1 /*
2 * Copyright (c) 2012 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 #include "audio/channel_send.h"
12
13 #include <algorithm>
14 #include <map>
15 #include <memory>
16 #include <string>
17 #include <utility>
18 #include <vector>
19
20 #include "api/array_view.h"
21 #include "api/call/transport.h"
22 #include "api/crypto/frame_encryptor_interface.h"
23 #include "api/rtc_event_log/rtc_event_log.h"
24 #include "api/sequence_checker.h"
25 #include "audio/channel_send_frame_transformer_delegate.h"
26 #include "audio/utility/audio_frame_operations.h"
27 #include "call/rtp_transport_controller_send_interface.h"
28 #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
29 #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
30 #include "modules/audio_coding/include/audio_coding_module.h"
31 #include "modules/audio_processing/rms_level.h"
32 #include "modules/pacing/packet_router.h"
33 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
34 #include "modules/utility/include/process_thread.h"
35 #include "rtc_base/checks.h"
36 #include "rtc_base/event.h"
37 #include "rtc_base/format_macros.h"
38 #include "rtc_base/location.h"
39 #include "rtc_base/logging.h"
40 #include "rtc_base/numerics/safe_conversions.h"
41 #include "rtc_base/race_checker.h"
42 #include "rtc_base/rate_limiter.h"
43 #include "rtc_base/synchronization/mutex.h"
44 #include "rtc_base/task_queue.h"
45 #include "rtc_base/time_utils.h"
46 #include "system_wrappers/include/clock.h"
47 #include "system_wrappers/include/field_trial.h"
48 #include "system_wrappers/include/metrics.h"
49
50 namespace webrtc {
51 namespace voe {
52
53 namespace {
54
55 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
56 constexpr int64_t kMinRetransmissionWindowMs = 30;
57
58 class RtpPacketSenderProxy;
59 class TransportSequenceNumberProxy;
60 class VoERtcpObserver;
61
62 class ChannelSend : public ChannelSendInterface,
63 public AudioPacketizationCallback { // receive encoded
64 // packets from the ACM
65 public:
66 // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
67 // declaration.
68 friend class VoERtcpObserver;
69
70 ChannelSend(Clock* clock,
71 TaskQueueFactory* task_queue_factory,
72 ProcessThread* module_process_thread,
73 Transport* rtp_transport,
74 RtcpRttStats* rtcp_rtt_stats,
75 RtcEventLog* rtc_event_log,
76 FrameEncryptorInterface* frame_encryptor,
77 const webrtc::CryptoOptions& crypto_options,
78 bool extmap_allow_mixed,
79 int rtcp_report_interval_ms,
80 uint32_t ssrc,
81 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
82 TransportFeedbackObserver* feedback_observer);
83
84 ~ChannelSend() override;
85
86 // Send using this encoder, with this payload type.
87 void SetEncoder(int payload_type,
88 std::unique_ptr<AudioEncoder> encoder) override;
89 void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
90 modifier) override;
91 void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) override;
92
93 // API methods
94 void StartSend() override;
95 void StopSend() override;
96
97 // Codecs
98 void OnBitrateAllocation(BitrateAllocationUpdate update) override;
99 int GetBitrate() const override;
100
101 // Network
102 void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
103
104 // Muting, Volume and Level.
105 void SetInputMute(bool enable) override;
106
107 // Stats.
108 ANAStats GetANAStatistics() const override;
109
110 // Used by AudioSendStream.
111 RtpRtcpInterface* GetRtpRtcp() const override;
112
113 void RegisterCngPayloadType(int payload_type, int payload_frequency) override;
114
115 // DTMF.
116 bool SendTelephoneEventOutband(int event, int duration_ms) override;
117 void SetSendTelephoneEventPayloadType(int payload_type,
118 int payload_frequency) override;
119
120 // RTP+RTCP
121 void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
122
123 void RegisterSenderCongestionControlObjects(
124 RtpTransportControllerSendInterface* transport,
125 RtcpBandwidthObserver* bandwidth_observer) override;
126 void ResetSenderCongestionControlObjects() override;
127 void SetRTCP_CNAME(absl::string_view c_name) override;
128 std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const override;
129 CallSendStatistics GetRTCPStatistics() const override;
130
131 // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
132 // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
133 // the actual processing of the audio takes place. The processing mainly
134 // consists of encoding and preparing the result for sending by adding it to a
135 // send queue.
136 // The main reason for using a task queue here is to release the native,
137 // OS-specific, audio capture thread as soon as possible to ensure that it
138 // can go back to sleep and be prepared to deliver an new captured audio
139 // packet.
140 void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
141
142 int64_t GetRTT() const override;
143
144 // E2EE Custom Audio Frame Encryption
145 void SetFrameEncryptor(
146 rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
147
148 // Sets a frame transformer between encoder and packetizer, to transform
149 // encoded frames before sending them out the network.
150 void SetEncoderToPacketizerFrameTransformer(
151 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
152 override;
153
154 private:
155 // From AudioPacketizationCallback in the ACM
156 int32_t SendData(AudioFrameType frameType,
157 uint8_t payloadType,
158 uint32_t rtp_timestamp,
159 const uint8_t* payloadData,
160 size_t payloadSize,
161 int64_t absolute_capture_timestamp_ms) override;
162
163 void OnUplinkPacketLossRate(float packet_loss_rate);
164 bool InputMute() const;
165
166 int32_t SendRtpAudio(AudioFrameType frameType,
167 uint8_t payloadType,
168 uint32_t rtp_timestamp,
169 rtc::ArrayView<const uint8_t> payload,
170 int64_t absolute_capture_timestamp_ms)
171 RTC_RUN_ON(encoder_queue_);
172
173 void OnReceivedRtt(int64_t rtt_ms);
174
175 void InitFrameTransformerDelegate(
176 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
177
178 // Thread checkers document and lock usage of some methods on voe::Channel to
179 // specific threads we know about. The goal is to eventually split up
180 // voe::Channel into parts with single-threaded semantics, and thereby reduce
181 // the need for locks.
182 SequenceChecker worker_thread_checker_;
183 SequenceChecker module_process_thread_checker_;
184 // Methods accessed from audio and video threads are checked for sequential-
185 // only access. We don't necessarily own and control these threads, so thread
186 // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
187 // audio thread to another, but access is still sequential.
188 rtc::RaceChecker audio_thread_race_checker_;
189
190 mutable Mutex volume_settings_mutex_;
191
192 bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
193
194 RtcEventLog* const event_log_;
195
196 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
197 std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
198
199 std::unique_ptr<AudioCodingModule> audio_coding_;
200 uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
201
202 // uses
203 ProcessThread* const _moduleProcessThreadPtr;
204 RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
205 bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_);
206 bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
207 // VoeRTP_RTCP
208 // TODO(henrika): can today be accessed on the main thread and on the
209 // task queue; hence potential race.
210 bool _includeAudioLevelIndication;
211
212 // RtcpBandwidthObserver
213 const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
214
215 PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
216 nullptr;
217 TransportFeedbackObserver* const feedback_observer_;
218 const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
219 const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
220
221 SequenceChecker construction_thread_;
222
223 bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
224
225 // E2EE Audio Frame Encryption
226 rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
227 RTC_GUARDED_BY(encoder_queue_);
228 // E2EE Frame Encryption Options
229 const webrtc::CryptoOptions crypto_options_;
230
231 // Delegates calls to a frame transformer to transform audio, and
232 // receives callbacks with the transformed frames; delegates calls to
233 // ChannelSend::SendRtpAudio to send the transformed audio.
234 rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
235 frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_);
236
237 mutable Mutex bitrate_mutex_;
238 int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_mutex_) = 0;
239
240 // Defined last to ensure that there are no running tasks when the other
241 // members are destroyed.
242 rtc::TaskQueue encoder_queue_;
243
244 const bool fixing_timestamp_stall_;
245 };
246
247 const int kTelephoneEventAttenuationdB = 10;
248
249 class RtpPacketSenderProxy : public RtpPacketSender {
250 public:
RtpPacketSenderProxy()251 RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
252
SetPacketPacer(RtpPacketSender * rtp_packet_pacer)253 void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
254 RTC_DCHECK(thread_checker_.IsCurrent());
255 MutexLock lock(&mutex_);
256 rtp_packet_pacer_ = rtp_packet_pacer;
257 }
258
EnqueuePackets(std::vector<std::unique_ptr<RtpPacketToSend>> packets)259 void EnqueuePackets(
260 std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
261 MutexLock lock(&mutex_);
262 rtp_packet_pacer_->EnqueuePackets(std::move(packets));
263 }
264
265 private:
266 SequenceChecker thread_checker_;
267 Mutex mutex_;
268 RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
269 };
270
271 class VoERtcpObserver : public RtcpBandwidthObserver {
272 public:
VoERtcpObserver(ChannelSend * owner)273 explicit VoERtcpObserver(ChannelSend* owner)
274 : owner_(owner), bandwidth_observer_(nullptr) {}
~VoERtcpObserver()275 ~VoERtcpObserver() override {}
276
SetBandwidthObserver(RtcpBandwidthObserver * bandwidth_observer)277 void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) {
278 MutexLock lock(&mutex_);
279 bandwidth_observer_ = bandwidth_observer;
280 }
281
OnReceivedEstimatedBitrate(uint32_t bitrate)282 void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
283 MutexLock lock(&mutex_);
284 if (bandwidth_observer_) {
285 bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
286 }
287 }
288
OnReceivedRtcpReceiverReport(const ReportBlockList & report_blocks,int64_t rtt,int64_t now_ms)289 void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
290 int64_t rtt,
291 int64_t now_ms) override {
292 {
293 MutexLock lock(&mutex_);
294 if (bandwidth_observer_) {
295 bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt,
296 now_ms);
297 }
298 }
299 // TODO(mflodman): Do we need to aggregate reports here or can we jut send
300 // what we get? I.e. do we ever get multiple reports bundled into one RTCP
301 // report for VoiceEngine?
302 if (report_blocks.empty())
303 return;
304
305 int fraction_lost_aggregate = 0;
306 int total_number_of_packets = 0;
307
308 // If receiving multiple report blocks, calculate the weighted average based
309 // on the number of packets a report refers to.
310 for (ReportBlockList::const_iterator block_it = report_blocks.begin();
311 block_it != report_blocks.end(); ++block_it) {
312 // Find the previous extended high sequence number for this remote SSRC,
313 // to calculate the number of RTP packets this report refers to. Ignore if
314 // we haven't seen this SSRC before.
315 std::map<uint32_t, uint32_t>::iterator seq_num_it =
316 extended_max_sequence_number_.find(block_it->source_ssrc);
317 int number_of_packets = 0;
318 if (seq_num_it != extended_max_sequence_number_.end()) {
319 number_of_packets =
320 block_it->extended_highest_sequence_number - seq_num_it->second;
321 }
322 fraction_lost_aggregate += number_of_packets * block_it->fraction_lost;
323 total_number_of_packets += number_of_packets;
324
325 extended_max_sequence_number_[block_it->source_ssrc] =
326 block_it->extended_highest_sequence_number;
327 }
328 int weighted_fraction_lost = 0;
329 if (total_number_of_packets > 0) {
330 weighted_fraction_lost =
331 (fraction_lost_aggregate + total_number_of_packets / 2) /
332 total_number_of_packets;
333 }
334 owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f);
335 }
336
337 private:
338 ChannelSend* owner_;
339 // Maps remote side ssrc to extended highest sequence number received.
340 std::map<uint32_t, uint32_t> extended_max_sequence_number_;
341 Mutex mutex_;
342 RtcpBandwidthObserver* bandwidth_observer_ RTC_GUARDED_BY(mutex_);
343 };
344
SendData(AudioFrameType frameType,uint8_t payloadType,uint32_t rtp_timestamp,const uint8_t * payloadData,size_t payloadSize,int64_t absolute_capture_timestamp_ms)345 int32_t ChannelSend::SendData(AudioFrameType frameType,
346 uint8_t payloadType,
347 uint32_t rtp_timestamp,
348 const uint8_t* payloadData,
349 size_t payloadSize,
350 int64_t absolute_capture_timestamp_ms) {
351 RTC_DCHECK_RUN_ON(&encoder_queue_);
352 rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
353 if (frame_transformer_delegate_) {
354 // Asynchronously transform the payload before sending it. After the payload
355 // is transformed, the delegate will call SendRtpAudio to send it.
356 frame_transformer_delegate_->Transform(
357 frameType, payloadType, rtp_timestamp, rtp_rtcp_->StartTimestamp(),
358 payloadData, payloadSize, absolute_capture_timestamp_ms,
359 rtp_rtcp_->SSRC());
360 return 0;
361 }
362 return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
363 absolute_capture_timestamp_ms);
364 }
365
SendRtpAudio(AudioFrameType frameType,uint8_t payloadType,uint32_t rtp_timestamp,rtc::ArrayView<const uint8_t> payload,int64_t absolute_capture_timestamp_ms)366 int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
367 uint8_t payloadType,
368 uint32_t rtp_timestamp,
369 rtc::ArrayView<const uint8_t> payload,
370 int64_t absolute_capture_timestamp_ms) {
371 if (_includeAudioLevelIndication) {
372 // Store current audio level in the RTP sender.
373 // The level will be used in combination with voice-activity state
374 // (frameType) to add an RTP header extension
375 rtp_sender_audio_->SetAudioLevel(rms_level_.Average());
376 }
377
378 // E2EE Custom Audio Frame Encryption (This is optional).
379 // Keep this buffer around for the lifetime of the send call.
380 rtc::Buffer encrypted_audio_payload;
381 // We don't invoke encryptor if payload is empty, which means we are to send
382 // DTMF, or the encoder entered DTX.
383 // TODO(minyue): see whether DTMF packets should be encrypted or not. In
384 // current implementation, they are not.
385 if (!payload.empty()) {
386 if (frame_encryptor_ != nullptr) {
387 // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
388 // Allocate a buffer to hold the maximum possible encrypted payload.
389 size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
390 cricket::MEDIA_TYPE_AUDIO, payload.size());
391 encrypted_audio_payload.SetSize(max_ciphertext_size);
392
393 // Encrypt the audio payload into the buffer.
394 size_t bytes_written = 0;
395 int encrypt_status = frame_encryptor_->Encrypt(
396 cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
397 /*additional_data=*/nullptr, payload, encrypted_audio_payload,
398 &bytes_written);
399 if (encrypt_status != 0) {
400 RTC_DLOG(LS_ERROR)
401 << "Channel::SendData() failed encrypt audio payload: "
402 << encrypt_status;
403 return -1;
404 }
405 // Resize the buffer to the exact number of bytes actually used.
406 encrypted_audio_payload.SetSize(bytes_written);
407 // Rewrite the payloadData and size to the new encrypted payload.
408 payload = encrypted_audio_payload;
409 } else if (crypto_options_.sframe.require_frame_encryption) {
410 RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
411 "A frame encryptor is required but one is not set.";
412 return -1;
413 }
414 }
415
416 // Push data from ACM to RTP/RTCP-module to deliver audio frame for
417 // packetization.
418 if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp,
419 // Leaving the time when this frame was
420 // received from the capture device as
421 // undefined for voice for now.
422 -1, payloadType,
423 /*force_sender_report=*/false)) {
424 return -1;
425 }
426
427 // RTCPSender has it's own copy of the timestamp offset, added in
428 // RTCPSender::BuildSR, hence we must not add the in the offset for the above
429 // call.
430 // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
431 // knowledge of the offset to a single place.
432
433 // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
434 if (!rtp_sender_audio_->SendAudio(
435 frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
436 payload.data(), payload.size(), absolute_capture_timestamp_ms)) {
437 RTC_DLOG(LS_ERROR)
438 << "ChannelSend::SendData() failed to send data to RTP/RTCP module";
439 return -1;
440 }
441
442 return 0;
443 }
444
ChannelSend(Clock * clock,TaskQueueFactory * task_queue_factory,ProcessThread * module_process_thread,Transport * rtp_transport,RtcpRttStats * rtcp_rtt_stats,RtcEventLog * rtc_event_log,FrameEncryptorInterface * frame_encryptor,const webrtc::CryptoOptions & crypto_options,bool extmap_allow_mixed,int rtcp_report_interval_ms,uint32_t ssrc,rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,TransportFeedbackObserver * feedback_observer)445 ChannelSend::ChannelSend(
446 Clock* clock,
447 TaskQueueFactory* task_queue_factory,
448 ProcessThread* module_process_thread,
449 Transport* rtp_transport,
450 RtcpRttStats* rtcp_rtt_stats,
451 RtcEventLog* rtc_event_log,
452 FrameEncryptorInterface* frame_encryptor,
453 const webrtc::CryptoOptions& crypto_options,
454 bool extmap_allow_mixed,
455 int rtcp_report_interval_ms,
456 uint32_t ssrc,
457 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
458 TransportFeedbackObserver* feedback_observer)
459 : event_log_(rtc_event_log),
460 _timeStamp(0), // This is just an offset, RTP module will add it's own
461 // random offset
462 _moduleProcessThreadPtr(module_process_thread),
463 input_mute_(false),
464 previous_frame_muted_(false),
465 _includeAudioLevelIndication(false),
466 rtcp_observer_(new VoERtcpObserver(this)),
467 feedback_observer_(feedback_observer),
468 rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
469 retransmission_rate_limiter_(
470 new RateLimiter(clock, kMaxRetransmissionWindowMs)),
471 frame_encryptor_(frame_encryptor),
472 crypto_options_(crypto_options),
473 encoder_queue_(task_queue_factory->CreateTaskQueue(
474 "AudioEncoder",
475 TaskQueueFactory::Priority::NORMAL)),
476 fixing_timestamp_stall_(
477 !field_trial::IsDisabled("WebRTC-Audio-FixTimestampStall")) {
478 RTC_DCHECK(module_process_thread);
479 module_process_thread_checker_.Detach();
480
481 audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
482
483 RtpRtcpInterface::Configuration configuration;
484 configuration.bandwidth_callback = rtcp_observer_.get();
485 configuration.transport_feedback_callback = feedback_observer_;
486 configuration.clock = (clock ? clock : Clock::GetRealTimeClock());
487 configuration.audio = true;
488 configuration.outgoing_transport = rtp_transport;
489
490 configuration.paced_sender = rtp_packet_pacer_proxy_.get();
491
492 configuration.event_log = event_log_;
493 configuration.rtt_stats = rtcp_rtt_stats;
494 configuration.retransmission_rate_limiter =
495 retransmission_rate_limiter_.get();
496 configuration.extmap_allow_mixed = extmap_allow_mixed;
497 configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
498
499 configuration.local_media_ssrc = ssrc;
500
501 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
502 rtp_rtcp_->SetSendingMediaStatus(false);
503
504 rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock,
505 rtp_rtcp_->RtpSender());
506
507 _moduleProcessThreadPtr->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
508
509 // Ensure that RTCP is enabled by default for the created channel.
510 rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
511
512 int error = audio_coding_->RegisterTransportCallback(this);
513 RTC_DCHECK_EQ(0, error);
514 if (frame_transformer)
515 InitFrameTransformerDelegate(std::move(frame_transformer));
516 }
517
~ChannelSend()518 ChannelSend::~ChannelSend() {
519 RTC_DCHECK(construction_thread_.IsCurrent());
520
521 // Resets the delegate's callback to ChannelSend::SendRtpAudio.
522 if (frame_transformer_delegate_)
523 frame_transformer_delegate_->Reset();
524
525 StopSend();
526 int error = audio_coding_->RegisterTransportCallback(NULL);
527 RTC_DCHECK_EQ(0, error);
528
529 if (_moduleProcessThreadPtr)
530 _moduleProcessThreadPtr->DeRegisterModule(rtp_rtcp_.get());
531 }
532
StartSend()533 void ChannelSend::StartSend() {
534 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
535 RTC_DCHECK(!sending_);
536 sending_ = true;
537
538 rtp_rtcp_->SetSendingMediaStatus(true);
539 int ret = rtp_rtcp_->SetSendingStatus(true);
540 RTC_DCHECK_EQ(0, ret);
541 // It is now OK to start processing on the encoder task queue.
542 encoder_queue_.PostTask([this] {
543 RTC_DCHECK_RUN_ON(&encoder_queue_);
544 encoder_queue_is_active_ = true;
545 });
546 }
547
StopSend()548 void ChannelSend::StopSend() {
549 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
550 if (!sending_) {
551 return;
552 }
553 sending_ = false;
554
555 rtc::Event flush;
556 encoder_queue_.PostTask([this, &flush]() {
557 RTC_DCHECK_RUN_ON(&encoder_queue_);
558 encoder_queue_is_active_ = false;
559 flush.Set();
560 });
561 flush.Wait(rtc::Event::kForever);
562
563 // Reset sending SSRC and sequence number and triggers direct transmission
564 // of RTCP BYE
565 if (rtp_rtcp_->SetSendingStatus(false) == -1) {
566 RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
567 }
568 rtp_rtcp_->SetSendingMediaStatus(false);
569 }
570
SetEncoder(int payload_type,std::unique_ptr<AudioEncoder> encoder)571 void ChannelSend::SetEncoder(int payload_type,
572 std::unique_ptr<AudioEncoder> encoder) {
573 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
574 RTC_DCHECK_GE(payload_type, 0);
575 RTC_DCHECK_LE(payload_type, 127);
576
577 // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
578 // as well as some other things, so we collect this info and send it along.
579 rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
580 encoder->RtpTimestampRateHz());
581 rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
582 encoder->RtpTimestampRateHz(),
583 encoder->NumChannels(), 0);
584
585 audio_coding_->SetEncoder(std::move(encoder));
586 }
587
ModifyEncoder(rtc::FunctionView<void (std::unique_ptr<AudioEncoder> *)> modifier)588 void ChannelSend::ModifyEncoder(
589 rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
590 // This method can be called on the worker thread, module process thread
591 // or network thread. Audio coding is thread safe, so we do not need to
592 // enforce the calling thread.
593 audio_coding_->ModifyEncoder(modifier);
594 }
595
CallEncoder(rtc::FunctionView<void (AudioEncoder *)> modifier)596 void ChannelSend::CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) {
597 ModifyEncoder([modifier](std::unique_ptr<AudioEncoder>* encoder_ptr) {
598 if (*encoder_ptr) {
599 modifier(encoder_ptr->get());
600 } else {
601 RTC_DLOG(LS_WARNING) << "Trying to call unset encoder.";
602 }
603 });
604 }
605
OnBitrateAllocation(BitrateAllocationUpdate update)606 void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
607 // This method can be called on the worker thread, module process thread
608 // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
609 // TODO(solenberg): Figure out a good way to check this or enforce calling
610 // rules.
611 // RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
612 // module_process_thread_checker_.IsCurrent());
613 MutexLock lock(&bitrate_mutex_);
614
615 CallEncoder([&](AudioEncoder* encoder) {
616 encoder->OnReceivedUplinkAllocation(update);
617 });
618 retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
619 configured_bitrate_bps_ = update.target_bitrate.bps();
620 }
621
GetBitrate() const622 int ChannelSend::GetBitrate() const {
623 MutexLock lock(&bitrate_mutex_);
624 return configured_bitrate_bps_;
625 }
626
OnUplinkPacketLossRate(float packet_loss_rate)627 void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) {
628 CallEncoder([&](AudioEncoder* encoder) {
629 encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
630 });
631 }
632
ReceivedRTCPPacket(const uint8_t * data,size_t length)633 void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
634 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
635
636 // Deliver RTCP packet to RTP/RTCP module for parsing
637 rtp_rtcp_->IncomingRtcpPacket(data, length);
638
639 int64_t rtt = GetRTT();
640 if (rtt == 0) {
641 // Waiting for valid RTT.
642 return;
643 }
644
645 int64_t nack_window_ms = rtt;
646 if (nack_window_ms < kMinRetransmissionWindowMs) {
647 nack_window_ms = kMinRetransmissionWindowMs;
648 } else if (nack_window_ms > kMaxRetransmissionWindowMs) {
649 nack_window_ms = kMaxRetransmissionWindowMs;
650 }
651 retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
652
653 OnReceivedRtt(rtt);
654 }
655
SetInputMute(bool enable)656 void ChannelSend::SetInputMute(bool enable) {
657 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
658 MutexLock lock(&volume_settings_mutex_);
659 input_mute_ = enable;
660 }
661
InputMute() const662 bool ChannelSend::InputMute() const {
663 MutexLock lock(&volume_settings_mutex_);
664 return input_mute_;
665 }
666
SendTelephoneEventOutband(int event,int duration_ms)667 bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
668 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
669 RTC_DCHECK_LE(0, event);
670 RTC_DCHECK_GE(255, event);
671 RTC_DCHECK_LE(0, duration_ms);
672 RTC_DCHECK_GE(65535, duration_ms);
673 if (!sending_) {
674 return false;
675 }
676 if (rtp_sender_audio_->SendTelephoneEvent(
677 event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
678 RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
679 return false;
680 }
681 return true;
682 }
683
RegisterCngPayloadType(int payload_type,int payload_frequency)684 void ChannelSend::RegisterCngPayloadType(int payload_type,
685 int payload_frequency) {
686 rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
687 rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency,
688 1, 0);
689 }
690
SetSendTelephoneEventPayloadType(int payload_type,int payload_frequency)691 void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
692 int payload_frequency) {
693 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
694 RTC_DCHECK_LE(0, payload_type);
695 RTC_DCHECK_GE(127, payload_type);
696 rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
697 rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
698 payload_frequency, 0, 0);
699 }
700
SetSendAudioLevelIndicationStatus(bool enable,int id)701 void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
702 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
703 _includeAudioLevelIndication = enable;
704 if (enable) {
705 rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::kUri, id);
706 } else {
707 rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::kUri);
708 }
709 }
710
RegisterSenderCongestionControlObjects(RtpTransportControllerSendInterface * transport,RtcpBandwidthObserver * bandwidth_observer)711 void ChannelSend::RegisterSenderCongestionControlObjects(
712 RtpTransportControllerSendInterface* transport,
713 RtcpBandwidthObserver* bandwidth_observer) {
714 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
715 RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
716 PacketRouter* packet_router = transport->packet_router();
717
718 RTC_DCHECK(rtp_packet_pacer);
719 RTC_DCHECK(packet_router);
720 RTC_DCHECK(!packet_router_);
721 rtcp_observer_->SetBandwidthObserver(bandwidth_observer);
722 rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer);
723 rtp_rtcp_->SetStorePacketsStatus(true, 600);
724 constexpr bool remb_candidate = false;
725 packet_router->AddSendRtpModule(rtp_rtcp_.get(), remb_candidate);
726 packet_router_ = packet_router;
727 }
728
ResetSenderCongestionControlObjects()729 void ChannelSend::ResetSenderCongestionControlObjects() {
730 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
731 RTC_DCHECK(packet_router_);
732 rtp_rtcp_->SetStorePacketsStatus(false, 600);
733 rtcp_observer_->SetBandwidthObserver(nullptr);
734 packet_router_->RemoveSendRtpModule(rtp_rtcp_.get());
735 packet_router_ = nullptr;
736 rtp_packet_pacer_proxy_->SetPacketPacer(nullptr);
737 }
738
SetRTCP_CNAME(absl::string_view c_name)739 void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
740 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
741 // Note: SetCNAME() accepts a c string of length at most 255.
742 const std::string c_name_limited(c_name.substr(0, 255));
743 int ret = rtp_rtcp_->SetCNAME(c_name_limited.c_str()) != 0;
744 RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
745 }
746
GetRemoteRTCPReportBlocks() const747 std::vector<ReportBlock> ChannelSend::GetRemoteRTCPReportBlocks() const {
748 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
749 // Get the report blocks from the latest received RTCP Sender or Receiver
750 // Report. Each element in the vector contains the sender's SSRC and a
751 // report block according to RFC 3550.
752 std::vector<ReportBlock> report_blocks;
753 for (const ReportBlockData& data : rtp_rtcp_->GetLatestReportBlockData()) {
754 ReportBlock report_block;
755 report_block.sender_SSRC = data.report_block().sender_ssrc;
756 report_block.source_SSRC = data.report_block().source_ssrc;
757 report_block.fraction_lost = data.report_block().fraction_lost;
758 report_block.cumulative_num_packets_lost = data.report_block().packets_lost;
759 report_block.extended_highest_sequence_number =
760 data.report_block().extended_highest_sequence_number;
761 report_block.interarrival_jitter = data.report_block().jitter;
762 report_block.last_SR_timestamp =
763 data.report_block().last_sender_report_timestamp;
764 report_block.delay_since_last_SR =
765 data.report_block().delay_since_last_sender_report;
766 report_blocks.push_back(report_block);
767 }
768 return report_blocks;
769 }
770
GetRTCPStatistics() const771 CallSendStatistics ChannelSend::GetRTCPStatistics() const {
772 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
773 CallSendStatistics stats = {0};
774 stats.rttMs = GetRTT();
775
776 StreamDataCounters rtp_stats;
777 StreamDataCounters rtx_stats;
778 rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
779 stats.payload_bytes_sent =
780 rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
781 stats.header_and_padding_bytes_sent =
782 rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
783 rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
784
785 // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
786 // separate outbound-rtp stream objects.
787 stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
788 stats.packetsSent =
789 rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
790 stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
791 stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData();
792
793 return stats;
794 }
795
ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame)796 void ChannelSend::ProcessAndEncodeAudio(
797 std::unique_ptr<AudioFrame> audio_frame) {
798 RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
799 RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
800 RTC_DCHECK_LE(audio_frame->num_channels_, 8);
801
802 // Profile time between when the audio frame is added to the task queue and
803 // when the task is actually executed.
804 audio_frame->UpdateProfileTimeStamp();
805 encoder_queue_.PostTask(
806 [this, audio_frame = std::move(audio_frame)]() mutable {
807 RTC_DCHECK_RUN_ON(&encoder_queue_);
808 if (!encoder_queue_is_active_) {
809 if (fixing_timestamp_stall_) {
810 _timeStamp +=
811 static_cast<uint32_t>(audio_frame->samples_per_channel_);
812 }
813 return;
814 }
815 // Measure time between when the audio frame is added to the task queue
816 // and when the task is actually executed. Goal is to keep track of
817 // unwanted extra latency added by the task queue.
818 RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
819 audio_frame->ElapsedProfileTimeMs());
820
821 bool is_muted = InputMute();
822 AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_,
823 is_muted);
824
825 if (_includeAudioLevelIndication) {
826 size_t length =
827 audio_frame->samples_per_channel_ * audio_frame->num_channels_;
828 RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
829 if (is_muted && previous_frame_muted_) {
830 rms_level_.AnalyzeMuted(length);
831 } else {
832 rms_level_.Analyze(
833 rtc::ArrayView<const int16_t>(audio_frame->data(), length));
834 }
835 }
836 previous_frame_muted_ = is_muted;
837
838 // Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
839
840 // The ACM resamples internally.
841 audio_frame->timestamp_ = _timeStamp;
842 // This call will trigger AudioPacketizationCallback::SendData if
843 // encoding is done and payload is ready for packetization and
844 // transmission. Otherwise, it will return without invoking the
845 // callback.
846 if (audio_coding_->Add10MsData(*audio_frame) < 0) {
847 RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
848 return;
849 }
850
851 _timeStamp += static_cast<uint32_t>(audio_frame->samples_per_channel_);
852 });
853 }
854
GetANAStatistics() const855 ANAStats ChannelSend::GetANAStatistics() const {
856 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
857 return audio_coding_->GetANAStats();
858 }
859
GetRtpRtcp() const860 RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
861 RTC_DCHECK(module_process_thread_checker_.IsCurrent());
862 return rtp_rtcp_.get();
863 }
864
GetRTT() const865 int64_t ChannelSend::GetRTT() const {
866 std::vector<ReportBlockData> report_blocks =
867 rtp_rtcp_->GetLatestReportBlockData();
868 if (report_blocks.empty()) {
869 return 0;
870 }
871
872 // We don't know in advance the remote ssrc used by the other end's receiver
873 // reports, so use the first report block for the RTT.
874 return report_blocks.front().last_rtt_ms();
875 }
876
SetFrameEncryptor(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor)877 void ChannelSend::SetFrameEncryptor(
878 rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
879 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
880 encoder_queue_.PostTask([this, frame_encryptor]() mutable {
881 RTC_DCHECK_RUN_ON(&encoder_queue_);
882 frame_encryptor_ = std::move(frame_encryptor);
883 });
884 }
885
SetEncoderToPacketizerFrameTransformer(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)886 void ChannelSend::SetEncoderToPacketizerFrameTransformer(
887 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
888 RTC_DCHECK_RUN_ON(&worker_thread_checker_);
889 if (!frame_transformer)
890 return;
891
892 encoder_queue_.PostTask(
893 [this, frame_transformer = std::move(frame_transformer)]() mutable {
894 RTC_DCHECK_RUN_ON(&encoder_queue_);
895 InitFrameTransformerDelegate(std::move(frame_transformer));
896 });
897 }
898
OnReceivedRtt(int64_t rtt_ms)899 void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
900 // Invoke audio encoders OnReceivedRtt().
901 CallEncoder(
902 [rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); });
903 }
904
InitFrameTransformerDelegate(rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)905 void ChannelSend::InitFrameTransformerDelegate(
906 rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
907 RTC_DCHECK_RUN_ON(&encoder_queue_);
908 RTC_DCHECK(frame_transformer);
909 RTC_DCHECK(!frame_transformer_delegate_);
910
911 // Pass a callback to ChannelSend::SendRtpAudio, to be called by the delegate
912 // to send the transformed audio.
913 ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback =
914 [this](AudioFrameType frameType, uint8_t payloadType,
915 uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload,
916 int64_t absolute_capture_timestamp_ms) {
917 RTC_DCHECK_RUN_ON(&encoder_queue_);
918 return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
919 absolute_capture_timestamp_ms);
920 };
921 frame_transformer_delegate_ =
922 new rtc::RefCountedObject<ChannelSendFrameTransformerDelegate>(
923 std::move(send_audio_callback), std::move(frame_transformer),
924 &encoder_queue_);
925 frame_transformer_delegate_->Init();
926 }
927
928 } // namespace
929
CreateChannelSend(Clock * clock,TaskQueueFactory * task_queue_factory,ProcessThread * module_process_thread,Transport * rtp_transport,RtcpRttStats * rtcp_rtt_stats,RtcEventLog * rtc_event_log,FrameEncryptorInterface * frame_encryptor,const webrtc::CryptoOptions & crypto_options,bool extmap_allow_mixed,int rtcp_report_interval_ms,uint32_t ssrc,rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,TransportFeedbackObserver * feedback_observer)930 std::unique_ptr<ChannelSendInterface> CreateChannelSend(
931 Clock* clock,
932 TaskQueueFactory* task_queue_factory,
933 ProcessThread* module_process_thread,
934 Transport* rtp_transport,
935 RtcpRttStats* rtcp_rtt_stats,
936 RtcEventLog* rtc_event_log,
937 FrameEncryptorInterface* frame_encryptor,
938 const webrtc::CryptoOptions& crypto_options,
939 bool extmap_allow_mixed,
940 int rtcp_report_interval_ms,
941 uint32_t ssrc,
942 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
943 TransportFeedbackObserver* feedback_observer) {
944 return std::make_unique<ChannelSend>(
945 clock, task_queue_factory, module_process_thread, rtp_transport,
946 rtcp_rtt_stats, rtc_event_log, frame_encryptor, crypto_options,
947 extmap_allow_mixed, rtcp_report_interval_ms, ssrc,
948 std::move(frame_transformer), feedback_observer);
949 }
950
951 } // namespace voe
952 } // namespace webrtc
953