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pbos@webrtc.org2a9108f2013-05-16 12:08:031/*
2 * Copyright (c) 2013 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
pbos@webrtc.org3f45c2e2013-08-05 16:22:5311#include <string.h>
12
pbos@webrtc.org2a9108f2013-05-16 12:08:0313#include <map>
14#include <vector>
15
Peter Boströmbf9f73c2015-09-25 11:58:3016#include "webrtc/audio/audio_receive_stream.h"
pbos@webrtc.org6eaf09a2015-03-23 13:12:2417#include "webrtc/base/checks.h"
kwiberg@webrtc.orgdeb9dae2015-02-26 14:34:5518#include "webrtc/base/scoped_ptr.h"
pbos@webrtc.orgd54aa962014-09-24 06:05:0019#include "webrtc/base/thread_annotations.h"
pbos@webrtc.org24e20892013-10-28 16:32:0120#include "webrtc/call.h"
Peter Boströmbf9f73c2015-09-25 11:58:3021#include "webrtc/call/rtc_event_log.h"
stefan@webrtc.org47f0c412013-12-04 10:24:2622#include "webrtc/common.h"
pbos@webrtc.orgd05597a2013-12-05 12:11:4723#include "webrtc/config.h"
pbos@webrtc.org24e20892013-10-28 16:32:0124#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
sprang@webrtc.orgcce642c2015-01-28 12:37:3625#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
Peter Boström8e07f3a2015-05-08 11:54:3826#include "webrtc/modules/utility/interface/process_thread.h"
Peter Boström8e07f3a2015-05-08 11:54:3827#include "webrtc/system_wrappers/interface/cpu_info.h"
pbos@webrtc.orgb5d2d162013-10-02 13:36:0928#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
pbos@webrtc.orgf076e172015-01-15 10:09:3929#include "webrtc/system_wrappers/interface/logging.h"
pbos@webrtc.org24e20892013-10-28 16:32:0130#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
pbos@webrtc.orgb5d2d162013-10-02 13:36:0931#include "webrtc/system_wrappers/interface/trace.h"
pbos@webrtc.orge2863932015-01-29 12:33:0732#include "webrtc/system_wrappers/interface/trace_event.h"
pbos@webrtc.org24e20892013-10-28 16:32:0133#include "webrtc/video/video_receive_stream.h"
34#include "webrtc/video/video_send_stream.h"
ivoc35fd7532015-09-09 07:09:4335#include "webrtc/voice_engine/include/voe_codec.h"
pbos@webrtc.org2a9108f2013-05-16 12:08:0336
37namespace webrtc {
pbos@webrtc.org1c655452014-09-17 09:02:2538
pbos@webrtc.org7da30672014-10-14 11:52:1039const int Call::Config::kDefaultStartBitrateBps = 300000;
40
pbos@webrtc.org24e20892013-10-28 16:32:0141namespace internal {
asapersson@webrtc.org8ef65482014-01-31 10:05:0742
pbos@webrtc.org24e20892013-10-28 16:32:0143class Call : public webrtc::Call, public PacketReceiver {
44 public:
Peter Boström8e07f3a2015-05-08 11:54:3845 explicit Call(const Call::Config& config);
pbos@webrtc.org24e20892013-10-28 16:32:0146 virtual ~Call();
47
kjellander@webrtc.org860ac532015-03-04 12:58:3548 PacketReceiver* Receiver() override;
pbos@webrtc.org24e20892013-10-28 16:32:0149
Fredrik Solenbergee9b72e2015-06-08 11:04:5650 webrtc::AudioSendStream* CreateAudioSendStream(
51 const webrtc::AudioSendStream::Config& config) override;
52 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
53
Fredrik Solenbergad867862015-04-29 13:24:0154 webrtc::AudioReceiveStream* CreateAudioReceiveStream(
55 const webrtc::AudioReceiveStream::Config& config) override;
56 void DestroyAudioReceiveStream(
57 webrtc::AudioReceiveStream* receive_stream) override;
pbos@webrtc.org24e20892013-10-28 16:32:0158
Fredrik Solenbergad867862015-04-29 13:24:0159 webrtc::VideoSendStream* CreateVideoSendStream(
60 const webrtc::VideoSendStream::Config& config,
61 const VideoEncoderConfig& encoder_config) override;
kjellander@webrtc.org860ac532015-03-04 12:58:3562 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
pbos@webrtc.org24e20892013-10-28 16:32:0163
Fredrik Solenbergad867862015-04-29 13:24:0164 webrtc::VideoReceiveStream* CreateVideoReceiveStream(
65 const webrtc::VideoReceiveStream::Config& config) override;
kjellander@webrtc.org860ac532015-03-04 12:58:3566 void DestroyVideoReceiveStream(
67 webrtc::VideoReceiveStream* receive_stream) override;
pbos@webrtc.org24e20892013-10-28 16:32:0168
kjellander@webrtc.org860ac532015-03-04 12:58:3569 Stats GetStats() const override;
pbos@webrtc.org24e20892013-10-28 16:32:0170
stefan30bf7782015-09-08 12:36:1571 DeliveryStatus DeliverPacket(MediaType media_type,
72 const uint8_t* packet,
73 size_t length,
74 const PacketTime& packet_time) override;
pbos@webrtc.org24e20892013-10-28 16:32:0175
kjellander@webrtc.org860ac532015-03-04 12:58:3576 void SetBitrateConfig(
77 const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
78 void SignalNetworkState(NetworkState state) override;
pbos@webrtc.org9b707ca2014-09-03 16:17:1279
pbos@webrtc.org24e20892013-10-28 16:32:0180 private:
Fredrik Solenbergad867862015-04-29 13:24:0181 DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
82 size_t length);
stefan30bf7782015-09-08 12:36:1583 DeliveryStatus DeliverRtp(MediaType media_type,
84 const uint8_t* packet,
85 size_t length,
86 const PacketTime& packet_time);
pbos@webrtc.org24e20892013-10-28 16:32:0187
Peter Boström8e07f3a2015-05-08 11:54:3888 void SetBitrateControllerConfig(
89 const webrtc::Call::Config::BitrateConfig& bitrate_config);
90
pbos495b3502015-07-15 15:02:5891 void ConfigureSync(const std::string& sync_group)
92 EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
93
Peter Boström8e07f3a2015-05-08 11:54:3894 const int num_cpu_cores_;
95 const rtc::scoped_ptr<ProcessThread> module_process_thread_;
96 const rtc::scoped_ptr<ChannelGroup> channel_group_;
Peter Boström8e07f3a2015-05-08 11:54:3897 volatile int next_channel_id_;
pbos@webrtc.org24e20892013-10-28 16:32:0198 Call::Config config_;
99
pbos@webrtc.org9b707ca2014-09-03 16:17:12100 // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
101 // ensures that we have a consistent network state signalled to all senders
102 // and receivers.
Peter Boström7d6ce5f2015-05-01 11:00:41103 rtc::CriticalSection network_enabled_crit_;
pbos@webrtc.org9b707ca2014-09-03 16:17:12104 bool network_enabled_ GUARDED_BY(network_enabled_crit_);
pbos@webrtc.org24e20892013-10-28 16:32:01105
kwiberg@webrtc.orgdeb9dae2015-02-26 14:34:55106 rtc::scoped_ptr<RWLockWrapper> receive_crit_;
Fredrik Solenbergad867862015-04-29 13:24:01107 std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
pbos@webrtc.org9b707ca2014-09-03 16:17:12108 GUARDED_BY(receive_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01109 std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
110 GUARDED_BY(receive_crit_);
111 std::set<VideoReceiveStream*> video_receive_streams_
112 GUARDED_BY(receive_crit_);
pbos495b3502015-07-15 15:02:58113 std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
114 GUARDED_BY(receive_crit_);
pbos@webrtc.org9b707ca2014-09-03 16:17:12115
kwiberg@webrtc.orgdeb9dae2015-02-26 14:34:55116 rtc::scoped_ptr<RWLockWrapper> send_crit_;
Fredrik Solenbergad867862015-04-29 13:24:01117 std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
118 std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
pbos@webrtc.org24e20892013-10-28 16:32:01119
Fredrik Solenbergad867862015-04-29 13:24:01120 VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
pbos@webrtc.org2fd91bd2014-07-07 13:06:48121
ivoc35fd7532015-09-09 07:09:43122 RtcEventLog* event_log_;
123
henrikg9199c0e2015-09-16 12:37:44124 RTC_DISALLOW_COPY_AND_ASSIGN(Call);
pbos@webrtc.org24e20892013-10-28 16:32:01125};
pbos@webrtc.orgd05597a2013-12-05 12:11:47126} // namespace internal
pbos@webrtc.orgc2014fd2013-08-14 13:52:52127
stefan@webrtc.org47f0c412013-12-04 10:24:26128Call* Call::Create(const Call::Config& config) {
Peter Boström8e07f3a2015-05-08 11:54:38129 return new internal::Call(config);
pbos@webrtc.orgc2014fd2013-08-14 13:52:52130}
pbos@webrtc.orgc2014fd2013-08-14 13:52:52131
pbos@webrtc.org2a9108f2013-05-16 12:08:03132namespace internal {
133
Peter Boström8e07f3a2015-05-08 11:54:38134Call::Call(const Call::Config& config)
135 : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
stefan63ac87f2015-09-11 16:52:15136 module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
Peter Boströma28bdde2015-05-28 12:10:39137 channel_group_(new ChannelGroup(module_process_thread_.get())),
pboscce61412015-07-23 13:58:33138 next_channel_id_(0),
Peter Boström8e07f3a2015-05-08 11:54:38139 config_(config),
pbos@webrtc.org9b707ca2014-09-03 16:17:12140 network_enabled_(true),
141 receive_crit_(RWLockWrapper::CreateRWLock()),
ivoc35fd7532015-09-09 07:09:43142 send_crit_(RWLockWrapper::CreateRWLock()),
143 event_log_(nullptr) {
henrikg5c075c82015-09-17 07:24:34144 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
145 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
146 config.bitrate_config.min_bitrate_bps);
Stefan Holmerca55fa12015-03-26 10:11:06147 if (config.bitrate_config.max_bitrate_bps != -1) {
henrikg5c075c82015-09-17 07:24:34148 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
149 config.bitrate_config.start_bitrate_bps);
pbos@webrtc.org5e3f6b42014-11-25 14:03:34150 }
ivoc35fd7532015-09-09 07:09:43151 if (config.voice_engine) {
152 VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
153 if (voe_codec) {
154 event_log_ = voe_codec->GetEventLog();
155 voe_codec->Release();
156 }
157 }
pbos@webrtc.org5e3f6b42014-11-25 14:03:34158
Peter Boström8e07f3a2015-05-08 11:54:38159 Trace::CreateTrace();
160 module_process_thread_->Start();
161
Peter Boström8e07f3a2015-05-08 11:54:38162 SetBitrateControllerConfig(config_.bitrate_config);
pbos@webrtc.org2a9108f2013-05-16 12:08:03163}
164
pbos@webrtc.orgbf6d5722013-09-09 15:04:25165Call::~Call() {
henrikg5c075c82015-09-17 07:24:34166 RTC_CHECK_EQ(0u, video_send_ssrcs_.size());
167 RTC_CHECK_EQ(0u, video_send_streams_.size());
168 RTC_CHECK_EQ(0u, audio_receive_ssrcs_.size());
169 RTC_CHECK_EQ(0u, video_receive_ssrcs_.size());
170 RTC_CHECK_EQ(0u, video_receive_streams_.size());
pbos@webrtc.orgf71c9742015-02-12 10:48:23171
Peter Boström8e07f3a2015-05-08 11:54:38172 module_process_thread_->Stop();
173 Trace::ReturnTrace();
pbos@webrtc.org2a9108f2013-05-16 12:08:03174}
175
pbos@webrtc.orgbf6d5722013-09-09 15:04:25176PacketReceiver* Call::Receiver() { return this; }
pbos@webrtc.org2a9108f2013-05-16 12:08:03177
Fredrik Solenbergee9b72e2015-06-08 11:04:56178webrtc::AudioSendStream* Call::CreateAudioSendStream(
179 const webrtc::AudioSendStream::Config& config) {
180 return nullptr;
181}
182
183void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
184}
185
Fredrik Solenbergad867862015-04-29 13:24:01186webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
187 const webrtc::AudioReceiveStream::Config& config) {
188 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
189 LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString();
190 AudioReceiveStream* receive_stream = new AudioReceiveStream(
191 channel_group_->GetRemoteBitrateEstimator(), config);
192 {
193 WriteLockScoped write_lock(*receive_crit_);
henrikg5c075c82015-09-17 07:24:34194 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
195 audio_receive_ssrcs_.end());
Fredrik Solenbergad867862015-04-29 13:24:01196 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
pbos495b3502015-07-15 15:02:58197 ConfigureSync(config.sync_group);
Fredrik Solenbergad867862015-04-29 13:24:01198 }
199 return receive_stream;
200}
201
202void Call::DestroyAudioReceiveStream(
203 webrtc::AudioReceiveStream* receive_stream) {
204 TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
henrikg5c075c82015-09-17 07:24:34205 RTC_DCHECK(receive_stream != nullptr);
Fredrik Solenbergad867862015-04-29 13:24:01206 AudioReceiveStream* audio_receive_stream =
207 static_cast<AudioReceiveStream*>(receive_stream);
208 {
209 WriteLockScoped write_lock(*receive_crit_);
210 size_t num_deleted = audio_receive_ssrcs_.erase(
211 audio_receive_stream->config().rtp.remote_ssrc);
henrikg5c075c82015-09-17 07:24:34212 RTC_DCHECK(num_deleted == 1);
pbos495b3502015-07-15 15:02:58213 const std::string& sync_group = audio_receive_stream->config().sync_group;
214 const auto it = sync_stream_mapping_.find(sync_group);
215 if (it != sync_stream_mapping_.end() &&
216 it->second == audio_receive_stream) {
217 sync_stream_mapping_.erase(it);
218 ConfigureSync(sync_group);
219 }
Fredrik Solenbergad867862015-04-29 13:24:01220 }
221 delete audio_receive_stream;
222}
223
224webrtc::VideoSendStream* Call::CreateVideoSendStream(
225 const webrtc::VideoSendStream::Config& config,
pbos@webrtc.org58b51402014-09-19 12:30:25226 const VideoEncoderConfig& encoder_config) {
pbos@webrtc.orge2863932015-01-29 12:33:07227 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
pbos@webrtc.orgf076e172015-01-15 10:09:39228 LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
henrikg5c075c82015-09-17 07:24:34229 RTC_DCHECK(!config.rtp.ssrcs.empty());
pbos@webrtc.org63988b22013-06-10 13:48:26230
mflodman@webrtc.orgf89ce462014-06-16 08:57:39231 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
232 // the call has already started.
solenbergbe5952f2015-09-08 12:13:22233 VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
Peter Boström8e07f3a2015-05-08 11:54:38234 module_process_thread_.get(), channel_group_.get(),
235 rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
236 suspended_video_send_ssrcs_);
pbos@webrtc.org63988b22013-06-10 13:48:26237
pbos@webrtc.org9b707ca2014-09-03 16:17:12238 // This needs to be taken before send_crit_ as both locks need to be held
239 // while changing network state.
Peter Boström7d6ce5f2015-05-01 11:00:41240 rtc::CritScope lock(&network_enabled_crit_);
pbos@webrtc.org9b707ca2014-09-03 16:17:12241 WriteLockScoped write_lock(*send_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01242 for (uint32_t ssrc : config.rtp.ssrcs) {
henrikg5c075c82015-09-17 07:24:34243 RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
Fredrik Solenbergad867862015-04-29 13:24:01244 video_send_ssrcs_[ssrc] = send_stream;
pbos@webrtc.org2a9108f2013-05-16 12:08:03245 }
Fredrik Solenbergad867862015-04-29 13:24:01246 video_send_streams_.insert(send_stream);
247
ivoc35fd7532015-09-09 07:09:43248 if (event_log_)
249 event_log_->LogVideoSendStreamConfig(config);
250
pbos@webrtc.org9b707ca2014-09-03 16:17:12251 if (!network_enabled_)
252 send_stream->SignalNetworkState(kNetworkDown);
pbos@webrtc.org2a9108f2013-05-16 12:08:03253 return send_stream;
254}
255
pbos@webrtc.org12a93e02013-11-21 13:49:43256void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
pbos@webrtc.orge2863932015-01-29 12:33:07257 TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
henrikg5c075c82015-09-17 07:24:34258 RTC_DCHECK(send_stream != nullptr);
pbos@webrtc.org00208582013-09-05 12:38:54259
pbos@webrtc.org2fd91bd2014-07-07 13:06:48260 send_stream->Stop();
261
pbos@webrtc.org6eaf09a2015-03-23 13:12:24262 VideoSendStream* send_stream_impl = nullptr;
pbos@webrtc.org00208582013-09-05 12:38:54263 {
pbos@webrtc.org9b707ca2014-09-03 16:17:12264 WriteLockScoped write_lock(*send_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01265 auto it = video_send_ssrcs_.begin();
266 while (it != video_send_ssrcs_.end()) {
pbos@webrtc.org00208582013-09-05 12:38:54267 if (it->second == static_cast<VideoSendStream*>(send_stream)) {
268 send_stream_impl = it->second;
Fredrik Solenbergad867862015-04-29 13:24:01269 video_send_ssrcs_.erase(it++);
pbos@webrtc.org2fd91bd2014-07-07 13:06:48270 } else {
271 ++it;
pbos@webrtc.org00208582013-09-05 12:38:54272 }
273 }
Fredrik Solenbergad867862015-04-29 13:24:01274 video_send_streams_.erase(send_stream_impl);
pbos@webrtc.org2a9108f2013-05-16 12:08:03275 }
henrikg5c075c82015-09-17 07:24:34276 RTC_CHECK(send_stream_impl != nullptr);
pbos@webrtc.org00208582013-09-05 12:38:54277
pbos@webrtc.org2fd91bd2014-07-07 13:06:48278 VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates();
279
280 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
281 it != rtp_state.end();
282 ++it) {
Fredrik Solenbergad867862015-04-29 13:24:01283 suspended_video_send_ssrcs_[it->first] = it->second;
pbos@webrtc.org2fd91bd2014-07-07 13:06:48284 }
285
pbos@webrtc.org00208582013-09-05 12:38:54286 delete send_stream_impl;
pbos@webrtc.org2a9108f2013-05-16 12:08:03287}
288
Fredrik Solenbergad867862015-04-29 13:24:01289webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
290 const webrtc::VideoReceiveStream::Config& config) {
pbos@webrtc.orge2863932015-01-29 12:33:07291 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
pbos@webrtc.orgf076e172015-01-15 10:09:39292 LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
Peter Boströme7c0a782015-04-24 13:16:03293 VideoReceiveStream* receive_stream = new VideoReceiveStream(
pboscce61412015-07-23 13:58:33294 num_cpu_cores_, channel_group_.get(),
Peter Boström8e07f3a2015-05-08 11:54:38295 rtc::AtomicOps::Increment(&next_channel_id_), config,
solenbergdce9bf32015-08-28 11:07:10296 config_.voice_engine);
pbos@webrtc.org2a9108f2013-05-16 12:08:03297
pbos@webrtc.org9b707ca2014-09-03 16:17:12298 // This needs to be taken before receive_crit_ as both locks need to be held
299 // while changing network state.
Peter Boström7d6ce5f2015-05-01 11:00:41300 rtc::CritScope lock(&network_enabled_crit_);
pbos@webrtc.org9b707ca2014-09-03 16:17:12301 WriteLockScoped write_lock(*receive_crit_);
henrikg5c075c82015-09-17 07:24:34302 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
303 video_receive_ssrcs_.end());
Fredrik Solenbergad867862015-04-29 13:24:01304 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
pbos@webrtc.orgc71929d2014-01-24 09:30:53305 // TODO(pbos): Configure different RTX payloads per receive payload.
306 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
307 config.rtp.rtx.begin();
308 if (it != config.rtp.rtx.end())
Fredrik Solenbergad867862015-04-29 13:24:01309 video_receive_ssrcs_[it->second.ssrc] = receive_stream;
310 video_receive_streams_.insert(receive_stream);
pbos@webrtc.orgc71929d2014-01-24 09:30:53311
pbos495b3502015-07-15 15:02:58312 ConfigureSync(config.sync_group);
313
pbos@webrtc.org9b707ca2014-09-03 16:17:12314 if (!network_enabled_)
315 receive_stream->SignalNetworkState(kNetworkDown);
pbos495b3502015-07-15 15:02:58316
ivoc35fd7532015-09-09 07:09:43317 if (event_log_)
318 event_log_->LogVideoReceiveStreamConfig(config);
319
pbos@webrtc.org2a9108f2013-05-16 12:08:03320 return receive_stream;
321}
322
pbos@webrtc.org12a93e02013-11-21 13:49:43323void Call::DestroyVideoReceiveStream(
324 webrtc::VideoReceiveStream* receive_stream) {
pbos@webrtc.orge2863932015-01-29 12:33:07325 TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
henrikg5c075c82015-09-17 07:24:34326 RTC_DCHECK(receive_stream != nullptr);
pbos@webrtc.org6eaf09a2015-03-23 13:12:24327 VideoReceiveStream* receive_stream_impl = nullptr;
pbos@webrtc.org00208582013-09-05 12:38:54328 {
pbos@webrtc.org9b707ca2014-09-03 16:17:12329 WriteLockScoped write_lock(*receive_crit_);
pbos@webrtc.orgc71929d2014-01-24 09:30:53330 // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
331 // separate SSRC there can be either one or two.
Fredrik Solenbergad867862015-04-29 13:24:01332 auto it = video_receive_ssrcs_.begin();
333 while (it != video_receive_ssrcs_.end()) {
pbos@webrtc.org00208582013-09-05 12:38:54334 if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
pbos@webrtc.org6eaf09a2015-03-23 13:12:24335 if (receive_stream_impl != nullptr)
henrikg5c075c82015-09-17 07:24:34336 RTC_DCHECK(receive_stream_impl == it->second);
pbos@webrtc.org00208582013-09-05 12:38:54337 receive_stream_impl = it->second;
Fredrik Solenbergad867862015-04-29 13:24:01338 video_receive_ssrcs_.erase(it++);
pbos@webrtc.orgc71929d2014-01-24 09:30:53339 } else {
340 ++it;
pbos@webrtc.org00208582013-09-05 12:38:54341 }
342 }
Fredrik Solenbergad867862015-04-29 13:24:01343 video_receive_streams_.erase(receive_stream_impl);
henrikg5c075c82015-09-17 07:24:34344 RTC_CHECK(receive_stream_impl != nullptr);
pbos495b3502015-07-15 15:02:58345 ConfigureSync(receive_stream_impl->config().sync_group);
pbos@webrtc.org2a9108f2013-05-16 12:08:03346 }
pbos@webrtc.org00208582013-09-05 12:38:54347 delete receive_stream_impl;
pbos@webrtc.org2a9108f2013-05-16 12:08:03348}
349
stefan@webrtc.org52322672014-11-05 14:05:29350Call::Stats Call::GetStats() const {
351 Stats stats;
Peter Boström8e07f3a2015-05-08 11:54:38352 // Fetch available send/receive bitrates.
stefan@webrtc.org52322672014-11-05 14:05:29353 uint32_t send_bandwidth = 0;
Peter Boström8e07f3a2015-05-08 11:54:38354 channel_group_->GetBitrateController()->AvailableBandwidth(&send_bandwidth);
355 std::vector<unsigned int> ssrcs;
stefan@webrtc.org52322672014-11-05 14:05:29356 uint32_t recv_bandwidth = 0;
Peter Boström8e07f3a2015-05-08 11:54:38357 channel_group_->GetRemoteBitrateEstimator()->LatestEstimate(&ssrcs,
358 &recv_bandwidth);
359 stats.send_bandwidth_bps = send_bandwidth;
stefan@webrtc.org52322672014-11-05 14:05:29360 stats.recv_bandwidth_bps = recv_bandwidth;
Peter Boströmad9dcaf2015-04-27 15:24:33361 stats.pacer_delay_ms = channel_group_->GetPacerQueuingDelayMs();
stefan@webrtc.org52322672014-11-05 14:05:29362 {
363 ReadLockScoped read_lock(*send_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01364 for (const auto& kv : video_send_ssrcs_) {
365 int rtt_ms = kv.second->GetRtt();
pbos@webrtc.org834b4752014-12-11 13:26:09366 if (rtt_ms > 0)
367 stats.rtt_ms = rtt_ms;
stefan@webrtc.org52322672014-11-05 14:05:29368 }
369 }
370 return stats;
pbos@webrtc.org2a9108f2013-05-16 12:08:03371}
372
pbos@webrtc.org5e3f6b42014-11-25 14:03:34373void Call::SetBitrateConfig(
374 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
pbos@webrtc.orge2863932015-01-29 12:33:07375 TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
henrikg5c075c82015-09-17 07:24:34376 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
pbos@webrtc.org6eaf09a2015-03-23 13:12:24377 if (bitrate_config.max_bitrate_bps != -1)
henrikg5c075c82015-09-17 07:24:34378 RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
Stefan Holmerca55fa12015-03-26 10:11:06379 if (config_.bitrate_config.min_bitrate_bps ==
pbos@webrtc.org5e3f6b42014-11-25 14:03:34380 bitrate_config.min_bitrate_bps &&
381 (bitrate_config.start_bitrate_bps <= 0 ||
Stefan Holmerca55fa12015-03-26 10:11:06382 config_.bitrate_config.start_bitrate_bps ==
pbos@webrtc.org5e3f6b42014-11-25 14:03:34383 bitrate_config.start_bitrate_bps) &&
Stefan Holmerca55fa12015-03-26 10:11:06384 config_.bitrate_config.max_bitrate_bps ==
pbos@webrtc.org5e3f6b42014-11-25 14:03:34385 bitrate_config.max_bitrate_bps) {
386 // Nothing new to set, early abort to avoid encoder reconfigurations.
387 return;
388 }
Stefan Holmerca55fa12015-03-26 10:11:06389 config_.bitrate_config = bitrate_config;
Peter Boström8e07f3a2015-05-08 11:54:38390 SetBitrateControllerConfig(bitrate_config);
391}
392
393void Call::SetBitrateControllerConfig(
394 const webrtc::Call::Config::BitrateConfig& bitrate_config) {
395 BitrateController* bitrate_controller =
396 channel_group_->GetBitrateController();
397 if (bitrate_config.start_bitrate_bps > 0)
398 bitrate_controller->SetStartBitrate(bitrate_config.start_bitrate_bps);
399 bitrate_controller->SetMinMaxBitrate(bitrate_config.min_bitrate_bps,
400 bitrate_config.max_bitrate_bps);
pbos@webrtc.org5e3f6b42014-11-25 14:03:34401}
402
pbos@webrtc.org9b707ca2014-09-03 16:17:12403void Call::SignalNetworkState(NetworkState state) {
404 // Take crit for entire function, it needs to be held while updating streams
405 // to guarantee a consistent state across streams.
Peter Boström7d6ce5f2015-05-01 11:00:41406 rtc::CritScope lock(&network_enabled_crit_);
pbos@webrtc.org9b707ca2014-09-03 16:17:12407 network_enabled_ = state == kNetworkUp;
408 {
409 ReadLockScoped write_lock(*send_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01410 for (auto& kv : video_send_ssrcs_) {
411 kv.second->SignalNetworkState(state);
pbos@webrtc.org9b707ca2014-09-03 16:17:12412 }
413 }
414 {
415 ReadLockScoped write_lock(*receive_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01416 for (auto& kv : video_receive_ssrcs_) {
417 kv.second->SignalNetworkState(state);
pbos@webrtc.org9b707ca2014-09-03 16:17:12418 }
419 }
420}
421
pbos495b3502015-07-15 15:02:58422void Call::ConfigureSync(const std::string& sync_group) {
423 // Set sync only if there was no previous one.
424 if (config_.voice_engine == nullptr || sync_group.empty())
425 return;
426
427 AudioReceiveStream* sync_audio_stream = nullptr;
428 // Find existing audio stream.
429 const auto it = sync_stream_mapping_.find(sync_group);
430 if (it != sync_stream_mapping_.end()) {
431 sync_audio_stream = it->second;
432 } else {
433 // No configured audio stream, see if we can find one.
434 for (const auto& kv : audio_receive_ssrcs_) {
435 if (kv.second->config().sync_group == sync_group) {
436 if (sync_audio_stream != nullptr) {
437 LOG(LS_WARNING) << "Attempting to sync more than one audio stream "
438 "within the same sync group. This is not "
439 "supported in the current implementation.";
440 break;
441 }
442 sync_audio_stream = kv.second;
443 }
444 }
445 }
446 if (sync_audio_stream)
447 sync_stream_mapping_[sync_group] = sync_audio_stream;
448 size_t num_synced_streams = 0;
449 for (VideoReceiveStream* video_stream : video_receive_streams_) {
450 if (video_stream->config().sync_group != sync_group)
451 continue;
452 ++num_synced_streams;
453 if (num_synced_streams > 1) {
454 // TODO(pbos): Support synchronizing more than one A/V pair.
455 // https://code.google.com/p/webrtc/issues/detail?id=4762
456 LOG(LS_WARNING) << "Attempting to sync more than one audio/video pair "
457 "within the same sync group. This is not supported in "
458 "the current implementation.";
459 }
460 // Only sync the first A/V pair within this sync group.
461 if (sync_audio_stream != nullptr && num_synced_streams == 1) {
462 video_stream->SetSyncChannel(config_.voice_engine,
463 sync_audio_stream->config().voe_channel_id);
464 } else {
465 video_stream->SetSyncChannel(config_.voice_engine, -1);
466 }
467 }
468}
469
Fredrik Solenbergad867862015-04-29 13:24:01470PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
471 const uint8_t* packet,
472 size_t length) {
pbos@webrtc.org2a9108f2013-05-16 12:08:03473 // TODO(pbos): Figure out what channel needs it actually.
474 // Do NOT broadcast! Also make sure it's a valid packet.
pbos@webrtc.orgbc57e0f2014-05-14 13:57:12475 // Return DELIVERY_UNKNOWN_SSRC if it can be determined that
476 // there's no receiver of the packet.
pbos@webrtc.org2a9108f2013-05-16 12:08:03477 bool rtcp_delivered = false;
Fredrik Solenbergad867862015-04-29 13:24:01478 if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
pbos@webrtc.org9b707ca2014-09-03 16:17:12479 ReadLockScoped read_lock(*receive_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01480 for (VideoReceiveStream* stream : video_receive_streams_) {
ivoc35fd7532015-09-09 07:09:43481 if (stream->DeliverRtcp(packet, length)) {
pbos@webrtc.org78ab5112013-08-05 12:49:22482 rtcp_delivered = true;
ivoc35fd7532015-09-09 07:09:43483 if (event_log_)
484 event_log_->LogRtcpPacket(true, media_type, packet, length);
485 }
pbos@webrtc.orgce851092013-08-05 12:01:36486 }
487 }
Fredrik Solenbergad867862015-04-29 13:24:01488 if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
pbos@webrtc.org9b707ca2014-09-03 16:17:12489 ReadLockScoped read_lock(*send_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01490 for (VideoSendStream* stream : video_send_streams_) {
ivoc35fd7532015-09-09 07:09:43491 if (stream->DeliverRtcp(packet, length)) {
pbos@webrtc.org78ab5112013-08-05 12:49:22492 rtcp_delivered = true;
ivoc35fd7532015-09-09 07:09:43493 if (event_log_)
494 event_log_->LogRtcpPacket(false, media_type, packet, length);
495 }
pbos@webrtc.org2a9108f2013-05-16 12:08:03496 }
497 }
pbos@webrtc.orgbc57e0f2014-05-14 13:57:12498 return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
pbos@webrtc.org2a9108f2013-05-16 12:08:03499}
500
Fredrik Solenbergad867862015-04-29 13:24:01501PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
502 const uint8_t* packet,
stefan30bf7782015-09-08 12:36:15503 size_t length,
504 const PacketTime& packet_time) {
pbos@webrtc.orgf8be3d22014-07-08 07:38:12505 // Minimum RTP header size.
506 if (length < 12)
507 return DELIVERY_PACKET_ERROR;
508
sprang@webrtc.orgcce642c2015-01-28 12:37:36509 uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
pbos@webrtc.orgf8be3d22014-07-08 07:38:12510
pbos@webrtc.org9b707ca2014-09-03 16:17:12511 ReadLockScoped read_lock(*receive_crit_);
Fredrik Solenbergad867862015-04-29 13:24:01512 if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
513 auto it = audio_receive_ssrcs_.find(ssrc);
514 if (it != audio_receive_ssrcs_.end()) {
ivoc35fd7532015-09-09 07:09:43515 auto status = it->second->DeliverRtp(packet, length, packet_time)
516 ? DELIVERY_OK
517 : DELIVERY_PACKET_ERROR;
518 if (status == DELIVERY_OK && event_log_)
519 event_log_->LogRtpHeader(true, media_type, packet, length);
520 return status;
Fredrik Solenbergad867862015-04-29 13:24:01521 }
522 }
523 if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
524 auto it = video_receive_ssrcs_.find(ssrc);
525 if (it != video_receive_ssrcs_.end()) {
ivoc35fd7532015-09-09 07:09:43526 auto status = it->second->DeliverRtp(packet, length, packet_time)
527 ? DELIVERY_OK
528 : DELIVERY_PACKET_ERROR;
529 if (status == DELIVERY_OK && event_log_)
530 event_log_->LogRtpHeader(true, media_type, packet, length);
531 return status;
Fredrik Solenbergad867862015-04-29 13:24:01532 }
533 }
534 return DELIVERY_UNKNOWN_SSRC;
pbos@webrtc.org2a9108f2013-05-16 12:08:03535}
536
stefan30bf7782015-09-08 12:36:15537PacketReceiver::DeliveryStatus Call::DeliverPacket(
538 MediaType media_type,
539 const uint8_t* packet,
540 size_t length,
541 const PacketTime& packet_time) {
pbos@webrtc.org6aae61c2014-07-08 12:10:51542 if (RtpHeaderParser::IsRtcp(packet, length))
Fredrik Solenbergad867862015-04-29 13:24:01543 return DeliverRtcp(media_type, packet, length);
pbos@webrtc.org2a9108f2013-05-16 12:08:03544
stefan30bf7782015-09-08 12:36:15545 return DeliverRtp(media_type, packet, length, packet_time);
pbos@webrtc.org2a9108f2013-05-16 12:08:03546}
547
548} // namespace internal
549} // namespace webrtc