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -826,25 +826,8 @@
: send_params_.max_bandwidth_bps;
}
- if (media_transport()) {
- webrtc::MediaTransportTargetRateConstraints constraints;
- if (bitrate_config_.start_bitrate_bps >= 0) {
- constraints.starting_bitrate =
- webrtc::DataRate::bps(bitrate_config_.start_bitrate_bps);
- }
- if (bitrate_config_.max_bitrate_bps > 0) {
- constraints.max_bitrate =
- webrtc::DataRate::bps(bitrate_config_.max_bitrate_bps);
- }
- if (bitrate_config_.min_bitrate_bps >= 0) {
- constraints.min_bitrate =
- webrtc::DataRate::bps(bitrate_config_.min_bitrate_bps);
- }
- media_transport()->SetTargetBitrateLimits(constraints);
- } else {
- call_->GetTransportControllerSend()->SetSdpBitrateParameters(
- bitrate_config_);
- }
+ call_->GetTransportControllerSend()->SetSdpBitrateParameters(
+ bitrate_config_);
}
for (auto& kv : send_streams_) {
@@ -1175,7 +1158,7 @@
for (uint32_t used_ssrc : sp.ssrcs)
send_ssrcs_.insert(used_ssrc);
- webrtc::VideoSendStream::Config config(this, media_transport());
+ webrtc::VideoSendStream::Config config(this);
for (const RidDescription& rid : sp.rids()) {
config.rtp.rids.push_back(rid.rid);
@@ -1308,7 +1291,7 @@
for (uint32_t used_ssrc : sp.ssrcs)
receive_ssrcs_.insert(used_ssrc);
- webrtc::VideoReceiveStream::Config config(this, media_transport_config());
+ webrtc::VideoReceiveStream::Config config(this);
webrtc::FlexfecReceiveStream::Config flexfec_config(this);
ConfigureReceiverRtp(&config, &flexfec_config, sp);
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 9d6e449..5c24454 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -22,7 +22,6 @@
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/default_task_queue_factory.h"
-#include "api/test/fake_media_transport.h"
#include "api/test/mock_video_bitrate_allocator.h"
#include "api/test/mock_video_bitrate_allocator_factory.h"
#include "api/test/mock_video_decoder_factory.h"
@@ -4249,82 +4248,6 @@
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
}
-// Test that when both the codec-specific bitrate params and max_bandwidth_bps
-// are present in the same send parameters, the settings are combined correctly.
-TEST_F(WebRtcVideoChannelTest,
- SetSendCodecsWithBitratesAndMaxSendBandwidthForMediaTransport) {
- // Same as SetSendCodecsWithBitratesAndMaxSendBandwidth but with Media
- // Transport.
- webrtc::MediaTransportSettings settings;
- settings.is_caller = true;
- webrtc::FakeMediaTransport fake_media_transport(settings);
- std::unique_ptr<cricket::FakeNetworkInterface> network_interface(
- new cricket::FakeNetworkInterface);
- channel_->SetInterface(network_interface.get(),
- webrtc::MediaTransportConfig(&fake_media_transport));
-
- send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
- send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
- send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
- send_parameters_.max_bandwidth_bps = 400000;
- {
- // We expect max_bandwidth_bps to take priority, if set.
- ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
- ASSERT_EQ(1u,
- fake_media_transport.target_rate_constraints_in_order().size());
- const webrtc::MediaTransportTargetRateConstraints& constraint =
- fake_media_transport.target_rate_constraints_in_order()[0];
- ASSERT_EQ(webrtc::DataRate::bps(100000), constraint.min_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(200000), constraint.starting_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(400000), constraint.max_bitrate);
- }
-
- {
- // Decrease max_bandwidth_bps.
- send_parameters_.max_bandwidth_bps = 350000;
- ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
- ASSERT_EQ(2u,
- fake_media_transport.target_rate_constraints_in_order().size());
- const webrtc::MediaTransportTargetRateConstraints& constraint =
- fake_media_transport.target_rate_constraints_in_order()[1];
-
- // Since the codec isn't changing, start_bitrate_bps should be 0.
- ASSERT_EQ(webrtc::DataRate::bps(100000), constraint.min_bitrate);
- ASSERT_EQ(absl::nullopt, constraint.starting_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(350000), constraint.max_bitrate);
- }
-
- {
- // Now try again with the values flipped around.
- send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400";
- send_parameters_.max_bandwidth_bps = 300000;
- ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
- ASSERT_EQ(3u,
- fake_media_transport.target_rate_constraints_in_order().size());
- const webrtc::MediaTransportTargetRateConstraints& constraint =
- fake_media_transport.target_rate_constraints_in_order()[2];
-
- ASSERT_EQ(webrtc::DataRate::bps(100000), constraint.min_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(200000), constraint.starting_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(300000), constraint.max_bitrate);
- }
-
- {
- // Now try again with the values flipped around.
- // If we change the codec max, max_bandwidth_bps should still apply.
- send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350";
- ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
- ASSERT_EQ(4u,
- fake_media_transport.target_rate_constraints_in_order().size());
- const webrtc::MediaTransportTargetRateConstraints& constraint =
- fake_media_transport.target_rate_constraints_in_order()[3];
-
- ASSERT_EQ(webrtc::DataRate::bps(100000), constraint.min_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(200000), constraint.starting_bitrate);
- ASSERT_EQ(webrtc::DataRate::bps(300000), constraint.max_bitrate);
- }
-}
-
TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldPreserveOtherBitrates) {
SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
200000);
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 201503a..eab2bc8 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -21,7 +21,6 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
-#include "api/transport/media/media_transport_interface.h"
#include "media/base/audio_source.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
@@ -697,13 +696,12 @@
const absl::optional<std::string>& audio_network_adaptor_config,
webrtc::Call* call,
webrtc::Transport* send_transport,
- const webrtc::MediaTransportConfig& media_transport_config,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
const webrtc::CryptoOptions& crypto_options)
: call_(call),
- config_(send_transport, media_transport_config),
+ config_(send_transport),
max_send_bitrate_bps_(max_send_bitrate_bps),
rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
RTC_DCHECK(call);
@@ -1052,7 +1050,6 @@
const std::vector<webrtc::RtpExtension>& extensions,
webrtc::Call* call,
webrtc::Transport* rtcp_send_transport,
- const webrtc::MediaTransportConfig& media_transport_config,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
@@ -1070,7 +1067,6 @@
config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
config_.rtp.extensions = extensions;
config_.rtcp_send_transport = rtcp_send_transport;
- config_.media_transport_config = media_transport_config;
config_.jitter_buffer_max_packets = jitter_buffer_max_packets;
config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
config_.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms;
@@ -1803,8 +1799,8 @@
ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(),
send_rtp_extensions_, max_send_bitrate_bps_,
audio_config_.rtcp_report_interval_ms, audio_network_adaptor_config,
- call_, this, media_transport_config(), engine()->encoder_factory_,
- codec_pair_id_, nullptr, crypto_options_);
+ call_, this, engine()->encoder_factory_, codec_pair_id_, nullptr,
+ crypto_options_);
send_streams_.insert(std::make_pair(ssrc, stream));
// At this point the stream's local SSRC has been updated. If it is the first
@@ -1884,9 +1880,8 @@
ssrc, new WebRtcAudioReceiveStream(
ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
- call_, this, media_transport_config(),
- engine()->decoder_factory_, decoder_map_, codec_pair_id_,
- engine()->audio_jitter_buffer_max_packets_,
+ call_, this, engine()->decoder_factory_, decoder_map_,
+ codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
engine()->audio_jitter_buffer_fast_accelerate_,
engine()->audio_jitter_buffer_min_delay_ms_,
engine()->audio_jitter_buffer_enable_rtx_handling_,
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index fd054df..f72fad7 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -21,7 +21,6 @@
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
-#include "api/transport/media/media_transport_config.h"
#include "call/call.h"
#include "media/base/fake_media_engine.h"
#include "media/base/fake_network_interface.h"
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 01bc47d..301b86c 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -47,8 +47,6 @@
"base/ice_transport_internal.h",
"base/mdns_message.cc",
"base/mdns_message.h",
- "base/no_op_dtls_transport.cc",
- "base/no_op_dtls_transport.h",
"base/p2p_constants.cc",
"base/p2p_constants.h",
"base/p2p_transport_channel.cc",
diff --git a/p2p/base/no_op_dtls_transport.cc b/p2p/base/no_op_dtls_transport.cc
deleted file mode 100644
index 0ce03b9..0000000
--- a/p2p/base/no_op_dtls_transport.cc
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright 2018 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 "p2p/base/no_op_dtls_transport.h"
-
-#include <algorithm>
-#include <memory>
-#include <utility>
-
-#include "absl/memory/memory.h"
-#include "api/rtc_event_log/rtc_event_log.h"
-#include "logging/rtc_event_log/events/rtc_event_dtls_transport_state.h"
-#include "logging/rtc_event_log/events/rtc_event_dtls_writable_state.h"
-#include "p2p/base/packet_transport_internal.h"
-#include "rtc_base/buffer.h"
-#include "rtc_base/checks.h"
-#include "rtc_base/dscp.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/message_queue.h"
-#include "rtc_base/rtc_certificate.h"
-#include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/stream.h"
-#include "rtc_base/thread.h"
-
-namespace cricket {
-
-NoOpDtlsTransport::NoOpDtlsTransport(
- IceTransportInternal* ice_transport,
- const webrtc::CryptoOptions& crypto_options)
- : crypto_options_(webrtc::CryptoOptions::NoGcm()),
- ice_transport_(ice_transport) {
- RTC_DCHECK(ice_transport_);
- ice_transport_->SignalWritableState.connect(
- this, &NoOpDtlsTransport::OnWritableState);
- ice_transport_->SignalReadyToSend.connect(this,
- &NoOpDtlsTransport::OnReadyToSend);
- ice_transport_->SignalReceivingState.connect(
- this, &NoOpDtlsTransport::OnReceivingState);
- ice_transport_->SignalNetworkRouteChanged.connect(
- this, &NoOpDtlsTransport::OnNetworkRouteChanged);
-}
-
-NoOpDtlsTransport::~NoOpDtlsTransport() {}
-const webrtc::CryptoOptions& NoOpDtlsTransport::crypto_options() const {
- return crypto_options_;
-}
-DtlsTransportState NoOpDtlsTransport::dtls_state() const {
- return DTLS_TRANSPORT_CONNECTED;
-}
-int NoOpDtlsTransport::component() const {
- return kNoOpDtlsTransportComponent;
-}
-bool NoOpDtlsTransport::IsDtlsActive() const {
- return true;
-}
-bool NoOpDtlsTransport::GetDtlsRole(rtc::SSLRole* role) const {
- return false;
-}
-bool NoOpDtlsTransport::SetDtlsRole(rtc::SSLRole role) {
- return false;
-}
-bool NoOpDtlsTransport::GetSslVersionBytes(int* version) const {
- return false;
-}
-bool NoOpDtlsTransport::GetSrtpCryptoSuite(int* cipher) {
- return false;
-}
-bool NoOpDtlsTransport::GetSslCipherSuite(int* cipher) {
- return false;
-}
-rtc::scoped_refptr<rtc::RTCCertificate> NoOpDtlsTransport::GetLocalCertificate()
- const {
- return rtc::scoped_refptr<rtc::RTCCertificate>();
-}
-bool NoOpDtlsTransport::SetLocalCertificate(
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
- return false;
-}
-std::unique_ptr<rtc::SSLCertChain> NoOpDtlsTransport::GetRemoteSSLCertChain()
- const {
- return std::unique_ptr<rtc::SSLCertChain>();
-}
-bool NoOpDtlsTransport::ExportKeyingMaterial(const std::string& label,
- const uint8_t* context,
- size_t context_len,
- bool use_context,
- uint8_t* result,
- size_t result_len) {
- return false;
-}
-bool NoOpDtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
- const uint8_t* digest,
- size_t digest_len) {
- return true;
-}
-bool NoOpDtlsTransport::SetSslMaxProtocolVersion(
- rtc::SSLProtocolVersion version) {
- return true;
-}
-IceTransportInternal* NoOpDtlsTransport::ice_transport() {
- return ice_transport_;
-}
-
-void NoOpDtlsTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- if (is_writable_) {
- SignalReadyToSend(this);
- }
-}
-
-void NoOpDtlsTransport::OnWritableState(
- rtc::PacketTransportInternal* transport) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- is_writable_ = ice_transport_->writable();
- if (is_writable_) {
- SignalWritableState(this);
- }
-}
-const std::string& NoOpDtlsTransport::transport_name() const {
- return ice_transport_->transport_name();
-}
-bool NoOpDtlsTransport::writable() const {
- return ice_transport_->writable();
-}
-bool NoOpDtlsTransport::receiving() const {
- return ice_transport_->receiving();
-}
-int NoOpDtlsTransport::SendPacket(const char* data,
- size_t len,
- const rtc::PacketOptions& options,
- int flags) {
- return 0;
-}
-
-int NoOpDtlsTransport::SetOption(rtc::Socket::Option opt, int value) {
- return ice_transport_->SetOption(opt, value);
-}
-
-int NoOpDtlsTransport::GetError() {
- return ice_transport_->GetError();
-}
-
-void NoOpDtlsTransport::OnNetworkRouteChanged(
- absl::optional<rtc::NetworkRoute> network_route) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- SignalNetworkRouteChanged(network_route);
-}
-
-void NoOpDtlsTransport::OnReceivingState(
- rtc::PacketTransportInternal* transport) {
- RTC_DCHECK_RUN_ON(&thread_checker_);
- SignalReceivingState(this);
-}
-
-} // namespace cricket
diff --git a/p2p/base/no_op_dtls_transport.h b/p2p/base/no_op_dtls_transport.h
deleted file mode 100644
index f8829db..0000000
--- a/p2p/base/no_op_dtls_transport.h
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * Copyright 2019 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.
- */
-
-#ifndef P2P_BASE_NO_OP_DTLS_TRANSPORT_H_
-#define P2P_BASE_NO_OP_DTLS_TRANSPORT_H_
-
-#include <memory>
-#include <string>
-#include <vector>
-
-#include "api/crypto/crypto_options.h"
-#include "p2p/base/dtls_transport_internal.h"
-#include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/packet_transport_internal.h"
-#include "rtc_base/buffer.h"
-#include "rtc_base/buffer_queue.h"
-#include "rtc_base/constructor_magic.h"
-#include "rtc_base/ssl_stream_adapter.h"
-#include "rtc_base/stream.h"
-#include "rtc_base/strings/string_builder.h"
-#include "rtc_base/thread_checker.h"
-
-namespace cricket {
-
-constexpr int kNoOpDtlsTransportComponent = -1;
-
-// This implementation wraps a cricket::DtlsTransport, and takes
-// ownership of it.
-// The implementation does not perform any operations, except of being
-// "connected". The purpose of this implementation is to disable RTP transport
-// while MediaTransport is used.
-//
-// This implementation is only temporary. Long-term we will refactor and disable
-// RTP transport entirely when MediaTransport is used. Always connected (after
-// ICE), no-op, dtls transport. This is used when DTLS is disabled.
-//
-// MaybeCreateJsepTransport controller expects DTLS connection to send a
-// 'connected' signal _after_ it is created (if it is created in a connected
-// state, that would not be noticed by jsep transport controller). Therefore,
-// the no-op dtls transport will wait for ICE event "writable", and then
-// immediately report that it's connected (emulating 0-rtt connection).
-//
-// We could simply not set a dtls to active (not set a certificate on the DTLS),
-// and it would use an underyling connection instead.
-// However, when MediaTransport is used, we want to entirely disable
-// dtls/srtp/rtp, in order to avoid multiplexing issues, such as "Failed to
-// unprotect RTCP packet".
-class NoOpDtlsTransport : public DtlsTransportInternal {
- public:
- NoOpDtlsTransport(IceTransportInternal* ice_transport,
- const webrtc::CryptoOptions& crypto_options);
-
- ~NoOpDtlsTransport() override;
- const webrtc::CryptoOptions& crypto_options() const override;
- DtlsTransportState dtls_state() const override;
- int component() const override;
- bool IsDtlsActive() const override;
- bool GetDtlsRole(rtc::SSLRole* role) const override;
- bool SetDtlsRole(rtc::SSLRole role) override;
- bool GetSslVersionBytes(int* version) const override;
- bool GetSrtpCryptoSuite(int* cipher) override;
- bool GetSslCipherSuite(int* cipher) override;
- rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override;
- bool SetLocalCertificate(
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) override;
- std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain() const override;
- bool ExportKeyingMaterial(const std::string& label,
- const uint8_t* context,
- size_t context_len,
- bool use_context,
- uint8_t* result,
- size_t result_len) override;
- bool SetRemoteFingerprint(const std::string& digest_alg,
- const uint8_t* digest,
- size_t digest_len) override;
- bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override;
- IceTransportInternal* ice_transport() override;
-
- const std::string& transport_name() const override;
- bool writable() const override;
- bool receiving() const override;
-
- private:
- void OnReadyToSend(rtc::PacketTransportInternal* transport);
- void OnWritableState(rtc::PacketTransportInternal* transport);
- void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
- void OnReceivingState(rtc::PacketTransportInternal* transport);
-
- int SendPacket(const char* data,
- size_t len,
- const rtc::PacketOptions& options,
- int flags) override;
- int SetOption(rtc::Socket::Option opt, int value) override;
- int GetError() override;
-
- rtc::ThreadChecker thread_checker_;
-
- webrtc::CryptoOptions crypto_options_;
- IceTransportInternal* ice_transport_;
- bool is_writable_ = false;
-};
-
-} // namespace cricket
-
-#endif // P2P_BASE_NO_OP_DTLS_TRANSPORT_H_
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 4d0b61f..bc44bbb 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -606,7 +606,6 @@
":libjingle_peerconnection",
":pc_test_utils",
"../api:callfactory_api",
- "../api:fake_media_transport",
"../api:rtc_event_log_output_file",
"../api:rtc_stats_api",
"../api:rtp_parameters",
diff --git a/pc/channel.cc b/pc/channel.cc
index 8392775..fc5337a 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -150,10 +150,6 @@
TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
RTC_DCHECK_RUN_ON(worker_thread_);
- if (media_transport_config_.media_transport) {
- media_transport_config_.media_transport->RemoveNetworkChangeCallback(this);
- }
-
// Eats any outstanding messages or packets.
worker_thread_->Clear(&invoker_);
worker_thread_->Clear(this);
@@ -171,15 +167,8 @@
}
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
-
- // If media transport is used, it's responsible for providing network
- // route changed callbacks.
- if (!media_transport_config_.media_transport) {
- rtp_transport_->SignalNetworkRouteChanged.connect(
- this, &BaseChannel::OnNetworkRouteChanged);
- }
- // TODO(bugs.webrtc.org/9719): Media transport should also be used to provide
- // 'writable' state here.
+ rtp_transport_->SignalNetworkRouteChanged.connect(
+ this, &BaseChannel::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&BaseChannel::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this,
@@ -208,12 +197,6 @@
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
media_channel_->SetInterface(this, media_transport_config);
-
- RTC_LOG(LS_INFO) << "BaseChannel::Init_w, media_transport_config="
- << media_transport_config.DebugString();
- if (media_transport_config_.media_transport) {
- media_transport_config_.media_transport->AddNetworkChangeCallback(this);
- }
}
void BaseChannel::Deinit() {
@@ -802,9 +785,6 @@
ssrc_generator) {}
VoiceChannel::~VoiceChannel() {
- if (media_transport()) {
- media_transport()->SetFirstAudioPacketReceivedObserver(nullptr);
- }
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
@@ -817,24 +797,10 @@
[this] { UpdateMediaSendRecvState_w(); });
}
-void BaseChannel::OnNetworkRouteChanged(
- const rtc::NetworkRoute& network_route) {
- OnNetworkRouteChanged(absl::make_optional(network_route));
-}
-
void VoiceChannel::Init_w(
webrtc::RtpTransportInternal* rtp_transport,
const webrtc::MediaTransportConfig& media_transport_config) {
BaseChannel::Init_w(rtp_transport, media_transport_config);
- if (media_transport_config.media_transport) {
- media_transport_config.media_transport->SetFirstAudioPacketReceivedObserver(
- this);
- }
-}
-
-void VoiceChannel::OnFirstAudioPacketReceived(int64_t channel_id) {
- has_received_packet_ = true;
- signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
}
void VoiceChannel::UpdateMediaSendRecvState_w() {
diff --git a/pc/channel.h b/pc/channel.h
index 62fcaa2..c2b9e40 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -74,8 +74,7 @@
public rtc::MessageHandler,
public sigslot::has_slots<>,
public MediaChannel::NetworkInterface,
- public webrtc::RtpPacketSinkInterface,
- public webrtc::MediaTransportNetworkChangeCallback {
+ public webrtc::RtpPacketSinkInterface {
public:
// If |srtp_required| is true, the channel will not send or receive any
// RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
@@ -156,11 +155,6 @@
// Fired on the network thread.
sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive;
- // Returns media transport, can be null if media transport is not available.
- webrtc::MediaTransportInterface* media_transport() {
- return media_transport_config_.media_transport;
- }
-
// From RtpTransport - public for testing only
void OnTransportReadyToSend(bool ready);
@@ -287,9 +281,6 @@
void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
bool IsReadyToSendMedia_n() const;
- // MediaTransportNetworkChangeCallback override.
- void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override;
-
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::Thread* const signaling_thread_;
@@ -337,8 +328,7 @@
// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
-class VoiceChannel : public BaseChannel,
- public webrtc::AudioPacketReceivedObserver {
+class VoiceChannel : public BaseChannel {
public:
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
@@ -372,8 +362,6 @@
webrtc::SdpType type,
std::string* error_desc) override;
- void OnFirstAudioPacketReceived(int64_t channel_id) override;
-
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_;
diff --git a/pc/channel_manager_unittest.cc b/pc/channel_manager_unittest.cc
index ab3b88b..9078513 100644
--- a/pc/channel_manager_unittest.cc
+++ b/pc/channel_manager_unittest.cc
@@ -13,7 +13,6 @@
#include <memory>
#include "api/rtc_error.h"
-#include "api/test/fake_media_transport.h"
#include "api/transport/media/media_transport_config.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "media/base/fake_media_engine.h"
@@ -74,18 +73,6 @@
return dtls_srtp_transport;
}
- std::unique_ptr<webrtc::MediaTransportInterface> CreateMediaTransport(
- rtc::PacketTransportInternal* packet_transport) {
- webrtc::MediaTransportSettings settings;
- settings.is_caller = true;
- auto media_transport_result =
- fake_media_transport_factory_.CreateMediaTransport(
- packet_transport, network_.get(),
- /*is_caller=*/settings);
- RTC_CHECK(media_transport_result.ok());
- return media_transport_result.MoveValue();
- }
-
void TestCreateDestroyChannels(
webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportConfig media_transport_config) {
@@ -122,7 +109,6 @@
cricket::FakeDataEngine* fdme_;
std::unique_ptr<cricket::ChannelManager> cm_;
cricket::FakeCall fake_call_;
- webrtc::FakeMediaTransportFactory fake_media_transport_factory_;
rtc::UniqueRandomIdGenerator ssrc_generator_;
};
@@ -192,14 +178,6 @@
webrtc::MediaTransportConfig());
}
-TEST_F(ChannelManagerTest, CreateDestroyChannelsWithMediaTransport) {
- EXPECT_TRUE(cm_->Init());
- auto rtp_transport = CreateDtlsSrtpTransport();
- auto media_transport = CreateMediaTransport(rtp_dtls_transport_.get());
- TestCreateDestroyChannels(
- rtp_transport.get(), webrtc::MediaTransportConfig(media_transport.get()));
-}
-
TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
network_->Start();
worker_->Start();
diff --git a/pc/datagram_rtp_transport.h b/pc/datagram_rtp_transport.h
index 8aadf97..f9684c6 100644
--- a/pc/datagram_rtp_transport.h
+++ b/pc/datagram_rtp_transport.h
@@ -18,6 +18,7 @@
#include "api/crypto/crypto_options.h"
#include "api/transport/datagram_transport_interface.h"
+#include "api/transport/media/media_transport_interface.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "p2p/base/ice_transport_internal.h"
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index 79b933c9..37f3162 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -111,7 +111,6 @@
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
std::unique_ptr<SctpTransportInternal> sctp_transport,
- std::unique_ptr<webrtc::MediaTransportInterface> media_transport,
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport,
webrtc::DataChannelTransportInterface* data_channel_transport)
: network_thread_(rtc::Thread::Current()),
@@ -139,7 +138,6 @@
? new rtc::RefCountedObject<webrtc::SctpTransport>(
std::move(sctp_transport))
: nullptr),
- media_transport_(std::move(media_transport)),
datagram_transport_(std::move(datagram_transport)),
datagram_rtp_transport_(std::move(datagram_rtp_transport)),
data_channel_transport_(data_channel_transport) {
@@ -149,7 +147,6 @@
// present.
RTC_DCHECK_EQ((rtcp_ice_transport_ != nullptr),
(rtcp_dtls_transport_ != nullptr));
- RTC_DCHECK(!datagram_transport_ || !media_transport_);
// Verify the "only one out of these three can be set" invariant.
if (unencrypted_rtp_transport_) {
RTC_DCHECK(!sdes_transport);
@@ -173,10 +170,6 @@
datagram_rtp_transport_.get(), default_rtp_transport()});
}
- if (media_transport_) {
- media_transport_->SetMediaTransportStateCallback(this);
- }
-
if (data_channel_transport_ && sctp_data_channel_transport_) {
composite_data_channel_transport_ =
std::make_unique<webrtc::CompositeDataChannelTransport>(
@@ -186,11 +179,6 @@
}
JsepTransport::~JsepTransport() {
- // Disconnect media transport state callbacks.
- if (media_transport_) {
- media_transport_->SetMediaTransportStateCallback(nullptr);
- }
-
if (sctp_transport_) {
sctp_transport_->Clear();
}
@@ -784,18 +772,6 @@
return true;
}
-void JsepTransport::OnStateChanged(webrtc::MediaTransportState state) {
- // TODO(bugs.webrtc.org/9719) This method currently fires on the network
- // thread, but media transport does not make such guarantees. We need to make
- // sure this callback is guaranteed to be executed on the network thread.
- RTC_DCHECK_RUN_ON(network_thread_);
- {
- rtc::CritScope scope(&accessor_lock_);
- media_transport_state_ = state;
- }
- SignalMediaTransportStateChanged();
-}
-
void JsepTransport::NegotiateDatagramTransport(SdpType type) {
RTC_DCHECK(type == SdpType::kAnswer || type == SdpType::kPrAnswer);
rtc::CritScope lock(&accessor_lock_);
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 658e8e7..5f7d46f 100644
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@@ -21,7 +21,6 @@
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/transport/datagram_transport_interface.h"
-#include "api/transport/media/media_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/p2p_constants.h"
@@ -89,16 +88,11 @@
//
// On Threading: JsepTransport performs work solely on the network thread, and
// so its methods should only be called on the network thread.
-class JsepTransport : public sigslot::has_slots<>,
- public webrtc::MediaTransportStateCallback {
+class JsepTransport : public sigslot::has_slots<> {
public:
// |mid| is just used for log statements in order to identify the Transport.
// Note that |local_certificate| is allowed to be null since a remote
// description may be set before a local certificate is generated.
- //
- // |media_trasport| is optional (experimental). If available it will be used
- // to send / receive encoded audio and video frames instead of RTP.
- // Currently |media_transport| can co-exist with RTP / RTCP transports.
JsepTransport(
const std::string& mid,
const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
@@ -111,7 +105,6 @@
std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
std::unique_ptr<SctpTransportInternal> sctp_transport,
- std::unique_ptr<webrtc::MediaTransportInterface> media_transport,
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport,
webrtc::DataChannelTransportInterface* data_channel_transport);
@@ -246,34 +239,17 @@
return data_channel_transport_;
}
- // Returns media transport, if available.
- // Note that media transport is owned by jseptransport and the pointer
- // to media transport will becomes invalid after destruction of jseptransport.
- webrtc::MediaTransportInterface* media_transport() const {
- rtc::CritScope scope(&accessor_lock_);
- return media_transport_.get();
- }
-
// Returns datagram transport, if available.
webrtc::DatagramTransportInterface* datagram_transport() const {
rtc::CritScope scope(&accessor_lock_);
return datagram_transport_.get();
}
- // Returns the latest media transport state.
- webrtc::MediaTransportState media_transport_state() const {
- rtc::CritScope scope(&accessor_lock_);
- return media_transport_state_;
- }
-
// This is signaled when RTCP-mux becomes active and
// |rtcp_dtls_transport_| is destroyed. The JsepTransportController will
// handle the signal and update the aggregate transport states.
sigslot::signal<> SignalRtcpMuxActive;
- // This is signaled for changes in |media_transport_| state.
- sigslot::signal<> SignalMediaTransportStateChanged;
-
// Signals that a data channel transport was negotiated and may be used to
// send data. The first parameter is |this|. The second parameter is the
// transport that was negotiated, or null if negotiation rejected the data
@@ -338,9 +314,6 @@
bool GetTransportStats(DtlsTransportInternal* dtls_transport,
TransportStats* stats);
- // Invoked whenever the state of the media transport changes.
- void OnStateChanged(webrtc::MediaTransportState state) override;
-
// Deactivates, signals removal, and deletes |composite_rtp_transport_| if the
// current state of negotiation is sufficient to determine which rtp_transport
// and data channel transport to use.
@@ -418,10 +391,6 @@
absl::optional<std::vector<int>> recv_extension_ids_
RTC_GUARDED_BY(network_thread_);
- // Optional media transport (experimental).
- std::unique_ptr<webrtc::MediaTransportInterface> media_transport_
- RTC_GUARDED_BY(accessor_lock_);
-
// Optional datagram transport (experimental).
std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport_
RTC_GUARDED_BY(accessor_lock_);
@@ -429,9 +398,8 @@
std::unique_ptr<webrtc::RtpTransportInternal> datagram_rtp_transport_
RTC_GUARDED_BY(accessor_lock_);
- // Non-SCTP data channel transport. Set to one of |media_transport_| or
- // |datagram_transport_| if that transport should be used for data chanels.
- // Unset if neither should be used for data channels.
+ // Non-SCTP data channel transport. Set to |datagram_transport_| if that
+ // transport should be used for data chanels. Unset otherwise.
webrtc::DataChannelTransportInterface* data_channel_transport_
RTC_GUARDED_BY(accessor_lock_) = nullptr;
@@ -439,15 +407,6 @@
std::unique_ptr<webrtc::CompositeDataChannelTransport>
composite_data_channel_transport_ RTC_GUARDED_BY(accessor_lock_);
- // If |media_transport_| is provided, this variable represents the state of
- // media transport.
- //
- // NOTE: datagram transport state is handled by DatagramDtlsAdaptor, because
- // DatagramDtlsAdaptor owns DatagramTransport. This state only represents
- // media transport.
- webrtc::MediaTransportState media_transport_state_
- RTC_GUARDED_BY(accessor_lock_) = webrtc::MediaTransportState::kPending;
-
RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
};
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 590aa6b..f62cd87 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -18,7 +18,6 @@
#include "api/transport/datagram_transport_interface.h"
#include "api/transport/media/media_transport_interface.h"
#include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/no_op_dtls_transport.h"
#include "p2p/base/port.h"
#include "pc/datagram_rtp_transport.h"
#include "pc/srtp_filter.h"
@@ -148,22 +147,12 @@
return MediaTransportConfig();
}
- MediaTransportInterface* media_transport = nullptr;
- if (config_.use_media_transport_for_media) {
- media_transport = jsep_transport->media_transport();
- }
-
DatagramTransportInterface* datagram_transport = nullptr;
if (config_.use_datagram_transport) {
datagram_transport = jsep_transport->datagram_transport();
}
- // Media transport and datagram transports can not be used together.
- RTC_DCHECK(!media_transport || !datagram_transport);
-
- if (media_transport) {
- return MediaTransportConfig(media_transport);
- } else if (datagram_transport) {
+ if (datagram_transport) {
return MediaTransportConfig(
/*rtp_max_packet_size=*/datagram_transport->GetLargestDatagramSize());
} else {
@@ -180,15 +169,6 @@
return jsep_transport->data_channel_transport();
}
-MediaTransportState JsepTransportController::GetMediaTransportState(
- const std::string& mid) const {
- auto jsep_transport = GetJsepTransportForMid(mid);
- if (!jsep_transport) {
- return MediaTransportState::kPending;
- }
- return jsep_transport->media_transport_state();
-}
-
cricket::DtlsTransportInternal* JsepTransportController::GetDtlsTransport(
const std::string& mid) {
auto jsep_transport = GetJsepTransportForMid(mid);
@@ -446,26 +426,9 @@
}
void JsepTransportController::SetMediaTransportSettings(
- bool use_media_transport_for_media,
- bool use_media_transport_for_data_channels,
bool use_datagram_transport,
bool use_datagram_transport_for_data_channels,
bool use_datagram_transport_for_data_channels_receive_only) {
- RTC_DCHECK(use_media_transport_for_media ==
- config_.use_media_transport_for_media ||
- jsep_transports_by_name_.empty())
- << "You can only change media transport configuration before creating "
- "the first transport.";
-
- RTC_DCHECK(use_media_transport_for_data_channels ==
- config_.use_media_transport_for_data_channels ||
- jsep_transports_by_name_.empty())
- << "You can only change media transport configuration before creating "
- "the first transport.";
-
- config_.use_media_transport_for_media = use_media_transport_for_media;
- config_.use_media_transport_for_data_channels =
- use_media_transport_for_data_channels;
config_.use_datagram_transport = use_datagram_transport;
config_.use_datagram_transport_for_data_channels =
use_datagram_transport_for_data_channels;
@@ -514,14 +477,6 @@
if (datagram_transport) {
RTC_DCHECK(config_.use_datagram_transport ||
config_.use_datagram_transport_for_data_channels);
- } else if (config_.media_transport_factory &&
- config_.use_media_transport_for_media &&
- config_.use_media_transport_for_data_channels) {
- // If media transport is used for both media and data channels,
- // then we don't need to create DTLS.
- // Otherwise, DTLS is still created.
- dtls = std::make_unique<cricket::NoOpDtlsTransport>(ice,
- config_.crypto_options);
} else if (config_.dtls_transport_factory) {
dtls = config_.dtls_transport_factory->CreateDtlsTransport(
ice, config_.crypto_options);
@@ -916,13 +871,12 @@
mid_to_transport_[mid] = jsep_transport;
return config_.transport_observer->OnTransportChanged(
mid, jsep_transport->rtp_transport(), jsep_transport->RtpDtlsTransport(),
- jsep_transport->media_transport(),
jsep_transport->data_channel_transport());
}
void JsepTransportController::RemoveTransportForMid(const std::string& mid) {
- bool ret = config_.transport_observer->OnTransportChanged(
- mid, nullptr, nullptr, nullptr, nullptr);
+ bool ret = config_.transport_observer->OnTransportChanged(mid, nullptr,
+ nullptr, nullptr);
// Calling OnTransportChanged with nullptr should always succeed, since it is
// only expected to fail when adding media to a transport (not removing).
RTC_DCHECK(ret);
@@ -1102,76 +1056,6 @@
return (it == jsep_transports_by_name_.end()) ? nullptr : it->second.get();
}
-std::unique_ptr<webrtc::MediaTransportInterface>
-JsepTransportController::MaybeCreateMediaTransport(
- const cricket::ContentInfo& content_info,
- const cricket::SessionDescription& description,
- bool local) {
- if (config_.media_transport_factory == nullptr) {
- return nullptr;
- }
-
- if (!config_.use_media_transport_for_media &&
- !config_.use_media_transport_for_data_channels) {
- return nullptr;
- }
-
- // Caller (offerer) media transport.
- if (local) {
- if (offer_media_transport_) {
- RTC_LOG(LS_INFO) << "Offered media transport has now been activated.";
- return std::move(offer_media_transport_);
- } else {
- RTC_LOG(LS_INFO)
- << "Not returning media transport. Either SDES wasn't enabled, or "
- "media transport didn't return an offer earlier.";
- // Offer wasn't generated. Either because media transport didn't want it,
- // or because SDES wasn't enabled.
- return nullptr;
- }
- }
-
- // Remote offer. If no x-mt lines, do not create media transport.
- if (description.MediaTransportSettings().empty()) {
- return nullptr;
- }
-
- // When bundle is enabled, two JsepTransports are created, and then
- // the second transport is destroyed (right away).
- // For media transport, we don't want to create the second
- // media transport in the first place.
- RTC_LOG(LS_INFO) << "Returning new, client media transport.";
-
- RTC_DCHECK(!local)
- << "If media transport is used, you must call "
- "GenerateOrGetLastMediaTransportOffer before SetLocalDescription. You "
- "also "
- "must use kRtcpMuxPolicyRequire and kBundlePolicyMaxBundle with media "
- "transport.";
- MediaTransportSettings settings;
- settings.is_caller = local;
- if (config_.use_media_transport_for_media) {
- settings.event_log = config_.event_log;
- }
-
- // Assume there is only one media transport (or if more, use the first one).
- if (!local && !description.MediaTransportSettings().empty() &&
- config_.media_transport_factory->GetTransportName() ==
- description.MediaTransportSettings()[0].transport_name) {
- settings.remote_transport_parameters =
- description.MediaTransportSettings()[0].transport_setting;
- }
-
- auto media_transport_result =
- config_.media_transport_factory->CreateMediaTransport(network_thread_,
- settings);
-
- // TODO(sukhanov): Proper error handling.
- RTC_CHECK(media_transport_result.ok());
-
- return media_transport_result.MoveValue();
-}
-
// TODO(sukhanov): Refactor to avoid code duplication for Media and Datagram
// transports setup.
std::unique_ptr<webrtc::DatagramTransportInterface>
@@ -1259,13 +1143,6 @@
CreateIceTransport(content_info.name, /*rtcp=*/false);
RTC_DCHECK(ice);
- std::unique_ptr<MediaTransportInterface> media_transport =
- MaybeCreateMediaTransport(content_info, description, local);
- if (media_transport) {
- media_transport_created_once_ = true;
- media_transport->Connect(ice->internal());
- }
-
std::unique_ptr<DatagramTransportInterface> datagram_transport =
MaybeCreateDatagramTransport(content_info, description, local);
if (datagram_transport) {
@@ -1285,7 +1162,6 @@
if (config_.rtcp_mux_policy !=
PeerConnectionInterface::kRtcpMuxPolicyRequire &&
content_info.type == cricket::MediaProtocolType::kRtp) {
- RTC_DCHECK(media_transport == nullptr);
RTC_DCHECK(datagram_transport == nullptr);
rtcp_ice = CreateIceTransport(content_info.name, /*rtcp=*/true);
rtcp_dtls_transport =
@@ -1335,8 +1211,6 @@
DataChannelTransportInterface* data_channel_transport = nullptr;
if (config_.use_datagram_transport_for_data_channels) {
data_channel_transport = datagram_transport.get();
- } else if (config_.use_media_transport_for_data_channels) {
- data_channel_transport = media_transport.get();
}
std::unique_ptr<cricket::JsepTransport> jsep_transport =
@@ -1345,16 +1219,14 @@
std::move(unencrypted_rtp_transport), std::move(sdes_transport),
std::move(dtls_srtp_transport), std::move(datagram_rtp_transport),
std::move(rtp_dtls_transport), std::move(rtcp_dtls_transport),
- std::move(sctp_transport), std::move(media_transport),
- std::move(datagram_transport), data_channel_transport);
+ std::move(sctp_transport), std::move(datagram_transport),
+ data_channel_transport);
jsep_transport->rtp_transport()->SignalRtcpPacketReceived.connect(
this, &JsepTransportController::OnRtcpPacketReceived_n);
jsep_transport->SignalRtcpMuxActive.connect(
this, &JsepTransportController::UpdateAggregateStates_n);
- jsep_transport->SignalMediaTransportStateChanged.connect(
- this, &JsepTransportController::OnMediaTransportStateChanged_n);
jsep_transport->SignalDataChannelTransportNegotiated.connect(
this, &JsepTransportController::OnDataChannelTransportNegotiated_n);
SetTransportForMid(content_info.name, jsep_transport.get());
@@ -1387,8 +1259,8 @@
RTC_DCHECK(network_thread_->IsCurrent());
for (const auto& jsep_transport : jsep_transports_by_name_) {
- config_.transport_observer->OnTransportChanged(
- jsep_transport.first, nullptr, nullptr, nullptr, nullptr);
+ config_.transport_observer->OnTransportChanged(jsep_transport.first,
+ nullptr, nullptr, nullptr);
}
jsep_transports_by_name_.clear();
@@ -1559,10 +1431,6 @@
UpdateAggregateStates_n();
}
-void JsepTransportController::OnMediaTransportStateChanged_n() {
- UpdateAggregateStates_n();
-}
-
void JsepTransportController::OnDataChannelTransportNegotiated_n(
cricket::JsepTransport* transport,
DataChannelTransportInterface* data_channel_transport) {
@@ -1570,7 +1438,7 @@
if (it.second == transport) {
config_.transport_observer->OnTransportChanged(
it.first, transport->rtp_transport(), transport->RtpDtlsTransport(),
- transport->media_transport(), data_channel_transport);
+ data_channel_transport);
}
}
}
@@ -1587,10 +1455,6 @@
PeerConnectionInterface::PeerConnectionState::kNew;
cricket::IceGatheringState new_gathering_state = cricket::kIceGatheringNew;
bool any_failed = false;
-
- // TODO(http://bugs.webrtc.org/9719) If(when) media_transport disables
- // dtls_transports entirely, the below line will have to be changed to account
- // for the fact that dtls transports might be absent.
bool all_connected = !dtls_transports.empty();
bool all_completed = !dtls_transports.empty();
bool any_gathering = false;
@@ -1620,35 +1484,6 @@
ice_state_counts[dtls->ice_transport()->GetIceTransportState()]++;
}
- // Don't indicate that the call failed or isn't connected due to media
- // transport state unless the media transport is used for media. If it's only
- // used for data channels, it will signal those separately.
- if (config_.use_media_transport_for_media || config_.use_datagram_transport) {
- for (auto it = jsep_transports_by_name_.begin();
- it != jsep_transports_by_name_.end(); ++it) {
- auto jsep_transport = it->second.get();
- if (!jsep_transport->media_transport()) {
- continue;
- }
-
- // There is no 'kIceConnectionDisconnected', so we only need to handle
- // connected and completed.
- // We treat kClosed as failed, because if it happens before shutting down
- // media transports it means that there was a failure.
- // MediaTransportInterface allows to flip back and forth between kWritable
- // and kPending, but there does not exist an implementation that does
- // that, and the contract of jsep transport controller doesn't quite
- // expect that. When this happens, we would go from connected to
- // connecting state, but this may change in future.
- any_failed |= jsep_transport->media_transport_state() ==
- webrtc::MediaTransportState::kClosed;
- all_completed &= jsep_transport->media_transport_state() ==
- webrtc::MediaTransportState::kWritable;
- all_connected &= jsep_transport->media_transport_state() ==
- webrtc::MediaTransportState::kWritable;
- }
- }
-
if (any_failed) {
new_connection_state = cricket::kIceConnectionFailed;
} else if (all_completed) {
@@ -1809,67 +1644,6 @@
SignalDtlsHandshakeError(error);
}
-absl::optional<cricket::SessionDescription::MediaTransportSetting>
-JsepTransportController::GenerateOrGetLastMediaTransportOffer() {
- if (media_transport_created_once_) {
- RTC_LOG(LS_INFO) << "Not regenerating media transport for the new offer in "
- "existing session.";
- return media_transport_offer_settings_;
- }
-
- RTC_LOG(LS_INFO) << "Generating media transport offer!";
-
- absl::optional<std::string> transport_parameters;
-
- // Check that media transport is supposed to be used.
- // Note that ICE is not available when media transport is created. It will
- // only be available in 'Connect'. This may be a potential server config, if
- // we decide to use this peer connection as a caller, not as a callee.
- // TODO(sukhanov): Avoid code duplication with CreateMedia/MediaTransport.
- if (config_.use_media_transport_for_media ||
- config_.use_media_transport_for_data_channels) {
- RTC_DCHECK(config_.media_transport_factory != nullptr);
- RTC_DCHECK(!config_.use_datagram_transport);
- webrtc::MediaTransportSettings settings;
- settings.is_caller = true;
- settings.pre_shared_key = rtc::CreateRandomString(32);
- if (config_.use_media_transport_for_media) {
- settings.event_log = config_.event_log;
- }
- auto media_transport_or_error =
- config_.media_transport_factory->CreateMediaTransport(network_thread_,
- settings);
-
- if (media_transport_or_error.ok()) {
- offer_media_transport_ = std::move(media_transport_or_error.value());
- transport_parameters =
- offer_media_transport_->GetTransportParametersOffer();
- } else {
- RTC_LOG(LS_INFO) << "Unable to create media transport, error="
- << media_transport_or_error.error().message();
- }
- }
-
- if (!offer_media_transport_) {
- RTC_LOG(LS_INFO) << "Media and data transports do not exist";
- return absl::nullopt;
- }
-
- if (!transport_parameters) {
- RTC_LOG(LS_INFO) << "Media transport didn't generate the offer";
- // Media transport didn't generate the offer, and is not supposed to be
- // used. Destroy the temporary media transport.
- offer_media_transport_ = nullptr;
- return absl::nullopt;
- }
-
- cricket::SessionDescription::MediaTransportSetting setting;
- setting.transport_name = config_.media_transport_factory->GetTransportName();
- setting.transport_setting = *transport_parameters;
- media_transport_offer_settings_ = setting;
- return setting;
-}
-
absl::optional<cricket::OpaqueTransportParameters>
JsepTransportController::GetTransportParameters(const std::string& mid) {
if (!(config_.use_datagram_transport ||
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index b7121e7..9c3f691 100644
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@@ -23,7 +23,6 @@
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/media/media_transport_config.h"
-#include "api/transport/media/media_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/dtls_transport.h"
#include "p2p/base/dtls_transport_factory.h"
@@ -72,7 +71,6 @@
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
- MediaTransportInterface* media_transport,
DataChannelTransportInterface* data_channel_transport) = 0;
};
@@ -106,12 +104,6 @@
// Factory for SCTP transports.
cricket::SctpTransportInternalFactory* sctp_factory = nullptr;
- // Whether media transport is used for media.
- bool use_media_transport_for_media = false;
-
- // Whether media transport is used for data channels.
- bool use_media_transport_for_data_channels = false;
-
// Whether an RtpMediaTransport should be created as default, when no
// MediaTransportFactory is provided.
bool use_rtp_media_transport = false;
@@ -128,13 +120,13 @@
bool use_datagram_transport_for_data_channels_receive_only = false;
// Optional media transport factory (experimental). If provided it will be
- // used to create media_transport (as long as either
- // |use_media_transport_for_media| or
- // |use_media_transport_for_data_channels| is set to true). However, whether
- // it will be used to send / receive audio and video frames instead of RTP
- // is determined by |use_media_transport_for_media|. Note that currently
- // media_transport co-exists with RTP / RTCP transports and may use the same
- // underlying ICE transport.
+ // used to create datagram_transport (as long as either
+ // |use_datagram_transport| or
+ // |use_datagram_transport_for_data_channels| is set to true). However,
+ // whether it will be used to send / receive audio and video frames instead
+ // of RTP is determined by |use_datagram_transport|. Note that currently
+ // datagram_transport co-exists with RTP / RTCP transports and may use the
+ // same underlying ICE transport.
MediaTransportFactory* media_transport_factory = nullptr;
};
@@ -174,13 +166,6 @@
DataChannelTransportInterface* GetDataChannelTransport(
const std::string& mid) const;
- // TODO(sukhanov): Deprecate, return only config.
- MediaTransportInterface* GetMediaTransport(const std::string& mid) const {
- return GetMediaTransportConfig(mid).media_transport;
- }
-
- MediaTransportState GetMediaTransportState(const std::string& mid) const;
-
/*********************
* ICE-related methods
********************/
@@ -235,8 +220,6 @@
// you did not call 'GetMediaTransport' or 'MaybeCreateJsepTransport'. Once
// Jsep transport is created, you can't change this setting.
void SetMediaTransportSettings(
- bool use_media_transport_for_media,
- bool use_media_transport_for_data_channels,
bool use_datagram_transport,
bool use_datagram_transport_for_data_channels,
bool use_datagram_transport_for_data_channels_receive_only);
@@ -245,13 +228,6 @@
// and deletes unused transports, but doesn't consider anything more complex.
void RollbackTransportForMids(const std::vector<std::string>& mids);
- // If media transport is present enabled and supported,
- // when this method is called, it creates a media transport and generates its
- // offer. The new offer is then returned, and the created media transport will
- // subsequently be used.
- absl::optional<cricket::SessionDescription::MediaTransportSetting>
- GenerateOrGetLastMediaTransportOffer();
-
// Gets the transport parameters for the transport identified by |mid|.
// If |mid| is bundled, returns the parameters for the bundled transport.
// If the transport for |mid| has not been created yet, it may be allocated in
@@ -371,16 +347,6 @@
const cricket::ContentInfo& content_info,
const cricket::SessionDescription& description);
- // Creates media transport if config wants to use it, and a=x-mt line is
- // present for the current media transport. Returned MediaTransportInterface
- // is not connected, and must be connected to ICE. You must call
- // |GenerateOrGetLastMediaTransportOffer| on the caller before calling
- // MaybeCreateMediaTransport.
- std::unique_ptr<webrtc::MediaTransportInterface> MaybeCreateMediaTransport(
- const cricket::ContentInfo& content_info,
- const cricket::SessionDescription& description,
- bool local);
-
// Creates datagram transport if config wants to use it, and a=x-mt line is
// present for the current media transport. Returned
// DatagramTransportInterface is not connected, and must be connected to ICE.
@@ -441,7 +407,6 @@
const cricket::Candidates& candidates);
void OnTransportRoleConflict_n(cricket::IceTransportInternal* transport);
void OnTransportStateChanged_n(cricket::IceTransportInternal* transport);
- void OnMediaTransportStateChanged_n();
void OnTransportCandidatePairChanged_n(
const cricket::CandidatePairChangeEvent& event);
void OnDataChannelTransportNegotiated_n(
@@ -480,21 +445,6 @@
Config config_;
- // Early on in the call we don't know if media transport is going to be used,
- // but we need to get the server-supported parameters to add to an SDP.
- // This server media transport will be promoted to the used media transport
- // after the local description is set, and the ownership will be transferred
- // to the actual JsepTransport.
- // This "offer" media transport is not created if it's done on the party that
- // provides answer. This offer media transport is only created once at the
- // beginning of the connection, and never again.
- std::unique_ptr<MediaTransportInterface> offer_media_transport_ = nullptr;
-
- // Contains the offer of the |offer_media_transport_|, in case if it needs to
- // be repeated.
- absl::optional<cricket::SessionDescription::MediaTransportSetting>
- media_transport_offer_settings_;
-
// Early on in the call we don't know if datagram transport is going to be
// used, but we need to get the server-supported parameters to add to an SDP.
// This server datagram transport will be promoted to the used datagram
@@ -506,24 +456,6 @@
std::unique_ptr<DatagramTransportInterface> offer_datagram_transport_ =
nullptr;
- // Contains the offer of the |offer_datagram_transport_|, in case if it needs
- // to be repeated.
- absl::optional<cricket::SessionDescription::MediaTransportSetting>
- datagram_transport_offer_settings_;
-
- // When the new offer is regenerated (due to upgrade), we don't want to
- // re-create media transport. New streams might be created; but media
- // transport stays the same. This flag prevents re-creation of the transport
- // on the offerer.
- // The first media transport is created in jsep transport controller as the
- // |offer_media_transport_|, and then the ownership is moved to the
- // appropriate JsepTransport, at which point |offer_media_transport_| is
- // zeroed out. On the callee (answerer), the first media transport is not even
- // assigned to |offer_media_transport_|. Both offerer and answerer can
- // recreate the Offer (e.g. after adding streams in Plan B), and so we want to
- // prevent recreation of the media transport when that happens.
- bool media_transport_created_once_ = false;
-
const cricket::SessionDescription* local_desc_ = nullptr;
const cricket::SessionDescription* remote_desc_ = nullptr;
absl::optional<bool> initial_offerer_;
diff --git a/pc/jsep_transport_controller_unittest.cc b/pc/jsep_transport_controller_unittest.cc
index b96a999..196be13 100644
--- a/pc/jsep_transport_controller_unittest.cc
+++ b/pc/jsep_transport_controller_unittest.cc
@@ -19,7 +19,6 @@
#include "p2p/base/dtls_transport_factory.h"
#include "p2p/base/fake_dtls_transport.h"
#include "p2p/base/fake_ice_transport.h"
-#include "p2p/base/no_op_dtls_transport.h"
#include "p2p/base/transport_info.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
@@ -331,7 +330,6 @@
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
- MediaTransportInterface* media_transport,
DataChannelTransportInterface* data_channel_transport) override {
changed_rtp_transport_by_mid_[mid] = rtp_transport;
if (dtls_transport) {
@@ -339,7 +337,6 @@
} else {
changed_dtls_transport_by_mid_[mid] = nullptr;
}
- changed_media_transport_by_mid_[mid] = media_transport;
return true;
}
@@ -373,8 +370,6 @@
std::map<std::string, RtpTransportInternal*> changed_rtp_transport_by_mid_;
std::map<std::string, cricket::DtlsTransportInternal*>
changed_dtls_transport_by_mid_;
- std::map<std::string, MediaTransportInterface*>
- changed_media_transport_by_mid_;
// Transport controller needs to be destroyed first, because it may issue
// callbacks that modify the changed_*_by_mid in the destructor.
@@ -443,46 +438,6 @@
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
- EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
-}
-
-TEST_F(JsepTransportControllerTest,
- DtlsIsStillCreatedIfMediaTransportIsOnlyUsedForDataChannels) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
-
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
-
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetDataChannelTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- // After SetLocalDescription, media transport should be created as caller.
- EXPECT_TRUE(media_transport->is_caller());
- EXPECT_TRUE(media_transport->pre_shared_key().has_value());
-
- // Return nullptr for non-existing mids.
- EXPECT_EQ(nullptr,
- transport_controller_->GetDataChannelTransport(kVideoMid2));
-
- EXPECT_EQ(cricket::ICE_CANDIDATE_COMPONENT_RTP,
- transport_controller_->GetDtlsTransport(kAudioMid1)->component())
- << "Media transport for media was not enabled, and so DTLS transport "
- "should be created.";
}
TEST_F(JsepTransportControllerTest,
@@ -575,339 +530,6 @@
.ok());
}
-TEST_F(JsepTransportControllerTest, GetMediaTransportInCaller) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
-
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
-
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- // After SetLocalDescription, media transport should be created as caller.
- EXPECT_TRUE(media_transport->is_caller());
- // We set the pre-shared key on the caller.
- EXPECT_TRUE(media_transport->pre_shared_key().has_value());
- EXPECT_TRUE(media_transport->is_connected());
-
- // Return nullptr for non-existing mids.
- EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
-
- EXPECT_EQ(cricket::kNoOpDtlsTransportComponent,
- transport_controller_->GetDtlsTransport(kAudioMid1)->component())
- << "Because media transport is used, expected no-op DTLS transport.";
-}
-
-TEST_F(JsepTransportControllerTest,
- GetMediaTransportOfferInTheConfigOnSubsequentCalls) {
- FakeMediaTransportFactory fake_media_transport_factory;
- WrapperMediaTransportFactory wrapping_factory(&fake_media_transport_factory);
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &wrapping_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
-
- absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
- ASSERT_NE(absl::nullopt, settings);
-
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- absl::optional<cricket::SessionDescription::MediaTransportSetting>
- new_settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
- ASSERT_NE(absl::nullopt, new_settings);
- EXPECT_EQ(settings->transport_name, new_settings->transport_name);
- EXPECT_EQ(settings->transport_setting, new_settings->transport_setting);
- EXPECT_EQ(1, wrapping_factory.created_transport_count());
-}
-
-TEST_F(JsepTransportControllerTest, GetMediaTransportInCallee) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- description->AddMediaTransportSetting("fake", "fake-remote-settings");
- EXPECT_TRUE(transport_controller_
- ->SetRemoteDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- // After SetRemoteDescription, media transport should be created as callee.
- EXPECT_FALSE(media_transport->is_caller());
- // We do not set pre-shared key on the callee, it comes in media transport
- // settings.
- EXPECT_EQ(absl::nullopt, media_transport->settings().pre_shared_key);
- EXPECT_TRUE(media_transport->is_connected());
-
- // Return nullptr for non-existing mids.
- EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
-
- EXPECT_EQ(cricket::kNoOpDtlsTransportComponent,
- transport_controller_->GetDtlsTransport(kAudioMid1)->component())
- << "Because media transport is used, expected no-op DTLS transport.";
-}
-
-TEST_F(JsepTransportControllerTest, GetMediaTransportInCalleePassesSdp) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- description->AddMediaTransportSetting("fake", "this-is-a-test-setting");
- EXPECT_TRUE(transport_controller_
- ->SetRemoteDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- EXPECT_EQ("this-is-a-test-setting",
- media_transport->settings().remote_transport_parameters);
-}
-
-// Caller generates the offer if media transport returns empty offer (no
-// parameters).
-TEST_F(JsepTransportControllerTest, MediaTransportGeneratesSessionDescription) {
- FakeMediaTransportFactory fake_media_transport_factory(
- /*transport_offer=*/"");
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
-
- ASSERT_TRUE(settings.has_value());
- EXPECT_EQ("fake", settings->transport_name);
- // Fake media transport returns empty settings (but not nullopt settings!)
- EXPECT_EQ("", settings->transport_setting);
-}
-
-// Caller generates the offer if media transport returns offer with parameters.
-TEST_F(JsepTransportControllerTest,
- MediaTransportGeneratesSessionDescriptionWithOfferParams) {
- FakeMediaTransportFactory fake_media_transport_factory(
- /*transport_offer=*/"offer-params");
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
-
- ASSERT_TRUE(settings.has_value());
- EXPECT_EQ("fake", settings->transport_name);
- EXPECT_EQ("offer-params", settings->transport_setting);
-}
-
-// Caller skips the offer if media transport requests it.
-TEST_F(JsepTransportControllerTest,
- MediaTransportGeneratesSkipsSessionDescription) {
- FakeMediaTransportFactory fake_media_transport_factory(
- /*transport_offer=*/absl::nullopt);
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- absl::optional<cricket::SessionDescription::MediaTransportSetting> settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
-
- // Fake media transport returns nullopt settings
- ASSERT_EQ(absl::nullopt, settings);
-}
-
-// Caller ignores its own outgoing parameters.
-TEST_F(JsepTransportControllerTest,
- GetMediaTransportInCallerIgnoresXmtSection) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- // Remote parameters are nullopt, because we are the offerer (we don't)
- // have the remote transport parameters, only ours.
- EXPECT_EQ(absl::nullopt,
- media_transport->settings().remote_transport_parameters);
-}
-
-TEST_F(JsepTransportControllerTest,
- GetMediaTransportInCalleeIgnoresDifferentTransport) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- description->AddMediaTransportSetting("not-a-fake-transport",
- "this-is-a-test-setting");
- EXPECT_TRUE(transport_controller_
- ->SetRemoteDescription(SdpType::kOffer, description.get())
- .ok());
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- EXPECT_EQ(absl::nullopt,
- media_transport->settings().remote_transport_parameters);
-}
-
-TEST_F(JsepTransportControllerTest, GetMediaTransportIsNotSetIfNoSdes) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- EXPECT_TRUE(transport_controller_
- ->SetRemoteDescription(SdpType::kOffer, description.get())
- .ok());
-
- EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
-
- // Even if we set local description with crypto now (after the remote offer
- // was set), media transport won't be provided.
- auto description2 = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description2.get());
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kAnswer, description2.get())
- .ok());
-
- EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
- EXPECT_EQ(cricket::ICE_CANDIDATE_COMPONENT_RTP,
- transport_controller_->GetDtlsTransport(kAudioMid1)->component())
- << "Because media transport is NOT used (fallback to RTP), expected "
- "actual DTLS transport for RTP";
-}
-
-TEST_F(JsepTransportControllerTest,
- AfterSettingAnswerTheSameMediaTransportIsReturnedCallee) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
-
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.media_transport_factory = &fake_media_transport_factory;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- description->AddMediaTransportSetting("fake", "fake-settings");
- EXPECT_TRUE(transport_controller_
- ->SetRemoteDescription(SdpType::kOffer, description.get())
- .ok());
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
- EXPECT_NE(nullptr, media_transport);
- EXPECT_FALSE(media_transport->pre_shared_key().has_value())
- << "On the callee, preshared key is passed through the media-transport "
- "settings (x-mt)";
-
- // Even if we set local description with crypto now (after the remote offer
- // was set), media transport won't be provided.
- auto description2 = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description2.get());
-
- RTCError result = transport_controller_->SetLocalDescription(
- SdpType::kAnswer, description2.get());
- EXPECT_TRUE(result.ok()) << result.message();
-
- // Media transport did not change.
- EXPECT_EQ(media_transport,
- transport_controller_->GetMediaTransport(kAudioMid1));
-}
-
TEST_F(JsepTransportControllerTest, SetIceConfig) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
@@ -1190,164 +812,6 @@
EXPECT_EQ(3, combined_connection_state_signal_count_);
}
-TEST_F(JsepTransportControllerTest,
- SignalConnectionStateConnectedWithMediaTransportAndNoDtlsCaller) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
- config.media_transport_factory = &fake_media_transport_factory;
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.use_media_transport_for_data_channels = true;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
-
- // Media Transport is only used with bundle.
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
- auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
- EXPECT_EQ(fake_audio_ice, fake_video_ice);
- fake_audio_ice->SetConnectionCount(2);
- fake_audio_ice->SetConnectionCount(1);
- fake_video_ice->SetConnectionCount(2);
- fake_video_ice->SetConnectionCount(1);
- fake_audio_ice->SetWritable(true);
- fake_video_ice->SetWritable(true);
-
- // Still not connected, because we are waiting for media transport.
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
- kTimeout);
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- media_transport->SetState(webrtc::MediaTransportState::kWritable);
- // Only one media transport.
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
-}
-
-TEST_F(JsepTransportControllerTest,
- SignalConnectionStateConnectedWithMediaTransportCaller) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
- config.media_transport_factory = &fake_media_transport_factory;
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
-
- // Media Transport is only used with bundle.
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
- transport_controller_->GetDtlsTransport(kAudioMid1));
- auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
- transport_controller_->GetDtlsTransport(kVideoMid1));
-
- auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
- auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
- fake_audio_ice->SetConnectionCount(2);
- fake_audio_ice->SetConnectionCount(1);
- fake_video_ice->SetConnectionCount(2);
- fake_video_ice->SetConnectionCount(1);
- fake_audio_ice->SetWritable(true);
- fake_video_ice->SetWritable(true);
- fake_audio_dtls->SetWritable(true);
- fake_video_dtls->SetWritable(true);
-
- // Still not connected, because we are waiting for media transport.
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
- kTimeout);
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
-
- ASSERT_NE(nullptr, media_transport);
-
- media_transport->SetState(webrtc::MediaTransportState::kWritable);
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
- kTimeout);
-
- // Still waiting for the second media transport.
- media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kVideoMid1));
- media_transport->SetState(webrtc::MediaTransportState::kWritable);
-
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
-}
-
-TEST_F(JsepTransportControllerTest,
- SignalConnectionStateFailedWhenMediaTransportClosedCaller) {
- FakeMediaTransportFactory fake_media_transport_factory;
- JsepTransportController::Config config;
- config.media_transport_factory = &fake_media_transport_factory;
- config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- config.use_media_transport_for_media = true;
- CreateJsepTransportController(config);
- auto description = CreateSessionDescriptionWithBundleGroup();
- AddCryptoSettings(description.get());
- EXPECT_NE(absl::nullopt,
- transport_controller_->GenerateOrGetLastMediaTransportOffer());
- EXPECT_TRUE(transport_controller_
- ->SetLocalDescription(SdpType::kOffer, description.get())
- .ok());
-
- auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
- transport_controller_->GetDtlsTransport(kAudioMid1));
- auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
- transport_controller_->GetDtlsTransport(kVideoMid1));
-
- auto fake_audio_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
- auto fake_video_ice = static_cast<cricket::FakeIceTransport*>(
- transport_controller_->GetDtlsTransport(kVideoMid1)->ice_transport());
- fake_audio_ice->SetWritable(true);
- fake_video_ice->SetWritable(true);
- // Decreasing connection count from 2 to 1 triggers connection state event.
- fake_audio_ice->SetConnectionCount(2);
- fake_audio_ice->SetConnectionCount(1);
- fake_video_ice->SetConnectionCount(2);
- fake_video_ice->SetConnectionCount(1);
-
- fake_audio_dtls->SetWritable(true);
- fake_video_dtls->SetWritable(true);
-
- FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kAudioMid1));
- ASSERT_NE(nullptr, media_transport);
- media_transport->SetState(webrtc::MediaTransportState::kWritable);
-
- media_transport = static_cast<FakeMediaTransport*>(
- transport_controller_->GetMediaTransport(kVideoMid1));
- ASSERT_NE(nullptr, media_transport);
-
- media_transport->SetState(webrtc::MediaTransportState::kWritable);
-
- EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
-
- media_transport->SetState(webrtc::MediaTransportState::kClosed);
- EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
-}
-
TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
CreateJsepTransportController(JsepTransportController::Config());
auto description = CreateSessionDescriptionWithoutBundle();
diff --git a/pc/jsep_transport_unittest.cc b/pc/jsep_transport_unittest.cc
index 87d6e87..c4193e5 100644
--- a/pc/jsep_transport_unittest.cc
+++ b/pc/jsep_transport_unittest.cc
@@ -114,11 +114,6 @@
RTC_NOTREACHED();
}
- // TODO(sukhanov): Currently there is no media_transport specific
- // logic in jseptransport, so jseptransport unittests are created with
- // media_transport = nullptr. In the future we will probably add
- // more logic that require unit tests. Note that creation of media_transport
- // is covered in jseptransportcontroller_unittest.
auto jsep_transport = std::make_unique<JsepTransport>(
kTransportName, /*local_certificate=*/nullptr, std::move(ice),
std::move(rtcp_ice), std::move(unencrypted_rtp_transport),
@@ -126,7 +121,6 @@
/*datagram_rtp_transport=*/nullptr, std::move(rtp_dtls_transport),
std::move(rtcp_dtls_transport),
/*sctp_transport=*/nullptr,
- /*media_transport=*/nullptr,
/*datagram_transport=*/nullptr,
/*data_channel_transport=*/nullptr);
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 873f27d..59f140f 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -1523,12 +1523,6 @@
offer->set_extmap_allow_mixed(session_options.offer_extmap_allow_mixed);
- if (session_options.media_transport_settings.has_value()) {
- offer->AddMediaTransportSetting(
- session_options.media_transport_settings->transport_name,
- session_options.media_transport_settings->transport_setting);
- }
-
return offer;
}
diff --git a/pc/media_session.h b/pc/media_session.h
index f91729a..235945c 100644
--- a/pc/media_session.h
+++ b/pc/media_session.h
@@ -115,11 +115,6 @@
std::vector<MediaDescriptionOptions> media_description_options;
std::vector<IceParameters> pooled_ice_credentials;
- // An optional media transport settings.
- // In the future we may consider using a vector here, to indicate multiple
- // supported transports.
- absl::optional<cricket::SessionDescription::MediaTransportSetting>
- media_transport_settings;
// Use the draft-ietf-mmusic-sctp-sdp-03 obsolete syntax for SCTP
// datachannels.
// Default is true for backwards compatibility with clients that use
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index f5f51c4..c24bd2e 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1135,7 +1135,6 @@
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
- RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTCError config_error = ValidateConfiguration(configuration);
@@ -1260,37 +1259,15 @@
use_datagram_transport_for_data_channels_receive_only_ =
configuration.use_datagram_transport_for_data_channels_receive_only
.value_or(datagram_transport_data_channel_config_.receive_only);
- if (use_datagram_transport_ || use_datagram_transport_for_data_channels_ ||
- configuration.use_media_transport ||
- configuration.use_media_transport_for_data_channels) {
+ if (use_datagram_transport_ || use_datagram_transport_for_data_channels_) {
if (!factory_->media_transport_factory()) {
RTC_DCHECK(false)
- << "PeerConnecton is initialized with use_media_transport = true or "
- << "use_media_transport_for_data_channels = true "
+ << "PeerConnecton is initialized with use_datagram_transport = true "
+ "or use_datagram_transport_for_data_channels = true "
<< "but media transport factory is not set in PeerConnectionFactory";
return false;
}
- if (configuration.use_media_transport ||
- configuration.use_media_transport_for_data_channels) {
- // TODO(bugs.webrtc.org/9719): This check will eventually go away, when
- // RTP media transport is introduced. But until then, we require SDES to
- // be enabled.
- if (configuration.enable_dtls_srtp.has_value() &&
- configuration.enable_dtls_srtp.value()) {
- RTC_LOG(LS_WARNING)
- << "When media transport is used, SDES must be enabled. Set "
- "configuration.enable_dtls_srtp to false. use_media_transport="
- << configuration.use_media_transport
- << ", use_media_transport_for_data_channels="
- << configuration.use_media_transport_for_data_channels;
- return false;
- }
- }
-
- config.use_media_transport_for_media = configuration.use_media_transport;
- config.use_media_transport_for_data_channels =
- configuration.use_media_transport_for_data_channels;
config.use_datagram_transport = use_datagram_transport_;
config.use_datagram_transport_for_data_channels =
use_datagram_transport_for_data_channels_;
@@ -1336,14 +1313,6 @@
data_channel_type_ = cricket::DCT_DATA_CHANNEL_TRANSPORT_SCTP;
config.sctp_factory = sctp_factory_.get();
}
- } else if (configuration.use_media_transport_for_data_channels) {
- if (configuration.enable_rtp_data_channel) {
- RTC_LOG(LS_ERROR) << "enable_rtp_data_channel and "
- "use_media_transport_for_data_channels are "
- "incompatible and cannot both be set to true";
- return false;
- }
- data_channel_type_ = cricket::DCT_MEDIA_TRANSPORT;
} else if (configuration.enable_rtp_data_channel) {
// Enable creation of RTP data channels if the kEnableRtpDataChannels is
// set. It takes precendence over the disable_sctp_data_channels
@@ -1385,7 +1354,6 @@
stats_collector_ = RTCStatsCollector::Create(this);
configuration_ = configuration;
- use_media_transport_ = configuration.use_media_transport;
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
@@ -3928,7 +3896,6 @@
RTCError PeerConnection::SetConfiguration(
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(signaling_thread());
- RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
@@ -3946,36 +3913,6 @@
}
if (local_description() &&
- configuration.use_media_transport != configuration_.use_media_transport) {
- LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
- "Can't change media_transport after calling "
- "SetLocalDescription.");
- }
-
- if (remote_description() &&
- configuration.use_media_transport != configuration_.use_media_transport) {
- LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
- "Can't change media_transport after calling "
- "SetRemoteDescription.");
- }
-
- if (local_description() &&
- configuration.use_media_transport_for_data_channels !=
- configuration_.use_media_transport_for_data_channels) {
- LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
- "Can't change media_transport_for_data_channels "
- "after calling SetLocalDescription.");
- }
-
- if (remote_description() &&
- configuration.use_media_transport_for_data_channels !=
- configuration_.use_media_transport_for_data_channels) {
- LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
- "Can't change media_transport_for_data_channels "
- "after calling SetRemoteDescription.");
- }
-
- if (local_description() &&
configuration.crypto_options != configuration_.crypto_options) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change crypto_options after calling "
@@ -4034,9 +3971,7 @@
"after calling SetRemoteDescription.");
}
- if (configuration.use_media_transport_for_data_channels ||
- configuration.use_media_transport ||
- (configuration.use_datagram_transport &&
+ if ((configuration.use_datagram_transport &&
*configuration.use_datagram_transport) ||
(configuration.use_datagram_transport_for_data_channels &&
*configuration.use_datagram_transport_for_data_channels)) {
@@ -4072,9 +4007,6 @@
modified_config.network_preference = configuration.network_preference;
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
- modified_config.use_media_transport = configuration.use_media_transport;
- modified_config.use_media_transport_for_data_channels =
- configuration.use_media_transport_for_data_channels;
modified_config.use_datagram_transport = configuration.use_datagram_transport;
modified_config.use_datagram_transport_for_data_channels =
configuration.use_datagram_transport_for_data_channels;
@@ -4158,8 +4090,6 @@
modified_config.use_datagram_transport_for_data_channels_receive_only
.value_or(datagram_transport_data_channel_config_.receive_only);
transport_controller_->SetMediaTransportSettings(
- modified_config.use_media_transport,
- modified_config.use_media_transport_for_data_channels,
use_datagram_transport_, use_datagram_transport_for_data_channels_,
use_datagram_transport_for_data_channels_receive_only_);
@@ -4178,7 +4108,6 @@
}
configuration_ = modified_config;
- use_media_transport_ = configuration.use_media_transport;
return RTCError::OK();
}
@@ -4967,12 +4896,6 @@
session_options->offer_extmap_allow_mixed =
configuration_.offer_extmap_allow_mixed;
- if (configuration_.use_media_transport ||
- configuration_.use_media_transport_for_data_channels) {
- session_options->media_transport_settings =
- transport_controller_->GenerateOrGetLastMediaTransportOffer();
- }
-
// If datagram transport is in use, add opaque transport parameters.
if (use_datagram_transport_ || use_datagram_transport_for_data_channels_) {
for (auto& options : session_options->media_description_options) {
@@ -5476,7 +5399,6 @@
}
return rtp_data_channel_->content_name();
case cricket::DCT_SCTP:
- case cricket::DCT_MEDIA_TRANSPORT:
case cricket::DCT_DATA_CHANNEL_TRANSPORT:
case cricket::DCT_DATA_CHANNEL_TRANSPORT_SCTP:
return sctp_mid_;
@@ -7106,7 +7028,6 @@
case cricket::DCT_SCTP:
case cricket::DCT_DATA_CHANNEL_TRANSPORT_SCTP:
case cricket::DCT_DATA_CHANNEL_TRANSPORT:
- case cricket::DCT_MEDIA_TRANSPORT:
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetupDataChannelTransport_n, this,
@@ -7777,20 +7698,14 @@
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
- MediaTransportInterface* media_transport,
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
- RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
bool ret = true;
auto base_channel = GetChannel(mid);
if (base_channel) {
ret = base_channel->SetRtpTransport(rtp_transport);
}
- if (use_media_transport_) {
- RTC_LOG(LS_ERROR) << "Media transport isn't supported.";
- }
-
if (data_channel_transport_ && mid == sctp_mid_ &&
data_channel_transport_ != data_channel_transport) {
// Changed which data channel transport is used for |sctp_mid_| (eg. now
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 3126348..9bc6119 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -20,7 +20,6 @@
#include "api/peer_connection_interface.h"
#include "api/transport/data_channel_transport_interface.h"
-#include "api/transport/media/media_transport_interface.h"
#include "api/turn_customizer.h"
#include "pc/ice_server_parsing.h"
#include "pc/jsep_transport_controller.h"
@@ -1201,7 +1200,6 @@
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
- MediaTransportInterface* media_transport,
DataChannelTransportInterface* data_channel_transport) override;
// RtpSenderBase::SetStreamsObserver override.
@@ -1289,14 +1287,6 @@
bool use_datagram_transport_for_data_channels_receive_only_
RTC_GUARDED_BY(signaling_thread()) = false;
- // Cache configuration_.use_media_transport so that we can access it from
- // other threads.
- // TODO(bugs.webrtc.org/9987): Caching just this bool and allowing the data
- // it's derived from to change is not necessarily sound. Stop doing it.
- rtc::RaceChecker use_media_transport_race_checker_;
- bool use_media_transport_ RTC_GUARDED_BY(use_media_transport_race_checker_) =
- configuration_.use_media_transport;
-
// TODO(zstein): |async_resolver_factory_| can currently be nullptr if it
// is not injected. It should be required once chromium supplies it.
std::unique_ptr<AsyncResolverFactory> async_resolver_factory_
diff --git a/pc/peer_connection_data_channel_unittest.cc b/pc/peer_connection_data_channel_unittest.cc
index a902c76..b063c39 100644
--- a/pc/peer_connection_data_channel_unittest.cc
+++ b/pc/peer_connection_data_channel_unittest.cc
@@ -22,8 +22,6 @@
#include "api/peer_connection_proxy.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
-#include "api/test/fake_media_transport.h"
-#include "api/transport/media/media_transport_interface.h"
#include "media/base/codec.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_constants.h"
@@ -65,8 +63,7 @@
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
- std::unique_ptr<CallFactoryInterface> call_factory,
- std::unique_ptr<MediaTransportFactory> media_transport_factory) {
+ std::unique_ptr<CallFactoryInterface> call_factory) {
PeerConnectionFactoryDependencies deps;
deps.network_thread = network_thread;
deps.worker_thread = worker_thread;
@@ -74,7 +71,6 @@
deps.task_queue_factory = CreateDefaultTaskQueueFactory();
deps.media_engine = std::move(media_engine);
deps.call_factory = std::move(call_factory);
- deps.media_transport_factory = std::move(media_transport_factory);
return deps;
}
@@ -90,8 +86,7 @@
rtc::Thread::Current(),
rtc::Thread::Current(),
std::make_unique<cricket::FakeMediaEngine>(),
- CreateCallFactory(),
- std::make_unique<FakeMediaTransportFactory>())) {}
+ CreateCallFactory())) {}
std::unique_ptr<cricket::SctpTransportInternalFactory>
CreateSctpTransportInternalFactory() {
@@ -385,50 +380,6 @@
EXPECT_EQ(kNewRecvPort, callee_transport->local_port());
}
-TEST_P(PeerConnectionDataChannelTest,
- NoSctpTransportCreatedIfMediaTransportDataChannelsEnabled) {
- RTCConfiguration config;
- config.use_media_transport_for_data_channels = true;
- config.enable_dtls_srtp = false; // SDES is required to use media transport.
- auto caller = CreatePeerConnectionWithDataChannel(config);
-
- ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
- EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
-}
-
-TEST_P(PeerConnectionDataChannelTest,
- MediaTransportDataChannelCreatedEvenIfSctpAvailable) {
- RTCConfiguration config;
- config.use_media_transport_for_data_channels = true;
- config.enable_dtls_srtp = false; // SDES is required to use media transport.
- PeerConnectionFactoryInterface::Options options;
- options.disable_sctp_data_channels = false;
- auto caller = CreatePeerConnectionWithDataChannel(config, options);
-
- ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
- EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
-}
-
-TEST_P(PeerConnectionDataChannelTest,
- CannotEnableBothMediaTransportAndRtpDataChannels) {
- RTCConfiguration config;
- config.enable_rtp_data_channel = true;
- config.use_media_transport_for_data_channels = true;
- config.enable_dtls_srtp = false; // SDES is required to use media transport.
- EXPECT_EQ(CreatePeerConnection(config), nullptr);
-}
-
-// This test now DCHECKs, instead of failing to SetLocalDescription.
-TEST_P(PeerConnectionDataChannelTest, MediaTransportWithoutSdesFails) {
- RTCConfiguration config;
- config.use_media_transport_for_data_channels = true;
- config.enable_dtls_srtp = true; // Disables SDES for data sections.
-
- auto caller = CreatePeerConnectionWithDataChannel(config);
-
- EXPECT_EQ(nullptr, caller);
-}
-
TEST_P(PeerConnectionDataChannelTest, ModernSdpSyntaxByDefault) {
PeerConnectionInterface::RTCOfferAnswerOptions options;
auto caller = CreatePeerConnectionWithDataChannel();
diff --git a/pc/peer_connection_integrationtest.cc b/pc/peer_connection_integrationtest.cc
index e59ce9a..dd06b65 100644
--- a/pc/peer_connection_integrationtest.cc
+++ b/pc/peer_connection_integrationtest.cc
@@ -4278,331 +4278,6 @@
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
-// This test sets up a call between two parties with a media transport data
-// channel.
-TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelEndToEnd) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- rtc_config.use_media_transport_for_data_channels = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Expect that data channel created on caller side will show up for callee as
- // well.
- caller()->CreateDataChannel();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Caller data channel should already exist (it created one). Callee data
- // channel may not exist yet, since negotiation happens in-band, not in SDP.
- ASSERT_NE(nullptr, caller()->data_channel());
- ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
- EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
- EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-
- // Ensure data can be sent in both directions.
- std::string data = "hello world";
- caller()->data_channel()->Send(DataBuffer(data));
- EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
- kDefaultTimeout);
- callee()->data_channel()->Send(DataBuffer(data));
- EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
- kDefaultTimeout);
-}
-
-// Tests that 'zero-rtt' data channel transports (which are ready-to-send as
-// soon as they're created) work correctly.
-TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelZeroRtt) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- rtc_config.use_media_transport_for_data_channels = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Ensure that the callee's media transport is ready-to-send immediately.
- // Note that only the callee can become writable in zero RTTs. The caller
- // must wait for the callee's answer.
- loopback_media_transports()->SetSecondStateAfterConnect(
- webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Expect that data channel created on caller side will show up for callee as
- // well.
- caller()->CreateDataChannel();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- loopback_media_transports()->SetFirstState(
- webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Caller data channel should already exist (it created one). Callee data
- // channel may not exist yet, since negotiation happens in-band, not in SDP.
- ASSERT_NE(nullptr, caller()->data_channel());
- ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
- EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
- EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-
- // Ensure data can be sent in both directions.
- std::string data = "hello world";
- caller()->data_channel()->Send(DataBuffer(data));
- EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
- kDefaultTimeout);
- callee()->data_channel()->Send(DataBuffer(data));
- EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
- kDefaultTimeout);
-}
-
-// Ensure that when the callee closes a media transport data channel, the
-// closing procedure results in the data channel being closed for the caller
-// as well.
-TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelCalleeCloses) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.use_media_transport_for_data_channels = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Create a data channel on the caller and signal it to the callee.
- caller()->CreateDataChannel();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Data channels exist and open on both ends of the connection.
- ASSERT_NE(nullptr, caller()->data_channel());
- ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
- ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
- ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
-
- // Close the data channel on the callee side, and wait for it to reach the
- // "closed" state on both sides.
- callee()->data_channel()->Close();
- EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
- EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
-}
-
-TEST_P(PeerConnectionIntegrationTest,
- MediaTransportDataChannelConfigSentToOtherSide) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.use_media_transport_for_data_channels = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Create a data channel with a non-default configuration and signal it to the
- // callee.
- webrtc::DataChannelInit init;
- init.id = 53;
- init.maxRetransmits = 52;
- caller()->CreateDataChannel("data-channel", &init);
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Ensure that the data channel exists on the callee with the correct
- // configuration.
- ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
- ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
- // Since "negotiate" is false, the "id" parameter is ignored.
- EXPECT_NE(init.id, callee()->data_channel()->id());
- EXPECT_EQ("data-channel", callee()->data_channel()->label());
- EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
- EXPECT_FALSE(callee()->data_channel()->negotiated());
-}
-
-TEST_P(PeerConnectionIntegrationTest, MediaTransportOfferUpgrade) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- rtc_config.use_media_transport = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Do initial offer/answer with just a video track.
- caller()->AddVideoTrack();
- callee()->AddVideoTrack();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Now add an audio track and do another offer/answer.
- caller()->AddAudioTrack();
- callee()->AddAudioTrack();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure both audio and video frames are received end-to-end.
- MediaExpectations media_expectations;
- media_expectations.ExpectBidirectionalAudioAndVideo();
- ASSERT_TRUE(ExpectNewFrames(media_expectations));
-
- // The second offer should not have generated another media transport.
- // Media transport was kept alive, and was not recreated.
- EXPECT_EQ(1, loopback_media_transports()->first_factory_transport_count());
- EXPECT_EQ(1, loopback_media_transports()->second_factory_transport_count());
-}
-
-TEST_P(PeerConnectionIntegrationTest, MediaTransportOfferUpgradeOnTheCallee) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- rtc_config.use_media_transport = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- // Do initial offer/answer with just a video track.
- caller()->AddVideoTrack();
- callee()->AddVideoTrack();
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- // Now add an audio track and do another offer/answer.
- caller()->AddAudioTrack();
- callee()->AddAudioTrack();
- callee()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure both audio and video frames are received end-to-end.
- MediaExpectations media_expectations;
- media_expectations.ExpectBidirectionalAudioAndVideo();
- ASSERT_TRUE(ExpectNewFrames(media_expectations));
-
- // The second offer should not have generated another media transport.
- // Media transport was kept alive, and was not recreated.
- EXPECT_EQ(1, loopback_media_transports()->first_factory_transport_count());
- EXPECT_EQ(1, loopback_media_transports()->second_factory_transport_count());
-}
-
-TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalAudio) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
- rtc_config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
- rtc_config.use_media_transport = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- caller()->AddAudioTrack();
- callee()->AddAudioTrack();
- // Start offer/answer exchange and wait for it to complete.
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- MediaExpectations media_expectations;
- media_expectations.ExpectBidirectionalAudio();
- ASSERT_TRUE(ExpectNewFrames(media_expectations));
-
- webrtc::MediaTransportPair::Stats first_stats =
- loopback_media_transports()->FirstStats();
- webrtc::MediaTransportPair::Stats second_stats =
- loopback_media_transports()->SecondStats();
-
- EXPECT_GT(first_stats.received_audio_frames, 0);
- EXPECT_GE(second_stats.sent_audio_frames, first_stats.received_audio_frames);
-
- EXPECT_GT(second_stats.received_audio_frames, 0);
- EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
-}
-
-TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalVideo) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.use_media_transport = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- caller()->AddVideoTrack();
- callee()->AddVideoTrack();
- // Start offer/answer exchange and wait for it to complete.
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- // Ensure that the media transport is ready.
- loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
- loopback_media_transports()->FlushAsyncInvokes();
-
- MediaExpectations media_expectations;
- media_expectations.ExpectBidirectionalVideo();
- ASSERT_TRUE(ExpectNewFrames(media_expectations));
-
- webrtc::MediaTransportPair::Stats first_stats =
- loopback_media_transports()->FirstStats();
- webrtc::MediaTransportPair::Stats second_stats =
- loopback_media_transports()->SecondStats();
-
- EXPECT_GT(first_stats.received_video_frames, 0);
- EXPECT_GE(second_stats.sent_video_frames, first_stats.received_video_frames);
-
- EXPECT_GT(second_stats.received_video_frames, 0);
- EXPECT_GE(first_stats.sent_video_frames, second_stats.received_video_frames);
-}
-
-TEST_P(PeerConnectionIntegrationTest,
- MediaTransportDataChannelUsesRtpBidirectionalVideo) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.use_media_transport = false;
- rtc_config.use_media_transport_for_data_channels = true;
- rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
- ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
- rtc_config, rtc_config, loopback_media_transports()->first_factory(),
- loopback_media_transports()->second_factory()));
- ConnectFakeSignaling();
-
- caller()->AddVideoTrack();
- callee()->AddVideoTrack();
- // Start offer/answer exchange and wait for it to complete.
- caller()->CreateAndSetAndSignalOffer();
- ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-
- MediaExpectations media_expectations;
- media_expectations.ExpectBidirectionalVideo();
- ASSERT_TRUE(ExpectNewFrames(media_expectations));
-}
-
// Test that the ICE connection and gathering states eventually reach
// "complete".
TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc
index 5a01430..7f42b8c 100644
--- a/pc/peer_connection_interface_unittest.cc
+++ b/pc/peer_connection_interface_unittest.cc
@@ -1421,15 +1421,15 @@
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
- config.use_media_transport = true;
- config.use_media_transport_for_data_channels = true;
+ config.use_datagram_transport = true;
+ config.use_datagram_transport_for_data_channels = true;
EXPECT_TRUE(pc_->SetConfiguration(config).ok());
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
- EXPECT_TRUE(returned_config.use_media_transport);
- EXPECT_TRUE(returned_config.use_media_transport_for_data_channels);
+ EXPECT_TRUE(returned_config.use_datagram_transport);
+ EXPECT_TRUE(returned_config.use_datagram_transport_for_data_channels);
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
diff --git a/pc/peer_connection_media_unittest.cc b/pc/peer_connection_media_unittest.cc
index 62368a2..077c4a3 100644
--- a/pc/peer_connection_media_unittest.cc
+++ b/pc/peer_connection_media_unittest.cc
@@ -20,7 +20,6 @@
#include "api/call/call_factory_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
-#include "api/test/fake_media_transport.h"
#include "media/base/fake_media_engine.h"
#include "p2p/base/fake_port_allocator.h"
#include "pc/media_session.h"
@@ -85,8 +84,6 @@
}
// Creates PeerConnectionFactory and PeerConnection for given configuration.
- // Note that PeerConnectionFactory is created with MediaTransportFactory,
- // because some tests pass config.use_media_transport = true.
WrapperPtr CreatePeerConnection(
const RTCConfiguration& config,
std::unique_ptr<FakeMediaEngine> media_engine) {
@@ -103,8 +100,6 @@
factory_dependencies.event_log_factory =
std::make_unique<RtcEventLogFactory>(
factory_dependencies.task_queue_factory.get());
- factory_dependencies.media_transport_factory =
- std::make_unique<FakeMediaTransportFactory>();
auto pc_factory =
CreateModularPeerConnectionFactory(std::move(factory_dependencies));
@@ -1244,128 +1239,6 @@
audio_options.combined_audio_video_bwe);
}
-TEST_P(PeerConnectionMediaTest, MediaTransportPropagatedToVoiceEngine) {
- RTCConfiguration config;
-
- // Setup PeerConnection to use media transport.
- config.use_media_transport = true;
-
- // Force SDES.
- config.enable_dtls_srtp = false;
-
- auto caller = CreatePeerConnectionWithAudio(config);
- auto callee = CreatePeerConnectionWithAudio(config);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- auto answer = callee->CreateAnswer();
- ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- ASSERT_TRUE(callee_voice);
-
- // Make sure media transport is propagated to voice channel.
- FakeMediaTransport* caller_voice_media_transport =
- static_cast<FakeMediaTransport*>(caller_voice->media_transport());
- FakeMediaTransport* callee_voice_media_transport =
- static_cast<FakeMediaTransport*>(callee_voice->media_transport());
- ASSERT_NE(nullptr, caller_voice_media_transport);
- ASSERT_NE(nullptr, callee_voice_media_transport);
-
- // Make sure media transport is created with correct is_caller.
- EXPECT_TRUE(caller_voice_media_transport->is_caller());
- EXPECT_FALSE(callee_voice_media_transport->is_caller());
-
- // TODO(sukhanov): Propagate media transport to video channel.
- // This test does NOT set up video channels, because currently it causes
- // us to create two media transports.
-}
-
-TEST_P(PeerConnectionMediaTest, MediaTransportOnlyForDataChannels) {
- RTCConfiguration config;
-
- // Setup PeerConnection to use media transport for data channels.
- config.use_media_transport_for_data_channels = true;
-
- // Force SDES.
- config.enable_dtls_srtp = false;
-
- auto caller = CreatePeerConnectionWithAudio(config);
- auto callee = CreatePeerConnectionWithAudio(config);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- ASSERT_TRUE(callee_voice);
-
- // Make sure media transport is not propagated to voice channel.
- EXPECT_EQ(nullptr, caller_voice->media_transport());
- EXPECT_EQ(nullptr, callee_voice->media_transport());
-}
-
-TEST_P(PeerConnectionMediaTest, MediaTransportForMediaAndDataChannels) {
- RTCConfiguration config;
-
- // Setup PeerConnection to use media transport for both media and data
- // channels.
- config.use_media_transport = true;
- config.use_media_transport_for_data_channels = true;
-
- // Force SDES.
- config.enable_dtls_srtp = false;
-
- auto caller = CreatePeerConnectionWithAudio(config);
- auto callee = CreatePeerConnectionWithAudio(config);
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- ASSERT_TRUE(callee_voice);
-
- // Make sure media transport is propagated to voice channel.
- FakeMediaTransport* caller_voice_media_transport =
- static_cast<FakeMediaTransport*>(caller_voice->media_transport());
- FakeMediaTransport* callee_voice_media_transport =
- static_cast<FakeMediaTransport*>(callee_voice->media_transport());
- ASSERT_NE(nullptr, caller_voice_media_transport);
- ASSERT_NE(nullptr, callee_voice_media_transport);
-
- // Make sure media transport is created with correct is_caller.
- EXPECT_TRUE(caller_voice_media_transport->is_caller());
- EXPECT_FALSE(callee_voice_media_transport->is_caller());
-}
-
-TEST_P(PeerConnectionMediaTest, MediaTransportNotPropagatedToVoiceEngine) {
- auto caller = CreatePeerConnectionWithAudioVideo();
- auto callee = CreatePeerConnectionWithAudioVideo();
-
- ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
- auto answer = callee->CreateAnswer();
- ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
-
- auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
- auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
- ASSERT_TRUE(caller_voice);
- ASSERT_TRUE(callee_voice);
-
- // Since we did not setup PeerConnection to use media transport, media
- // transport should not be created / propagated to the voice engine.
- ASSERT_EQ(nullptr, caller_voice->media_transport());
- ASSERT_EQ(nullptr, callee_voice->media_transport());
-
- auto caller_video = caller->media_engine()->GetVideoChannel(0);
- auto callee_video = callee->media_engine()->GetVideoChannel(0);
- ASSERT_EQ(nullptr, caller_video->media_transport());
- ASSERT_EQ(nullptr, callee_video->media_transport());
-}
-
template <typename C>
bool CompareCodecs(const std::vector<webrtc::RtpCodecCapability>& capabilities,
const std::vector<C>& codecs) {
diff --git a/pc/session_description.h b/pc/session_description.h
index f5e3635..7546d12 100644
--- a/pc/session_description.h
+++ b/pc/session_description.h
@@ -515,8 +515,6 @@
std::unique_ptr<SessionDescription> Clone() const;
- struct MediaTransportSetting;
-
// Content accessors.
const ContentInfos& contents() const { return contents_; }
ContentInfos& contents() { return contents_; }
@@ -627,32 +625,6 @@
}
bool extmap_allow_mixed() const { return extmap_allow_mixed_; }
- // Adds the media transport setting.
- // Media transport name uniquely identifies the type of media transport.
- // The name cannot be empty, or repeated in the previously added transport
- // settings.
- void AddMediaTransportSetting(const std::string& media_transport_name,
- const std::string& media_transport_setting) {
- RTC_DCHECK(!media_transport_name.empty());
- for (const auto& setting : media_transport_settings_) {
- RTC_DCHECK(media_transport_name != setting.transport_name)
- << "MediaTransportSetting was already registered, transport_name="
- << setting.transport_name;
- }
- media_transport_settings_.push_back(
- {media_transport_name, media_transport_setting});
- }
-
- // Gets the media transport settings, in order of preference.
- const std::vector<MediaTransportSetting>& MediaTransportSettings() const {
- return media_transport_settings_;
- }
-
- struct MediaTransportSetting {
- std::string transport_name;
- std::string transport_setting;
- };
-
private:
SessionDescription(const SessionDescription&);
@@ -669,8 +641,6 @@
// correctly. If it's included in offer to us we will respond that we support
// it.
bool extmap_allow_mixed_ = false;
-
- std::vector<MediaTransportSetting> media_transport_settings_;
};
// Indicates whether a session description was sent by the local client or
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index 7a42dca..c0e959a 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -229,13 +229,6 @@
static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
-// This is a non-standardized media transport settings.
-// This setting is going to be set in the offer. There may be one or more
-// a=x-mt: settings, and they are in the priority order (the most preferred on
-// top). x-mt setting format depends on the media transport, and is generated by
-// |MediaTransportInterface::GetTransportParametersOffer|.
-static const char kMediaTransportSettingLine[] = "x-mt";
-
// This is a non-standardized setting for plugin transports.
static const char kOpaqueTransportParametersLine[] = "x-opaque";
@@ -530,17 +523,6 @@
InitLine(kLineTypeAttributes, attribute, os);
}
-// Writes an x-mt SDP attribute line based on the media transport settings.
-static void AddMediaTransportLine(
- const cricket::SessionDescription::MediaTransportSetting& setting,
- std::string* message) {
- rtc::StringBuilder os;
- InitAttrLine(kMediaTransportSettingLine, &os);
- os << kSdpDelimiterColon << setting.transport_name << kSdpDelimiterColon
- << rtc::Base64::Encode(setting.transport_setting);
- AddLine(os.str(), message);
-}
-
// Adds an x-otp SDP attribute line based on opaque transport parameters.
static void AddOpaqueTransportLine(
const cricket::OpaqueTransportParameters params,
@@ -902,11 +884,6 @@
// Time Description.
AddLine(kTimeDescription, &message);
- for (const cricket::SessionDescription::MediaTransportSetting& settings :
- desc->MediaTransportSettings()) {
- AddMediaTransportLine(settings, &message);
- }
-
// Group
if (desc->HasGroup(cricket::GROUP_TYPE_BUNDLE)) {
std::string group_line = kAttrGroup;
@@ -2122,28 +2099,6 @@
return true;
}
-bool ParseMediaTransportLine(const std::string& line,
- std::string* transport_name,
- std::string* transport_setting,
- SdpParseError* error) {
- std::string value;
- if (!GetValue(line, kMediaTransportSettingLine, &value, error)) {
- return false;
- }
- std::string media_transport_settings_base64;
- if (!rtc::tokenize_first(value, kSdpDelimiterColonChar, transport_name,
- &media_transport_settings_base64)) {
- return ParseFailedGetValue(line, kMediaTransportSettingLine, error);
- }
- if (!rtc::Base64::Decode(media_transport_settings_base64,
- rtc::Base64::DO_STRICT, transport_setting,
- nullptr)) {
- return ParseFailedGetValue(line, kMediaTransportSettingLine, error);
- }
-
- return true;
-}
-
bool ParseOpaqueTransportLine(const std::string& line,
std::string* protocol,
std::string* transport_parameters,
@@ -2327,24 +2282,6 @@
return false;
}
session_extmaps->push_back(extmap);
- } else if (HasAttribute(line, kMediaTransportSettingLine)) {
- std::string transport_name;
- std::string transport_setting;
- if (!ParseMediaTransportLine(line, &transport_name, &transport_setting,
- error)) {
- return false;
- }
-
- for (const auto& setting : desc->MediaTransportSettings()) {
- if (setting.transport_name == transport_name) {
- // Ignore repeated transport names rather than failing to parse so
- // that in the future the same transport could have multiple configs.
- RTC_LOG(INFO) << "x-mt line with repeated transport, transport_name="
- << transport_name;
- return true;
- }
- }
- desc->AddMediaTransportSetting(transport_name, transport_setting);
}
}
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 5c7e783..e8e937a 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -4640,121 +4640,6 @@
Field("name", &cricket::ContentInfo::name, "")));
}
-// Test that the media transport name and base64-decoded setting is parsed from
-// an a=x-mt line.
-TEST_F(WebRtcSdpTest, ParseMediaTransport) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp += "a=x-mt:rtp:dGVzdDY0\r\n";
- SdpParseError error;
-
- ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error))
- << error.description;
- const auto& settings = output.description()->MediaTransportSettings();
- ASSERT_EQ(1u, settings.size());
- EXPECT_EQ("rtp", settings[0].transport_name);
- EXPECT_EQ("test64", settings[0].transport_setting);
-}
-
-// Test that an a=x-mt line fails to parse if its setting is invalid base 64.
-TEST_F(WebRtcSdpTest, ParseMediaTransportInvalidBase64) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp += "a=x-mt:rtp:ThisIsInvalidBase64\r\n";
- SdpParseError error;
-
- ASSERT_FALSE(webrtc::SdpDeserialize(sdp, &output, &error));
-}
-
-// Test that multiple a=x-mt lines are parsed in the order of preference (the
-// order of the lines in the SDP).
-TEST_F(WebRtcSdpTest, ParseMediaTransportMultipleLines) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp +=
- "a=x-mt:rtp:dGVzdDY0\r\n"
- "a=x-mt:generic:Z2VuZXJpY3NldHRpbmc=\r\n";
- SdpParseError error;
-
- ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error))
- << error.description;
- const auto& settings = output.description()->MediaTransportSettings();
- ASSERT_EQ(2u, settings.size());
- EXPECT_EQ("rtp", settings[0].transport_name);
- EXPECT_EQ("test64", settings[0].transport_setting);
- EXPECT_EQ("generic", settings[1].transport_name);
- EXPECT_EQ("genericsetting", settings[1].transport_setting);
-}
-
-// Test that only the first a=x-mt line associated with a transport name is
-// parsed and the rest ignored.
-TEST_F(WebRtcSdpTest, ParseMediaTransportSkipRepeatedTransport) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp +=
- "a=x-mt:rtp:dGVzdDY0\r\n"
- "a=x-mt:rtp:Z2VuZXJpY3NldHRpbmc=\r\n";
- SdpParseError error;
-
- // Repeated 'rtp' transport setting. We still parse the SDP successfully,
- // but ignore the repeated transport.
- ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error));
- const auto& settings = output.description()->MediaTransportSettings();
- EXPECT_EQ("test64", settings[0].transport_setting);
-}
-
-// Test that an a=x-mt line fails to parse if it is missing a setting.
-TEST_F(WebRtcSdpTest, ParseMediaTransportMalformedLine) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp += "a=x-mt:rtp\r\n";
- SdpParseError error;
-
- ASSERT_FALSE(webrtc::SdpDeserialize(sdp, &output, &error));
-}
-
-// Test that an a=x-mt line fails to parse if its missing a name and setting.
-TEST_F(WebRtcSdpTest, ParseMediaTransportMalformedLine2) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp += "a=x-mt\r\n";
- SdpParseError error;
-
- ASSERT_FALSE(webrtc::SdpDeserialize(sdp, &output, &error));
-}
-
-TEST_F(WebRtcSdpTest, ParseMediaTransportIgnoreNonsenseAttributeLines) {
- JsepSessionDescription output(kDummyType);
- std::string sdp = kSdpSessionString;
- sdp += "a=x-nonsense:rtp:dGVzdDY0\r\n";
- SdpParseError error;
-
- ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error))
- << error.description;
- EXPECT_TRUE(output.description()->MediaTransportSettings().empty());
-}
-
-TEST_F(WebRtcSdpTest, SerializeMediaTransportSettings) {
- auto description = std::make_unique<cricket::SessionDescription>();
-
- JsepSessionDescription output(SdpType::kOffer);
- // JsepSessionDescription takes ownership of the description.
- output.Initialize(std::move(description), "session_id", "session_version");
- output.description()->AddMediaTransportSetting("foo", "bar");
- std::string serialized_out;
- output.ToString(&serialized_out);
- ASSERT_THAT(serialized_out, ::testing::HasSubstr("\r\na=x-mt:foo:YmFy\r\n"));
-}
-
-TEST_F(WebRtcSdpTest, SerializeMediaTransportSettingsTestCopy) {
- cricket::SessionDescription description;
- description.AddMediaTransportSetting("name", "setting");
- std::unique_ptr<cricket::SessionDescription> copy = description.Clone();
- ASSERT_EQ(1u, copy->MediaTransportSettings().size());
- EXPECT_EQ("name", copy->MediaTransportSettings()[0].transport_name);
- EXPECT_EQ("setting", copy->MediaTransportSettings()[0].transport_setting);
-}
-
TEST_F(WebRtcSdpTest, SerializeWithDefaultSctpProtocol) {
AddSctpDataChannel(false); // Don't use sctpmap
JsepSessionDescription jsep_desc(kDummyType);
diff --git a/test/call_test.cc b/test/call_test.cc
index 9f26cc6..10b631a 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -36,7 +36,7 @@
task_queue_factory_(CreateDefaultTaskQueueFactory()),
send_event_log_(std::make_unique<RtcEventLogNull>()),
recv_event_log_(std::make_unique<RtcEventLogNull>()),
- audio_send_config_(/*send_transport=*/nullptr, MediaTransportConfig()),
+ audio_send_config_(/*send_transport=*/nullptr),
audio_send_stream_(nullptr),
frame_generator_capturer_(nullptr),
fake_encoder_factory_([this]() {
@@ -275,8 +275,7 @@
RTC_DCHECK_LE(num_audio_streams, 1);
RTC_DCHECK_LE(num_flexfec_streams, 1);
if (num_audio_streams > 0) {
- AudioSendStream::Config audio_send_config(send_transport,
- MediaTransportConfig());
+ AudioSendStream::Config audio_send_config(send_transport);
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
diff --git a/test/peer_scenario/peer_scenario_client.cc b/test/peer_scenario/peer_scenario_client.cc
index 28cbb6e..d8f2b65 100644
--- a/test/peer_scenario/peer_scenario_client.cc
+++ b/test/peer_scenario/peer_scenario_client.cc
@@ -185,7 +185,6 @@
pcf_deps.fec_controller_factory = nullptr;
pcf_deps.network_controller_factory = nullptr;
pcf_deps.network_state_predictor_factory = nullptr;
- pcf_deps.media_transport_factory = nullptr;
pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index f5d2116..2738f69 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -73,8 +73,7 @@
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
Transport* send_transport)
: sender_(sender), config_(config) {
- AudioSendStream::Config send_config(send_transport,
- webrtc::MediaTransportConfig());
+ AudioSendStream::Config send_config(send_transport);
ssrc_ = sender->GetNextAudioSsrc();
send_config.rtp.ssrc = ssrc_;
SdpAudioFormat::Parameters sdp_params;
diff --git a/test/scenario/stats_collection.h b/test/scenario/stats_collection.h
index 64cb58c..908385e 100644
--- a/test/scenario/stats_collection.h
+++ b/test/scenario/stats_collection.h
@@ -15,6 +15,7 @@
#include "absl/types/optional.h"
#include "call/call.h"
+#include "rtc_base/thread.h"
#include "test/logging/log_writer.h"
#include "test/scenario/performance_stats.h"
diff --git a/video/encoder_rtcp_feedback.cc b/video/encoder_rtcp_feedback.cc
index 19a8f64..a736d83 100644
--- a/video/encoder_rtcp_feedback.cc
+++ b/video/encoder_rtcp_feedback.cc
@@ -67,16 +67,6 @@
video_stream_encoder_->SendKeyFrame();
}
-void EncoderRtcpFeedback::OnKeyFrameRequested(uint64_t channel_id) {
- if (channel_id != ssrcs_[0]) {
- RTC_LOG(LS_INFO) << "Key frame request on unknown channel id " << channel_id
- << " expected " << ssrcs_[0];
- return;
- }
-
- video_stream_encoder_->SendKeyFrame();
-}
-
void EncoderRtcpFeedback::OnReceivedLossNotification(
uint32_t ssrc,
uint16_t seq_num_of_last_decodable,
diff --git a/video/encoder_rtcp_feedback.h b/video/encoder_rtcp_feedback.h
index 21624db..b5dd028 100644
--- a/video/encoder_rtcp_feedback.h
+++ b/video/encoder_rtcp_feedback.h
@@ -12,7 +12,6 @@
#include <vector>
-#include "api/transport/media/media_transport_interface.h"
#include "api/video/video_stream_encoder_interface.h"
#include "call/rtp_video_sender_interface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -24,12 +23,9 @@
class VideoStreamEncoderInterface;
// This class passes feedback (such as key frame requests or loss notifications)
-// from either Mediatransport or the RtpRtcp module.
-// TODO(bugs.webrtc.org/9719): Should be eliminated when RtpMediaTransport is
-// implemented.
+// from the RtpRtcp module.
class EncoderRtcpFeedback : public RtcpIntraFrameObserver,
- public RtcpLossNotificationObserver,
- public MediaTransportKeyFrameRequestCallback {
+ public RtcpLossNotificationObserver {
public:
EncoderRtcpFeedback(Clock* clock,
const std::vector<uint32_t>& ssrcs,
@@ -40,9 +36,6 @@
void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
- // Implements MediaTransportKeyFrameRequestCallback
- void OnKeyFrameRequested(uint64_t channel_id) override;
-
// Implements RtcpLossNotificationObserver.
void OnReceivedLossNotification(uint32_t ssrc,
uint16_t seq_num_of_last_decodable,
diff --git a/video/encoder_rtcp_feedback_unittest.cc b/video/encoder_rtcp_feedback_unittest.cc
index b49a0b9..81ac22b 100644
--- a/video/encoder_rtcp_feedback_unittest.cc
+++ b/video/encoder_rtcp_feedback_unittest.cc
@@ -55,9 +55,4 @@
encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
}
-TEST_F(VieKeyRequestTest, TriggerRequestFromMediaTransport) {
- EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
- encoder_rtcp_feedback_.OnKeyFrameRequested(kSsrc);
-}
-
} // namespace webrtc
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 8f7d612..ad8c808 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -22,7 +22,6 @@
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_base.h"
-#include "api/transport/media/media_transport_config.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/fake_network_pipe.h"
@@ -1410,8 +1409,7 @@
}
void VideoQualityTest::SetupAudio(Transport* transport) {
- AudioSendStream::Config audio_send_config(transport,
- webrtc::MediaTransportConfig());
+ AudioSendStream::Config audio_send_config(transport);
audio_send_config.rtp.ssrc = kAudioSendSsrc;
// Add extension to enable audio send side BWE, and allow audio bit rate
diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc
index a683f7d..8213c64 100644
--- a/video/video_receive_stream.cc
+++ b/video/video_receive_stream.cc
@@ -125,40 +125,6 @@
const char* ImplementationName() const override { return "NullVideoDecoder"; }
};
-// Inherit video_coding::EncodedFrame, which is the class used by
-// video_coding::FrameBuffer and other components in the receive pipeline. It's
-// a subclass of EncodedImage, and it always owns the buffer.
-class EncodedFrameForMediaTransport : public video_coding::EncodedFrame {
- public:
- explicit EncodedFrameForMediaTransport(
- MediaTransportEncodedVideoFrame frame) {
- // TODO(nisse): This is ugly. We copy the EncodedImage (a base class of
- // ours, in several steps), to get all the meta data. We should be using
- // std::move in some way. Then we also need to handle the case of an unowned
- // buffer, in which case we need to make an owned copy.
- *static_cast<class EncodedImage*>(this) = frame.encoded_image();
-
- // If we don't already own the buffer, make a copy.
- Retain();
-
- _payloadType = static_cast<uint8_t>(frame.payload_type());
-
- // TODO(nisse): frame_id and picture_id are probably not the same thing. For
- // a single layer, this should be good enough.
- id.picture_id = frame.frame_id();
- id.spatial_layer = frame.encoded_image().SpatialIndex().value_or(0);
- num_references = std::min(static_cast<size_t>(kMaxFrameReferences),
- frame.referenced_frame_ids().size());
- for (size_t i = 0; i < num_references; i++) {
- references[i] = frame.referenced_frame_ids()[i];
- }
- }
-
- // TODO(nisse): Implement. Not sure how they are used.
- int64_t ReceivedTime() const override { return 0; }
- int64_t RenderTime() const override { return 0; }
-};
-
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Maximum time between frames before resetting the FrameBuffer to avoid RTP
// timestamps wraparound to affect FrameBuffer.
@@ -238,23 +204,18 @@
new video_coding::FrameBuffer(clock_, timing_.get(), &stats_proxy_));
process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE);
- if (config_.media_transport()) {
- config_.media_transport()->SetReceiveVideoSink(this);
- config_.media_transport()->AddRttObserver(this);
+ // Register with RtpStreamReceiverController.
+ media_receiver_ = receiver_controller->CreateReceiver(
+ config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
+ if (config_.rtp.rtx_ssrc) {
+ rtx_receive_stream_ = std::make_unique<RtxReceiveStream>(
+ &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
+ config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
+ rtx_receiver_ = receiver_controller->CreateReceiver(
+ config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
} else {
- // Register with RtpStreamReceiverController.
- media_receiver_ = receiver_controller->CreateReceiver(
- config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
- if (config_.rtp.rtx_ssrc) {
- rtx_receive_stream_ = std::make_unique<RtxReceiveStream>(
- &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
- config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
- rtx_receiver_ = receiver_controller->CreateReceiver(
- config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
- } else {
- rtp_receive_statistics_->EnableRetransmitDetection(config.rtp.remote_ssrc,
- true);
- }
+ rtp_receive_statistics_->EnableRetransmitDetection(config.rtp.remote_ssrc,
+ true);
}
}
@@ -281,10 +242,6 @@
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
Stop();
- if (config_.media_transport()) {
- config_.media_transport()->SetReceiveVideoSink(nullptr);
- config_.media_transport()->RemoveRttObserver(this);
- }
process_thread_->DeRegisterModule(&rtp_stream_sync_);
}
@@ -536,11 +493,7 @@
}
void VideoReceiveStream::RequestKeyFrame(int64_t timestamp_ms) {
- if (config_.media_transport()) {
- config_.media_transport()->RequestKeyFrame(config_.rtp.remote_ssrc);
- } else {
- rtp_video_stream_receiver_.RequestKeyFrame();
- }
+ rtp_video_stream_receiver_.RequestKeyFrame();
last_keyframe_request_ms_ = timestamp_ms;
}
@@ -573,22 +526,12 @@
rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
}
-void VideoReceiveStream::OnData(uint64_t channel_id,
- MediaTransportEncodedVideoFrame frame) {
- OnCompleteFrame(
- std::make_unique<EncodedFrameForMediaTransport>(std::move(frame)));
-}
-
void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
RTC_DCHECK_RUN_ON(&module_process_sequence_checker_);
frame_buffer_->UpdateRtt(max_rtt_ms);
rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
}
-void VideoReceiveStream::OnRttUpdated(int64_t rtt_ms) {
- frame_buffer_->UpdateRtt(rtt_ms);
-}
-
int VideoReceiveStream::id() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
return config_.rtp.remote_ssrc;
diff --git a/video/video_receive_stream.h b/video/video_receive_stream.h
index 7c68563..2a4e0d1 100644
--- a/video/video_receive_stream.h
+++ b/video/video_receive_stream.h
@@ -15,7 +15,6 @@
#include <vector>
#include "api/task_queue/task_queue_factory.h"
-#include "api/transport/media/media_transport_interface.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.h"
@@ -49,9 +48,7 @@
public NackSender,
public video_coding::OnCompleteFrameCallback,
public Syncable,
- public CallStatsObserver,
- public MediaTransportVideoSinkInterface,
- public MediaTransportRttObserver {
+ public CallStatsObserver {
public:
VideoReceiveStream(TaskQueueFactory* task_queue_factory,
RtpStreamReceiverControllerInterface* receiver_controller,
@@ -110,17 +107,9 @@
void OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) override;
- // Implements MediaTransportVideoSinkInterface, converts the received frame to
- // OnCompleteFrameCallback
- void OnData(uint64_t channel_id,
- MediaTransportEncodedVideoFrame frame) override;
-
// Implements CallStatsObserver::OnRttUpdate
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
- // Implements MediaTransportRttObserver::OnRttUpdated
- void OnRttUpdated(int64_t rtt_ms) override;
-
// Implements Syncable.
int id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 12d9dac..8fae407 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -103,7 +103,7 @@
event_log, &config_, encoder_config.max_bitrate_bps,
encoder_config.bitrate_priority, suspended_ssrcs,
suspended_payload_states, encoder_config.content_type,
- std::move(fec_controller), config_.media_transport));
+ std::move(fec_controller)));
},
[this]() { thread_sync_event_.Set(); }));
diff --git a/video/video_send_stream_impl.cc b/video/video_send_stream_impl.cc
index 31dddcc..97f3bb7 100644
--- a/video/video_send_stream_impl.cc
+++ b/video/video_send_stream_impl.cc
@@ -181,8 +181,7 @@
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type,
- std::unique_ptr<FecController> fec_controller,
- MediaTransportInterface* media_transport)
+ std::unique_ptr<FecController> fec_controller)
: clock_(clock),
has_alr_probing_(config->periodic_alr_bandwidth_probing ||
GetAlrSettings(content_type)),
@@ -216,8 +215,7 @@
event_log,
std::move(fec_controller),
CreateFrameEncryptionConfig(config_))),
- weak_ptr_factory_(this),
- media_transport_(media_transport) {
+ weak_ptr_factory_(this) {
video_stream_encoder->SetFecControllerOverride(rtp_video_sender_);
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
@@ -225,14 +223,7 @@
encoder_feedback_.SetRtpVideoSender(rtp_video_sender_);
- if (media_transport_) {
- // The configured ssrc is interpreted as a channel id, so there must be
- // exactly one.
- RTC_DCHECK_EQ(config_->rtp.ssrcs.size(), 1);
- media_transport_->SetKeyFrameRequestCallback(&encoder_feedback_);
- } else {
- RTC_DCHECK(!config_->rtp.ssrcs.empty());
- }
+ RTC_DCHECK(!config_->rtp.ssrcs.empty());
RTC_DCHECK(call_stats_);
RTC_DCHECK(transport_);
RTC_DCHECK_NE(initial_encoder_max_bitrate, 0);
@@ -310,9 +301,6 @@
<< "VideoSendStreamImpl::Stop not called";
RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
transport_->DestroyRtpVideoSender(rtp_video_sender_);
- if (media_transport_) {
- media_transport_->SetKeyFrameRequestCallback(nullptr);
- }
}
void VideoSendStreamImpl::RegisterProcessThread(
@@ -581,31 +569,8 @@
}
EncodedImageCallback::Result result(EncodedImageCallback::Result::OK);
- if (media_transport_) {
- int64_t frame_id;
- {
- // TODO(nisse): Responsibility for allocation of frame ids should move to
- // VideoStreamEncoder.
- rtc::CritScope cs(&media_transport_id_lock_);
- frame_id = media_transport_frame_id_++;
- }
- // TODO(nisse): Responsibility for reference meta data should be moved
- // upstream, ideally close to the encoders, but probably VideoStreamEncoder
- // will need to do some translation to produce reference info using frame
- // ids.
- std::vector<int64_t> referenced_frame_ids;
- if (encoded_image._frameType != VideoFrameType::kVideoFrameKey) {
- RTC_DCHECK_GT(frame_id, 0);
- referenced_frame_ids.push_back(frame_id - 1);
- }
- media_transport_->SendVideoFrame(
- config_->rtp.ssrcs[0], webrtc::MediaTransportEncodedVideoFrame(
- frame_id, referenced_frame_ids,
- config_->rtp.payload_type, encoded_image));
- } else {
- result = rtp_video_sender_->OnEncodedImage(
- encoded_image, codec_specific_info, fragmentation);
- }
+ result = rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info,
+ fragmentation);
// Check if there's a throttled VideoBitrateAllocation that we should try
// sending.
rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
diff --git a/video/video_send_stream_impl.h b/video/video_send_stream_impl.h
index 091ac0f..4195efc 100644
--- a/video/video_send_stream_impl.h
+++ b/video/video_send_stream_impl.h
@@ -87,8 +87,7 @@
std::map<uint32_t, RtpState> suspended_ssrcs,
std::map<uint32_t, RtpPayloadState> suspended_payload_states,
VideoEncoderConfig::ContentType content_type,
- std::unique_ptr<FecController> fec_controller,
- MediaTransportInterface* media_transport);
+ std::unique_ptr<FecController> fec_controller);
~VideoSendStreamImpl() override;
// RegisterProcessThread register |module_process_thread| with those objects
@@ -199,10 +198,6 @@
};
absl::optional<VbaSendContext> video_bitrate_allocation_context_
RTC_GUARDED_BY(worker_queue_);
- MediaTransportInterface* const media_transport_;
- rtc::CriticalSection media_transport_id_lock_;
- int64_t media_transport_frame_id_ RTC_GUARDED_BY(media_transport_id_lock_) =
- 0;
};
} // namespace internal
} // namespace webrtc
diff --git a/video/video_send_stream_impl_unittest.cc b/video/video_send_stream_impl_unittest.cc
index ce88a36..1c44cc8 100644
--- a/video/video_send_stream_impl_unittest.cc
+++ b/video/video_send_stream_impl_unittest.cc
@@ -137,8 +137,7 @@
&video_stream_encoder_, &event_log_, &config_,
initial_encoder_max_bitrate, initial_encoder_bitrate_priority,
suspended_ssrcs, suspended_payload_states, content_type,
- std::make_unique<FecControllerDefault>(&clock_),
- /*media_transport=*/nullptr);
+ std::make_unique<FecControllerDefault>(&clock_));
}
protected:
