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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.
*/
#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/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:
// 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.
// `HeaderExtensionsToNegotiate` is used for initializing the return value of
// HeaderExtensionsToNegotiate().
RtpTransceiver(
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver,
ConnectionContext* context,
std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToNegotiate,
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();
// 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> GetHeaderExtensionsToNegotiate()
const override;
std::vector<RtpHeaderExtensionCapability> GetNegotiatedHeaderExtensions()
const override;
RTCError SetHeaderExtensionsToNegotiate(
rtc::ArrayView<const RtpHeaderExtensionCapability> header_extensions)
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_negotiate_;
// `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>,
GetHeaderExtensionsToNegotiate)
PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
GetNegotiatedHeaderExtensions)
PROXY_METHOD1(webrtc::RTCError,
SetHeaderExtensionsToNegotiate,
rtc::ArrayView<const RtpHeaderExtensionCapability>)
END_PROXY_MAP(RtpTransceiver)
} // namespace webrtc
#endif // PC_RTP_TRANSCEIVER_H_