Merge mozilla-central to autoland. a=merge CLOSED TREE
[gecko.git] / third_party / libwebrtc / pc / rtp_transceiver.h
blob88febb94295e5233e82ab1266243a97b388d530f
1 /*
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved.
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
11 #ifndef PC_RTP_TRANSCEIVER_H_
12 #define PC_RTP_TRANSCEIVER_H_
14 #include <stddef.h>
16 #include <functional>
17 #include <memory>
18 #include <string>
19 #include <vector>
21 #include "absl/strings/string_view.h"
22 #include "absl/types/optional.h"
23 #include "api/array_view.h"
24 #include "api/audio_options.h"
25 #include "api/crypto/crypto_options.h"
26 #include "api/jsep.h"
27 #include "api/media_types.h"
28 #include "api/rtc_error.h"
29 #include "api/rtp_parameters.h"
30 #include "api/rtp_receiver_interface.h"
31 #include "api/rtp_sender_interface.h"
32 #include "api/rtp_transceiver_direction.h"
33 #include "api/rtp_transceiver_interface.h"
34 #include "api/scoped_refptr.h"
35 #include "api/task_queue/pending_task_safety_flag.h"
36 #include "api/task_queue/task_queue_base.h"
37 #include "api/video/video_bitrate_allocator_factory.h"
38 #include "media/base/media_channel.h"
39 #include "media/base/media_config.h"
40 #include "media/base/media_engine.h"
41 #include "pc/channel_interface.h"
42 #include "pc/connection_context.h"
43 #include "pc/proxy.h"
44 #include "pc/rtp_receiver.h"
45 #include "pc/rtp_receiver_proxy.h"
46 #include "pc/rtp_sender.h"
47 #include "pc/rtp_sender_proxy.h"
48 #include "pc/rtp_transport_internal.h"
49 #include "pc/session_description.h"
50 #include "rtc_base/thread_annotations.h"
52 namespace cricket {
53 class MediaEngineInterface;
56 namespace webrtc {
58 class PeerConnectionSdpMethods;
60 // Implementation of the public RtpTransceiverInterface.
62 // The RtpTransceiverInterface is only intended to be used with a PeerConnection
63 // that enables Unified Plan SDP. Thus, the methods that only need to implement
64 // public API features and are not used internally can assume exactly one sender
65 // and receiver.
67 // Since the RtpTransceiver is used internally by PeerConnection for tracking
68 // RtpSenders, RtpReceivers, and BaseChannels, and PeerConnection needs to be
69 // backwards compatible with Plan B SDP, this implementation is more flexible
70 // than that required by the WebRTC specification.
72 // With Plan B SDP, an RtpTransceiver can have any number of senders and
73 // receivers which map to a=ssrc lines in the m= section.
74 // With Unified Plan SDP, an RtpTransceiver will have exactly one sender and one
75 // receiver which are encapsulated by the m= section.
77 // This class manages the RtpSenders, RtpReceivers, and BaseChannel associated
78 // with this m= section. Since the transceiver, senders, and receivers are
79 // reference counted and can be referenced from JavaScript (in Chromium), these
80 // objects must be ready to live for an arbitrary amount of time. The
81 // BaseChannel is not reference counted, so
82 // the PeerConnection must take care of creating/deleting the BaseChannel.
84 // The RtpTransceiver is specialized to either audio or video according to the
85 // MediaType specified in the constructor. Audio RtpTransceivers will have
86 // AudioRtpSenders, AudioRtpReceivers, and a VoiceChannel. Video RtpTransceivers
87 // will have VideoRtpSenders, VideoRtpReceivers, and a VideoChannel.
88 class RtpTransceiver : public RtpTransceiverInterface {
89 public:
90 // Construct a Plan B-style RtpTransceiver with no senders, receivers, or
91 // channel set.
92 // `media_type` specifies the type of RtpTransceiver (and, by transitivity,
93 // the type of senders, receivers, and channel). Can either by audio or video.
94 RtpTransceiver(cricket::MediaType media_type, ConnectionContext* context);
95 // Construct a Unified Plan-style RtpTransceiver with the given sender and
96 // receiver. The media type will be derived from the media types of the sender
97 // and receiver. The sender and receiver should have the same media type.
98 // `HeaderExtensionsToNegotiate` is used for initializing the return value of
99 // HeaderExtensionsToNegotiate().
100 RtpTransceiver(
101 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
102 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
103 receiver,
104 ConnectionContext* context,
105 std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToNegotiate,
106 std::function<void()> on_negotiation_needed);
107 ~RtpTransceiver() override;
109 // Not copyable or movable.
110 RtpTransceiver(const RtpTransceiver&) = delete;
111 RtpTransceiver& operator=(const RtpTransceiver&) = delete;
112 RtpTransceiver(RtpTransceiver&&) = delete;
113 RtpTransceiver& operator=(RtpTransceiver&&) = delete;
115 // Returns the Voice/VideoChannel set for this transceiver. May be null if
116 // the transceiver is not in the currently set local/remote description.
117 cricket::ChannelInterface* channel() const { return channel_.get(); }
119 // Creates the Voice/VideoChannel and sets it.
120 RTCError CreateChannel(
121 absl::string_view mid,
122 Call* call_ptr,
123 const cricket::MediaConfig& media_config,
124 bool srtp_required,
125 CryptoOptions crypto_options,
126 const cricket::AudioOptions& audio_options,
127 const cricket::VideoOptions& video_options,
128 VideoBitrateAllocatorFactory* video_bitrate_allocator_factory,
129 std::function<RtpTransportInternal*(absl::string_view)> transport_lookup);
131 // Sets the Voice/VideoChannel. The caller must pass in the correct channel
132 // implementation based on the type of the transceiver. The call must
133 // furthermore be made on the signaling thread.
135 // `channel`: The channel instance to be associated with the transceiver.
136 // This must be a valid pointer.
137 // The state of the object
138 // is expected to be newly constructed and not initalized for network
139 // activity (see next parameter for more).
141 // The transceiver takes ownership of `channel`.
143 // `transport_lookup`: This
144 // callback function will be used to look up the `RtpTransport` object
145 // to associate with the channel via `BaseChannel::SetRtpTransport`.
146 // The lookup function will be called on the network thread, synchronously
147 // during the call to `SetChannel`. This means that the caller of
148 // `SetChannel()` may provide a callback function that references state
149 // that exists within the calling scope of SetChannel (e.g. a variable
150 // on the stack).
151 // The reason for this design is to limit the number of times we jump
152 // synchronously to the network thread from the signaling thread.
153 // The callback allows us to combine the transport lookup with network
154 // state initialization of the channel object.
155 // ClearChannel() must be used before calling SetChannel() again.
156 void SetChannel(std::unique_ptr<cricket::ChannelInterface> channel,
157 std::function<RtpTransportInternal*(const std::string&)>
158 transport_lookup);
160 // Clear the association between the transceiver and the channel.
161 void ClearChannel();
163 // Adds an RtpSender of the appropriate type to be owned by this transceiver.
164 // Must not be null.
165 void AddSender(
166 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender);
168 // Removes the given RtpSender. Returns false if the sender is not owned by
169 // this transceiver.
170 bool RemoveSender(RtpSenderInterface* sender);
172 // Returns a vector of the senders owned by this transceiver.
173 std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
174 senders() const {
175 return senders_;
178 // Adds an RtpReceiver of the appropriate type to be owned by this
179 // transceiver. Must not be null.
180 void AddReceiver(
181 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
182 receiver);
184 // Removes the given RtpReceiver. Returns false if the sender is not owned by
185 // this transceiver.
186 bool RemoveReceiver(RtpReceiverInterface* receiver);
188 // Returns a vector of the receivers owned by this transceiver.
189 std::vector<
190 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
191 receivers() const {
192 return receivers_;
195 // Returns the backing object for the transceiver's Unified Plan sender.
196 rtc::scoped_refptr<RtpSenderInternal> sender_internal() const;
198 // Returns the backing object for the transceiver's Unified Plan receiver.
199 rtc::scoped_refptr<RtpReceiverInternal> receiver_internal() const;
201 // RtpTransceivers are not associated until they have a corresponding media
202 // section set in SetLocalDescription or SetRemoteDescription. Therefore,
203 // when setting a local offer we need a way to remember which transceiver was
204 // used to create which media section in the offer. Storing the mline index
205 // in CreateOffer is specified in JSEP to allow us to do that.
206 absl::optional<size_t> mline_index() const { return mline_index_; }
207 void set_mline_index(absl::optional<size_t> mline_index) {
208 mline_index_ = mline_index;
211 // Sets the MID for this transceiver. If the MID is not null, then the
212 // transceiver is considered "associated" with the media section that has the
213 // same MID.
214 void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; }
216 // Sets the intended direction for this transceiver. Intended to be used
217 // internally over SetDirection since this does not trigger a negotiation
218 // needed callback.
219 void set_direction(RtpTransceiverDirection direction) {
220 direction_ = direction;
223 // Sets the current direction for this transceiver as negotiated in an offer/
224 // answer exchange. The current direction is null before an answer with this
225 // transceiver has been set.
226 void set_current_direction(RtpTransceiverDirection direction);
228 // Sets the fired direction for this transceiver. The fired direction is null
229 // until SetRemoteDescription is called or an answer is set (either local or
230 // remote) after which the only valid reason to go back to null is rollback.
231 void set_fired_direction(absl::optional<RtpTransceiverDirection> direction);
233 // According to JSEP rules for SetRemoteDescription, RtpTransceivers can be
234 // reused only if they were added by AddTrack.
235 void set_created_by_addtrack(bool created_by_addtrack) {
236 created_by_addtrack_ = created_by_addtrack;
238 // If AddTrack has been called then transceiver can't be removed during
239 // rollback.
240 void set_reused_for_addtrack(bool reused_for_addtrack) {
241 reused_for_addtrack_ = reused_for_addtrack;
244 bool created_by_addtrack() const { return created_by_addtrack_; }
246 bool reused_for_addtrack() const { return reused_for_addtrack_; }
248 // Returns true if this transceiver has ever had the current direction set to
249 // sendonly or sendrecv.
250 bool has_ever_been_used_to_send() const {
251 return has_ever_been_used_to_send_;
254 // Informs the transceiver that its owning
255 // PeerConnection is closed.
256 void SetPeerConnectionClosed();
258 // Executes the "stop the RTCRtpTransceiver" procedure from
259 // the webrtc-pc specification, described under the stop() method.
260 void StopTransceiverProcedure();
262 // RtpTransceiverInterface implementation.
263 cricket::MediaType media_type() const override;
264 absl::optional<std::string> mid() const override;
265 rtc::scoped_refptr<RtpSenderInterface> sender() const override;
266 rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
267 bool stopped() const override;
268 bool stopping() const override;
269 RtpTransceiverDirection direction() const override;
270 RTCError SetDirectionWithError(
271 RtpTransceiverDirection new_direction) override;
272 absl::optional<RtpTransceiverDirection> current_direction() const override;
273 absl::optional<RtpTransceiverDirection> fired_direction() const override;
274 RTCError StopStandard() override;
275 void StopInternal() override;
276 RTCError SetCodecPreferences(
277 rtc::ArrayView<RtpCodecCapability> codecs) override;
278 std::vector<RtpCodecCapability> codec_preferences() const override {
279 return codec_preferences_;
281 std::vector<RtpHeaderExtensionCapability> GetHeaderExtensionsToNegotiate()
282 const override;
283 std::vector<RtpHeaderExtensionCapability> GetNegotiatedHeaderExtensions()
284 const override;
285 RTCError SetHeaderExtensionsToNegotiate(
286 rtc::ArrayView<const RtpHeaderExtensionCapability> header_extensions)
287 override;
289 // Called on the signaling thread when the local or remote content description
290 // is updated. Used to update the negotiated header extensions.
291 // TODO(tommi): The implementation of this method is currently very simple and
292 // only used for updating the negotiated headers. However, we're planning to
293 // move all the updates done on the channel from the transceiver into this
294 // method. This will happen with the ownership of the channel object being
295 // moved into the transceiver.
296 void OnNegotiationUpdate(SdpType sdp_type,
297 const cricket::MediaContentDescription* content);
299 private:
300 cricket::MediaEngineInterface* media_engine() const {
301 return context_->media_engine();
303 ConnectionContext* context() const { return context_; }
304 void OnFirstPacketReceived();
305 void StopSendingAndReceiving();
306 // Delete a channel, and ensure that references to its media channel
307 // are updated before deleting it.
308 void PushNewMediaChannelAndDeleteChannel(
309 std::unique_ptr<cricket::ChannelInterface> channel_to_delete);
311 // Enforce that this object is created, used and destroyed on one thread.
312 TaskQueueBase* const thread_;
313 const bool unified_plan_;
314 const cricket::MediaType media_type_;
315 rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_thread_safety_;
316 std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
317 senders_;
318 std::vector<
319 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
320 receivers_;
322 bool stopped_ RTC_GUARDED_BY(thread_) = false;
323 bool stopping_ RTC_GUARDED_BY(thread_) = false;
324 bool is_pc_closed_ = false;
325 RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive;
326 absl::optional<RtpTransceiverDirection> current_direction_;
327 absl::optional<RtpTransceiverDirection> fired_direction_;
328 absl::optional<std::string> mid_;
329 absl::optional<size_t> mline_index_;
330 bool created_by_addtrack_ = false;
331 bool reused_for_addtrack_ = false;
332 bool has_ever_been_used_to_send_ = false;
334 // Accessed on both thread_ and the network thread. Considered safe
335 // because all access on the network thread is within an invoke()
336 // from thread_.
337 std::unique_ptr<cricket::ChannelInterface> channel_ = nullptr;
338 ConnectionContext* const context_;
339 std::vector<RtpCodecCapability> codec_preferences_;
340 std::vector<RtpHeaderExtensionCapability> header_extensions_to_negotiate_;
342 // `negotiated_header_extensions_` is read and written to on the signaling
343 // thread from the SdpOfferAnswerHandler class (e.g.
344 // PushdownMediaDescription().
345 cricket::RtpHeaderExtensions negotiated_header_extensions_
346 RTC_GUARDED_BY(thread_);
348 const std::function<void()> on_negotiation_needed_;
351 BEGIN_PRIMARY_PROXY_MAP(RtpTransceiver)
353 PROXY_PRIMARY_THREAD_DESTRUCTOR()
354 BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
355 PROXY_CONSTMETHOD0(absl::optional<std::string>, mid)
356 PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender)
357 PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver)
358 PROXY_CONSTMETHOD0(bool, stopped)
359 PROXY_CONSTMETHOD0(bool, stopping)
360 PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction)
361 PROXY_METHOD1(RTCError, SetDirectionWithError, RtpTransceiverDirection)
362 PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction)
363 PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction)
364 PROXY_METHOD0(RTCError, StopStandard)
365 PROXY_METHOD0(void, StopInternal)
366 PROXY_METHOD1(RTCError, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>)
367 PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences)
368 PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
369 GetHeaderExtensionsToNegotiate)
370 PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
371 GetNegotiatedHeaderExtensions)
372 PROXY_METHOD1(RTCError,
373 SetHeaderExtensionsToNegotiate,
374 rtc::ArrayView<const RtpHeaderExtensionCapability>)
375 END_PROXY_MAP(RtpTransceiver)
377 } // namespace webrtc
379 #endif // PC_RTP_TRANSCEIVER_H_