blob: 924c764ccde2e6bddfcad5ceecc857c8646ec4c5 [file] [log] [blame]
Tomas Gunnarssonf25761d2020-06-03 20:55:331/*
2 * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
3 *
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 */
10
11#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
12#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_
13
14#include <memory>
15#include <string>
16#include <vector>
17
Ali Tofighd14e8892022-05-13 09:42:1618#include "absl/strings/string_view.h"
Tomas Gunnarssonf25761d2020-06-03 20:55:3319#include "absl/types/optional.h"
Jonas Orelande62c2f22022-03-29 09:04:4820#include "api/field_trials_view.h"
Tomas Gunnarssonf25761d2020-06-03 20:55:3321#include "api/frame_transformer_interface.h"
22#include "api/scoped_refptr.h"
Danil Chapovalov630c40d2023-07-17 14:42:4523#include "api/units/time_delta.h"
Tomas Gunnarssonf25761d2020-06-03 20:55:3324#include "api/video/video_bitrate_allocation.h"
25#include "modules/rtp_rtcp/include/receive_statistics.h"
26#include "modules/rtp_rtcp/include/report_block_data.h"
27#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
28#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
29#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
30#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
31#include "modules/rtp_rtcp/source/video_fec_generator.h"
Alessio Bazzicabc1c93d2021-03-12 16:45:2632#include "system_wrappers/include/ntp_time.h"
Tomas Gunnarssonf25761d2020-06-03 20:55:3333
34namespace webrtc {
35
36// Forward declarations.
37class FrameEncryptorInterface;
38class RateLimiter;
Tomas Gunnarssonf25761d2020-06-03 20:55:3339class RtcEventLog;
40class RTPSender;
41class Transport;
42class VideoBitrateAllocationObserver;
43
44class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
45 public:
46 struct Configuration {
47 Configuration() = default;
48 Configuration(Configuration&& rhs) = default;
49
Byoungchan Lee604fd2f2022-01-21 00:49:3950 Configuration(const Configuration&) = delete;
51 Configuration& operator=(const Configuration&) = delete;
52
Tomas Gunnarssonf25761d2020-06-03 20:55:3353 // True for a audio version of the RTP/RTCP module object false will create
54 // a video version.
55 bool audio = false;
56 bool receiver_only = false;
57
58 // The clock to use to read time. If nullptr then system clock will be used.
59 Clock* clock = nullptr;
60
61 ReceiveStatisticsProvider* receive_statistics = nullptr;
62
63 // Transport object that will be called when packets are ready to be sent
64 // out on the network.
65 Transport* outgoing_transport = nullptr;
66
67 // Called when the receiver requests an intra frame.
68 RtcpIntraFrameObserver* intra_frame_callback = nullptr;
69
70 // Called when the receiver sends a loss notification.
71 RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr;
72
Danil Chapovalov52518632023-05-09 15:11:4973 // Called when receive an RTCP message related to the link in general, e.g.
74 // bandwidth estimation related message.
75 NetworkLinkRtcpObserver* network_link_rtcp_observer = nullptr;
76
Tomas Gunnarssonf25761d2020-06-03 20:55:3377 NetworkStateEstimateObserver* network_state_estimate_observer = nullptr;
78 TransportFeedbackObserver* transport_feedback_callback = nullptr;
79 VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
80 RtcpRttStats* rtt_stats = nullptr;
81 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
82 // Called on receipt of RTCP report block from remote side.
Tomas Gunnarssonf25761d2020-06-03 20:55:3383 // TODO(bugs.webrtc.org/10679): Consider whether we want to use
84 // only getters or only callbacks. If we decide on getters, the
85 // ReportBlockDataObserver should also be removed in favor of
86 // GetLatestReportBlockData().
Tomas Gunnarssonf25761d2020-06-03 20:55:3387 RtcpCnameCallback* rtcp_cname_callback = nullptr;
88 ReportBlockDataObserver* report_block_data_observer = nullptr;
89
Tomas Gunnarssonf25761d2020-06-03 20:55:3390 // Spread any bursts of packets into smaller bursts to minimize packet loss.
91 RtpPacketSender* paced_sender = nullptr;
92
93 // Generates FEC packets.
94 // TODO(sprang): Wire up to RtpSenderEgress.
95 VideoFecGenerator* fec_generator = nullptr;
96
97 BitrateStatisticsObserver* send_bitrate_observer = nullptr;
Tomas Gunnarssonf25761d2020-06-03 20:55:3398 RtcEventLog* event_log = nullptr;
99 SendPacketObserver* send_packet_observer = nullptr;
100 RateLimiter* retransmission_rate_limiter = nullptr;
101 StreamDataCountersCallback* rtp_stats_callback = nullptr;
102
103 int rtcp_report_interval_ms = 0;
104
105 // Update network2 instead of pacer_exit field of video timing extension.
106 bool populate_network2_timestamp = false;
107
108 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer;
109
110 // E2EE Custom Video Frame Encryption
111 FrameEncryptorInterface* frame_encryptor = nullptr;
112 // Require all outgoing frames to be encrypted with a FrameEncryptor.
113 bool require_frame_encryption = false;
114
115 // Corresponds to extmap-allow-mixed in SDP negotiation.
116 bool extmap_allow_mixed = false;
117
118 // If true, the RTP sender will always annotate outgoing packets with
119 // MID and RID header extensions, if provided and negotiated.
120 // If false, the RTP sender will stop sending MID and RID header extensions,
121 // when it knows that the receiver is ready to demux based on SSRC. This is
122 // done by RTCP RR acking.
123 bool always_send_mid_and_rid = false;
124
Danil Chapovalov7bb93222023-03-24 16:10:31125 // If set, field trials are read from `field_trials`.
Jonas Orelande62c2f22022-03-29 09:04:48126 const FieldTrialsView* field_trials = nullptr;
Tomas Gunnarssonf25761d2020-06-03 20:55:33127
128 // SSRCs for media and retransmission, respectively.
Artem Titov913cfa72021-07-28 21:57:33129 // FlexFec SSRC is fetched from `flexfec_sender`.
Tomas Gunnarssonf25761d2020-06-03 20:55:33130 uint32_t local_media_ssrc = 0;
131 absl::optional<uint32_t> rtx_send_ssrc;
132
133 bool need_rtp_packet_infos = false;
134
Niels Möllerbe810cba2020-12-02 13:25:03135 // Estimate RTT as non-sender as described in
136 // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
137 bool non_sender_rtt_measurement = false;
Niels Mölleraf785d92022-05-31 08:45:41138
139 // If non-empty, sets the value for sending in the RID (and Repaired) RTP
140 // header extension. RIDs are used to identify an RTP stream if SSRCs are
141 // not negotiated. If the RID and Repaired RID extensions are not
142 // registered, the RID will not be sent.
143 std::string rid;
Markus Handellc8c4a282023-05-08 14:46:21144
145 // Enables send packet batching from the egress RTP sender.
146 bool enable_send_packet_batching = false;
Tomas Gunnarssonf25761d2020-06-03 20:55:33147 };
148
Alessio Bazzicabc1c93d2021-03-12 16:45:26149 // Stats for RTCP sender reports (SR) for a specific SSRC.
150 // Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1.
151 struct SenderReportStats {
Ivo Creusen2562cf02021-09-03 14:51:22152 // Arrival NTP timestamp for the last received RTCP SR.
Alessio Bazzicabc1c93d2021-03-12 16:45:26153 NtpTime last_arrival_timestamp;
154 // Received (a.k.a., remote) NTP timestamp for the last received RTCP SR.
155 NtpTime last_remote_timestamp;
Danil Chapovalov9f397212023-02-27 18:49:31156 // Received (a.k.a., remote) RTP timestamp from the last received RTCP SR.
Danil Chapovalov0f43da22023-02-28 16:03:21157 uint32_t last_remote_rtp_timestamp = 0;
Alessio Bazzicabc1c93d2021-03-12 16:45:26158 // Total number of RTP data packets transmitted by the sender since starting
159 // transmission up until the time this SR packet was generated. The count
160 // should be reset if the sender changes its SSRC identifier.
Danil Chapovalov0f43da22023-02-28 16:03:21161 uint32_t packets_sent = 0;
Alessio Bazzicabc1c93d2021-03-12 16:45:26162 // Total number of payload octets (i.e., not including header or padding)
163 // transmitted in RTP data packets by the sender since starting transmission
164 // up until the time this SR packet was generated. The count should be reset
165 // if the sender changes its SSRC identifier.
Danil Chapovalov0f43da22023-02-28 16:03:21166 uint64_t bytes_sent = 0;
Alessio Bazzicabc1c93d2021-03-12 16:45:26167 // Total number of RTCP SR blocks received.
168 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent.
Danil Chapovalov0f43da22023-02-28 16:03:21169 uint64_t reports_count = 0;
Alessio Bazzicabc1c93d2021-03-12 16:45:26170 };
Ivo Creusen2562cf02021-09-03 14:51:22171 // Stats about the non-sender SSRC, based on RTCP extended reports (XR).
172 // Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2.
173 struct NonSenderRttStats {
174 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
175 absl::optional<TimeDelta> round_trip_time;
176 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
177 TimeDelta total_round_trip_time = TimeDelta::Zero();
178 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
179 int round_trip_time_measurements = 0;
180 };
Alessio Bazzicabc1c93d2021-03-12 16:45:26181
Tomas Gunnarssonf25761d2020-06-03 20:55:33182 // **************************************************************************
183 // Receiver functions
184 // **************************************************************************
185
Harald Alvestrand1f206b82023-02-01 11:12:46186 virtual void IncomingRtcpPacket(
187 rtc::ArrayView<const uint8_t> incoming_packet) = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33188
189 virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
190
Tommi08be9ba2021-06-15 21:01:57191 // Called when the local ssrc changes (post initialization) for receive
192 // streams to match with send. Called on the packet receive thread/tq.
193 virtual void SetLocalSsrc(uint32_t ssrc) = 0;
194
Tomas Gunnarssonf25761d2020-06-03 20:55:33195 // **************************************************************************
196 // Sender
197 // **************************************************************************
198
199 // Sets the maximum size of an RTP packet, including RTP headers.
200 virtual void SetMaxRtpPacketSize(size_t size) = 0;
201
202 // Returns max RTP packet size. Takes into account RTP headers and
203 // FEC/ULP/RED overhead (when FEC is enabled).
204 virtual size_t MaxRtpPacketSize() const = 0;
205
206 virtual void RegisterSendPayloadFrequency(int payload_type,
207 int payload_frequency) = 0;
208
209 // Unregisters a send payload.
Artem Titov913cfa72021-07-28 21:57:33210 // `payload_type` - payload type of codec
Tomas Gunnarssonf25761d2020-06-03 20:55:33211 // Returns -1 on failure else 0.
212 virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0;
213
214 virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
215
216 // Register extension by uri, triggers CHECK on falure.
217 virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0;
218
Tomas Gunnarssonf25761d2020-06-03 20:55:33219 virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0;
220
221 // Returns true if RTP module is send media, and any of the extensions
222 // required for bandwidth estimation is registered.
223 virtual bool SupportsPadding() const = 0;
224 // Same as SupportsPadding(), but additionally requires that
225 // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option
226 // enabled.
227 virtual bool SupportsRtxPayloadPadding() const = 0;
228
229 // Returns start timestamp.
230 virtual uint32_t StartTimestamp() const = 0;
231
232 // Sets start timestamp. Start timestamp is set to a random value if this
233 // function is never called.
234 virtual void SetStartTimestamp(uint32_t timestamp) = 0;
235
236 // Returns SequenceNumber.
237 virtual uint16_t SequenceNumber() const = 0;
238
239 // Sets SequenceNumber, default is a random number.
240 virtual void SetSequenceNumber(uint16_t seq) = 0;
241
242 virtual void SetRtpState(const RtpState& rtp_state) = 0;
243 virtual void SetRtxState(const RtpState& rtp_state) = 0;
244 virtual RtpState GetRtpState() const = 0;
245 virtual RtpState GetRtxState() const = 0;
246
Ivo Creusen8c40d512021-07-13 12:53:22247 // This can be used to enable/disable receive-side RTT.
248 virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
249
Tomas Gunnarssonf25761d2020-06-03 20:55:33250 // Returns SSRC.
251 virtual uint32_t SSRC() const = 0;
252
Tomas Gunnarssonf25761d2020-06-03 20:55:33253 // Sets the value for sending in the MID RTP header extension.
254 // The MID RTP header extension should be registered for this to do anything.
255 // Once set, this value can not be changed or removed.
Ali Tofighd14e8892022-05-13 09:42:16256 virtual void SetMid(absl::string_view mid) = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33257
Tomas Gunnarssonf25761d2020-06-03 20:55:33258 // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination
259 // of values of the enumerator RtxMode.
260 virtual void SetRtxSendStatus(int modes) = 0;
261
262 // Returns status of sending RTX (RFC 4588). The returned value can be
263 // a combination of values of the enumerator RtxMode.
264 virtual int RtxSendStatus() const = 0;
265
266 // Returns the SSRC used for RTX if set, otherwise a nullopt.
267 virtual absl::optional<uint32_t> RtxSsrc() const = 0;
268
269 // Sets the payload type to use when sending RTX packets. Note that this
270 // doesn't enable RTX, only the payload type is set.
271 virtual void SetRtxSendPayloadType(int payload_type,
272 int associated_payload_type) = 0;
273
274 // Returns the FlexFEC SSRC, if there is one.
275 virtual absl::optional<uint32_t> FlexfecSsrc() const = 0;
276
Philipp Hancke8602f602022-08-30 17:53:47277 // Sets sending status.
Tomas Gunnarssonf25761d2020-06-03 20:55:33278 // Returns -1 on failure else 0.
279 virtual int32_t SetSendingStatus(bool sending) = 0;
280
281 // Returns current sending status.
282 virtual bool Sending() const = 0;
283
284 // Starts/Stops media packets. On by default.
285 virtual void SetSendingMediaStatus(bool sending) = 0;
286
287 // Returns current media sending status.
288 virtual bool SendingMedia() const = 0;
289
290 // Returns whether audio is configured (i.e. Configuration::audio = true).
291 virtual bool IsAudioConfigured() const = 0;
292
293 // Indicate that the packets sent by this module should be counted towards the
294 // bitrate estimate since the stream participates in the bitrate allocation.
295 virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;
296
Tomas Gunnarssonf25761d2020-06-03 20:55:33297 // Returns bitrate sent (post-pacing) per packet type.
298 virtual RtpSendRates GetSendRates() const = 0;
299
300 virtual RTPSender* RtpSender() = 0;
301 virtual const RTPSender* RtpSender() const = 0;
302
303 // Record that a frame is about to be sent. Returns true on success, and false
304 // if the module isn't ready to send.
305 virtual bool OnSendingRtpFrame(uint32_t timestamp,
306 int64_t capture_time_ms,
307 int payload_type,
308 bool force_sender_report) = 0;
309
310 // Try to send the provided packet. Returns true iff packet matches any of
311 // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the
312 // transport.
Markus Handellcb838e22023-05-05 12:41:30313 virtual bool TrySendPacket(std::unique_ptr<RtpPacketToSend> packet,
Tomas Gunnarssonf25761d2020-06-03 20:55:33314 const PacedPacketInfo& pacing_info) = 0;
315
Markus Handellc8c4a282023-05-08 14:46:21316 // Notifies that a batch of packet sends is completed. The implementation can
317 // use this to optimize packet sending.
318 virtual void OnBatchComplete() = 0;
319
Erik Språng1d50cb62020-07-02 15:41:32320 // Update the FEC protection parameters to use for delta- and key-frames.
321 // Only used when deferred FEC is active.
322 virtual void SetFecProtectionParams(
323 const FecProtectionParams& delta_params,
324 const FecProtectionParams& key_params) = 0;
325
326 // If deferred FEC generation is enabled, this method should be called after
327 // calling TrySendPacket(). Any generated FEC packets will be removed and
328 // returned from the FEC generator.
329 virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0;
330
Erik Språng50459492022-09-08 14:53:06331 virtual void OnAbortedRetransmissions(
332 rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
333
Tomas Gunnarssonf25761d2020-06-03 20:55:33334 virtual void OnPacketsAcknowledged(
335 rtc::ArrayView<const uint16_t> sequence_numbers) = 0;
336
337 virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
338 size_t target_size_bytes) = 0;
339
340 virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
341 rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
342
343 // Returns an expected per packet overhead representing the main RTP header,
344 // any CSRCs, and the registered header extensions that are expected on all
345 // packets (i.e. disregarding things like abs capture time which is only
346 // populated on a subset of packets, but counting MID/RID type extensions
347 // when we expect to send them).
348 virtual size_t ExpectedPerPacketOverhead() const = 0;
349
Erik Språngb6bbdeb2021-08-13 14:12:41350 // Access to packet state (e.g. sequence numbering) must only be access by
351 // one thread at a time. It may be only one thread, or a construction thread
352 // that calls SetRtpState() - handing over to a pacer thread that calls
353 // TrySendPacket() - and at teardown ownership is handed to a destruciton
354 // thread that calls GetRtpState().
355 // This method is used to signal that "ownership" of the rtp state is being
356 // transferred to another thread.
357 virtual void OnPacketSendingThreadSwitched() = 0;
358
Tomas Gunnarssonf25761d2020-06-03 20:55:33359 // **************************************************************************
360 // RTCP
361 // **************************************************************************
362
363 // Returns RTCP status.
364 virtual RtcpMode RTCP() const = 0;
365
366 // Sets RTCP status i.e on(compound or non-compound)/off.
Artem Titov913cfa72021-07-28 21:57:33367 // `method` - RTCP method to use.
Tomas Gunnarssonf25761d2020-06-03 20:55:33368 virtual void SetRTCPStatus(RtcpMode method) = 0;
369
370 // Sets RTCP CName (i.e unique identifier).
371 // Returns -1 on failure else 0.
Ali Tofighd14e8892022-05-13 09:42:16372 virtual int32_t SetCNAME(absl::string_view cname) = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33373
Tomas Gunnarssonf25761d2020-06-03 20:55:33374 // Returns current RTT (round-trip time) estimate.
Danil Chapovalov8095d022023-05-09 07:59:46375 virtual absl::optional<TimeDelta> LastRtt() const = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33376
377 // Returns the estimated RTT, with fallback to a default value.
Danil Chapovalov630c40d2023-07-17 14:42:45378 virtual TimeDelta ExpectedRetransmissionTime() const = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33379
380 // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the
381 // process function.
382 // Returns -1 on failure else 0.
383 virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0;
384
Tomas Gunnarssonf25761d2020-06-03 20:55:33385 // Returns send statistics for the RTP and RTX stream.
386 virtual void GetSendStreamDataCounters(
387 StreamDataCounters* rtp_counters,
388 StreamDataCounters* rtx_counters) const = 0;
389
Tomas Gunnarssonf25761d2020-06-03 20:55:33390 // A snapshot of Report Blocks with additional data of interest to statistics.
391 // Within this list, the sender-source SSRC pair is unique and per-pair the
392 // ReportBlockData represents the latest Report Block that was received for
393 // that pair.
394 virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0;
Alessio Bazzicabc1c93d2021-03-12 16:45:26395 // Returns stats based on the received RTCP SRs.
396 virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0;
Ivo Creusen2562cf02021-09-03 14:51:22397 // Returns non-sender RTT stats, based on DLRR.
398 virtual absl::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0;
Tomas Gunnarssonf25761d2020-06-03 20:55:33399
Tomas Gunnarssonf25761d2020-06-03 20:55:33400 // (REMB) Receiver Estimated Max Bitrate.
401 // Schedules sending REMB on next and following sender/receiver reports.
402 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0;
403 // Stops sending REMB on next and following sender/receiver reports.
404 void UnsetRemb() override = 0;
405
Tomas Gunnarssonf25761d2020-06-03 20:55:33406 // (NACK)
407
408 // Sends a Negative acknowledgement packet.
409 // Returns -1 on failure else 0.
410 // TODO(philipel): Deprecate this and start using SendNack instead, mostly
411 // because we want a function that actually send NACK for the specified
412 // packets.
413 virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0;
414
415 // Sends NACK for the packets specified.
416 // Note: This assumes the caller keeps track of timing and doesn't rely on
417 // the RTP module to do this.
418 virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
419
420 // Store the sent packets, needed to answer to a Negative acknowledgment
421 // requests.
422 virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
423
Tomas Gunnarssonf25761d2020-06-03 20:55:33424 virtual void SetVideoBitrateAllocation(
425 const VideoBitrateAllocation& bitrate) = 0;
426
427 // **************************************************************************
428 // Video
429 // **************************************************************************
430
431 // Requests new key frame.
432 // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1
433 void SendPictureLossIndication() { SendRTCP(kRtcpPli); }
434 // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2
435 void SendFullIntraRequest() { SendRTCP(kRtcpFir); }
436
437 // Sends a LossNotification RTCP message.
438 // Returns -1 on failure else 0.
439 virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num,
440 uint16_t last_received_seq_num,
441 bool decodability_flag,
442 bool buffering_allowed) = 0;
443};
444
445} // namespace webrtc
446
447#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_