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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 |
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34 #include "webrtc/video/audio_receive_stream.h" | 34 #include "webrtc/video/audio_receive_stream.h" |
35 #include "webrtc/video/video_receive_stream.h" | 35 #include "webrtc/video/video_receive_stream.h" |
36 #include "webrtc/video/video_send_stream.h" | 36 #include "webrtc/video/video_send_stream.h" |
37 | 37 |
38 namespace webrtc { | 38 namespace webrtc { |
39 | 39 |
40 const int Call::Config::kDefaultStartBitrateBps = 300000; | 40 const int Call::Config::kDefaultStartBitrateBps = 300000; |
41 | 41 |
42 namespace internal { | 42 namespace internal { |
43 | 43 |
44 class CpuOveruseObserverProxy : public webrtc::CpuOveruseObserver { | |
45 public: | |
46 explicit CpuOveruseObserverProxy(LoadObserver* overuse_callback) | |
47 : overuse_callback_(overuse_callback) { | |
48 DCHECK(overuse_callback != nullptr); | |
49 } | |
50 | |
51 virtual ~CpuOveruseObserverProxy() {} | |
52 | |
53 void OveruseDetected() override { | |
54 rtc::CritScope lock(&crit_); | |
55 overuse_callback_->OnLoadUpdate(LoadObserver::kOveruse); | |
56 } | |
57 | |
58 void NormalUsage() override { | |
59 rtc::CritScope lock(&crit_); | |
60 overuse_callback_->OnLoadUpdate(LoadObserver::kUnderuse); | |
61 } | |
62 | |
63 private: | |
64 rtc::CriticalSection crit_; | |
65 LoadObserver* overuse_callback_ GUARDED_BY(crit_); | |
66 }; | |
67 | |
68 class Call : public webrtc::Call, public PacketReceiver { | 44 class Call : public webrtc::Call, public PacketReceiver { |
69 public: | 45 public: |
70 explicit Call(const Call::Config& config); | 46 explicit Call(const Call::Config& config); |
71 virtual ~Call(); | 47 virtual ~Call(); |
72 | 48 |
73 PacketReceiver* Receiver() override; | 49 PacketReceiver* Receiver() override; |
74 | 50 |
75 webrtc::AudioSendStream* CreateAudioSendStream( | 51 webrtc::AudioSendStream* CreateAudioSendStream( |
76 const webrtc::AudioSendStream::Config& config) override; | 52 const webrtc::AudioSendStream::Config& config) override; |
77 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; | 53 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
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131 GUARDED_BY(receive_crit_); | 107 GUARDED_BY(receive_crit_); |
132 std::set<VideoReceiveStream*> video_receive_streams_ | 108 std::set<VideoReceiveStream*> video_receive_streams_ |
133 GUARDED_BY(receive_crit_); | 109 GUARDED_BY(receive_crit_); |
134 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ | 110 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ |
135 GUARDED_BY(receive_crit_); | 111 GUARDED_BY(receive_crit_); |
136 | 112 |
137 rtc::scoped_ptr<RWLockWrapper> send_crit_; | 113 rtc::scoped_ptr<RWLockWrapper> send_crit_; |
138 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_); | 114 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_); |
139 std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_); | 115 std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_); |
140 | 116 |
141 rtc::scoped_ptr<CpuOveruseObserverProxy> overuse_observer_proxy_; | |
142 | |
143 VideoSendStream::RtpStateMap suspended_video_send_ssrcs_; | 117 VideoSendStream::RtpStateMap suspended_video_send_ssrcs_; |
144 | 118 |
145 DISALLOW_COPY_AND_ASSIGN(Call); | 119 DISALLOW_COPY_AND_ASSIGN(Call); |
146 }; | 120 }; |
147 } // namespace internal | 121 } // namespace internal |
148 | 122 |
149 Call* Call::Create(const Call::Config& config) { | 123 Call* Call::Create(const Call::Config& config) { |
150 return new internal::Call(config); | 124 return new internal::Call(config); |
151 } | 125 } |
152 | 126 |
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165 DCHECK_GE(config.bitrate_config.start_bitrate_bps, | 139 DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
166 config.bitrate_config.min_bitrate_bps); | 140 config.bitrate_config.min_bitrate_bps); |
167 if (config.bitrate_config.max_bitrate_bps != -1) { | 141 if (config.bitrate_config.max_bitrate_bps != -1) { |
168 DCHECK_GE(config.bitrate_config.max_bitrate_bps, | 142 DCHECK_GE(config.bitrate_config.max_bitrate_bps, |
169 config.bitrate_config.start_bitrate_bps); | 143 config.bitrate_config.start_bitrate_bps); |
170 } | 144 } |
171 | 145 |
172 Trace::CreateTrace(); | 146 Trace::CreateTrace(); |
173 module_process_thread_->Start(); | 147 module_process_thread_->Start(); |
174 | 148 |
175 if (config.overuse_callback) { | |
176 overuse_observer_proxy_.reset( | |
177 new CpuOveruseObserverProxy(config.overuse_callback)); | |
178 } | |
179 | |
180 SetBitrateControllerConfig(config_.bitrate_config); | 149 SetBitrateControllerConfig(config_.bitrate_config); |
181 } | 150 } |
182 | 151 |
183 Call::~Call() { | 152 Call::~Call() { |
184 CHECK_EQ(0u, video_send_ssrcs_.size()); | 153 CHECK_EQ(0u, video_send_ssrcs_.size()); |
185 CHECK_EQ(0u, video_send_streams_.size()); | 154 CHECK_EQ(0u, video_send_streams_.size()); |
186 CHECK_EQ(0u, audio_receive_ssrcs_.size()); | 155 CHECK_EQ(0u, audio_receive_ssrcs_.size()); |
187 CHECK_EQ(0u, video_receive_ssrcs_.size()); | 156 CHECK_EQ(0u, video_receive_ssrcs_.size()); |
188 CHECK_EQ(0u, video_receive_streams_.size()); | 157 CHECK_EQ(0u, video_receive_streams_.size()); |
189 | 158 |
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241 | 210 |
242 webrtc::VideoSendStream* Call::CreateVideoSendStream( | 211 webrtc::VideoSendStream* Call::CreateVideoSendStream( |
243 const webrtc::VideoSendStream::Config& config, | 212 const webrtc::VideoSendStream::Config& config, |
244 const VideoEncoderConfig& encoder_config) { | 213 const VideoEncoderConfig& encoder_config) { |
245 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); | 214 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); |
246 LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString(); | 215 LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString(); |
247 DCHECK(!config.rtp.ssrcs.empty()); | 216 DCHECK(!config.rtp.ssrcs.empty()); |
248 | 217 |
249 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if | 218 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
250 // the call has already started. | 219 // the call has already started. |
251 VideoSendStream* send_stream = new VideoSendStream( | 220 VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_, |
252 overuse_observer_proxy_.get(), num_cpu_cores_, | |
253 module_process_thread_.get(), channel_group_.get(), | 221 module_process_thread_.get(), channel_group_.get(), |
254 rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config, | 222 rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config, |
255 suspended_video_send_ssrcs_); | 223 suspended_video_send_ssrcs_); |
256 | 224 |
257 // This needs to be taken before send_crit_ as both locks need to be held | 225 // This needs to be taken before send_crit_ as both locks need to be held |
258 // while changing network state. | 226 // while changing network state. |
259 rtc::CritScope lock(&network_enabled_crit_); | 227 rtc::CritScope lock(&network_enabled_crit_); |
260 WriteLockScoped write_lock(*send_crit_); | 228 WriteLockScoped write_lock(*send_crit_); |
261 for (uint32_t ssrc : config.rtp.ssrcs) { | 229 for (uint32_t ssrc : config.rtp.ssrcs) { |
262 DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end()); | 230 DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end()); |
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536 const uint8_t* packet, | 504 const uint8_t* packet, |
537 size_t length) { | 505 size_t length) { |
538 if (RtpHeaderParser::IsRtcp(packet, length)) | 506 if (RtpHeaderParser::IsRtcp(packet, length)) |
539 return DeliverRtcp(media_type, packet, length); | 507 return DeliverRtcp(media_type, packet, length); |
540 | 508 |
541 return DeliverRtp(media_type, packet, length); | 509 return DeliverRtp(media_type, packet, length); |
542 } | 510 } |
543 | 511 |
544 } // namespace internal | 512 } // namespace internal |
545 } // namespace webrtc | 513 } // namespace webrtc |
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