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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <stddef.h>
#include <algorithm>
#include <functional>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_base.h"
#include "pc/channel_interface.h"
#include "pc/channel_manager.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 "rtc_base/ref_counted_object.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// 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 and is owned by the ChannelManager, so
// the PeerConnection must take care of creating/deleting the BaseChannel and
// setting the channel reference in the transceiver to null when it has been
// deleted.
// 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 final
: public rtc::RefCountedObject<RtpTransceiverInterface>,
public sigslot::has_slots<> {
// 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,
cricket::ChannelManager* channel_manager);
// 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().
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
cricket::ChannelManager* channel_manager,
std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer,
std::function<void()> on_negotiation_needed);
~RtpTransceiver() override;
// 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_; }
// Sets the Voice/VideoChannel. The caller must pass in the correct channel
// implementation based on the type of the transceiver.
void SetChannel(cricket::ChannelInterface* channel);
// 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.
senders() const {
return senders_;
// Adds an RtpReceiver of the appropriate type to be owned by this
// transceiver. Must not be null.
void AddReceiver(
// 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.
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).
void set_fired_direction(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);
void OnFirstPacketReceived();
void StopSendingAndReceiving();
// 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_;
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;
cricket::ChannelInterface* channel_ = nullptr;
cricket::ChannelManager* channel_manager_ = nullptr;
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_
const std::function<void()> on_negotiation_needed_;
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_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences)
rtc::ArrayView<const RtpHeaderExtensionCapability>)
} // namespace webrtc