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Diffstat (limited to 'third_party/libwebrtc/pc/rtp_transceiver.h')
| -rw-r--r-- | third_party/libwebrtc/pc/rtp_transceiver.h | 383 |
1 files changed, 383 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/rtp_transceiver.h b/third_party/libwebrtc/pc/rtp_transceiver.h new file mode 100644 index 0000000000..0844b349b6 --- /dev/null +++ b/third_party/libwebrtc/pc/rtp_transceiver.h @@ -0,0 +1,383 @@ +/* + * Copyright 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef PC_RTP_TRANSCEIVER_H_ +#define PC_RTP_TRANSCEIVER_H_ + +#include <stddef.h> + +#include <functional> +#include <memory> +#include <string> +#include <vector> + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "api/audio_options.h" +#include "api/jsep.h" +#include "api/media_types.h" +#include "api/rtc_error.h" +#include "api/rtp_parameters.h" +#include "api/rtp_receiver_interface.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_direction.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "api/task_queue/task_queue_base.h" +#include "api/video/video_bitrate_allocator_factory.h" +#include "media/base/media_channel.h" +#include "pc/channel_interface.h" +#include "pc/connection_context.h" +#include "pc/proxy.h" +#include "pc/rtp_receiver.h" +#include "pc/rtp_receiver_proxy.h" +#include "pc/rtp_sender.h" +#include "pc/rtp_sender_proxy.h" +#include "pc/rtp_transport_internal.h" +#include "pc/session_description.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread_annotations.h" + +namespace cricket { +class MediaEngineInterface; +} + +namespace webrtc { + +class PeerConnectionSdpMethods; + +// Implementation of the public RtpTransceiverInterface. +// +// The RtpTransceiverInterface is only intended to be used with a PeerConnection +// that enables Unified Plan SDP. Thus, the methods that only need to implement +// public API features and are not used internally can assume exactly one sender +// and receiver. +// +// Since the RtpTransceiver is used internally by PeerConnection for tracking +// RtpSenders, RtpReceivers, and BaseChannels, and PeerConnection needs to be +// backwards compatible with Plan B SDP, this implementation is more flexible +// than that required by the WebRTC specification. +// +// With Plan B SDP, an RtpTransceiver can have any number of senders and +// receivers which map to a=ssrc lines in the m= section. +// With Unified Plan SDP, an RtpTransceiver will have exactly one sender and one +// receiver which are encapsulated by the m= section. +// +// This class manages the RtpSenders, RtpReceivers, and BaseChannel associated +// with this m= section. Since the transceiver, senders, and receivers are +// reference counted and can be referenced from JavaScript (in Chromium), these +// objects must be ready to live for an arbitrary amount of time. The +// BaseChannel is not reference counted, so +// the PeerConnection must take care of creating/deleting the BaseChannel. +// +// The RtpTransceiver is specialized to either audio or video according to the +// MediaType specified in the constructor. Audio RtpTransceivers will have +// AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers +// will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel. +class RtpTransceiver : public RtpTransceiverInterface, + public sigslot::has_slots<> { + public: + // Construct a Plan B-style RtpTransceiver with no senders, receivers, or + // channel set. + // `media_type` specifies the type of RtpTransceiver (and, by transitivity, + // the type of senders, receivers, and channel). Can either by audio or video. + RtpTransceiver(cricket::MediaType media_type, ConnectionContext* context); + // Construct a Unified Plan-style RtpTransceiver with the given sender and + // receiver. The media type will be derived from the media types of the sender + // and receiver. The sender and receiver should have the same media type. + // `HeaderExtensionsToOffer` is used for initializing the return value of + // HeaderExtensionsToOffer(). + RtpTransceiver( + rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender, + rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> + receiver, + ConnectionContext* context, + std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer, + std::function<void()> on_negotiation_needed); + ~RtpTransceiver() override; + + // Not copyable or movable. + RtpTransceiver(const RtpTransceiver&) = delete; + RtpTransceiver& operator=(const RtpTransceiver&) = delete; + RtpTransceiver(RtpTransceiver&&) = delete; + RtpTransceiver& operator=(RtpTransceiver&&) = delete; + + // Returns the Voice/VideoChannel set for this transceiver. May be null if + // the transceiver is not in the currently set local/remote description. + cricket::ChannelInterface* channel() const { return channel_.get(); } + + // Creates the Voice/VideoChannel and sets it. + RTCError CreateChannel( + absl::string_view mid, + Call* call_ptr, + const cricket::MediaConfig& media_config, + bool srtp_required, + CryptoOptions crypto_options, + const cricket::AudioOptions& audio_options, + const cricket::VideoOptions& video_options, + VideoBitrateAllocatorFactory* video_bitrate_allocator_factory, + std::function<RtpTransportInternal*(absl::string_view)> transport_lookup); + + // Sets the Voice/VideoChannel. The caller must pass in the correct channel + // implementation based on the type of the transceiver. The call must + // furthermore be made on the signaling thread. + // + // `channel`: The channel instance to be associated with the transceiver. + // This must be a valid pointer. + // The state of the object + // is expected to be newly constructed and not initalized for network + // activity (see next parameter for more). + // + // The transceiver takes ownership of `channel`. + // + // `transport_lookup`: This + // callback function will be used to look up the `RtpTransport` object + // to associate with the channel via `BaseChannel::SetRtpTransport`. + // The lookup function will be called on the network thread, synchronously + // during the call to `SetChannel`. This means that the caller of + // `SetChannel()` may provide a callback function that references state + // that exists within the calling scope of SetChannel (e.g. a variable + // on the stack). + // The reason for this design is to limit the number of times we jump + // synchronously to the network thread from the signaling thread. + // The callback allows us to combine the transport lookup with network + // state initialization of the channel object. + // ClearChannel() must be used before calling SetChannel() again. + void SetChannel(std::unique_ptr<cricket::ChannelInterface> channel, + std::function<RtpTransportInternal*(const std::string&)> + transport_lookup); + + // Clear the association between the transceiver and the channel. + void ClearChannel(); + + // Adds an RtpSender of the appropriate type to be owned by this transceiver. + // Must not be null. + void AddSender( + rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender); + + // Removes the given RtpSender. Returns false if the sender is not owned by + // this transceiver. + bool RemoveSender(RtpSenderInterface* sender); + + // Returns a vector of the senders owned by this transceiver. + std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> + senders() const { + return senders_; + } + + // Adds an RtpReceiver of the appropriate type to be owned by this + // transceiver. Must not be null. + void AddReceiver( + rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> + receiver); + + // Removes the given RtpReceiver. Returns false if the sender is not owned by + // this transceiver. + bool RemoveReceiver(RtpReceiverInterface* receiver); + + // Returns a vector of the receivers owned by this transceiver. + std::vector< + rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> + receivers() const { + return receivers_; + } + + // Returns the backing object for the transceiver's Unified Plan sender. + rtc::scoped_refptr<RtpSenderInternal> sender_internal() const; + + // Returns the backing object for the transceiver's Unified Plan receiver. + rtc::scoped_refptr<RtpReceiverInternal> receiver_internal() const; + + // RtpTransceivers are not associated until they have a corresponding media + // section set in SetLocalDescription or SetRemoteDescription. Therefore, + // when setting a local offer we need a way to remember which transceiver was + // used to create which media section in the offer. Storing the mline index + // in CreateOffer is specified in JSEP to allow us to do that. + absl::optional<size_t> mline_index() const { return mline_index_; } + void set_mline_index(absl::optional<size_t> mline_index) { + mline_index_ = mline_index; + } + + // Sets the MID for this transceiver. If the MID is not null, then the + // transceiver is considered "associated" with the media section that has the + // same MID. + void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; } + + // Sets the intended direction for this transceiver. Intended to be used + // internally over SetDirection since this does not trigger a negotiation + // needed callback. + void set_direction(RtpTransceiverDirection direction) { + direction_ = direction; + } + + // Sets the current direction for this transceiver as negotiated in an offer/ + // answer exchange. The current direction is null before an answer with this + // transceiver has been set. + void set_current_direction(RtpTransceiverDirection direction); + + // Sets the fired direction for this transceiver. The fired direction is null + // until SetRemoteDescription is called or an answer is set (either local or + // remote) after which the only valid reason to go back to null is rollback. + void set_fired_direction(absl::optional<RtpTransceiverDirection> direction); + + // According to JSEP rules for SetRemoteDescription, RtpTransceivers can be + // reused only if they were added by AddTrack. + void set_created_by_addtrack(bool created_by_addtrack) { + created_by_addtrack_ = created_by_addtrack; + } + // If AddTrack has been called then transceiver can't be removed during + // rollback. + void set_reused_for_addtrack(bool reused_for_addtrack) { + reused_for_addtrack_ = reused_for_addtrack; + } + + bool created_by_addtrack() const { return created_by_addtrack_; } + + bool reused_for_addtrack() const { return reused_for_addtrack_; } + + // Returns true if this transceiver has ever had the current direction set to + // sendonly or sendrecv. + bool has_ever_been_used_to_send() const { + return has_ever_been_used_to_send_; + } + + // Informs the transceiver that its owning + // PeerConnection is closed. + void SetPeerConnectionClosed(); + + // Executes the "stop the RTCRtpTransceiver" procedure from + // the webrtc-pc specification, described under the stop() method. + void StopTransceiverProcedure(); + + // Fired when the RtpTransceiver state changes such that negotiation is now + // needed (e.g., in response to a direction change). + // sigslot::signal0<> SignalNegotiationNeeded; + + // RtpTransceiverInterface implementation. + cricket::MediaType media_type() const override; + absl::optional<std::string> mid() const override; + rtc::scoped_refptr<RtpSenderInterface> sender() const override; + rtc::scoped_refptr<RtpReceiverInterface> receiver() const override; + bool stopped() const override; + bool stopping() const override; + RtpTransceiverDirection direction() const override; + RTCError SetDirectionWithError( + RtpTransceiverDirection new_direction) override; + absl::optional<RtpTransceiverDirection> current_direction() const override; + absl::optional<RtpTransceiverDirection> fired_direction() const override; + RTCError StopStandard() override; + void StopInternal() override; + RTCError SetCodecPreferences( + rtc::ArrayView<RtpCodecCapability> codecs) override; + std::vector<RtpCodecCapability> codec_preferences() const override { + return codec_preferences_; + } + std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer() + const override; + std::vector<RtpHeaderExtensionCapability> HeaderExtensionsNegotiated() + const override; + RTCError SetOfferedRtpHeaderExtensions( + rtc::ArrayView<const RtpHeaderExtensionCapability> + header_extensions_to_offer) override; + + // Called on the signaling thread when the local or remote content description + // is updated. Used to update the negotiated header extensions. + // TODO(tommi): The implementation of this method is currently very simple and + // only used for updating the negotiated headers. However, we're planning to + // move all the updates done on the channel from the transceiver into this + // method. This will happen with the ownership of the channel object being + // moved into the transceiver. + void OnNegotiationUpdate(SdpType sdp_type, + const cricket::MediaContentDescription* content); + + private: + cricket::MediaEngineInterface* media_engine() const { + return context_->media_engine(); + } + ConnectionContext* context() const { return context_; } + void OnFirstPacketReceived(); + void StopSendingAndReceiving(); + // Delete a channel, and ensure that references to its media channel + // are updated before deleting it. + void PushNewMediaChannelAndDeleteChannel( + std::unique_ptr<cricket::ChannelInterface> channel_to_delete); + + // Enforce that this object is created, used and destroyed on one thread. + TaskQueueBase* const thread_; + const bool unified_plan_; + const cricket::MediaType media_type_; + rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_; + std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> + senders_; + std::vector< + rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> + receivers_; + + bool stopped_ RTC_GUARDED_BY(thread_) = false; + bool stopping_ RTC_GUARDED_BY(thread_) = false; + bool is_pc_closed_ = false; + RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive; + absl::optional<RtpTransceiverDirection> current_direction_; + absl::optional<RtpTransceiverDirection> fired_direction_; + absl::optional<std::string> mid_; + absl::optional<size_t> mline_index_; + bool created_by_addtrack_ = false; + bool reused_for_addtrack_ = false; + bool has_ever_been_used_to_send_ = false; + + // Accessed on both thread_ and the network thread. Considered safe + // because all access on the network thread is within an invoke() + // from thread_. + std::unique_ptr<cricket::ChannelInterface> channel_ = nullptr; + ConnectionContext* const context_; + std::vector<RtpCodecCapability> codec_preferences_; + std::vector<RtpHeaderExtensionCapability> header_extensions_to_offer_; + + // `negotiated_header_extensions_` is read and written to on the signaling + // thread from the SdpOfferAnswerHandler class (e.g. + // PushdownMediaDescription(). + cricket::RtpHeaderExtensions negotiated_header_extensions_ + RTC_GUARDED_BY(thread_); + + const std::function<void()> on_negotiation_needed_; +}; + +BEGIN_PRIMARY_PROXY_MAP(RtpTransceiver) + +PROXY_PRIMARY_THREAD_DESTRUCTOR() +BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type) +PROXY_CONSTMETHOD0(absl::optional<std::string>, mid) +PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender) +PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver) +PROXY_CONSTMETHOD0(bool, stopped) +PROXY_CONSTMETHOD0(bool, stopping) +PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction) +PROXY_METHOD1(webrtc::RTCError, SetDirectionWithError, RtpTransceiverDirection) +PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction) +PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction) +PROXY_METHOD0(webrtc::RTCError, StopStandard) +PROXY_METHOD0(void, StopInternal) +PROXY_METHOD1(webrtc::RTCError, + SetCodecPreferences, + rtc::ArrayView<RtpCodecCapability>) +PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences) +PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>, + HeaderExtensionsToOffer) +PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>, + HeaderExtensionsNegotiated) +PROXY_METHOD1(webrtc::RTCError, + SetOfferedRtpHeaderExtensions, + rtc::ArrayView<const RtpHeaderExtensionCapability>) +END_PROXY_MAP(RtpTransceiver) + +} // namespace webrtc + +#endif // PC_RTP_TRANSCEIVER_H_ |