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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include <string.h> | 11 #include <string.h> |
12 #include <algorithm> | 12 #include <algorithm> |
13 #include <map> | 13 #include <map> |
14 #include <memory> | 14 #include <memory> |
15 #include <set> | 15 #include <set> |
16 #include <utility> | 16 #include <utility> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/audio/audio_receive_stream.h" | 19 #include "webrtc/audio/audio_receive_stream.h" |
20 #include "webrtc/audio/audio_send_stream.h" | 20 #include "webrtc/audio/audio_send_stream.h" |
21 #include "webrtc/audio/audio_state.h" | 21 #include "webrtc/audio/audio_state.h" |
22 #include "webrtc/audio/scoped_voe_interface.h" | 22 #include "webrtc/audio/scoped_voe_interface.h" |
23 #include "webrtc/base/basictypes.h" | 23 #include "webrtc/base/basictypes.h" |
24 #include "webrtc/base/checks.h" | 24 #include "webrtc/base/checks.h" |
25 #include "webrtc/base/constructormagic.h" | 25 #include "webrtc/base/constructormagic.h" |
26 #include "webrtc/base/location.h" | 26 #include "webrtc/base/location.h" |
27 #include "webrtc/base/logging.h" | 27 #include "webrtc/base/logging.h" |
28 #include "webrtc/base/optional.h" | 28 #include "webrtc/base/optional.h" |
| 29 #include "webrtc/base/ptr_util.h" |
29 #include "webrtc/base/task_queue.h" | 30 #include "webrtc/base/task_queue.h" |
30 #include "webrtc/base/thread_annotations.h" | 31 #include "webrtc/base/thread_annotations.h" |
31 #include "webrtc/base/thread_checker.h" | 32 #include "webrtc/base/thread_checker.h" |
32 #include "webrtc/base/trace_event.h" | 33 #include "webrtc/base/trace_event.h" |
33 #include "webrtc/call/bitrate_allocator.h" | 34 #include "webrtc/call/bitrate_allocator.h" |
34 #include "webrtc/call/call.h" | 35 #include "webrtc/call/call.h" |
35 #include "webrtc/call/flexfec_receive_stream_impl.h" | 36 #include "webrtc/call/flexfec_receive_stream_impl.h" |
36 #include "webrtc/call/rtp_transport_controller_send.h" | 37 #include "webrtc/call/rtp_transport_controller_send.h" |
37 #include "webrtc/config.h" | 38 #include "webrtc/config.h" |
38 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 39 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
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90 | 91 |
91 namespace internal { | 92 namespace internal { |
92 | 93 |
93 class Call : public webrtc::Call, | 94 class Call : public webrtc::Call, |
94 public PacketReceiver, | 95 public PacketReceiver, |
95 public RecoveredPacketReceiver, | 96 public RecoveredPacketReceiver, |
96 public SendSideCongestionController::Observer, | 97 public SendSideCongestionController::Observer, |
97 public BitrateAllocator::LimitObserver { | 98 public BitrateAllocator::LimitObserver { |
98 public: | 99 public: |
99 Call(const Call::Config& config, | 100 Call(const Call::Config& config, |
100 std::unique_ptr<RtpTransportControllerSend> transport_send); | 101 std::unique_ptr<RtpTransportControllerSendInterface> transport_send); |
101 virtual ~Call(); | 102 virtual ~Call(); |
102 | 103 |
103 // Implements webrtc::Call. | 104 // Implements webrtc::Call. |
104 PacketReceiver* Receiver() override; | 105 PacketReceiver* Receiver() override; |
105 | 106 |
106 webrtc::AudioSendStream* CreateAudioSendStream( | 107 webrtc::AudioSendStream* CreateAudioSendStream( |
107 const webrtc::AudioSendStream::Config& config) override; | 108 const webrtc::AudioSendStream::Config& config) override; |
108 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; | 109 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
109 | 110 |
110 webrtc::AudioReceiveStream* CreateAudioReceiveStream( | 111 webrtc::AudioReceiveStream* CreateAudioReceiveStream( |
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291 ss << "send_bw_bps: " << send_bandwidth_bps << ", "; | 292 ss << "send_bw_bps: " << send_bandwidth_bps << ", "; |
292 ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; | 293 ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; |
293 ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; | 294 ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; |
294 ss << "pacer_delay_ms: " << pacer_delay_ms << ", "; | 295 ss << "pacer_delay_ms: " << pacer_delay_ms << ", "; |
295 ss << "rtt_ms: " << rtt_ms; | 296 ss << "rtt_ms: " << rtt_ms; |
296 ss << '}'; | 297 ss << '}'; |
297 return ss.str(); | 298 return ss.str(); |
298 } | 299 } |
299 | 300 |
300 Call* Call::Create(const Call::Config& config) { | 301 Call* Call::Create(const Call::Config& config) { |
301 return new internal::Call( | 302 return new internal::Call(config, |
302 config, std::unique_ptr<RtpTransportControllerSend>( | 303 rtc::MakeUnique<RtpTransportControllerSend>( |
303 new RtpTransportControllerSend(Clock::GetRealTimeClock(), | 304 Clock::GetRealTimeClock(), config.event_log)); |
304 config.event_log))); | 305 } |
| 306 |
| 307 Call* Call::Create( |
| 308 const Call::Config& config, |
| 309 std::unique_ptr<RtpTransportControllerSendInterface> transport_send) { |
| 310 return new internal::Call(config, std::move(transport_send)); |
305 } | 311 } |
306 | 312 |
307 namespace internal { | 313 namespace internal { |
308 | 314 |
309 Call::Call(const Call::Config& config, | 315 Call::Call(const Call::Config& config, |
310 std::unique_ptr<RtpTransportControllerSend> transport_send) | 316 std::unique_ptr<RtpTransportControllerSendInterface> transport_send) |
311 : clock_(Clock::GetRealTimeClock()), | 317 : clock_(Clock::GetRealTimeClock()), |
312 num_cpu_cores_(CpuInfo::DetectNumberOfCores()), | 318 num_cpu_cores_(CpuInfo::DetectNumberOfCores()), |
313 module_process_thread_(ProcessThread::Create("ModuleProcessThread")), | 319 module_process_thread_(ProcessThread::Create("ModuleProcessThread")), |
314 pacer_thread_(ProcessThread::Create("PacerThread")), | 320 pacer_thread_(ProcessThread::Create("PacerThread")), |
315 call_stats_(new CallStats(clock_)), | 321 call_stats_(new CallStats(clock_)), |
316 bitrate_allocator_(new BitrateAllocator(this)), | 322 bitrate_allocator_(new BitrateAllocator(this)), |
317 config_(config), | 323 config_(config), |
318 audio_network_state_(kNetworkDown), | 324 audio_network_state_(kNetworkDown), |
319 video_network_state_(kNetworkDown), | 325 video_network_state_(kNetworkDown), |
320 receive_crit_(RWLockWrapper::CreateRWLock()), | 326 receive_crit_(RWLockWrapper::CreateRWLock()), |
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335 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 341 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
336 RTC_DCHECK(config.event_log != nullptr); | 342 RTC_DCHECK(config.event_log != nullptr); |
337 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); | 343 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
338 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, | 344 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
339 config.bitrate_config.min_bitrate_bps); | 345 config.bitrate_config.min_bitrate_bps); |
340 if (config.bitrate_config.max_bitrate_bps != -1) { | 346 if (config.bitrate_config.max_bitrate_bps != -1) { |
341 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, | 347 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, |
342 config.bitrate_config.start_bitrate_bps); | 348 config.bitrate_config.start_bitrate_bps); |
343 } | 349 } |
344 Trace::CreateTrace(); | 350 Trace::CreateTrace(); |
345 transport_send->RegisterNetworkObserver(this); | 351 transport_send->send_side_cc()->RegisterNetworkObserver(this); |
346 transport_send_ = std::move(transport_send); | 352 transport_send_ = std::move(transport_send); |
347 transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown); | 353 transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown); |
348 transport_send_->send_side_cc()->SetBweBitrates( | 354 transport_send_->send_side_cc()->SetBweBitrates( |
349 config_.bitrate_config.min_bitrate_bps, | 355 config_.bitrate_config.min_bitrate_bps, |
350 config_.bitrate_config.start_bitrate_bps, | 356 config_.bitrate_config.start_bitrate_bps, |
351 config_.bitrate_config.max_bitrate_bps); | 357 config_.bitrate_config.max_bitrate_bps); |
352 call_stats_->RegisterStatsObserver(&receive_side_cc_); | 358 call_stats_->RegisterStatsObserver(&receive_side_cc_); |
353 call_stats_->RegisterStatsObserver(transport_send_->send_side_cc()); | 359 call_stats_->RegisterStatsObserver(transport_send_->send_side_cc()); |
354 | 360 |
355 module_process_thread_->Start(); | 361 module_process_thread_->Start(); |
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1282 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1288 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1283 receive_side_cc_.OnReceivedPacket( | 1289 receive_side_cc_.OnReceivedPacket( |
1284 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1290 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1285 header); | 1291 header); |
1286 } | 1292 } |
1287 } | 1293 } |
1288 | 1294 |
1289 } // namespace internal | 1295 } // namespace internal |
1290 | 1296 |
1291 } // namespace webrtc | 1297 } // namespace webrtc |
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