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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 "webrtc/video/video_send_stream.h" | 11 #include "webrtc/video/video_send_stream.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <sstream> | 14 #include <sstream> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
20 #include "webrtc/modules/pacing/include/packet_router.h" | 20 #include "webrtc/modules/pacing/include/packet_router.h" |
21 #include "webrtc/system_wrappers/interface/logging.h" | 21 #include "webrtc/system_wrappers/interface/logging.h" |
22 #include "webrtc/system_wrappers/interface/trace_event.h" | 22 #include "webrtc/system_wrappers/interface/trace_event.h" |
23 #include "webrtc/video/video_capture_input.h" | 23 #include "webrtc/video/video_capture_input.h" |
24 #include "webrtc/video_engine/call_stats.h" | |
25 #include "webrtc/video_engine/payload_router.h" | |
24 #include "webrtc/video_engine/vie_channel.h" | 26 #include "webrtc/video_engine/vie_channel.h" |
25 #include "webrtc/video_engine/vie_channel_group.h" | 27 #include "webrtc/video_engine/vie_channel_group.h" |
26 #include "webrtc/video_engine/vie_defines.h" | 28 #include "webrtc/video_engine/vie_defines.h" |
27 #include "webrtc/video_engine/vie_encoder.h" | 29 #include "webrtc/video_engine/vie_encoder.h" |
28 #include "webrtc/video_send_stream.h" | 30 #include "webrtc/video_send_stream.h" |
29 | 31 |
30 namespace webrtc { | 32 namespace webrtc { |
33 | |
34 class BitrateAllocator; | |
35 class PacedSender; | |
36 class RtcpIntraFrameObserver; | |
37 class TransportFeedbackObserver; | |
38 | |
31 std::string | 39 std::string |
32 VideoSendStream::Config::EncoderSettings::ToString() const { | 40 VideoSendStream::Config::EncoderSettings::ToString() const { |
33 std::stringstream ss; | 41 std::stringstream ss; |
34 ss << "{payload_name: " << payload_name; | 42 ss << "{payload_name: " << payload_name; |
35 ss << ", payload_type: " << payload_type; | 43 ss << ", payload_type: " << payload_type; |
36 ss << ", encoder: " << (encoder != nullptr ? "(VideoEncoder)" : "nullptr"); | 44 ss << ", encoder: " << (encoder != nullptr ? "(VideoEncoder)" : "nullptr"); |
37 ss << '}'; | 45 ss << '}'; |
38 return ss.str(); | 46 return ss.str(); |
39 } | 47 } |
40 | 48 |
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107 int channel_id, | 115 int channel_id, |
108 const VideoSendStream::Config& config, | 116 const VideoSendStream::Config& config, |
109 const VideoEncoderConfig& encoder_config, | 117 const VideoEncoderConfig& encoder_config, |
110 const std::map<uint32_t, RtpState>& suspended_ssrcs) | 118 const std::map<uint32_t, RtpState>& suspended_ssrcs) |
111 : transport_adapter_(config.send_transport), | 119 : transport_adapter_(config.send_transport), |
112 encoded_frame_proxy_(config.post_encode_callback), | 120 encoded_frame_proxy_(config.post_encode_callback), |
113 config_(config), | 121 config_(config), |
114 suspended_ssrcs_(suspended_ssrcs), | 122 suspended_ssrcs_(suspended_ssrcs), |
115 module_process_thread_(module_process_thread), | 123 module_process_thread_(module_process_thread), |
116 channel_group_(channel_group), | 124 channel_group_(channel_group), |
117 channel_id_(channel_id), | |
118 use_config_bitrate_(true), | 125 use_config_bitrate_(true), |
119 stats_proxy_(Clock::GetRealTimeClock(), config) { | 126 stats_proxy_(Clock::GetRealTimeClock(), config) { |
120 RTC_DCHECK(!config_.rtp.ssrcs.empty()); | 127 RTC_DCHECK(!config_.rtp.ssrcs.empty()); |
121 RTC_CHECK(channel_group->CreateSendChannel( | 128 |
122 channel_id_, &transport_adapter_, &stats_proxy_, | 129 // Set up Call-wide sequence numbers, if configured for this send stream. |
123 config.pre_encode_callback, num_cpu_cores, config_)); | 130 TransportFeedbackObserver* transport_feedback_observer = nullptr; |
124 vie_channel_ = channel_group_->GetChannel(channel_id_); | 131 for (const RtpExtension& extension : config.rtp.extensions) { |
125 vie_encoder_ = channel_group_->GetEncoder(channel_id_); | 132 if (extension.name == RtpExtension::kTransportSequenceNumber) { |
133 transport_feedback_observer = | |
134 channel_group_->GetTransportFeedbackObserver(); | |
135 break; | |
136 } | |
137 } | |
138 | |
139 const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs; | |
140 RTC_DCHECK(!ssrcs.empty()); | |
stefan-webrtc
2015/10/15 11:33:45
No need to DCHECK this here and on line 127.
mflodman
2015/10/15 14:39:22
Thanks, copy paste generated from channel group.
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141 | |
142 vie_encoder_.reset(new ViEEncoder( | |
143 channel_id, num_cpu_cores, module_process_thread_, &stats_proxy_, | |
144 config.pre_encode_callback, channel_group_->pacer(), | |
pbos-webrtc
2015/10/15 12:07:55
Do we want pacer/allocator through constructor ins
mflodman
2015/10/15 14:39:21
Kept it as is for now, since I need to leave for t
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145 channel_group_->bitrate_allocator())); | |
146 RTC_CHECK(vie_encoder_->Init()); | |
147 | |
148 vie_channel_.reset(new ViEChannel( | |
149 num_cpu_cores, config.send_transport, module_process_thread_, | |
150 channel_group_->GetRtcpIntraFrameObserver(), | |
151 channel_group_->GetBitrateController()->CreateRtcpBandwidthObserver(), | |
152 transport_feedback_observer, | |
153 channel_group_->GetRemoteBitrateEstimator(), | |
154 channel_group_->GetCallStats()->rtcp_rtt_stats(), channel_group_->pacer(), | |
155 channel_group_->packet_router(), ssrcs.size(), true)); | |
156 RTC_CHECK(vie_channel_->Init() == 0); | |
pbos-webrtc
2015/10/15 12:07:55
Can we make this one a bool while we're at it?
mflodman
2015/10/15 14:39:22
I did that change, but reverted. I'm not a big fan
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157 | |
158 vie_encoder_->StartThreadsAndSetSharedMembers( | |
159 vie_channel_->send_payload_router(), | |
160 vie_channel_->vcm_protection_callback()); | |
161 | |
162 std::vector<uint32_t> first_ssrc(1, ssrcs[0]); | |
163 vie_encoder_->SetSsrcs(first_ssrc); | |
stefan-webrtc
2015/10/15 11:33:45
It's not entirely clear to me why we only set the
pbos-webrtc
2015/10/15 12:07:55
Pref setting all the ssrcs, and then vie_encoder_
mflodman
2015/10/15 14:39:21
This is just copy paste, I can try to figure out a
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126 | 164 |
127 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { | 165 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { |
128 const std::string& extension = config_.rtp.extensions[i].name; | 166 const std::string& extension = config_.rtp.extensions[i].name; |
129 int id = config_.rtp.extensions[i].id; | 167 int id = config_.rtp.extensions[i].id; |
130 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 168 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
131 RTC_DCHECK_GE(id, 1); | 169 RTC_DCHECK_GE(id, 1); |
132 RTC_DCHECK_LE(id, 14); | 170 RTC_DCHECK_LE(id, 14); |
133 if (extension == RtpExtension::kTOffset) { | 171 if (extension == RtpExtension::kTOffset) { |
134 RTC_CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id)); | 172 RTC_CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id)); |
135 } else if (extension == RtpExtension::kAbsSendTime) { | 173 } else if (extension == RtpExtension::kAbsSendTime) { |
136 RTC_CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id)); | 174 RTC_CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id)); |
137 } else if (extension == RtpExtension::kVideoRotation) { | 175 } else if (extension == RtpExtension::kVideoRotation) { |
138 RTC_CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id)); | 176 RTC_CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id)); |
139 } else if (extension == RtpExtension::kTransportSequenceNumber) { | 177 } else if (extension == RtpExtension::kTransportSequenceNumber) { |
140 RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id)); | 178 RTC_CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id)); |
141 } else { | 179 } else { |
142 RTC_NOTREACHED() << "Registering unsupported RTP extension."; | 180 RTC_NOTREACHED() << "Registering unsupported RTP extension."; |
143 } | 181 } |
144 } | 182 } |
145 | 183 |
146 // TODO(pbos): Consider configuring REMB in Call. | 184 // TODO(pbos): Consider configuring REMB in Call. |
147 channel_group_->SetChannelRembStatus(true, false, vie_channel_); | 185 channel_group_->SetChannelRembStatus(true, false, vie_channel_.get()); |
148 | 186 |
149 // Enable NACK, FEC or both. | 187 // Enable NACK, FEC or both. |
150 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; | 188 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
151 const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; | 189 const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; |
152 // TODO(changbin): Should set RTX for RED mapping in RTP sender in future. | 190 // TODO(changbin): Should set RTX for RED mapping in RTP sender in future. |
153 vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec, | 191 vie_channel_->SetProtectionMode(enable_protection_nack, enable_protection_fec, |
154 config_.rtp.fec.red_payload_type, | 192 config_.rtp.fec.red_payload_type, |
155 config_.rtp.fec.ulpfec_payload_type); | 193 config_.rtp.fec.ulpfec_payload_type); |
156 vie_encoder_->UpdateProtectionMethod(enable_protection_nack, | 194 vie_encoder_->UpdateProtectionMethod(enable_protection_nack, |
157 enable_protection_fec); | 195 enable_protection_fec); |
158 | 196 |
159 ConfigureSsrcs(); | 197 ConfigureSsrcs(); |
160 | 198 |
161 vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str()); | 199 vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str()); |
162 | 200 |
163 input_.reset(new internal::VideoCaptureInput( | 201 input_.reset(new internal::VideoCaptureInput( |
164 module_process_thread_, vie_encoder_, config_.local_renderer, | 202 module_process_thread_, vie_encoder_.get(), config_.local_renderer, |
165 &stats_proxy_, this, config_.encoding_time_observer)); | 203 &stats_proxy_, this, config_.encoding_time_observer)); |
166 | 204 |
167 // 28 to match packet overhead in ModuleRtpRtcpImpl. | 205 // 28 to match packet overhead in ModuleRtpRtcpImpl. |
168 RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28)); | 206 RTC_DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28)); |
169 vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28)); | 207 vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28)); |
170 | 208 |
171 RTC_DCHECK(config.encoder_settings.encoder != nullptr); | 209 RTC_DCHECK(config.encoder_settings.encoder != nullptr); |
172 RTC_DCHECK_GE(config.encoder_settings.payload_type, 0); | 210 RTC_DCHECK_GE(config.encoder_settings.payload_type, 0); |
173 RTC_DCHECK_LE(config.encoder_settings.payload_type, 127); | 211 RTC_DCHECK_LE(config.encoder_settings.payload_type, 127); |
174 RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder( | 212 RTC_CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder( |
175 config.encoder_settings.encoder, | 213 config.encoder_settings.encoder, |
176 config.encoder_settings.payload_type, | 214 config.encoder_settings.payload_type, |
177 config.encoder_settings.internal_source)); | 215 config.encoder_settings.internal_source)); |
178 | 216 |
179 RTC_CHECK(ReconfigureVideoEncoder(encoder_config)); | 217 RTC_CHECK(ReconfigureVideoEncoder(encoder_config)); |
180 | 218 |
181 vie_channel_->RegisterSendSideDelayObserver(&stats_proxy_); | 219 vie_channel_->RegisterSendSideDelayObserver(&stats_proxy_); |
182 | 220 |
183 if (config_.post_encode_callback) | 221 if (config_.post_encode_callback) |
184 vie_encoder_->RegisterPostEncodeImageCallback(&encoded_frame_proxy_); | 222 vie_encoder_->RegisterPostEncodeImageCallback(&encoded_frame_proxy_); |
185 | 223 |
186 if (config_.suspend_below_min_bitrate) | 224 if (config_.suspend_below_min_bitrate) |
187 vie_encoder_->SuspendBelowMinBitrate(); | 225 vie_encoder_->SuspendBelowMinBitrate(); |
188 | 226 |
227 channel_group_->AddEncoder(ssrcs, vie_encoder_.get()); | |
228 | |
189 vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_); | 229 vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_); |
190 vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_); | 230 vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_); |
191 vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_); | 231 vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_); |
192 vie_channel_->RegisterSendBitrateObserver(&stats_proxy_); | 232 vie_channel_->RegisterSendBitrateObserver(&stats_proxy_); |
193 vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_); | 233 vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_); |
194 } | 234 } |
195 | 235 |
196 VideoSendStream::~VideoSendStream() { | 236 VideoSendStream::~VideoSendStream() { |
197 vie_channel_->RegisterSendFrameCountObserver(nullptr); | 237 vie_channel_->RegisterSendFrameCountObserver(nullptr); |
198 vie_channel_->RegisterSendBitrateObserver(nullptr); | 238 vie_channel_->RegisterSendBitrateObserver(nullptr); |
199 vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr); | 239 vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr); |
200 vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr); | 240 vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr); |
201 vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr); | 241 vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr); |
202 | 242 |
203 // Remove capture input (thread) so that it's not running after the current | 243 // Remove capture input (thread) so that it's not running after the current |
204 // channel is deleted. | 244 // channel is deleted. |
205 input_.reset(); | 245 input_.reset(); |
206 | 246 |
207 vie_encoder_->DeRegisterExternalEncoder( | 247 vie_encoder_->DeRegisterExternalEncoder( |
208 config_.encoder_settings.payload_type); | 248 config_.encoder_settings.payload_type); |
209 | 249 |
210 channel_group_->DeleteChannel(channel_id_); | 250 channel_group_->GetCallStats()->DeregisterStatsObserver( |
251 vie_channel_->GetStatsObserver()); | |
252 channel_group_->SetChannelRembStatus(false, false, vie_channel_.get()); | |
253 | |
254 // Remove the feedback, stop all encoding threads and processing. This must be | |
255 // done before deleting the channel. | |
256 channel_group_->RemoveEncoder(vie_encoder_.get()); | |
257 vie_encoder_->StopThreadsAndRemoveSharedMembers(); | |
258 | |
259 unsigned int remote_ssrc = 0; | |
pbos-webrtc
2015/10/15 12:07:55
Is this not an uint32_t?
mflodman
2015/10/15 14:39:21
Yes, should be. Copy paste from ChannelManager.
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260 vie_channel_->GetRemoteSSRC(&remote_ssrc); | |
pbos-webrtc
2015/10/15 12:07:55
Can this fail? If so should we remove the stream z
mflodman
2015/10/15 14:39:21
No, it can't fail.
| |
261 | |
262 channel_group_->GetRemoteBitrateEstimator()->RemoveStream(remote_ssrc); | |
211 } | 263 } |
212 | 264 |
213 VideoCaptureInput* VideoSendStream::Input() { | 265 VideoCaptureInput* VideoSendStream::Input() { |
214 return input_.get(); | 266 return input_.get(); |
215 } | 267 } |
216 | 268 |
217 void VideoSendStream::Start() { | 269 void VideoSendStream::Start() { |
218 transport_adapter_.Enable(); | 270 transport_adapter_.Enable(); |
219 vie_encoder_->Pause(); | 271 vie_encoder_->Pause(); |
220 if (vie_channel_->StartSend() == 0) { | 272 if (vie_channel_->StartSend() == 0) { |
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496 vie_channel_->IsSendingFecEnabled()); | 548 vie_channel_->IsSendingFecEnabled()); |
497 | 549 |
498 // Restart the media flow | 550 // Restart the media flow |
499 vie_encoder_->Restart(); | 551 vie_encoder_->Restart(); |
500 | 552 |
501 return true; | 553 return true; |
502 } | 554 } |
503 | 555 |
504 } // namespace internal | 556 } // namespace internal |
505 } // namespace webrtc | 557 } // namespace webrtc |
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